2010-09-13  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.36 Released.

2010-08-23  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.36-rc1 Released.

2010-08-20 16:46 +0000 [r283048-283123]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Merged revision 278274 from
	  https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
	  | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
	  line Reference correct struct member for unlikely event
	  PRI_EVENT_CONFIG_ERR. ..........

	* channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending
	  ALERTING is a protocol error The PRI layer in chan_dadhi will
	  check if a PROGRESS message has already been sent, and not allow
	  sending another (although that is technically allowed by the Q931
	  spec), however it does not protect against sending an ALERTING
	  and then sending a PROGRESS message, which is a violation of the
	  specification. Most switches don't seem to care too deeply about
	  this, but some do, and will disconnect the call when receiving
	  this invalid sequence. Protocol specification reference:
	  T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
	  protocol control (network side) point-point (sheet 3 of 8)"
	  (closes issue #17874) Reported by: nic_bellamy Patches:
	  asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
	  nic bellamy (license 299)
	  asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299)
	  asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299)

2010-08-19 21:03 +0000 [r282893]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: tos_sip option was not being set correctly
	  When tos_sip is used, the tos of the sip socket is only set
	  correctly if the socket binding changes on a reload. If the
	  binding stays the same but the TOS changes, the new tos value
	  would not take into effect. This patch fixes that. (closes issue
	  #17712) Reported by: nickb

2010-08-19 02:12 +0000 [r282729]  Terry Wilson <twilson@digium.com>

	* configs/sip.conf.sample: Add some documentation about codec
	  negotiation to sip.conf

2010-08-16 17:06 +0000 [r282430]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Send a SRCCHANGE indication when we masquerade
	  Masquerading a channel means that the src of the audio is
	  potentially changing, so send a SRCCHANGE so that RTP-based media
	  streams can get a new SSRC generated to reflect the change.
	  Original patch by addix (along with lots of testing--thanks!).
	  (closes issue #17007) Reported by: addix Patches:
	  1001-reset-SSRC-original-channel.diff uploaded by addix (license
	  1006) srcchange.diff uploaded by twilson (license 396) Tested by:
	  addix, twilson Review: https://reviewboard.asterisk.org/r/862/

2010-08-12 22:49 +0000 [r282129]  Jason Parker <jparker@digium.com>

	* pbx/pbx_config.c: Register CLI commands before parsing config, in
	  case there is a config error.

2010-08-12 03:00 +0000 [r281911]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Ensure SSRC is changed when media source is
	  changed to resolve audio delay. This change causes the SSRC to
	  change right before the channels are bridged, which is what used
	  to happen. It seems that fixes were made to attempt limiting SSRC
	  changes, targeted mainly at sending DTMF. DTMF is not affecting
	  the SSRC with this change. There are two other control frames
	  sent in ast_channel_bridge that probably should also be changed
	  to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this
	  change up to the discretion of resolving issue #17007. For
	  reference - old review implementing new control frame SRCCHANGE:
	  https://reviewboard.asterisk.org/r/540 (closes issue #17404)
	  Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
	  (license 325) Tested by: sdolloff

2010-08-11 18:28 +0000 [r281762-281819]  Leif Madsen <lmadsen@digium.com>

	* configs/say.conf.sample: Add Danish support to say.conf.sample
	  (closes issue #17836) Reported by: RoadKill Patches:
	  say.conf.sample.patch.dk uploaded by RoadKill (license 933)

	* configs/say.conf.sample: Allow say.conf to handle large numbers
	  ending with multiple zeros. (closes issue #17833) Reported by:
	  RoadKill Patches: say.conf.sample.patch.largenumbers uploaded by
	  RoadKill (license 933)

2010-08-10 17:45 +0000 [r281566]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: Reset visible indication after answer. (closes
	  issue #17641) Reported by: klaus3000 Patches:
	  ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
	  klaus3000 (license 65) Tested by: schmidts

2010-08-09 20:04 +0000 [r281390]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c: Prevent loss of Caller ID information set
	  on local channel after masquerade. Caller ID set on the channel
	  before a masquerade occurs when using a local channel would cause
	  the information to be lost. The problem was that the information
	  was set on a channel destined to be hung up. The somewhat
	  confusing fix is to detect if any Caller ID has been set on the
	  channel and if so preswap the Caller ID data so that basically
	  the masquerade puts the data back. (closes issue #17138) Reported
	  by: kobaz Review: https://reviewboard.asterisk.org/r/847/

2010-08-06 21:34 +0000 [r281185]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: chan_sip: fixes provisional keepalive
	  scheduled item crash There is a scheduler item in chan_sip that
	  keeps sending the last provisional message in response to an
	  INVITE Request for a period of time until a final response to
	  that INVITE is sent. Because of the way this scheduler item
	  works, it requires a reference to a sip_pvt pointer to work
	  properly. The problem with this is that it is currently possible
	  (but rare) for the sip_pvt to get destroyed and that scheduler
	  item to still exist. When this occurs, the scheduler event fires
	  and attempts to access a freed sip_pvt which causes a crash.
	  (closes issue #17497) Reported by: anonymouz666 Patches:
	  keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671)
	  Review: https://reviewboard.asterisk.org/r/849/

2010-08-05 07:28 +0000 [r280982]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Change context lock back to a mutex, because
	  functionality depends upon the lock being recursive. (closes
	  issue #17643) Reported by: zerohalo Patches:
	  20100726__issue17643.diff.txt uploaded by tilghman (license 14)
	  Tested by: zerohalo

2010-08-04 18:54 +0000 [r280944]  Russell Bryant <russell@digium.com>

	* contrib/scripts/astcli (added): Copy astcli back to 1.4 so it's
	  available for automated testing purposes.

2010-08-03 20:49 +0000 [r280811]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_callerid.c, channels/chan_dahdi.c: Prevent DAHDI
	  channels from overriding the callerid, once it's been set by the
	  user. (closes issue #16661) Reported by: jstapleton Patches:
	  20100414__issue16661.diff.txt uploaded by tilghman (license 14)
	  20100415__issue16661__1.6.2.diff.txt uploaded by tilghman
	  (license 14) Tested by: jstapleton

2010-07-29 19:04 +0000 [r280448]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes issue with translator frame not getting
	  freed A translator frame even if it local storage so the
	  translation path can be freed. This issue prevented g729 licenses
	  from being freed up. (closes issue #17630) Reported by: manvirr
	  Patches: encoder_fix.diff uploaded by dvossel (license 671)
	  Tested by: manvirr, dvossel

2010-07-29 15:52 +0000 [r280341]  Jean Galarneau <jgalarneau@digium.com>

	* apps/app_meetme.c: Fix a dsp structure leak occuring when a local
	  channel is put into a meetme conference, then masquaraded away.
	  ABE-2422

2010-07-28 13:50 +0000 [r280088]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/live_ast: Update help text to be less confusing.

2010-07-27 20:33 +0000 [r279945]  David Vossel <dvossel@digium.com>

	* main/channel.c, include/asterisk/audiohook.h, main/audiohook.c:
	  remove empty audiohook write list on channel If a channel has an
	  audiohook write list created on it, that list stays on the
	  channel until the channel is destroyed. There is no reason to
	  keep that list on the channel if it becomes empty. If it is empty
	  that just means we are doing needless translating for every
	  ast_read and ast_write. This patch removes the audiohook list
	  from the channel once it is detected to be empty on either a read
	  or write. If a audiohook is added back to the channel after this
	  list is destroyed, the list just gets recreated as if it never
	  existed to begin with. (closes issue #17630) Reported by: manvirr
	  Review: https://reviewboard.asterisk.org/r/799/

2010-07-24 23:57 +0000 [r279346]  Bradley Latus <brad.latus@gmail.com>

	* doc/asterisk.8: Minor update to man page

2010-07-24 23:27 +0000 [r279344]  Jeff Peeler <jpeeler@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Provide a default value for DAHDI_TRANSCODE so when DAHDI is not
	  installed menuselect doesn't get confused: Unknown value '' found
	  in build_tools/menuselect-deps for DAHDI_TRANSCODE

2010-07-23 21:56 +0000 [r279206]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, apps/app_queue.c: SIP promiscuous redirect could
	  fail to dial the redirect. The ast_channel was created with one
	  variable to ast_request() but the call to ast_call() that
	  initiates the outgoing call was using a different variable. The
	  two variables are not equivalent if the call_forward string
	  included a channel technology specifier. e.g., SIP/200

2010-07-23 18:04 +0000 [r279053]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Backport fixes for sip_uri_params_cmp() from
	  trunk.

2010-07-23 17:04 +0000 [r278981-278984]  Tilghman Lesher <tlesher@digium.com>

	* autoconf/ast_check_pwlib.m4, configure, configure.ac: Establish a
	  maximum version for openh323 (i.e. not opal), because chan_h323
	  will fail to load, even if it links. (issue #17679) Reported by:
	  am

	* main/asterisk.c: Avoid race with consolethread on shutdown (on
	  parallel processors). (closes issue #17080) Reported by:
	  sybasesql Patches: 20100721__issue17080.diff.txt uploaded by
	  tilghman (license 14) Tested by: sybasesql

2010-07-22 19:31 +0000 [r278701]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: DNID does not get cleard on a new call
	  when using immediate=yes with ISDN signaling. When you are using
	  chan_dahdi ISDN signaling with immediate=yes and a call comes in
	  without a DNID then you get the DNID of a previous call.
	  Chan_dahdi does not touch the DNID field on a new call if it does
	  not have a DNID. Made always copy the DNID from the new call. The
	  patches backport the relevant changes from trunk -r210387.
	  (closes issue #17568) Reported by: wuwu Patches:
	  issue17568_v1.4.patch uploaded by rmudgett (license 664)
	  issue17568_v1.6.2.patch uploaded by rmudgett (license 664)

2010-08-10  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.35 Released.

2010-07-22  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.35-rc1 Released.

2010-07-22 14:55 +0000 [r278618]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Allow PLC to function properly when channels use
	  SLIN for audio. If a channel involved in a bridge was using SLIN
	  audio, then translation paths were not guaranteed to be set up
	  properly since in all likelihood the number of translation steps
	  was only 1. This patch enforces the transcode_via_slin behavior
	  if transcode_via_slin or generic_plc is enabled and one of the
	  formats to make compatible is SLIN. AST-352

2010-07-20 22:23 +0000 [r278023-278261]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Delete IMAP messages in reverse order, to
	  ensure reordering after each expunge does not cause deletion of
	  the wrong message. (closes issue #16350) Reported by: noahisaac
	  Patches: 20100623__issue16350.diff.txt uploaded by tilghman
	  (license 14)

	* main/autoservice.c, res/res_features.c,
	  include/asterisk/channel.h: Do not queue up DTMF frames while a
	  call is on hold. (Fixes ABE-2110)

	* main/manager.c: Off-by-one error (closes issue #16506) Reported
	  by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by
	  tilghman (license 14)

2010-07-19 20:56 +0000 [r277944]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Regression with T.38 negotiation Prior to
	  1.4.26.3 T.38 negotiation worked properly, in the case of the
	  reporter. (issue #16852) Reported by: cfc (closes issue #16705)
	  Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
	  by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
	  samdell3 Review: https://reviewboard.asterisk.org/r/754/

2010-07-19 20:16 +0000 [r277906]  Jean Galarneau <jgalarneau@digium.com>

	* res/res_features.c: Avoid trying to pickup a parked extension
	  before the park operation is completed. A crash could occur if
	  the extension is picked up while the parking extension is being
	  announced. Testing pu->notquiteyet while searching for a parked
	  extension resolves this crash. (ABE-2418)

2010-07-17 16:59 +0000 [r277738]  Tilghman Lesher <tlesher@digium.com>

	* autoconf/ast_func_fork.m4, configure: Remove uclibc cross-compile
	  triplet, as uclibc has a working fork()... it's only uclinux that
	  does not. (closes issue #17616) Reported by: pprindeville

2010-07-16 22:43 +0000 [r277625]  Tim Ringenbach <tim.ringenbach@gmail.com>

	* res/res_features.c: Save and restore AST_FLAG_BRIDGE_HANGUP_DONT
	  on attended transfer. ast_bridge_call() clears
	  AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
	  ast_bridge_call() is called for a second bridge on the same
	  channel, and it clears that flag, which still needs to get set
	  for when the original ast_bridge_call() gets control back and
	  checks it. Review: https://reviewboard.asterisk.org/r/741

2010-07-16 21:54 +0000 [r277568]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, res/res_config_odbc.c: Since we split
	  values at the semicolon, we should store values with a semicolon
	  as an encoded value. (closes issue #17369) Reported by: gkservice
	  Patches: 20100625__issue17369.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman

2010-07-16 21:18 +0000 [r277497]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Default to no udptl error correction so that
	  error correction will be disabled in the event that the remote
	  end indicates that they do not support the error correction mode
	  we requested. FAX-128

2010-07-16 20:18 +0000 [r277419]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: priexclusive in chan_dahdi.conf ignored
	  when reloading dahdi module During a reload, the priexclusive and
	  outsignalling parameters are not read in from the config file as
	  intended. Unfortunately, they get set to defaults as a result.
	  This patch makes sure that they do not get set to defaults during
	  a reload. (closes issue #17441) Reported by: mtryfoss Patches:
	  issue17441_v1.4.patch uploaded by rmudgett (license 664)
	  issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
	  issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
	  by: rmudgett

2010-07-16 18:30 +0000 [r277327]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c: Interpret device state AST_DEVICE_UNKNOWN as
	  extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
	  Reported by: francesco_r Patches: pbx.c.patch uploaded by
	  viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar

2010-07-16 18:04 +0000 [r277261]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: If variable gotten is not set, will segfault on
	  Solaris. (closes issue #17636) Reported by: bklang

2010-07-16 17:29 +0000 [r277247]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: For pass through DTMF tones, measure the actual
	  duration between the begin and end packets on the wire. If it is
	  detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
	  emulation. AST-362

2010-07-16 17:10 +0000 [r277182]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_amd.c: Total analysis time error with SIP and silence
	  suppression When using app_amd with SIP providers that have
	  silence suppression on, the iTotalTime count increases
	  exponentially. (closes issue #17656) Reported by: juls

2010-07-15 13:48 +0000 [r276652]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: In a perfect world, the frame source would never
	  be NULL. In the meantime, don't crash when it is.

2010-07-14 11:49 +0000 [r276267]  Leif Madsen <lmadsen@digium.com>

	* configs/voicemail.conf.sample: Update documentation for
	  voicemail.conf externpass option.

2010-07-13 19:14 +0000 [r275994-276126]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Only reset a CDR that exists.

	* res/res_features.c: Use chan->cdr instead of chan_cdr (just like
	  peer->cdr instead of peer_cdr in the last commit).

	* res/res_features.c: Access peer->cdr directly instead of through
	  a saved off reference. At this point in the code, it is possible
	  that peer_cdr may be invalid. Specifically, in the blind transfer
	  code, CDRs are swapped between channels. So, peer_cdr is no
	  longer == peer->cdr. The scenario that exposed a crash in this
	  code was a blind transfer that hit the system call limit, causing
	  the transferee channel to get destroyed after the transfer
	  attempt failed. Even if it succeeds and this code doesn't crash,
	  this code was still trying to reset a CDR on a channel that was
	  now owned by a different thread, which is a BadThing(tm).
	  (ABE-2417)

2010-07-13 14:47 +0000 [r275909]  Tilghman Lesher <tlesher@digium.com>

	* contrib/realtime/mysql/sipfriends.sql,
	  contrib/realtime/mysql/voicemail.sql,
	  contrib/scripts/realtime_pgsql.sql (removed),
	  contrib/scripts/vmdb.sql (removed),
	  contrib/scripts/iax-friends.sql (removed),
	  contrib/realtime/mysql/iaxfriends.sql,
	  contrib/realtime/mysql/meetme.sql, contrib/scripts/meetme.sql
	  (removed), contrib/realtime (added), contrib/realtime/postgresql,
	  contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
	  contrib/realtime/oracle, contrib/realtime/sqlserver,
	  contrib/scripts/sip-friends.sql (removed): Move SQL scripts into
	  their own database-specific directories.

2010-07-12 20:34 +0000 [r275665-275773]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: Make user removals and traversals thread safe
	  in meetme. Race conditions present in meetme involving the user
	  list where a lack of locking has the potential for a user to be
	  removed during a traversal or as in the case of the reporter
	  after checking if the list is empty could cause a crash. Fixing
	  this was done by convering the userlist to an ao2 container.
	  (closes issue #17390) Reported by: Vince Review:
	  https://reviewboard.asterisk.org/r/746/

	* main/channel.c: Change ast_write to not stop generator when
	  called from ast_prod. For SIP channels configured with the
	  progressinband option on, the ringback was being immediately
	  stopped. This problem was due to ast_prod being moved for a
	  deadlock fix in 259858. Prodding the channel after setting up the
	  generator triggered the check in ast_write to stop the generator.
	  The fix here should write the frame the same as was done before
	  the call to ast_prod was moved. (closes issue #17372) Reported
	  by: tech_admin

2010-07-09 19:28 +0000 [r275241-275290]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/cli.c: fix tab-completion for unload command. (closes issue
	  #17536) Reported by: junky Patches: unload_vs_mod_unload.diff
	  uploaded by junky (license 177) Tested by: pabelanger

	* channels/chan_sip.c: Fix logging message for stale nonce. (closes
	  issue #17582) Reported by: kenner Patches: chan_sip.c.diff
	  uploaded by kenner (license 1040) Tested by: lmadsen

2010-07-09 18:23 +0000 [r275027-275182]  Matthew Nicholson <mnicholson@digium.com>

	* main/loader.c: give a better error message when attempting to
	  unload a module that is not loaded

	* main/loader.c: don't unload modules that returned
	  AST_MODULE_LOAD_DECLINE when they were loaded

	* apps/app_dial.c: Clear the AST_CDR_FLAG_DIALED flag for channels
	  going into the pbx via the G option in app_dial (closes issue
	  #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
	  uploaded by mnicholson (license 96) Tested by: jamicque,
	  mnicholson

2010-07-09 15:33 +0000 [r275021]  Russell Bryant <russell@digium.com>

	* include/asterisk/test.h, main/test.c: Document that a leading and
	  trailing slash is expected for test categories. Also, emit a
	  warning if a test is registered without one of these.

2010-07-07 18:12 +0000 [r274579]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Close the DAHDI FD on error when
	  processing chan_dahdi toneduration config parameter.

2010-07-07 06:13 +0000 [r274417]  Tilghman Lesher <tlesher@digium.com>

	* configs/say.conf.sample: Correct how 100, 200, 300, etc. is said.
	  Also add the crazy British numbers. (closes issue #16102)
	  Reported by: Delvar Patches: say.conf.fix.patch uploaded by
	  Delvar (license 908) (plus a few additional fixes and
	  simplifications by me)

2010-07-06 22:46 +0000 [r274283-274359]  Jeff Peeler <jpeeler@digium.com>

	* main/Makefile: Ensure file.o is built correctly. (related to
	  issue #15250)

	* configs/sip.conf.sample: Correct sip.conf.sample comments for
	  prematuremedia option. (closes issue #17513) Reported by: festr
	  Patches: patch uploaded by festr (license 443)

2010-07-06 22:08 +0000 [r274280]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Add option to not
	  do a call forward on 482 Loop Detected Asterisk has always set up
	  a forwarded call when receiving a 482 Loop Detected. This
	  prevents handling the call failure by just continuing on in the
	  dialplan. Since this would be a change in behavior, the new
	  option to disable this behavior is forwardloopdetected which
	  defaults to 'yes'. Review:
	  https://reviewboard.asterisk.org/r/764/

2010-07-06 14:29 +0000 [r274157]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c: Fix problem with RFC 2833 DTMF not being accepted. A
	  recent check was added to ensure that we did not erroneously
	  detect duplicate DTMF when we received packets out of order. The
	  problem was that the check did not account for the fact that the
	  seqno of an RTP stream will roll over back to 0 after hitting
	  65535. Now, we have a secondary check that will ensure that the
	  seqno rolling over will not cause us to stop accepting DTMF.
	  (closes issue #17571) Reported by: mdeneen Patches:
	  rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
	  Tested by: richardf, maxochoa, JJCinAZ

2010-07-06 13:52 +0000 [r274093]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: Make get_member_status return QUEUE_NO_MEMBERS
	  instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work
	  again. This regression was introduced in 273639. Also fixed
	  whitespace.

2010-07-05 19:48 +0000 [r273981]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_oss.c, channels/chan_iax2.c: Command 'stop
	  gracefully' doesn't.

2010-07-05 13:51 +0000 [r273884]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/config.c: Remove extra line breaks from 'core show config
	  mappings' (closes issue #17583) Reported by: pabelanger Patches:
	  issue17583.patch uploaded by pabelanger (license 224) Tested by:
	  lmadsen

2010-07-02 21:36 +0000 [r273717-273793]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, channels/chan_local.c, configure,
	  include/asterisk/autoconfig.h.in, channels/chan_agent.c,
	  configure.ac, channels/chan_h323.c, include/asterisk/lock.h,
	  include/asterisk/compiler.h: Have the DEADLOCK_AVOIDANCE macro
	  warn when an unlock fails, to help catch potentially large
	  software bugs. (closes issue #17407) Reported by: pdf Patches:
	  20100527__issue17407.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/751/

	* main/autoservice.c: Autoservice loop optimization causes a busy
	  loop, when channels are serviced while in hangup. (closes issue
	  #17564) Reported by: ramonpeek Patches:
	  20100630__issue17564.diff.txt uploaded by tilghman (license 14)
	  Tested by: ramonpeek

2010-07-02 15:54 +0000 [r273640]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* apps/app_voicemail.c, channels/chan_dahdi.c,
	  channels/chan_misdn.c, channels/chan_sip.c, res/res_agi.c,
	  res/res_jabber.c: Fix various typos, reported by Lintian

2010-07-02 15:46 +0000 [r273639]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: If all members are paused, the wrong status is
	  indicated. (closes issue #17576) Reported by: ramonpeek Patches:
	  diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek

2010-07-01 22:09 +0000 [r273565]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't return a partially initialized datastore.
	  If memory allocation fails in ast_strdup(), don't return a
	  partially initialized datastore. Bad things may happen. (related
	  to ABE-2415)

2010-07-01 20:19 +0000 [r273354-273474]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: Allow admin user to join conference without
	  using admin mode and no user pin. Configuring the conference in
	  meetme.conf like the following: conf => 2345,,6666 did not prompt
	  for pin when used without admin mode. This meant that the
	  conference could not be joined as an admin even if the user knew
	  the correct pin. The original bug report was submitted claiming
	  that the blank user pin should deny entry into the conference. I
	  think a better way to handle this would be with a feature
	  enhancement that used the following syntax: conf => 2345,X,6666 -
	  where X denotes no acceptable pin allowed (closes issue #15704)
	  Reported by: modelnine

	* apps/app_meetme.c: Ensure channel placed in meetme in ringing
	  state is properly hung up. An outgoing channel placed in meetme
	  while still ringing which was then hung up would not exit meetme
	  and the channel was not properly destroyed. Specifically checking
	  for this scenario by looking at the appropriate control frames
	  resolves the issue. (closes issue #15871) Reported by: Ivan
	  Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
	  (license 229)

2010-06-29 23:15 +0000 [r273057-273060]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Allow the "useragent" value to be restored
	  into memory from the realtime backend. This value is purely
	  informational. It does not alter configuration at all. (closes
	  issue #16029) Reported by: Guggemand Patches:
	  realtime-useragent.patch uploaded by Guggemand (license 897)
	  Tested by: Guggemand

	* main/channel.c: _Really_ skip the channel... don't just retry for
	  another 200 cycles. (Closes issue SWP-1652, ABE-2240)

2010-06-29 21:36 +0000 [r273017]  Russell Bryant <russell@digium.com>

	* /: Remove properties that were erroneously merged to 1.4 from one
	  of my branches.

2010-07-22  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.34 Released.

2010-07-07  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.34-rc2 Released.

	* Fix problem with RFC 2833 DTMF not being accepted.
  
	  A recent check was added to ensure that we did not erroneously
	  detect duplicate DTMF when we received packets out of order.
	  The problem was that the check did not account for the fact that
	  the seqno of an RTP stream will roll over back to 0 after hitting
	  65535. Now, we have a secondary check that will ensure that the
	  seqno rolling over will not cause us to stop accepting DTMF.
  
	  (closes issue 0017571)
	  Reported by: mdeneen
	  Patches:
	        rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
	  Tested by: richardf, maxochoa, JJCinAZ

	* Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
	  via the G option in app_dial
  
	  (closes issue 0017592)
	  Reported by: jamicque
	  Patches:
	        G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
	  Tested by: jamicque, mnicholson

2010-06-29  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.34-rc1 Released.

2010-06-28 21:50 +0000 [r272921-272925]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Don't change ownership/group/permissions on run
	  directory, if it already exists. (closes issue #17076) Reported
	  by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
	  tilghman (license 14) Tested by: stuarth

	* main/config.c: Also trim trailing blanks on #includes

	* main/config.c: Change the way that we read include files, to
	  accommodate for changes in GCC 4.4. (closes issue #17472)
	  Reported by: seandarcy Patches: config2.patch uploaded by nivan
	  (license 1066) Tested by: nivan

2010-06-28 18:47 +0000 [r272878-272881]  Russell Bryant <russell@digium.com>

	* tests/test_astobj2.c (added): Backport applicable parts of
	  test_astobj2.

	* main/asterisk.c, Makefile, include/asterisk/test.h (added),
	  build_tools/cflags-devmode.xml, include/asterisk.h,
	  tests/Makefile, tests/test_skel.c, /, main/Makefile, tests
	  (added), include/asterisk/linkedlists.h, main/test.c (added):
	  Backport unit test API to 1.4. Review:
	  https://reviewboard.asterisk.org/r/750/

2010-06-28 17:31 +0000 [r272804]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Decode URI in contact header of 302
	  response. ABE-2352

2010-06-28 17:11 +0000 [r272688-272763]  Russell Bryant <russell@digium.com>

	* Makefile: Force SILENTMAKE where it is needed.

	* Makefile: Backport method of setting SUBMAKE from trunk. By
	  setting the PRINT_DIR variable, SUBMAKE will print out the
	  directories it descends into, which is important for editors
	  (like vim) that watch the build output so that they can take you
	  to the file where an error occurred.

2010-06-25 20:17 +0000 [r272562]  Tilghman Lesher <tlesher@digium.com>

	* doc/voicemail_odbc_postgresql.txt: Make the structure of the
	  table specified before match the queries and results. (closes
	  issue #17557) Reported by: cmaj

2010-06-24 21:58 +0000 [r272446]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: ss_thread calls pri_grab without lock
	  during overlap dial Recent changes to chan_dahdi with relation to
	  overlap dialing call pri_grab without first obtaining a lock.
	  (closes issue #17414) Reported by: pdf Patches: bug17414.patch
	  uploaded by jpeeler (license 325)

2010-06-23 22:33 +0000 [r272367]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: Send AgentComplete manager events in the event
	  of blind and attended transfers. (closes issue #16819) Reported
	  by: elbriga Patches: app_queue.diff uploaded by elbriga (license
	  482)

2010-06-23 20:57 +0000 [r272255]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_meetme.c: First caller into a dynamic conference now
	  enter pin once. If MeetMe is configured to use dynamic conference
	  numbers, then the first caller (which creates the conference) had
	  to enter the PIN number twice. (closes issue #15878) Reported by:
	  shawkris Patches: issue15878.patch uploaded by pabelanger
	  (license 224) Tested by: pabelanger

2010-06-23 18:40 +0000 [r272147]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Backport part of revision 136715 to fix
	  callerid in voicemail text files (IMAP only). (closes issue
	  #16945) Reported by: mneuhauser

2010-06-22 17:31 +0000 [r271689-271902]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Decrease the module ref count in sip_hangup
	  when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep
	  the ref count correct. (closes issue #16815) Reported by: rain
	  Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
	  (modified) Tested by: rain

	* pbx/pbx_dundi.c: Allow users to specify a port for dundi peers.
	  (closes issue #17056) Reported by: klaus3000 Patches:
	  dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
	  Tested by: klaus3000

	* configs/sip_notify.conf.sample, channels/chan_sip.c: Modify
	  chan_sip's packet generation api to automatically calculate the
	  Content-Length. This is done by storing packet content in a
	  buffer until it is actually time to send the packet, at which
	  time the size of the packet is calculated. This change was made
	  to ensure that the Content-Length is always correct. (closes
	  issue #17326) Reported by: kenner Tested by: mnicholson, kenner
	  Review: https://reviewboard.asterisk.org/r/693/

2010-06-21 20:37 +0000 [r271552]  Jeff Peeler <jpeeler@digium.com>

	* pbx/pbx_ael.c: Do not use sizeof to calculate size of a heap
	  allocated character array. Change left out from 271399. (closes
	  issue #16053) Reported by: diLLec

2010-06-18 20:52 +0000 [r271399-271444]  Jeff Peeler <jpeeler@digium.com>

	* pbx/pbx_ael.c: Check for newly added memory allocation failures
	  gracefully during AEL2 parsing.

	* pbx/pbx_ael.c: Fix crash when parsing some heavily nested
	  statements in AEL on reload. Due to the recursion used when
	  compiling AEL in gen_prios, all the stack space was being
	  consumed when parsing some AEL that contained nesting 13 levels
	  deep. Changing a few large buffers to be heap allocated fixed the
	  crash, although I did not test how many more levels can now be
	  safely used. (closes issue #16053) Reported by: diLLec Tested by:
	  jpeeler

2010-06-18 18:54 +0000 [r271339-271340]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Remove an unnecessary assignment that
	  causes a DEBUG_THREADS build failure on mac os x.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Fix a build problem on Mac OS X with
	  DEBUG_THREADS enabled. This set of changes was already in trunk.

2010-06-18 18:33 +0000 [r271335]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Eliminate deadlock potential in
	  dahdi_fixup(). (This is a backport of 269307, committed to trunk
	  by rmudgett.) Calling dahdi_indicate() when the channel private
	  lock is already held can cause a deadlock if the PRI lock is
	  needed because dahdi_indicate() will also get the channel private
	  lock. The pri_grab() function assumes that the channel private
	  lock is held once to avoid deadlock. (closes issue #17261)
	  Reported by: aragon

2010-06-22  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.33.1 Released.

	* channels/chan_dahdi.c: Merge revision 270404 from the 1.4 branch.

	  fixes FXS port still ringing when answered, as reported by Tzafrir
	  on dev-list.

	  (issue #17067)
	  Reported by: tzafrir
	  Tested by: alecdavis

2010-06-17  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.33 Released.

2010-06-10  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.33-rc2 Released.

2010-06-10  Tilghman Lesher <tlesher@digium.com>

	* Ensure signals are not blocked inside other signal handlers.

	  This eliminates the annoying <beep> on the console.

	  (closes issue 0017477)
	   Reported by: jvandal
	   Patches:
	         20100610__issue17477.diff.txt uploaded by tilghman (license 14)

2010-06-09  Paul Belanger <paul.belanger@polybeacon.com>

	* Fix Debian init script to not use -c.

	  When using the init script as-is currently, it could cause issues on Debian
	  such as high CPU usage. This fix has worked for several people so I'm
	  implementing the change. We now handle color displays properly.

	  (closes issue 0016784)
	  Reported by: pabelanger
	  Patches:
	        20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
	  Tested by: pabelanger, tilghman

2010-06-01  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.33-rc1 Released.

2010-06-01 15:17 +0000 [r266585]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Prevent CLI prompt from distorting output of
	  lines shorter than the prompt. Uses the VT100 method of clearing
	  the line from the cursor position to the end of the line: Esc-0K
	  (closes issue #17160) Reported by: coolmig Patches:
	  20100531__issue17160.diff.txt uploaded by tilghman (license 14)
	  Tested by: coolmig

2010-06-01 14:57 +0000 [r266579-266580]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Fix formatting issue with previous patch.

	* channels/chan_sip.c: Missing fallback to audio fax feature when
	  T.38 re-INVITE failed When a T.38 re-INVITE failed with an 488 or
	  606 answer, we should fallback to audio fax by send a
	  re-re-INVITE without T.38. The function is backported from 1.6
	  asterisk. (closes issue #16795) Reported by: vrban (closes issue
	  #16692) Reported by: vrban Patches:
	  t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
	  Tested by: lmadsen, vrban, haggard
	  https://reviewboard.asterisk.org/r/514/

2010-05-30 04:43 +0000 [r266437]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk: Reverting patch and reopening
	  issue #16784, as patch breaks color display.

2010-05-26 21:11 +0000 [r266142]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, main/logger.c: Use sigaction for signals which
	  should persist past the initial trigger, not signal. If you call
	  signal() in a Solaris signal handler, instead of just resetting
	  the signal handler, it causes the signal to refire, because the
	  signal is not marked as handled prior to the signal handler being
	  called. This effectively causes Solaris to immediately exceed the
	  threadstack in recursive signal handlers and crash. (closes issue
	  #17000) Reported by: rmcgilvr Patches:
	  20100526__issue17000.diff.txt uploaded by tilghman (license 14)
	  Tested by: rmcgilvr

2010-05-26 20:33 +0000 [r266140]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c: add dahdi_func_write to zap_tech structure
	  This was supposed to be committed with r263292, the back-port of
	  teh DAHDI buffer policy dial string option

2010-05-26 18:21 +0000 [r266004]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make AgentComplete message more consistent. At
	  times, the "Member" field was not specified during the event.
	  It's there now. (closes issue #15638) Reported by: elbriga
	  Patches: patchAppQueueAgentComplete.diff uploaded by elbriga
	  (license 482)

2010-05-26 16:21 +0000 [r265910]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c: Not finding rows in the DB does not rise
	  to the level of a warning. (closes issue #17062) Reported by:
	  drookie Patches: 20100525__issue17062.diff.txt uploaded by
	  tilghman (license 14)

2010-05-25 17:11 +0000 [r265613]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c: fixes build issue with zaptel (closes
	  issue #17394) Reported by: aragon Patches: half_buffer_fix.diff
	  uploaded by dvossel (license 671) Tested by: aragon

2010-05-25 16:48 +0000 [r265610]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: Don't mark the cdr records of unanswered queue
	  calls with "NOANSWER". This restores the behavior prior to
	  r258670. (closes issue #17334) Reported by: jvandal Patches:
	  queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
	  by: aragon, jvandal

2010-05-25 13:33 +0000 [r265570]  Terry Wilson <twilson@digium.com>

	* include/asterisk/options.h, main/asterisk.c, Makefile,
	  doc/manager.txt, main/manager.c: Merged revisions 265320,265467
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24
	  May 2010) | 14 lines Add the FullyBooted AMI event It is possible
	  to connect to the manager interface before all Asterisk modules
	  are loaded. To ensure that an application does not send AMI
	  actions that might require a module that has not yet loaded, the
	  application can listen for the FullyBooted manager event. It will
	  be sent upon connection if all modules have been loaded, or as
	  soon as loading is complete. The event: Event: FullyBooted
	  Privilege: system,all Status: Fully Booted Review:
	  https://reviewboard.asterisk.org/r/639/ ........ r265467 |
	  twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
	  Merge the rest of the FullyBooted patch ........

2010-05-24 19:37 +0000 [r265365]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes segfault when using generic plc

2010-05-21 20:59 +0000 [r264996-265089]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/file.h, apps/app_queue.c: Don't hang up on a
	  queue caller if the file we attempt to play does not exist. This
	  also fixes a documentation mistake in file.h that made my
	  original attempt to correct this problem not work correctly.
	  (closes issue #17061) Reported by: RoadKill

	* include/asterisk/channel.h: Fix grammatical error in comment.

	* main/channel.c, main/autoservice.c, include/asterisk/channel.h:
	  Allow ast_safe_sleep to defer specific frames until after the
	  sleep has concluded. From reviewboard Background: A Digium
	  customer discovered a somewhat odd bug. The setup is that parties
	  A and B are bridged, and party A places party B on hold. While
	  party B is listening to hold music, he mashes a bunch of DTMF.
	  Party A takes party B off hold while this is happening, but party
	  B continues to hear hold music. I could reproduce this about 1 in
	  5 times. The issue: When DTMF features are enabled and a user
	  presses keys, the channel that the DTMF is streamed to is placed
	  in an ast_safe_sleep for 100 ms, the duration of the emulated
	  tone. If an AST_CONTROL_UNHOLD frame is read from the channel
	  during the sleep, the frame is dropped. Thus the unhold
	  indication is never made to the channel that was originally
	  placed on hold. The fix: Originally, I discussed with Kevin
	  possible ways of fixing the specific problem reported. However,
	  we determined that the same type of problem could happen in other
	  situations where ast_safe_sleep() is used. Using autoservice as a
	  model, I modified ast_safe_sleep_conditional() to defer specific
	  frame types so they can be re-queued once the sleep has finished.
	  I made a common function for determining if a frame should be
	  deferred so that there are not two identical switch blocks to
	  maintain. Review: https://reviewboard.asterisk.org/r/674/

2010-05-20 23:23 +0000 [r264820]  Richard Mudgett <rmudgett@digium.com>

	* main/callerid.c: ast_callerid_parse() had a path that left name
	  uninitialized. Several callers of ast_callerid_parse() do not
	  initialize the name parameter before calling thus there is the
	  potential to use an uninitialized pointer.

2010-05-20 15:59 +0000 [r264541]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/options.h, main/loader.c, main/channel.c,
	  include/asterisk/channel.h: 1.4 version of PLC fix. Analogous to
	  trunk revision 264452, but without the change to chan_sip since
	  it is not necessary in this branch.

2010-05-19 20:01 +0000 [r264334]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_speech_utils.c: Set quieted flag when receiving a dtmf
	  tone during playback in speechbackground. (closes issue #16966)
	  Reported by: asackheim

2010-05-19 17:41 +0000 [r264248]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/options.h, configure, configure.ac: Internal
	  timing is now on by default, if you're using DAHDI 2.3 or above.
	  The reason for ensuring DAHDI 2.3 or above is that this version
	  ensures that a timer is always available, whereas in previous
	  versions, it was possible for DAHDI to be loaded, but have no
	  drivers to actually generate timing. If internal_timing was
	  turned on in this circumstance, a complete lack of audio would
	  result. This is the reason why internal_timing was not on by
	  default. However, now that DAHDI ensures the availability of a
	  timer, there is no reason for this setting to be off (and in
	  fact, it solves a great many initial user problems). (closes
	  issue #15932) Reported by: dimas Patches:
	  20100519__issue15932.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman

2010-05-19 08:23 +0000 [r264056]  Alec L Davis <sivad.a@paradise.net.nz>

	* configs/indications.conf.sample: fix incorrectly typed
	  indications for [nz] stutter and dialrecall (closes issue #17359)
	  Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
	  alecdavis (license 585)

2010-05-19 06:32 +0000 [r263949]  Tilghman Lesher <tlesher@digium.com>

	* main/dsp.c: Because progress is called multiple times, across
	  several frames, we must persist states when detecting multitone
	  sequences. (closes issue #16749) Reported by: dant Patches:
	  dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
	  dant

2010-05-18 18:54 +0000 [r263769]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_directory.c: Modify directory name reading to be
	  interrupted with operator or pound escape. In the case of
	  accidentally entering the wrong first three letters for the
	  reading, users could be very frustrated if the name listing is
	  very long. This allows interrupting the reading by pressing 0 or
	  #. 0 will attempt to execute a configured operator (o) extension
	  and # will exit and proceed in the dialplan. ABE-2200

2010-05-17 22:00 +0000 [r263637-263639]  Mark Michelson <mmichelson@digium.com>

	* main/devicestate.c: Fix logic error when checking for a devstate
	  provider. When using strsep, if one of the list of specified
	  separators is not found, it is the first parameter to strsep
	  which is now NULL, not the pointer returned by strsep. This issue
	  isn't especially severe in that the worst it is likely to do is
	  waste some cycles when a device with no '/' and no ':' is passed
	  to ast_device_state.

	* main/pbx.c: Remove arbitrary size limitation for hints. (closes
	  issue #17257) Reported by: tim_ringenbach Patches:
	  hints_crash_fix.diff uploaded by tim ringenbach (license 540)

2010-05-17 14:35 +0000 [r263374-263456]  Leif Madsen <lmadsen@digium.com>

	* main/http.c: Manager cookies are not compatible with RFC2109. The
	  Version field in the cookies we're setting contain quotes around
	  the version number which is not compatible with RFC2109 and
	  breaks some implementations. (closes issue #17231) Reported by:
	  ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
	  ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
	  ecarruda (license 559) Tested by: ecarruda, russell

	* sounds/Makefile: Update link to new version of core sounds. The
	  latest version of the core sounds files 1.4.19 now includes the
	  missing queue-minute sound file which is called by app_queue but
	  which has been missing. (closes issue #17123) Reported by:
	  n8ideas

2010-05-17 13:01 +0000 [r263292]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c: backport of DAHDI buffer policy dial
	  string option

2010-05-13 23:08 +0000 [r263112]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, main/file.c: Fix internal timing not working with
	  Zaptel dahdi_compat.h was not being included in channel.c when
	  used with Zaptel and wasn't in file.c at all. (closes issue
	  #15250) Reported by: mneuhauser Patches: dahdi_compat.patch
	  uploaded by mneuhauser (license 425) Tested by: IgorG

2010-05-12 17:00 +0000 [r262662]  David Vossel <dvossel@digium.com>

	* apps/app_meetme.c: fixes app_meetme dsp error We attempted to
	  detect silence after translating a frame from signed linear. This
	  caused a flooding of errors. To resolve this the code to detect
	  silence was moved before the translation. (closes issue #17133)
	  Reported by: jsdyer

2010-05-11 19:55 +0000 [r262421]  Jason Parker <jparker@digium.com>

	* pbx/Makefile: Use a less silly method for modifying a
	  flex-generated file. The sed syntax that was used wasn't actually
	  valid, causing some versions to choke. This is the method that is
	  used in 1.6.x+ for similar changes. (closes issue #16696)
	  Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell
	  (license 4) Tested by: qwell

2010-05-11 17:22 +0000 [r262321]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, Makefile.rules: Fix issue #17302 a slightly
	  different way (mad props to Qwell)

2010-05-10 16:34 +0000 [r262151]  Tilghman Lesher <tlesher@digium.com>

	* Makefile.rules: Allow compilation on Mac OS X 10.4 (Tiger)
	  (closes issue #17297) Reported by: jcovert Patches:
	  20100506__issue17297.diff.txt uploaded by tilghman (license 14)
	  (closes issue #17302) Reported by: jcovert

2010-05-06 20:10 +0000 [r261698-261735]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Only allow the operator key to be accepted
	  after leaving a voicemail. Or rather disallow the operator key
	  from being accepted when not offered, such as after finishing a
	  recording from within the mailbox options menu. ABE-2121 SWP-1267

	* apps/app_voicemail.c: Revert 261698, code in trunk leads me to
	  believe unadvertised options are supported.

	* apps/app_voicemail.c: Remove some hidden broken code in the
	  voicemail mailbox options menu. After finishing a recording from
	  within the mailbox options menu, pressing 0 exhibited strange
	  behavior with operator=yes turned on. Pressing 0 was not even
	  advertised as an option and the options from the vm-saveoper
	  prompt: "Press 1 to accept this recording. Otherwise, please
	  continue to hold" did not function correctly. While this of
	  course could be fixed, it didn't really seem to make sense even
	  if it was working properly. ABE-2121 SWP-1267

2010-05-06 16:56 +0000 [r261608]  Jason Parker <jparker@digium.com>

	* sounds/Makefile: Use the versioned MOH tarballs, now that we have
	  them. This makes for more reproducibility. Prompted by a
	  discussion in #asterisk-dev

2010-06-01  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.32 Released

2010-05-26  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.32-rc2 Released

2010-05-26 10:56 -0500 [r265891]  Matt Nicholson <mnicholson@digium.com>

	* Merged r265610 from 1.4:

	  Don't mark the cdr records of unanswered queue calls with "NOANSWER".
	  This restores the behavior prior to r258670.

	  (closes issue #17334)
	  Reported by: jvandal
	  Patches:
	        queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
	  Tested by: aragon, jvandal

2010-05-06  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.32-rc1 Released

2010-05-05 16:42 +0000 [r261274]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Registration fix for SIP realtime. Make sure
	  realtime fields are not empty. (closes issue #17266) Reported by:
	  Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
	  Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
	  https://reviewboard.asterisk.org/r/643/

2010-05-04 23:47 +0000 [r261093-261094]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Add a tiny corner case to the previous commit

	* main/channel.c: Protect against overflow, when calculating how
	  long to wait for a frame. (closes issue #17128) Reported by:
	  under Patches: d.diff uploaded by under (license 914)

2010-05-04 18:46 +0000 [r260923]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Voicemail transfer to operator should occur
	  immediately, not after main menu. There were two scenarios in the
	  advanced options that while using the operator=yes and review=yes
	  options, the transfer occurred only after exiting the main menu
	  (after sending a reply or leaving a message for an extension).
	  Now after the audio is processed for the reply or message the
	  transfer occurs immediately as expected. ABE-2107 ABE-2108

2010-05-04 17:40 +0000 [r260887]  tringenbach <tringenbach@localhost>:

	* README-SERIOUSLY.bestpractices.txt: Fix FILTER() examples to work
	  in 1.4 Review: https://reviewboard.asterisk.org/r/644/

2010-05-04 15:49 +0000 [r260801]  Jason Parker <jparker@digium.com>

	* build_tools/make_build_h: Fix fallout from removing from
	  configure script. Pointed out by philipp64 on #asterisk-dev

2010-05-03 16:54 +0000 [r260661-260662]  Paul Belanger <paul.belanger@polybeacon.com>

	* Makefile: Should have removed /usr/lib/ part. Thanks Qwell.

	* Makefile: non-root make install PREFIX=/tmp fails. Prepend libdir
	  when executing mkpkgconfig allowing non-root installs to work.
	  (closes issue #17268) Reported by: pabelanger Patches:
	  issue17268.patch uploaded by pabelanger (license 224) Tested by:
	  pabelanger

2010-05-03 14:57 +0000 [r260569]  Leif Madsen <lmadsen@digium.com>

	* doc/HOWTO_collect_debug_information.txt: Minor typo pointed out
	  by pabelanger on IRC.

2010-04-30 22:22 +0000 [r260434]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Ensure channel state is not incorrectly
	  set in the case of a very early answer. The needringing bit was
	  being read in dahdi_read after answering thereby setting the
	  state to ringing from up. This clears needringing upon answering
	  so that is no longer possible. (closes issue #17067) Reported by:
	  tzafrir Patches: needringing.diff uploaded by tzafrir (license
	  46)

2010-04-30 20:08 +0000 [r260345]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c: Fix potential crash from race condition
	  due to accessing channel data without the channel locked. In
	  res_musiconhold.c, there are several places where a channel's
	  stream's existence is checked prior to calling ast_closestream on
	  it. The issue here is that in several cases, the channel was not
	  locked while checking the stream. The result was that if two
	  threads checked the state of the channel's stream at
	  approximately the same time, then there could be a situation
	  where both threads attempt to call ast_closestream on the
	  channel's stream. The result here is that the refcount for the
	  stream would go below 0, resulting in a crash. I have added
	  proper channel locking to res_musiconhold.c to ensure that we do
	  not try to check chan->stream without the channel locked. A
	  Digium customer has been using this patch for several weeks and
	  has not had any crashes since applying the patch. ABE-2147

2010-04-29 22:11 +0000 [r260195]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: DTMF CallerID detection problems. The code
	  handling DTMF CallerID drops digits on long CallerID numbers and
	  may timeout waiting for the first ring with shorter numbers. The
	  DTMF emulation mode was not turned off when processing DTMF
	  CallerID. When the emulation code gets behind in processing the
	  DTMF digits it can skip a digit. For shorter numbers, the timeout
	  may have been too short. I increased it from 2 seconds to 4
	  seconds. Four seconds is a typical time between rings for many
	  countries. (closes issue #16460) Reported by: sum Patches:
	  issue16460.patch uploaded by rmudgett (license 664)
	  issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested
	  by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/
	  JIRA SWP-562 JIRA AST-334 JIRA SWP-901

2010-04-29 15:31 +0000 [r259858-260049]  David Vossel <dvossel@digium.com>

	* include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
	  audiohook_write_list The middle_frame in the audiohook_write_list
	  function was being freed if a audiohook manipulator returned a
	  failure. This is incorrect logic. This patch resolves this and
	  adds detailed descriptions of how this function should work and
	  why manipulator failures must be ignored. (closes issue #17052)
	  Reported by: dvossel Tested by: dvossel (closes issue #16196)
	  Reported by: atis Review: https://reviewboard.asterisk.org/r/623/

	* main/channel.c, channels/chan_local.c: resolves deadlocks in
	  chan_local Issue_1. In the local_hangup() 3 locks must be held at
	  the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock
	  avoidance is done when the channel to hangup is the outbound
	  chan_local channel, but when it is not the outbound channel we
	  have an issue... We attempt to do deadlock avoidance only on the
	  tech pvt, when both the tech pvt and the pvt->owner are locked
	  coming into that loop. By never giving up the pvt->owner channel
	  deadlock avoidance is not entirely possible. This patch resolves
	  that by doing deadlock avoidance on both the pvt->owner and the
	  pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is
	  used in ast_activate_generator() to queue a frame on the channel
	  and make the channel's read function get called. This function is
	  used in ast_activate_generator() while the channel is locked,
	  which mean's the channel will have a lock both from the generator
	  code and the frame_queue code by the time it gets to
	  chan_local.c's local_queue_frame code... local_queue_frame
	  contains some of the same crazy deadlock avoidance that
	  local_hangup requires, and this recursive lock prevents that
	  deadlock avoidance from happening correctly. This patch removes
	  ast_prod() from the channel lock so only one lock is held during
	  the local_queue_frame function. (closes issue #17185) Reported
	  by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
	  (license 671) issue_17185_v2.diff uploaded by dvossel (license
	  671) Tested by: schmoozecom, GameGamer43 Review:
	  https://reviewboard.asterisk.org/r/631/

2010-04-28 21:07 +0000 [r259852]  Leif Madsen <lmadsen@digium.com>

	* config.guess: Update config.guess. Updating config.guess because
	  after installing Ubuntu Server 9.10 and running all the update
	  scripts, running ./configure would not continue because it was
	  unable to determine what kind of system I had. After updating
	  config.guess things started working again.

2010-04-28 20:30 +0000 [r259748-259847]  Jason Parker <jparker@digium.com>

	* configure, configure.ac: Add AC_CONFIG_AUX_DIR to configure
	  script, so systems without install can use install-sh from our
	  source dir.

	* makeopts.in: Missed this when removing $ID

	* Makefile, configure, configure.ac: Remove usage of `id` since it
	  isn't useful and was causing breakge. Solaris `id` doesn't
	  support the -u argument. Instead of figuring out how to fix this
	  to work on Solaris, I decided to check why it was necessary and
	  where else it was used. It was only used in one place, and it
	  hasn't been needed for a very long time (I question whether it
	  was ever needed).

2010-04-28 17:13 +0000 [r259664]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Do not play goodbye prompt after timeout of
	  message review. ABE-2124

2010-04-27 21:53 +0000 [r259531]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: DAHDI "WARNING" message is confusing and
	  vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
	  failed: Success" Changed the warning to "Failed to decode
	  CallerID on channel 'name'". The message before it is likely more
	  specific about why the CallerID decode failed. SWP-501 AST-283

2010-04-27 21:48 +0000 [r259526]  Leif Madsen <lmadsen@digium.com>

	* sounds/Makefile: Update sounds files. * Add additional sounds
	  prompts for say_enumeration * Update the English conference
	  sounds prompts so they are better quality and all sound more
	  consistent * Clean up the core-sounds-XX.txt and
	  extra-sounds-XX.txt files to include all present sound files Both
	  core (en, fr, es) and extra (en, fr) sounds files have been
	  updated. (closes issue #16200) Reported by: murf (closes issue
	  #17137) Reported by: lmadsen

2010-04-27 21:15 +0000 [r259352-259441]  Jason Parker <jparker@digium.com>

	* main/editline/configure, main/editline/configure.in: Add gar to
	  the check for AR for those silly OSes (Solaris) that don't have
	  ar.

	* configure, configure.ac: Support the silly OSes that don't have
	  ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when
	  path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
	  switch to AC_CHECK_TOOLS.

2010-04-27 18:14 +0000 [r259270]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
	  hidecalleridname parameter in chan_dahdi.conf Issue #7321
	  implements a new chan_dahdi configuration option. However, a
	  change mentioned in the issue was never implemented. This is the
	  change that will allow the feature to work. I added a note to
	  chan_dahdi.conf.sample about the feature. (closes issue #17143)
	  Reported by: djensen99 Patches: diff.txt uploaded by djensen99
	  (license NA) (One line change) Tested by: djensen99

2010-04-26 21:44 +0000 [r259018-259104]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Let compilation succeed warning-free when
	  DONT_OPTIMIZE is turned off.

	* main/channel.c: Prevent Newchannel manager events for dummy
	  channels. No Newchannel manager event will be fired for channels
	  that are allocated to not match a registered technology type.
	  Thus bogus channels allocated solely for variable substitution or
	  CDR operations do not result in a Newchannel event. (closes issue
	  #16957) Reported by: atis Review:
	  https://reviewboard.asterisk.org/r/601

2010-04-25 18:09 +0000 [r258775]  Tilghman Lesher <tlesher@digium.com>

	* res/res_monitor.c: When StopMonitor is called, ensure that it
	  will not be restarted by a channel event. (closes issue #16590)
	  Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff
	  uploaded by kkm (license 888)

2010-04-22 21:49 +0000 [r258670]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, main/channel.c, res/res_features.c: Fix broken CDR
	  behavior. This change allows a CDR record previously marked with
	  disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally
	  this change partially reverts r235635 and does not set the
	  AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call().
	  To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is
	  now cleared from all brige CDRs in ast_bridge_call(). (closes
	  issue #16797) Reported by: VarnishedOtter Tested by: mnicholson

2010-04-21 21:45 +0000 [r258432]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Fix looping forever when no input received
	  in certain voicemail menu scenarios. Specifically, prompting for
	  an extension (when leaving or forwarding a message) or when
	  prompting for a digit (when saving a message or changing
	  folders). ABE-2122 SWP-1268

2010-04-20 16:16 +0000 [r257856-258029]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Play correct prompt when voicemail store
	  failure occurs after attempted forward. If a user's mailbox was
	  full and a message was attempted to be forwarded to said box,
	  warnings on the console would indicate failure. However, the
	  played prompt was that of success (vm-msgsaved). Now storage
	  failure is taken into account and the correct prompt
	  (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262

	* apps/app_voicemail.c: make app_voicemail compile with
	  IMAP_STORAGE

2010-04-16 21:15 +0000 [r257686]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* apps/app_mixmonitor.c: Make the mixmonitor thread process audio
	  frames faster Mantis issue 17078 reports MixMonitor recordings
	  have shorter durations than the call duration. This was because
	  the mixmonitor thread was not processing frames from the
	  audiohook fast enough. The mixmonitor thread would slowly fall
	  behind the most recent audio frame and when the channel hangs up,
	  the mixmonitor thread would exit without processing the same
	  number of frames as the channel; leaving the mixmonitor recording
	  shorter than actual call duration. This revision fixes this issue
	  by moving the ast_audiohook_trigger_wait() and the subsequent
	  audiohook.status check into the block where the
	  ast_audiohook_read_frame() function returns NULL. (closes issue
	  #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
	  by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
	  https://reviewboard.asterisk.org/r/611/

2010-04-15 21:23 +0000 [r257467-257544]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, main/app.c: Allow application options
	  with arguments to contain parentheses, through a variety of
	  escaping techniques. Fixes SWP-1194 (ABE-2143). Review:
	  https://reviewboard.asterisk.org/r/604/

	* channels/chan_sip.c: Don't recreate peer, when responding to a
	  repeated deregistration attempt. When a reply to a deregistration
	  is lost in transmit, the client retries the deregistration.
	  Previously, this would cause a realtime/autocreate peer to be
	  loaded back into memory, after it had already been correctly
	  purged. Instead, we just want to resend the reply without loading
	  the peer. (closes issue #16908) Reported by: kkm Patches:
	  20100412__issue16908.diff.txt uploaded by tilghman (license 14)
	  Tested by: kkm

2010-04-15 19:40 +0000 [r257342-257426]  Leif Madsen <lmadsen@digium.com>

	* doc/backtrace.txt: Update backtrace.txt documentation. Update the
	  backtrace.txt documentation so it conforms to the same layout as
	  other documents we've been working on recently. Additionally, add
	  a bunch of new information about gathering backtraces for crashes
	  and deadlocks, along with ways of verifying your file before
	  uploading it. Create a couple of one line commands for people to
	  generate the files we need. (closes issue #17190) Reported by:
	  lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
	  (license 10) Tested by: lmadsen, pabelanger

	* doc/backtrace.txt: Update address of the bug tracker.

2010-04-14 23:08 +0000 [r257266]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: When forwarding a message, ensure that
	  prepending works correctly. This is a regression in 1.4, only.
	  (closes issue #17103) Reported by: mglazer Patches:
	  20100408__issue17103.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman

2010-04-13 16:46 +0000 [r257070]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c, configs/manager.conf.sample: Add an option to
	  restore past broken behavor of the Events manager action Before
	  r238915, certain values for the EventMask parameter of the Events
	  action would result in no response being returned. This patch
	  adds an option to restore that broken behavior. Also while fixing
	  this bug I discovered that passing an empty EventMasks parameter
	  would also result in no response being returned, this has been
	  fixed as well while being preserved when the broken behavior is
	  requested. (closes issue #17023) Reported by: nblasgen Review:
	  https://reviewboard.asterisk.org/r/602/

2010-04-12 17:29 +0000 [r256900]  Leif Madsen <lmadsen@digium.com>

	* doc/HOWTO_collect_debug_information.txt (added): Add How-To
	  document on collecting debugging info for issues.asterisk.org
	  Paul Belanger has been helping a lot with bug tracking recently
	  and created this document that we can now point to when
	  additional debugging information is required. This document will
	  help those filing issues to know how to get the information
	  required when filing their issues. This will make things easier
	  on the developers. Initial text and changes by pabelanger. Tweaks
	  and editing by myself. (closes issue #17159) Reported by:
	  pabelanger Patches: HOWTO_collect_debug_information.txt.patch
	  uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger,
	  lmadsen

2010-04-06 00:10 +0000 [r256225]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: DAHDI/PRI call to pri_channel_bridge() not
	  protected by PRI lock. SWP-1231 ABE-2163

2010-05-03  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.31 Released

2010-04-29  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.31-rc2 Released

2010-04-29 10:31 +0000 [r260049]  David Vossel <dvossel@digium.com>

	* include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
	  audiohook_write_list.  (closes issue 0017052) Reported by: dvossel
	  Tested by: dvossel. (closes issue 0016196) Reported by: atis.
	  Review: https://reviewboard.asterisk.org/r/623/

2010-04-28 10:31 +0000 [r259858]  David Vossel <dvossel@digium.com>

	* channels/chan_local.c, main/channel.c: Resolves deadlocks in
	  chan_local.  (closes issue 0017185) Reported by: schmoozecom
	  Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
	  issue_17185_v2.diff uploaded by dvossel (license 671) Tested
	  by: schmoozecom, GameGamer43
	  Review: https://reviewboard.asterisk.org/r/631/

2010-04-05  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.31-rc1 Released

2010-04-02 23:45 +0000 [r256009-256014]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c: Resolve a deadlock that occurs due to a
	  pointless call to ast_bridged_channel() (closes issue #16840)
	  Reported by: bzing2 Patches: patch.txt uploaded by bzing2
	  (license 902) issue_16840.rev1.diff uploaded by russell (license
	  2) Tested by: bzing2, russell

	* main/channel.c: Remove extremely verbose debug message.

2010-03-31 19:09 +0000 [r255591]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Ensure line terminators in email are
	  consistent. Fixes an issue with certain Mail Transport Agents,
	  where attachments are not interpreted correctly. (closes issue
	  #16557) Reported by: jcovert Patches:
	  20100308__issue16557__1.4.diff.txt uploaded by tilghman (license
	  14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman
	  (license 14) 20100308__issue16557__trunk.diff.txt uploaded by
	  tilghman (license 14) Tested by: ebroad, zktech Reviewboard:
	  https://reviewboard.asterisk.org/r/544/

2010-03-31 17:42 +0000 [r255503]  Leif Madsen <lmadsen@digium.com>

	* apps/app_dial.c, configs/sip.conf.sample: Add documentation
	  clarifying when 't' and 'T' can be used. (closes issue #17021)
	  Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad

2010-03-30 20:56 +0000 [r255322-255409]  Russell Bryant <russell@digium.com>

	* channels/chan_h323.c: Don't kill Asterisk if the H323 listener
	  does not start.

	* pbx/pbx_dundi.c: Don't make Asterisk not start if pbx_dundi fails
	  to initialize.

2010-03-25 20:41 +0000 [r254714-254800]  Jason Parker <jparker@digium.com>

	* utils/Makefile: Don't remove local copies of utils in uninstall.

	* main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
	  issue with out-of-tree modules. Take 2, without ABI breakage this
	  time. Review: https://reviewboard.asterisk.org/r/588/

2010-03-25 18:51 +0000 [r254639]  Russell Bryant <russell@digium.com>

	* Makefile, /: Update Asterisk 1.4 to use menuselect trunk. Review:
	  https://reviewboard.asterisk.org/r/590/

2010-03-25 17:33 +0000 [r254452-254552]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/acl.h: Add doxygen for acl.h Review:
	  https://reviewboard.asterisk.org/r/528

	* main/rtp.c: Several fixes regarding RFC2833 DTMF detection. Here
	  is a copy and paste of the details from my request on reviewboard
	  that dealt with these changes: Fix 1. The first change in place
	  is to fix Mantis issue 15811, which deals with a situation where
	  Asterisk will incorrectly interpret out of order RFC2833 frames
	  as duplicate DTMF digits. For instance, we would receive a
	  sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1
	  seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7:
	  DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we
	  received the frame with seqno 5, we would interpret this as a new
	  DTMF 1. With this patch, we will check the seqno of the incoming
	  digit and not process the frame if the seqno is lower than the
	  last recorded seqno. Note that we do not record the seqno of the
	  dropped DTMF frame for future processing. While the above
	  situation is what was designed to be fixed, the patch is written
	  in such a way that the following would also be fixed too: seqno
	  9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13:
	  DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2
	  (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this
	  second situation, the beginning of the DTMF 2 arrives before the
	  final end frame of the DTMF 1. With the patch, seqno 12 is no
	  processed and thus we properly interpret the DTMF. Fix 2. The
	  second change in place is to fix an issue like the following:
	  seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
	  lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
	  *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
	  code in place that was supposed to properly end the previously
	  unended DTMF 1. The problem was that the code was essentially a
	  no-op. The code would set up an end frame for the DTMF 1 but
	  would immediately overwrite the frame with the begin for DTMF 2.
	  I changed process_dtmf_rfc2833() so that instead of returning a
	  single frame, it is given as an output parameter a list of
	  frames. Each frame that needs to be returned is appended to this
	  list. Fix 3. The final change is a minor one where an
	  AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
	  DTMF or an RFC 3389 frame and no frame was returned, then we
	  would return &ast_null_frame. The problem is that earlier in the
	  function, we may have generated an AST_CONTROL_SRCCHANGE frame
	  and put it in the list of frames we wish to return. This frame
	  would be lost in such a case. The patch fixes this problem
	  Review: https://reviewboard.asterisk.org/r/558

2010-03-25 15:57 +0000 [r254451]  Terry Wilson <twilson@digium.com>

	* main/file.c: Handle new SRCCHANGE control message here too

2010-03-24 00:37 +0000 [r254235]  Jeff Peeler <jpeeler@digium.com>

	* res/res_monitor.c: Ensure that monitor recordings are written to
	  the correct location (again) This is an extension to 248860. As
	  such the dialplan test has been extended: ; non absolute path,
	  not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
	  exten => 5040, n, dial(sip/5001) ; absolute path, not combined
	  exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
	  5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
	  monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
	  combined: changemonitor from non absolute to no path (leaves
	  tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
	  exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
	  dial(sip/5001) ; combined: changemonitor from no path to non
	  absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
	  exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
	  wasn't possible before exten => 5044, n, dial(sip/5001) ; non
	  absolute path, combined exten => 5045, 1,
	  monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
	  dial(sip/5001) ; absolute path, combined exten => 5046, 1,
	  monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
	  dial(sip/5001) ; no path, combined exten => 5047, 1,
	  monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
	  combined: changemonitor from non absolute to absolute (leaves
	  tmp/jeff) exten => 5048, 1,
	  monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
	  changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
	  dial(sip/5001) ; combined: changemonitor from absolute to non
	  absolute (leaves /tmp/jeff) exten => 5049, 1,
	  monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
	  changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
	  dial(sip/5001) ; combined: changemonitor from no path to absolute
	  exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
	  changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
	  dial(sip/5001) ; combined: changemonitor from absolute to no path
	  (leaves /tmp/jeff) exten => 5051, 1,
	  monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
	  changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
	  not combined: changemonitor from non absolute to no path (leaves
	  tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
	  exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
	  dial(sip/5001) ; not combined: changemonitor from no path to non
	  absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
	  5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
	  dial(sip/5001) ; not combined: changemonitor from non absolute to
	  absolute (leaves tmp/jeff) exten => 5054, 1,
	  monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
	  changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
	  dial(sip/5001) ; not combined: changemonitor from absolute to non
	  absolute (leaves /tmp/jeff) exten => 5055, 1,
	  monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
	  changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
	  dial(sip/5001) ; not combined: changemonitor from no path to
	  absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
	  5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
	  n, dial(sip/5001) ; not combined: changemonitor from absolute to
	  no path (leaves /tmp/jeff) exten => 5057, 1,
	  monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
	  changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)

2010-03-23 22:45 +0000 [r254046-254161]  Jason Parker <jparker@digium.com>

	* main/astobj2.c, main/lock.c (removed), main/channel.c,
	  main/Makefile, include/asterisk/astobj2.h, UPGRADE.txt,
	  include/asterisk/lock.h: Revert revisions 254046 and 254098.

	* UPGRADE.txt: Add note about the out-of-tree module ABI changes.

	* main/astobj2.c, main/lock.c (added), main/channel.c,
	  main/Makefile, include/asterisk/astobj2.h,
	  include/asterisk/lock.h: Allow out-of-tree modules to load,
	  regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This
	  can be guaranteed by forcing the ABI to no longer change when
	  these compiler flags are set. An unfortunate side-effect to this
	  is that there is an ABI change here. However, there is some
	  mitigation. Existing modules *will* fail to load since they would
	  require functions that no longer exist. Review:
	  https://reviewboard.asterisk.org/r/508/

2010-03-22 19:50 +0000 [r253799]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_features.c: Unconditionally copy the caller's account
	  code to the called party. (related to issue #16331)

2010-03-21 14:26 +0000 [r253631-253670]  Russell Bryant <russell@digium.com>

	* main/Makefile: Fix final link on FreeBSD by adding the
	  PTHREAD_CFLAGS.

	* main/sched.c, Makefile, apps/app_dial.c, channels/chan_dahdi.c,
	  main/manager.c, res/res_features.c, main/http.c, main/utils.c,
	  pbx/pbx_dundi.c, apps/app_followme.c: Resolve a number of FreeBSD
	  build issues.

2010-03-18 17:57 +0000 [r253252-253349]  Leif Madsen <lmadsen@digium.com>

	* apps/app_userevent.c: Typo found while fixing issue #16961.

	* doc/localchannel.txt: Synchronize text in localchannels.txt and
	  localchannels.tex. (issue #16963)

	* doc/localchannel.txt: Update new Local channel documentation. The
	  original reporter, Kobaz, of an issue with a Local channel that
	  inspired the Local channel documentation provided some tweaks to
	  the documentation after testing what I had written. Hopefully
	  anything that was vague or unclear has been cleaned up by these
	  changes. (closes issue #16963) Reported by: kobaz Patches:
	  localchannel-2.txt uploaded by kobaz (license 834) Tested by:
	  kobaz, lmadsen

2010-03-17 16:25 +0000 [r253158]  Terry Wilson <twilson@digium.com>

	* main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_mgcp.c, channels/chan_sip.c,
	  include/asterisk/rtp.h: Revert API change in release branches
	  This re-renames ast_rtp_update_source to ast_rtp_new_source

2010-03-17 00:26 +0000 [r253018]  Leif Madsen <lmadsen@digium.com>

	* configs/say.conf.sample: Add french snipset to say.conf. Add the
	  french snipset to say.conf. (Closes issue #15799)

2010-03-16 20:52 +0000 [r252766-252928]  Russell Bryant <russell@digium.com>

	* Makefile.rules: Backport chan_sip build fix for Mac OSX 10.6 from
	  trunk.

	* codecs/gsm/Makefile: Use uname -s, as done in trunk.

	* codecs/gsm/Makefile: Apply codec_gsm Mac OS X 10.6 build fix that
	  is in trunk and 1.6.X.

	* utils/Makefile: Don't treat warnings as errors for muted. muted
	  supports OS X, but uses functions marked as deprecated in 10.6.
	  However, the functions are still supported, so just ignore the
	  warnings for now and allow the build to proceed.

2010-03-16 18:46 +0000 [r252761]  Leif Madsen <lmadsen@digium.com>

	* configs/extensions.ael.sample: Additional extensions.ael global
	  variable fixes. Fixing up a couple more overlapping global
	  variable namespaces shared with extensions.conf.sample. Also
	  noticed a few of the lines that were commented out didn't have
	  the closing semi-colon so I added that as well. (issue #17035)

2010-03-15 21:43 +0000 [r252617]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/org.asterisk.asterisk.plist: Uh, yeah. Umask. I'm
	  stupid.

2010-03-15 20:48 +0000 [r252531-252533]  Leif Madsen <lmadsen@digium.com>

	* configs/extensions.ael.sample: Update extensions.ael file to not
	  overlap extensions.conf. Updated the extensions.ael file so the
	  global variables don't overlap those that we have in
	  extensions.conf (sample files). This way unexpected things won't
	  happed hopefully if both pbx_ael and res_config are loaded.
	  (closes issue #17035) Reported by: pprindeville

	* configure, configs/extensions.ael.sample: Revert last commit that
	  had bad changed to configure.

	* configure, configs/extensions.ael.sample: Update extensions.ael
	  file to not overlap extensions.conf. Updated the extensions.ael
	  file so the global variables don't overlap those that we have in
	  extensions.conf (sample files). This way unexpected things won't
	  happed hopefully if both pbx_ael and res_config are loaded.
	  (closes issue #17035) Reported by: pprindeville

2010-03-15 01:39 +0000 [r252361-252366]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Typo

	* main/asterisk.c, Makefile,
	  contrib/init.d/org.asterisk.asterisk.plist (added): Launch
	  Asterisk on Mac OS X with launchd. Reviewboard:
	  https://reviewboard.asterisk.org/r/551/

2010-03-13 00:30 +0000 [r252175]  Terry Wilson <twilson@digium.com>

	* main/rtp.c, channels/chan_mgcp.c, main/channel.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  include/asterisk/rtp.h, channels/chan_h323.c,
	  configs/sip.conf.sample, include/asterisk/frame.h: Merged
	  revisions 252089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
	  twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
	  Only change the RTP ssrc when we see that it has changed This
	  change basically reverts the change reviewed in
	  https://reviewboard.asterisk.org/r/374/ and instead limits the
	  updating of the RTP synchronization source to only those times
	  when we detect that the other side of the conversation has
	  changed the ssrc. The problem is that SRCUPDATE control frames
	  are sent many times where we don't want a new ssrc, including
	  whenever Asterisk has to send DTMF in a normal bridge. This is
	  also not the first time that this mistake has been made. The
	  initial implementation of the ast_rtp_new_source function also
	  changed the ssrc--and then it was removed because of this same
	  issue. Then, we put it back in again to fix a different issue.
	  This patch attempts to only change the ssrc when we see that the
	  other side of the conversation has changed the ssrc. It also
	  renames some functions to make their purpose more clear. Review:
	  https://reviewboard.asterisk.org/r/540/ ........

2010-03-12 19:58 +0000 [r251986-251997]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Forward declaring dahdi_pri was already
	  done.

	* channels/chan_dahdi.c: Make chan_dahdi wakeup_sub() prototype not
	  conditional.

2010-03-11  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.30 released

2010-03-04  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.30-rc3 released

2010-03-03 21:28 +0000 [r250613]  Leif Madsen <lmadsen@digium.com>

	* doc/localchannel.txt: Update existing Local channel
	  documentation. A complete re-write of the Local channel
	  documentation has been performed, with the existing information
	  from localchannel.txt and localchannel.tex merged in. (issue
	  #16637) Reported by: kobaz Patches: localchannel.tex uploaded by
	  lmadsen (license 10) localchannel.txt uploaded by lmadsen
	  (license 10) Tested by: lmadsen, jsmith, mmichelson

2010-03-03 19:04 +0000 [r250480]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Make sure to clear red alarm after
	  polarity reversal. From the issue: The automatic overnight line
	  tests (or manual ones) used on UK (BT) lines causes a red alarm
	  on a dahdi / TDM400P connected channel. This is because the line
	  uses voltage tests (battery loss) and polarity reversal. The
	  polarity reversal causes chan_dahdi to initiate v23 CallerID
	  processing but during this the event DAHDI_EVENT_NOALARM is
	  ignored so that the alarm is never cleared. (closes issue #14163)
	  Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded
	  by jedi98 (license 653) Tested by: mattbrown, Chainsaw,
	  mikeeccleston

2010-03-03 18:02 +0000 [r250394]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes problem with duplicate TXREQ packets
	  When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
	  call store_by_transfercallno() to link the chan_iax2_pvt struct
	  into iax_transfercallno_pvts. If a duplicate TXREQ packet is
	  received for the same call, the pvt struct will be linked into
	  iax_transfercallno_pvts multiple times. This patch fixes this.
	  Thanks rain for debugging this and providing a patch! (closes
	  issue #16904) Reported by: rain Patches:
	  iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
	  by: rain, dvossel

2010-03-02 21:08 +0000 [r250041-250050]  Leif Madsen <lmadsen@digium.com>

	* doc/imapstorage.txt: Update IMAP documentation. Update the IMAP
	  documentation to make it clear that storing voicemails in the
	  same folder as a large number of emails could potentially cause
	  significant slow downs when writing or retrieving voicemails.
	  (closes issue #16704) Reported by: TimeHider Tested by: lmadsen,
	  TimeHider

	* configs/cdr.conf.sample: Update documentation to clarify purpose
	  of unanswered option. (closes issue #16267) Reported by: elsto
	  Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
	  10) Tested by: davidw, elsto

	* doc/configuration.txt: Update documentation to not imply we
	  support overriding options. (issue #16855) Reported by: davidw

2010-03-02 19:36 +0000 [r249845-249946]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_echo.c: revert ability to exit echo app caused a
	  regression, as only supported VOICE, not VIDEO etc. Left in small
	  formatting change. (issue #16880)

	* apps/app_echo.c: fixes ability to exit echo app when called from
	  a ISDN channel, null frames prevent '#' exit. Now only echo back
	  VOICE and DTMF frames (issue #16880) Reported by: alecdavis
	  Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis

2010-03-01 19:35 +0000 [r249671]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: Fix crash in app_voicemail related to
	  message counting. We were passing a 'struct inprocess **' and
	  treating it like a 'struct inprocess *' causing a segfault.
	  (closes issue #16921) Reported by: whardier Patches:
	  20100301_issue16921.patch uploaded by seanbright (license 71)
	  Tested by: whardier

2010-03-01 17:02 +0000 [r249536]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c: Modify queued frames from local channels
	  to not set the other side to up In this case, attended transfers
	  were broken due to ast_feature_request_and_dial detecting the
	  channel being set to up before the answer frame could be read and
	  therefore failing to mark the channel as ready. This fix is a
	  regression fix for 244785, which should continue to work properly
	  as well. (closes issue #16816) Reported by: jamhed Tested by:
	  jamhed, corruptor

2010-02-27 23:51 +0000 [r249365]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_dahdi.c: overlap receiving: automatically send CALL
	  PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
	  user has determined that sufficient call information has been
	  received the user shall stop T302 and send CALL PROCEEDING to the
	  network. Previously timeouts were possible if the dialplan took a
	  long time to issue any response back to the network. Verified
	  that our local TELCO also does the same. (issue #16789) Reported
	  by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt
	  uploaded by alecdavis (license 585) Tested by: alecdavis (closes
	  issue #16789)

2010-02-27 14:07 +0000 [r249234]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: add a reference to the now-published IAX2
	  RFC

2010-02-26 17:04 +0000 [r249100]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: For T.38 reINVITEs treat a 606 the same as a
	  488. (closes issue #16792) Reported by: vrban Patches:
	  t38_606.patch uploaded by vrban (license 756)

2010-02-25 21:22 +0000 [r248860]  Jeff Peeler <jpeeler@digium.com>

	* res/res_monitor.c: Ensure that monitor recordings are written to
	  the correct location (again) This is an extension to 248757. As
	  such the dialplan test has been extended: exten => 5040, 1,
	  monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
	  dial(sip/5001) exten => 5041, 1,
	  monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
	  dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
	  exten => 5042, n, dial(sip/5001) exten => 5043, 1,
	  monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
	  changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
	  exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
	  changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
	  design and emits a warning exten => 5044, n, dial(sip/5001)

2010-02-25 21:21 +0000 [r248859]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Some platforms clear /var/run at boot, which
	  makes connecting a remote console... difficult. Previously, we
	  only created the default /var/run/asterisk directory at install
	  time. While we could create it in the init script, that would not
	  work for those who start asterisk manually from the command line.
	  So the safest thing to do is to create it as part of the Asterisk
	  boot process. This also changes the ownership of the directory,
	  because the pid and ctl files are created after we setuid/setgid.
	  (closes issue #16802) Reported by: Brian Patches:
	  20100224__issue16802.diff.txt uploaded by tilghman (license 14)
	  Tested by: tzafrir

2010-02-25 18:06 +0000 [r248668-248757]  Jeff Peeler <jpeeler@digium.com>

	* res/res_monitor.c: Ensure that monitor recordings are written to
	  the correct location. Recordings should be placed in the monitor
	  directory when a non-absolute path is used. Exact dialplan used
	  for testing: exten => 5040, 1,
	  monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
	  dial(sip/5001) exten => 5041, 1,
	  monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
	  dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
	  exten => 5042, n, dial(sip/5001) ABE-2101

	* apps/app_voicemail.c: Make deletion of temporary greetings work
	  properly with IMAP_STORAGE This same patch was merged in 220833,
	  but was skipped in this branch erroneously. (closes issue #16170)
	  Reported by: francesco_r

2010-02-24 21:02 +0000 [r248582]  Tilghman Lesher <tlesher@digium.com>

	* main/logger.c: Remove color code sequences from verbose messages
	  that go to logfiles. (closes issue #16786) Reported by: dodo
	  Patches: logger2.patch uploaded by dodo (license 989) Tested by:
	  tilghman

2010-02-23 16:26 +0000 [r248396]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes invite with replaces deadlock (closes
	  issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
	  uploaded by dvossel (license 671) Tested by: pwalker, dvossel

2010-02-22 13:52 +0000 [r248268]  Olle Johansson <oej@edvina.net>

	* apps/app_meetme.c: Don't log to debug unless debug is turned on

2010-02-20 22:25 +0000 [r248106]  Olle Johansson <oej@edvina.net>

	* main/rtp.c: Make sure we support RTCP compound messages with zero
	  reports

2010-02-19 19:11 +0000 [r248012]  Tilghman Lesher <tlesher@digium.com>

	* main/loader.c, /: Backport crash fix from trunk to 1.4, whereby
	  'core show gracefully' could crash Asterisk. (closes issue
	  #16470) Reported by: kjotte

2010-02-19 17:18 +0000 [r247910]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Merged revision 247904 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
	  .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
	  19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
	  consistent with other channel technologies. The processing of
	  DTMF tones on the receiving side of an ISDN channel is
	  inconsistent with the way it is handled in other channels,
	  especially DAHDI analog. This causes DTMF tones sent from an ISDN
	  phone to be doubled at the connected party. We are using the
	  following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
	  Option one is necessary because the asterisk DSP DTMF detection
	  is better than mISDN's internal DSP. Not as many false positives.
	  Option two is necessary to transmit DTMF tones end to end when
	  mISDN channels are connected to SIP channels with out of band
	  DTMF for example. The symptom is that DTMF tones sent by an ISDN
	  phone are doubled on the way through asterisk when two mISDN
	  channels are connected with a Local channel in between or if it
	  is bridged to an analog channel. The doubling of DTMF tones is
	  because DTMF is passed inband to asterisk by the mISDN channel
	  and passed out of band once again after the release of the DTMF
	  tone. Passing it inband is wrong. Neither an analog channel nor
	  SIP channel passes DTMF inband if configured to inband DTMF.
	  Analog and SIP channels filter out the DTMF tones because they
	  use the voice frames returned by ast_dsp_process. But chan_misdn
	  passes the unfiltered input voice frames instead. To overcome one
	  aspect of the problem, the doubling of DTMF tones when two mISDN
	  channels are directly bridged, someone made an 'optimization',
	  where in that case the DTMF tone passed out-of-band to the peer
	  channel is not translated to an inband tone at the transmit side.
	  This optimization is bad because it does not work in general. For
	  example, analog channels or mISDN channels when bridged through
	  an intermediary local channel will generate DTMF tones from
	  out-of-band information. Also, of course, it must not be done
	  when there is no inband DTMF available. This patch fixes the
	  issue. Now chan_misdn will filter the received inband DTMF signal
	  the same as other channel types. Another change included: No need
	  to build an extra translation path because ast_process_dsp does
	  it if required. Patches: misdn-dtmf.patch JIRA ABE-2080

2010-02-18 19:38 +0000 [r247651]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_features.c: Copy the calling party's account code to the
	  called party if they don't already have one. (closes issue
	  #16331) Reported by: bluefox Tested by: mnicholson

2010-02-18 16:53 +0000 [r247502-247508]  Leif Madsen <lmadsen@digium.com>

	* README-SERIOUSLY.bestpractices.txt: Add additional link to best
	  practices document per jsmith.

	* README-SERIOUSLY.bestpractices.txt (added): Add best practices
	  documentation. (issue #16808) Reported by: lmadsen (issue #16810)
	  Reported by: Nick_Lewis Tested by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/507/

2010-02-18 04:19 +0000 [r247422]  Russell Bryant <russell@digium.com>

	* Makefile, sounds/Makefile: Tweak argument handling for wget in
	  the sounds Makefile. 1) Fix the check to see if we are using wget
	  to not be full of fail. The configure script populates this
	  variable with the absolute path to wget if it is found, so it
	  didn't work. 2) Allow some extra arguments to be passed in for
	  wget. This is just a simple change to allow our Bamboo build
	  script to tell wget to be quiet and not fill up our logs with
	  download status output.

2010-02-17 16:24 +0000 [r247168]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make sure that when autofill is disabled that
	  callers not in the front of the queue cannot place calls. (closes
	  issue #16834) Reported by: kebl0155 Patches:
	  app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356)

2010-02-15 23:42 +0000 [r246709]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Make the menuselect instructions correct by allowing
	  'make menuselect' to actually solve dependency problems.
	  (Previously, it would fail out again with the same message about
	  running 'make menuselect', which was NOT at all helpful.)

2010-02-12 23:30 +0000 [r246545]  David Vossel <dvossel@digium.com>

	* main/channel.c: lock channel during datastore removal On channel
	  destruction the channel's datastores are removed and destroyed.
	  Since there are public API calls to find and remove datastores on
	  a channel, a lock should be held whenever datastores are removed
	  and destroyed. This resolves a crash caused by a race condition
	  in app_chanspy.c. (closes issue #16678) Reported by:
	  tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
	  tim ringenbach (license 540) Tested by: dvossel

2010-02-12 18:52 +0000 [r246460]  Jason Parker <jparker@digium.com>

	* main/channel.c: Fix some silly formatting, and remove unnecessary
	  option_debug checks

2010-02-10 17:44 +0000 [r246115]  David Vossel <dvossel@digium.com>

	* apps/app_queue.c: fixes random deadlock in app_queue with
	  use_weight during reload (closes issue #16677) Reported by:
	  tim_ringenbach Patches: app_queue_use_weight_deadlock.diff
	  uploaded by tim ringenbach (license 540)

2010-02-10 13:37 +0000 [r245944]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample: Include examples of FILTER usage
	  in extension patterns where a "." may be a risk.

2010-02-10 08:24 +0000 [r245909]  Olle Johansson <oej@edvina.net>

	* res/res_smdi.c: Make sure that res_smdi loads regardless of
	  configuration, since chan_dahdi depends on res_smdi

2010-02-09 22:55 +0000 [r245792]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes iaxs and iaxsl size off by one issue.
	  2^15 = 32768 which is the maximum allowed iax2 callnumber.
	  Creating the iaxs and iaxsl array of size 32768 means the maximum
	  callnumber is actually out of bounds. This causes a nasty crash.
	  (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
	  uploaded by dvossel (license 671)

2010-02-08 20:39 +0000 [r245496]  Jason Parker <jparker@digium.com>

	* main/ast_expr2.fl, main/ast_expr2f.c: Remove reference of
	  documentation in source directory. People don't always build
	  Asterisk from source (distro packages, anybody?).

2010-02-08 11:57 +0000 [r245422]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Res_features depends on res_adsi in 1.4

2010-02-05 18:32 +0000 [r245044]  Kevin P. Fleming <kpfleming@digium.com>

	* contrib/firmware (removed), LICENSE: Remove contrib/firmware
	  directory as it is empty Remove explicit license for IAXy
	  firmware as it is no longer included in the tree

2010-02-05 17:03 +0000 [r244926]  Sean Bright <sean@malleable.com>

	* main/asterisk.c: Update main copyright date.

2010-02-04 23:20 +0000 [r244785]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c: Change channel state on local channels for
	  busy,answer,ring. Previously local channels channel state never
	  changed. This became problematic when the state of the other side
	  of the local channel was lost, for example during a masquerade.
	  Changing the state of the local channel allows for the scenario
	  to be detected when the channel state is set to ringing, but the
	  peer isn't ringing. The specific problem scenario is described in
	  164201. Although this was noted on one of the issues, here is the
	  tested dialplan verified to work: exten =>
	  9700,1,Dial(Local/*9700@default&Local/#9700@default) exten =>
	  *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
	  exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
	  *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did
	  not exten =>
	  #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
	  issue #14992) Reported by: davidw

2010-02-01 23:13 +0000 [r244070-244242]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Backup and restore original textfile, for
	  prosthesis (gerund of prepend). Also, fix menuselect such that
	  changing voicemail build options correctly causes rebuild.
	  (closes issue #16415) Reported by: tomo1657 Patches:
	  prepention.patch uploaded by tomo1657 (license 484) (with
	  modifications by me to backport to 1.4)

	* res/res_features.c: When a transferer hangs up during an attended
	  transfer BEFORE the transfer is answered, don't stop playing MOH.
	  (closes issue #16513) Reported by: litnimax Patches:
	  atxfer_moh_16513.patch uploaded by gknispel proformatique
	  (license 261) Tested by: litnimax

	* main/channel.c, channels/chan_local.c: Revert previous chan_local
	  fix (r236981) and fix instead by destroying expired frames in the
	  queue. (closes issue #16525) Reported by: kobaz Patches:
	  20100126__issue16525.diff.txt uploaded by tilghman (license 14)
	  20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
	  (license 14) Tested by: kobaz, atis (closes issue #16581)
	  Reported by: ZX81 (closes issue #16681) Reported by: alexr1

2010-01-28 18:48 +0000 [r243862-243863]  Leif Madsen <lmadsen@digium.com>

	* BUGS: Oops, correct wrong link (https vs. http) in previous
	  commit.

	* BUGS: Update location of bug tracker in documentation.

2010-01-28 15:03 +0000 [r243779]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a bogus third argument to
	  ast_copy_string().

2010-01-27 20:35 +0000 [r243570-243691]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_queue.c: Revert 243570, I should have looked at this
	  closer. Will reopen the issue, but am leaving the review closed
	  as the change was pointless. (issue #16488)

	* apps/app_queue.c: Extend announcement URL used with Queue from 80
	  chars to PATH_MAX. (closes issue #16488) Reported by: syspert
	  Patches: soundfilelen.pacth-2 uploaded by syspert (license 938)
	  Review: https://reviewboard.asterisk.org/r/475/

2010-01-27 18:06 +0000 [r243486]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Use a safe list traversal while checking for
	  duplicate vars in pbx_builtin_setvar_helper.

2010-01-26 23:55 +0000 [r243390]  David Vossel <dvossel@digium.com>

	* res/res_features.c: fixes bug with channel receiving wrong
	  privileges after call parking (closes issue #16429) Reported by:
	  Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro
	  Konishi (license 947) Tested by: dvossel

2010-01-26 18:19 +0000 [r243258]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Remove unnecessary code in ast_read as issue
	  16058 has been fully solved now.

2010-01-25 21:50 +0000 [r242852-242969]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, pbx/Makefile: Err, and use the new menuselect
	  define, too.

	* build_tools/cflags.xml, build_tools/menuselect-deps.in,
	  configure, configure.ac: Only rebuild parsers by an option in
	  menuselect

	* configure, main/Makefile, configure.ac, pbx/Makefile: Restore
	  FreeBSD to able-to-compile-ish-mode

2010-01-25 20:08 +0000 [r242850-242851]  Olle Johansson <oej@edvina.net>

	* main/manager.c: Remove debugging that indeed should have been
	  gone before commit. Sorry.

	* main/manager.c: Report error when writing to functions returns
	  error in AMI setvar action

2010-01-25 05:42 +0000 [r242520-242728]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, pbx/Makefile: Buildbot pointed out an error
	  (thanks, buildbot!)

	* main/Makefile, pbx/Makefile: Oops, should have used CMD_PREFIX,
	  not ECHO_PREFIX, for the commands.

	* main/Makefile: Make the build of the Asterisk expression parser
	  match that of the AEL parser.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  pbx/ael/ael_lex.c, pbx/Makefile, makeopts.in: Only rebuild bison
	  and flex source files on demand, if bison and flex are detected
	  by the configure script. Changed after discussion on the -dev
	  list about possible unnecessary build failures, due to
	  checkouts/untars causing these special source files to possibly
	  be newer than their resulting C files. This should additionally
	  ensure that nobody need learn about extra Makefile arguments to
	  ensure the proper files get rebuilt when changes are made to
	  these special source files.

2010-01-22 21:44 +0000 [r242423]  Tilghman Lesher <tlesher@digium.com>

	* pbx/Makefile: Rebuild from flex, bison sources when necessary.
	  (issue #14629) Reported by: Marquis Patches:
	  20100121__issue14629.diff.txt uploaded by tilghman (license 14)

2010-01-22 09:19 +0000 [r242226]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Initialize notify_types to NULL

2010-01-22 01:48 +0000 [r242142]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/cdr.c: Add Dialed Number Identifier (DNID) field to cdr.
	  Branch support, retains ABI, if backend CDR collector is adaptive
	  then database requires 'dnid' field to be added, otherwise no
	  functional changes. Reported by: alecdavis Tested by: alecdavis
	  Patch cdr_dnid.diff2.txt uploaded by alecdavis (license 585)
	  Review: https://reviewboard.asterisk.org/r/455/

2010-01-21 15:25 +0000 [r241932]  Sean Bright <sean@malleable.com>

	* configure, configure.ac: Fix configure check for
	  PTHREAD_ONCE_INIT when manually adding -Wall to CFLAGS. (closes
	  issue #16666) Reported by: romain_proformatique

2010-01-21 05:53 +0000 [r241765]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_math.c: Guard against division by zero.

2010-01-20 20:00 +0000 [r241626]  David Vossel <dvossel@digium.com>

	* Makefile: fixes parsing error in Makefile. Some echo lines were
	  missing "; . Thanks to jparker for pointing out the problem.

2010-01-20 14:12 +0000 [r241543-241544]  Sean Bright <sean@malleable.com>

	* pbx/pbx_spool.c: Modify fix for issue 16554 to be more inline
	  with what is already in trunk. I should have taken a closer look
	  at trunk/1.6.x, as this bug has already been fixed in a much more
	  simple manner, by just settings o->vars to NULL after the
	  ast_pbx_outgoing_* calls. (issue #16554) Reported by: mav3rick

	* pbx/pbx_spool.c: Fix a memory leak in pbx_spool when using SetVar
	  in a call file. In pbx_spool, when we are freeing our 'outgoing'
	  struct, we weren't deallocating the ast_variable list we had
	  built from SetVars in a call file. Adding a call to
	  ast_variables_destroy in our deallocation routine works, but only
	  if the variables have not already been passed into
	  ast_pbx_outgoing_app() or _exten(), both of which take care of
	  destroying the variable list for us. (closes issue #16554)
	  Reported by: mav3rick Patches: issue16554_20100119.patch uploaded
	  by seanbright (license 71) Tested by: mav3rick

2010-01-20 09:38 +0000 [r241458]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/pbx.c: Update CDR variables as pbx starts Allows CDR
	  variables added in cdr.c:set_one_cid to become visable during the
	  call, by executing ast_cdr_update() early in __ast_pbx_run. Based
	  on cdr_update.diff3.txt (issue #16638) Reported by: alecdavis
	  Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585)
	  Tested by: alecdavis

2010-01-19 17:41 +0000 [r241228]  Jason Parker <jparker@digium.com>

	* Makefile: Allow parallel make (-j) to work properly. 1.4 changes
	  are quite different from the others. (issue #16489) Reported by:
	  Chainsaw Tested by: qwell

2010-01-19 17:22 +0000 [r241227]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_agent.c: Fix deadlock in agent_read by removing
	  call to agent_logoff. One must always lock the agents list lock
	  before the agent private. agent_read locks the private
	  immediately, so locking the agents list lock is not an option
	  (which is what agent_logoff requires). Because agent_read already
	  has access to the agent private all that is necessary is to do
	  the required hanging up that agent_logoff performed. (closes
	  issue #16321) Reported by: valon24 Patches: bug16321.patch
	  uploaded by jpeeler (license 325)

2010-01-18 19:54 +0000 [r241015]  Sean Bright <sean@malleable.com>

	* main/config.c: Plug a memory leak when reading configs with their
	  comments. While reading through configuration files with the
	  intent of returning their full contents (comments specifically)
	  we allocated some memory and then forgot to free it. This doesn't
	  fix 16554 but clears up a leak I had in the lab. (issue #16554)
	  Reported by: mav3rick Patches: issue16554_20100118.patch uploaded
	  by seanbright (license 71) Tested by: seanbright

2010-01-18 16:51 +0000 [r240891]  David Vossel <dvossel@digium.com>

	* Makefile: updated transmit_silence option documentation in
	  asterisk.conf This patch updates the transmit_silence option to
	  better document why the option exists, and what it affects.
	  Thanks to russell for providing the verbage for this update.

2010-01-18 13:27 +0000 [r240768]  Olle Johansson <oej@edvina.net>

	* utils/Makefile: Fix muted compilation in 1.4 only

2010-01-15 23:06 +0000 [r240547]  Russell Bryant <russell@digium.com>

	* Makefile: Fix a spelling error in the asterisk.conf sample.

2010-01-15 20:52 +0000 [r240414]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Disallow leaving more than maxmsg
	  voicemails. This is a possibility because our previous method
	  assumed that no messages are left in parallel, which is not a
	  safe assumption. Due to the vmu structure duplication, it was
	  necessary to track in-process messages via a separate structure.
	  If at some point, we switch vmu to an ao2-reference-counted
	  structure, which would eliminate the prior noted duplication of
	  structures, then we could incorporate this new in-process
	  structure directly into vmu. (closes issue #16271) Reported by:
	  sohosys Patches: 20100108__issue16271.diff.txt uploaded by
	  tilghman (license 14) 20100108__issue16271__trunk.diff.txt
	  uploaded by tilghman (license 14)
	  20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
	  (license 14) Tested by: jsutton

2010-01-14  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.29

2010-01-08  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.29-rc1

2010-01-07 20:14 +0000 [r238409-238411]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes crash in "scheduled_destroy" in
	  chan_iax A signed short was used to represent a callnumber. This
	  is makes it possible to attempt to access the iaxs array with a
	  negative index. (closes issue #16565) Reported by: jensvb

	* channels/chan_sip.c: Change in sip show channels display format
	  allowing more digits for CID (closes issue 0016459) Reported by:
	  Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
	  (license 953)

2010-01-06 21:41 +0000 [r238230]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_cdr.c: Revise documentation on disposition values to
	  the actual values used. (closes issue #16289) Reported by:
	  wdoekes

2010-01-06 15:18 +0000 [r237697-238009]  Russell Bryant <russell@digium.com>

	* apps/app_mp3.c: Resolve a crash due to an ast_frame not being
	  fully initialized. (closes issue #16531) Reported by: john8675309
	  (closes SWP-615)

	* main/utils.c: Change a NOTICE log message to DEBUG where it
	  belongs. (closes issue #16479) Reported by: alexrecarey (closes
	  SWP-577)

2010-01-04 21:45 +0000 [r237318-237573]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Bounds checking for input string (closes issue
	  #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt
	  uploaded by tilghman (license 14)

	* main/pbx.c: Regression in issue #15421 - Pattern matching (closes
	  issue #16482) Reported by: wdoekes Patches:
	  astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
	  20091223__issue16482.diff.txt uploaded by tilghman (license 14)
	  Tested by: wdoekes, tilghman

	* main/pbx.c, res/res_agi.c, include/asterisk/channel.h: Add a flag
	  to disable the Background behavior, for AGI users. This is in a
	  section of code that relates to two other issues, namely issue
	  #14011 and issue #14940), one of which was the behavior of
	  Background when called with a context argument that matched the
	  current context. This fix broke FreePBX, however, in a post-Dial
	  situation. Needless to say, this is an extremely difficult
	  collision of several different issues. While the use of an
	  exception flag is ugly, fixing all of the issues linked is rather
	  difficult (although if someone would like to propose a better
	  solution, we're happy to entertain that suggestion). (closes
	  issue #16434) Reported by: rickead2000 Patches:
	  20091217__issue16434.diff.txt uploaded by tilghman (license 14)
	  20091222__issue16434__1.6.1.diff.txt uploaded by tilghman
	  (license 14) Tested by: rickead2000

	* channels/chan_local.c: It's also possible for the Local channel
	  to directly execute an Application. Reviewboard:
	  https://reviewboard.asterisk.org/r/452/

2010-01-02 09:52 +0000 [r237135]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Release memory of the contact acl before
	  unloading module

2009-12-30 21:57 +0000 [r236981]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Don't queue frames to channels that have
	  no means to process them. (closes issue #15609) Reported by:
	  aragon Patches:
	  20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
	  tilghman (license 14) Tested by: aragon Review:
	  https://reviewboard.asterisk.org/r/452/

2009-12-30 20:25 +0000 [r236890]  Jeff Peeler <jpeeler@digium.com>

	* utils/astman.c: Remove conflicting function definitions
	  (asterisk.h) so LOW_MEMORY compiles.

2009-12-28 15:12 +0000 [r236509-236585]  Sean Bright <sean@malleable.com>

	* include/asterisk/threadstorage.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Try a test
	  compile to see if PTHREAD_ONCE_INIT requires extra braces. There
	  was conditional code (based on build platform) to optioinally
	  wrap PTHREAD_ONCE_INIT in braces that was removed since it is
	  fixed in newer versions of Solaris/OpenSolaris, but I am still
	  running into it on Solaris 10 x86 so add a configure-time check
	  for it.

	* apps/app_meetme.c: Avoid a crash with large numbers of MeetMe
	  conferences. Similar to changes made to Queue(), when we have
	  large numbers of conferences in meetme.conf (1000s) and we use
	  alloca()/strdupa(), we can blow out the stack and crash, so
	  instead just use a single fixed buffer. (closes issue #16509)
	  Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
	  by seanbright (license 71) Tested by: seanbright

2009-12-27 18:19 +0000 [r236433]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk: Turn on colors in the daemon,
	  since there's many requests for it on Ubuntu.

2009-12-26 15:26 +0000 [r236357]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: update to latest releases with zero uid/gid

2009-12-23 15:21 +0000 [r236261]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Properly set T.38 attributes and don't
	  return before T.38 ports are configured when T.38 is found but no
	  audio stream is found. (closes issue #16318) Reported by:
	  bird_of_Luck Patches: t38-sdp-parsing-fix3.diff uploaded by
	  mnicholson (license 96), written by vrban and mnicholson Tested
	  by: vrban, mihaill

2009-12-23 02:55 +0000 [r236184]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: If EXEC only gets a single argument, don't crash
	  when the second is used. (closes issue #16504) Reported by:
	  bklang

2009-12-22 16:58 +0000 [r236062]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes issue with p->method incorrectly set
	  to ACK It is possible for a second ACK to come in for a
	  retransmitted message. If an ack does not match an unacked
	  message in our queue, restore the previous p->method as this ACK
	  is completely ignored. (closes issue #16295) Reported by:
	  omolenkamp Patches: issue16295_v2.diff uploaded by dvossel
	  (license 671)

2009-12-21 19:43 +0000 [r235940]  Jeff Peeler <jpeeler@digium.com>

	* res/res_monitor.c: Change Monitor to not assume file to write to
	  does not contain pathing. 227944 changed the fname_base argument
	  to always append the configured monitor path. This change was
	  necessary to properly compare files for uniqueness. If a full
	  path is given though, nothing needs to be appended and that is
	  handled correctly now. (closes issue #16377) (closes issue
	  #16376) Reported by: bcnit Patches:
	  res_monitor.c-issue16376-1.patch uploaded by dant (license 670)

2009-12-21 16:45 +0000 [r235821]  Tilghman Lesher <tlesher@digium.com>

	* res/res_features.c: Send parking lot announcement to the channel
	  which parked the call, not the park-ee. (closes issue #16234)
	  Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt
	  uploaded by tilghman (license 14)
	  20091221__issue16234__1.4.diff.txt uploaded by tilghman (license
	  14) Tested by: yeshuawatso

2009-12-18 22:39 +0000 [r235652]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Revise verbiage, per #asterisk-dev
	  discussion

2009-12-18 22:29 +0000 [r235635]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, include/asterisk/cdr.h: Correct CDR dispositions
	  for BUSY/FAILED This patch is simple in that it reorders the
	  disposition defines so that the fix for issue 12946 works
	  properly (the default CDR disposition was changed to
	  AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set
	  in ast_call to ensure all CDR records are written. The side
	  effects of CDR changes are scary, so I'm documenting the test
	  cases performed to attempt to catch any regressions. The
	  following tests were all performed using 1.4 rev 195881 vs head
	  (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A
	  (Both SIP and features) A calls B A blind transfers to C Hangup C
	  (Both SIP and features) A calls B A attended transfers to C
	  Hangup C A calls B A attended transfers to C (SIP) C blind
	  transfers to A (features) Hangup A All of the test scenario CDRs
	  matched. The following tests were performed just with the patch
	  to ensure proper operation (with unanswered=yes): exten
	  =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w)
	  exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue
	  #16180) Reported by: aatef Patches: bug16180.patch uploaded by
	  jpeeler (license 325)

2009-12-18 21:18 +0000 [r235572]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Point to the typical missing package,
	  not the cryptic "termcap support".

2009-12-17  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.28

2009-12-09  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.28-rc1

2009-12-09 19:58 +0000 [r233782-233879]  Russell Bryant <russell@digium.com>

	* main/loader.c: Fix breakage of the "module load <module>" CLI
	  command.

	* main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
	  include/asterisk/module.h, formats/format_pcm.c,
	  formats/format_h263.c, formats/format_g723.c,
	  formats/format_h264.c, formats/format_jpeg.c,
	  formats/format_g726.c, formats/format_gsm.c,
	  formats/format_g729.c, formats/format_sln.c,
	  formats/format_wav.c, formats/format_ogg_vorbis.c,
	  formats/format_wav_gsm.c: Set a module load priority for format
	  modules. A recent change to app_voicemail made it such that the
	  module now assumes that all format modules are available while
	  processing voicemail configuration. However, when autoloading
	  modules, it was possible that app_voicemail was loaded before the
	  format modules. Since format modules don't depend on anything,
	  set a module load priority on them to ensure that they get loaded
	  first when autoloading. This version of the patch is specific to
	  Asterisk 1.4 and 1.6.0. These versions did not already support
	  module load priority in the module API. This adds a trivial
	  version of this which is just a module flag to include it in a
	  pass before loading "everything". Thanks to mmichelson for the
	  review! (closes issue #16412) Reported by: jiddings Tested by:
	  russell Review: https://reviewboard.asterisk.org/r/445/

2009-12-08 00:02 +0000 [r233618]  Atis Lezdins <atis@iq-labs.net>

	* contrib/valgrind.supp: Merged revisions 233577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 |
	  atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix
	  compatibility with valgrind 3.3 and older. (noticed in issue
	  #16388) Reported by: parisioa Patches: valgrind.supp uloaded by
	  atis (license 242) Tested by: atis, parisioa ........

2009-12-07 23:24 +0000 [r233471-233609]  David Vossel <dvossel@digium.com>

	* main/utils.c: hex escape control and non 7-bit clean characters
	  in uri_encode In ast_uri_encode, non 7-bit clean characters were
	  being hex escaped correctly, but control characters were not.
	  (issue #16299)

	* channels/chan_sip.c: fixes missing Contact header angle brackets
	  (closes issue #16298) Reported by: mgernoth Patches:
	  reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
	  by: dvossel

2009-12-07 16:11 +0000 [r233392]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Allow SDP packets with only video session
	  information. (closes issue #16387) Reported by: zalex1953 Tested
	  by: mnicholson, zalex1953

2009-12-04 21:54 +0000 [r233116-233279]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample: clarify requirecalltoken option in
	  iax.sample.conf (closes issue #16223) Reported by: bklang
	  Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
	  (license 919)

	* apps/app_voicemail.c: document and rename strip_control() in
	  app_voicemail (closes issue #16291) Reported by: wdoekes

2009-12-04 17:12 +0000 [r233092]  Russell Bryant <russell@digium.com>

	* main/channel.c: Only do frame payload check for HOLD frames. This
	  code was added for helping to debug the source of invalid HOLD
	  frames. However, a side effect of this is that it will
	  incorrectly report errors for frames that have an integer
	  payload. Make the check for this block specific to the HOLD frame
	  case.

2009-12-04 16:59 +0000 [r233014-233091]  Matthias Nick <mnick@digium.com>

	* pbx/pbx_config.c: Parse global variables or expressions in hint
	  extensions Parse global variables or expressions in hint
	  extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
	  (closes issue #16166) Reported by: rmudgett Tested by: mnick,
	  rmudgett

	* main/dsp.c: Warning message gets displayed only once Added
	  additional field 'int display_inband_dtmf_warning', which when
	  set to '1' displays the warning ('Inband DTMF is not supported on
	  codec %s. Use RFC2833'), and when set to '0' doesn't display the
	  warning. Otherwise you would get hundreds of warnings every
	  second. (closes issue #15769) Reported by: falves11 Patches:
	  patch_15769_14.txt uploaded by mnick (license 874) Tested by:
	  mnick, falves11

2009-12-03 20:10 +0000 [r232820]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Deprecate "cz" in favor of "cs". Also,
	  change the use of language codes so that language registers as a
	  prefix, rather than an exact match. (closes issue #16272)
	  Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt
	  uploaded by tilghman (license 14)

2009-12-02 21:57 +0000 [r232581]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: Send ack (response/message) after receiving
	  manager action userevent (closes issue #16264) Reported by: dimas
	  Patches: event-ack.patch uploaded by dimas (license 88)

2009-12-02 19:03 +0000 [r232444]  David Vossel <dvossel@digium.com>

	* apps/app_queue.c: fixes app_queue ao2 error (closes issue #16369)
	  Reported by: vrban Patches: queue_issue_1.4.diff uploaded by
	  dvossel (license 671) Tested by: dvossel

2009-12-02 17:04 +0000 [r232355]  Joshua Colp <jcolp@digium.com>

	* apps/app_amd.c: Fix a bug where if you hung up very quickly after
	  calling AMD it would overwrite the AMDSTATUS of HANGUP with
	  TOOLONG. (closes issue #16239) Reported by: CGMChris

2009-12-02 16:59 +0000 [r232268-232350]  David Vossel <dvossel@digium.com>

	* main/acl.c: ast_outaddrfor doesn't do htons() on port, looks odd
	  in strace. (closes issue #16290) Reported by: wdoekes

	* funcs/func_groupcount.c: fixes segfault in func_groupcount closes
	  issue #16337) Reported by: Parantido Patches: issue_16337.diff
	  uploaded by dvossel (license 671) Tested by: Parantido, dvossel

2009-12-02 04:05 +0000 [r232165]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Fix compiling without devmode (closes issue
	  #16367) Reported by: falves11

2009-12-02 00:42 +0000 [r232090]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Do not modify the gain settings on data
	  calls. (The digital flag actually represents a data call.)
	  (closes issue #15972) Reported by: udosw Patches:
	  transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
	  Tested by: alecdavis

2009-12-01 23:25 +0000 [r232007]  Russell Bryant <russell@digium.com>

	* main/file.c: Fix a warning pointed out by buildbot.

2009-12-01 21:52 +0000 [r231911-231926]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: log channel name in dev mode as well

	* main/channel.c: Fix crash with invalid frame data The crash was
	  happening as a result of a frame containing an invalid data
	  pointer, but was set with data length of zero. The few times the
	  issue was reproduced it _seemed_ that the frame was queued
	  properly, that is the data pointer was set to NULL. I never could
	  reproduce the crash so as a last resort the crash has been fixed,
	  but a check in __ast_read has been added to give as much
	  information about the source of problematic frames in the future.
	  (closes issue #16058) Reported by: atis

2009-12-01 21:14 +0000 [r231853]  David Vossel <dvossel@digium.com>

	* main/pbx.c: WaitExten m option with no parameters generates frame
	  with zero datalen but non-null data ptr

2009-12-01 15:34 +0000 [r231614-231740]  Matthew Nicholson <mnicholson@digium.com>

	* main/file.c: Ignore unknown formats in ast_format_str_reduce()
	  and return an error if no know formats are found.

	* apps/app_voicemail.c, include/asterisk/file.h, main/file.c,
	  main/app.c: Remove duplicate entries from voicemail format lists.
	  This prevents app_voicemail from entering an infinite loop when
	  the same format is specified twice in the format list. (closes
	  issue #15625) Reported by: Shagg63 Tested by: mnicholson Review:
	  https://reviewboard.asterisk.org/r/429/

2009-11-30 17:14 +0000 [r231437-231441]  David Vossel <dvossel@digium.com>

	* main/rtp.c: fixes crash caused by RTP comfort noise payload
	  greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
	  by: amorsen Patches: issue16242.diff uploaded by oej (license
	  306) Tested by: amorsen, oej, dvossel

	* apps/app_queue.c: app_queue crashes randomly, often during
	  call-transfers In app_queue, it is possible for a call_queue to
	  be destroyed while another object still holds a pointer to it.
	  This patch converts call_queue objects to ao2 objects allowing
	  them to be ref counted. This makes it safe for the queue_ent
	  object in queue_exec() to reference it's parent call_queue even
	  after it has left the queue. (closes issue #15686) Reported by:
	  Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license
	  671) Tested by: dvossel, aragon Review:
	  https://reviewboard.asterisk.org/r/427/

2009-11-25 22:31 +0000 [r231298]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: After a frame duplication failure, unlock the
	  channel before returning.

2009-11-25 21:38 +0000 [r231233-231235]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c: fixes solaris segfault on dial with verbosity >=
	  3 (closes issue #16193) Reported by: asgaroth Patches:
	  bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
	  Tested by: asgaroth, snuffy

	* channels/chan_sip.c: fixes conditional jump or move depending on
	  uninitialised STACK value (closes issue #16261) Reported by:
	  edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917)

2009-11-23 15:31 +0000 [r230772-230875]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: When 'sip set debug' is enabled, and the
	  last line of an incoming SIP message is not properly newline
	  terminated, ensure that that line is included in the debug
	  output. (part of issue #16268)

	* main/editline/makelist.in, channels/chan_sip.c,
	  channels/ring_tone.h, channels/busy_tone.h: Correct fix for issue
	  #16268... the reporter's original patch was very close to
	  correct.

	* channels/chan_sip.c: Ensure that SDP parsing does not ignore the
	  last line of the SDP. (closes issue #16268) Reported by: sgimeno

2009-11-20 20:53 +0000 [r230627]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_features.c: Copy the peer CDR's userfield to the bridge
	  CDR if it exists. This is necessary for the recordagentcalls
	  option in chan_agent to store the recorded file name in the
	  bridge CDR. (closes issue #14590) Reported by: msetim Patches:
	  queue_agent_userfield.patch uploaded by Laureano (license 265)
	  Tested by: Laureano, mnicholson

2009-11-19 21:22 +0000 [r230508]  David Vossel <dvossel@digium.com>

	* apps/app_mixmonitor.c: fixes MixMonitor thread not exiting when
	  StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
	  Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
	  671) Tested by: dvossel, AlexMS Review:
	  https://reviewboard.asterisk.org/r/424/

2009-11-19 16:09 +0000 [r230469]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: Update copyright year in visible output. (cli)
	  Spotted by Stuart Henderson

2009-11-30  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27.1

	* AST-2009-010

	* SDP parser regression fix (issue #16268)

2009-11-18  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27

2009-11-13  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27-rc5

2009-11-12 16:41 +0000 [r229669]  David Vossel <dvossel@digium.com>

	* funcs/func_audiohookinherit.c: fixes merging error, datastore was
	  being freed in the wrong function. (closes issue #16219) Reported
	  by: aragon

2009-11-11 19:46 +0000 [r229498]  David Brooks <dbrooks@digium.com>

	* main/pbx.c: Solaris doesn't like NULL going to ast_log Solaris
	  will crash if NULL is passed to ast_log. This simple patch simply
	  uses S_OR to get around this. (closes issue #15392) Reported by:
	  yrashk

2009-11-10 22:09 +0000 [r229360]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: If two pattern classes start with the same digit and
	  have the same number of characters, they will compare equal. The
	  example given in the issue report is that of [234] and [246],
	  which have these characteristics, yet they are clearly not
	  equivalent. The code still uses these two characteristics, yet
	  when the two scores compare equal, an additional check will be
	  done to compare all characters within the class to verify
	  equality. (closes issue #15421) Reported by: jsmith Patches:
	  20091109__issue15421__2.diff.txt uploaded by tilghman (license
	  14) Tested by: jsmith, thedavidfactor

2009-11-10 21:45 +0000 [r229355]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt: Fix ExternalIVR Documentation Remove
	  documentation for event that doesn't function (closes issue
	  #16220) Reported by: thedavidfactor Patches:
	  externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
	  (license 903)

2009-11-10 20:03 +0000 [r229281]  Joshua Colp <jcolp@digium.com>

	* codecs/codec_g726.c: Remove broken support for direct transcoding
	  between G.726 RFC3551 and G.726 AAL2. On some systems the
	  translation core would actually consider g726aal2 -> g726 ->
	  signed linear to be a quicker path then g726aal2 -> signed linear
	  which exposed this problem. (closes issue #15504) Reported by:
	  globalnetinc

2009-11-10 17:23 +0000 [r229191]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt: Document ExternalIVR event tag collision
	  ExternalIVR uses the D tag for two different event types. This
	  documents that behavior and how to differentiate between the two
	  cases. Also includes a minor spelling fix and clarification
	  (closes issue #16211) Reported by: thedavidfactor Patches:
	  externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
	  (license 903)

2009-11-10 17:15 +0000 [r229167]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: don't crash on log message in solaris
	  AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
	  bklang

2009-11-10 15:22 +0000 [r229091]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Reverted revision 202022. (closes issue
	  #16175) Reported by: paul-tg

2009-11-09  Leif Madsen <lmadsen@digium.com>

	* Release Astersik 1.4.27-rc4

2009-11-09 15:37 +0000 [r228896]  Leif Madsen <lmadsen@digium.com>

	* main/channel.c: Update WARNING message. Update a WARNING message
	  to give a suggested fix when encountered. (closes issue #16198)
	  Reported by: atis Tested by: atis

2009-11-09 14:16 +0000 [r228827]  Matthew Nicholson <mnicholson@digium.com>

	* include/asterisk/lock.h: Perform limited bounds checking when
	  destroying ast_mutex_t structures to make sure we don't try to
	  use negative indices. (closes issue #15588) Reported by: zerohalo
	  Patches: 20090820__issue15588.diff.txt uploaded by tilghman
	  (license 14) Tested by: zerohalo

2009-11-06 22:33 +0000 [r228692]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes audiohook write crash occuring in chan_spy
	  whisper mode. After writing to the audiohook list in ast_write(),
	  frames were being freed incorrectly. Under certain conditions
	  this resulted in a double free crash. (closes issue #16133)
	  Reported by: wetwired (closes issue #16045) Reported by:
	  bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
	  671) Tested by: bluecrow76, dvossel, habile

2009-11-06 18:32 +0000 [r228547]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't overwrite caller ID name on a trunk
	  with the configured fullname when using users.conf (issue
	  ABE-1989)

2009-11-06  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27-rc3

2009-11-06 17:07 +0000 [r228418]  David Vossel <dvossel@digium.com>

	* codecs/codec_ilbc.c: fixes segfault in iLBC For reasons not yet
	  known, it appears possible for an ast_frame to have a datalen
	  greater than zero while the actual data is NULL during Packet
	  Loss Concealment. Most codecs don't support PLC so this doesn't
	  affect them. This patch catches the malformed frame and prevents
	  the crash from occuring. Additional efforts to determine why it
	  is possible for a frame to look like this are still being
	  investigated. (issue #16979)

2009-11-06 16:41 +0000 [r228409]  Joshua Colp <jcolp@digium.com>

	* main/abstract_jb.c: Fix a bug caused by a partially invalid frame
	  (from the jitterbuffer) passing through the Asterisk core.
	  (closes issue #15560) Reported by: jvandal (closes issue #15709)
	  Reported by: covici

2009-11-06 16:26 +0000 [r228378]  Matthew Nicholson <mnicholson@digium.com>

	* funcs/func_base64.c, main/utils.c: Properly handle '=' while
	  decoding base64 messages and null terminate strings returned from
	  BASE64_DECODE. (closes issue #15271) Reported by: chappell
	  Patches: base64_fix.patch uploaded by chappell (license 8) Tested
	  by: kobaz

2009-11-06 15:41 +0000 [r228272-228338]  David Vossel <dvossel@digium.com>

	* main/astfd.c: fixes crash in astfd.c (closes issue #15981)
	  Reported by: slavon

	* funcs/func_audiohookinherit.c: fixes memory leak in
	  func_audiohookinherit.c (closes issue 0015394) Reported by:
	  boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
	  (license 790) Tested by: dbrooks, boroda

2009-11-05 19:14 +0000 [r228079]  Jason Parker <jparker@digium.com>

	* channels/chan_vpb.cc: Fix crash on VPB exception when no hardware
	  is present. (closes issue #14970) Reported by: tzafrir Patches:
	  vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
	  markwaters

2009-11-05 18:59 +0000 [r228078]  David Brooks <dbrooks@digium.com>

	* channels/chan_misdn.c: chan_misdn Asterisk 1.4.27-rc2 crash Crash
	  related to chan_misdn connection. Patch submitted by
	  gknispel_proformatique, tested by francesco_r. "I have many crash
	  since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
	  bt." This patch zeros out an ast_frame. (closes issue #16041)
	  Reported by: francesco_r

2009-11-04 23:47 +0000 [r227944]  Jeff Peeler <jpeeler@digium.com>

	* res/res_monitor.c: Fix incorrect filename comparsion after
	  monitor file change The logic to detect if a requested file is
	  indeed a different file from the current file was incorrect. The
	  main issue being confusion of the use of filename_base which was
	  previously set without pathing information and then compared to
	  another full path. Robust file comparison logic has been added to
	  properly check if two files are the same even if symlinks are
	  used. (closes issue #15313) Reported by: caspy Patches:
	  20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
	  325) but mostly tilghman's work

2009-11-04 20:52 +0000 [r227758-227827]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_dial.c: This patch modifies the Dial application to
	  monitor the calling channel for hangups while playing back
	  announcements. (closes issue #16005) Reported by: falves11
	  Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
	  (license 96) Tested by: mnicholson, falves11 Review:
	  https://reviewboard.asterisk.org/r/407/

	* channels/chan_sip.c: Modify the SDP parsing code to parse session
	  and media level items separately. With the new code, media level
	  proprieties should no longer be confused with session level
	  proprieties. This change also reorganizes some of the SDP parsing
	  code which should make it easier to manage in the future. (closes
	  issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
	  file Review: https://reviewboard.asterisk.org/r/385/

2009-11-04 19:25 +0000 [r227700-227735]  Joshua Colp <jcolp@digium.com>

	* static-http/prototype.js: Fix a security issue where it may be
	  possible for someone to execute a cross-site AJAX request
	  exploit. (AST-2009-009)

	* channels/chan_sip.c: Fix a security issue where sending a
	  REGISTER with a differing username in the From URI and
	  Authorization header would reveal whether it was valid or not.
	  (AST-2009-008)

2009-11-03 17:55 +0000 [r227275]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Make sure the outgoing flag is cleared if
	  a new channel fails to get created for outgoing calls. This is
	  the relevant portion of asterisk/trunk -r226648

2009-11-03 15:36 +0000 [r227166]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where an RPID header could be
	  generated with a blank username in the URI. (closes issue #15909)
	  Reported by: kobaz

2009-11-03 10:48 +0000 [r227088-227090]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fixing bug before someone reports it...

	* channels/chan_sip.c: Adding IP address in Contact ACL log message
	  and removing redundant message (based on kpfleming's feedback)

	* channels/chan_sip.c: Use proper response code when violating
	  Contact ACL's. Review: https://reviewboard.asterisk.org/r/415/
	  Thanks kpfleming for a quick review. (EDVX-003)

2009-11-02 20:52 +0000 [r226972]  David Brooks <dbrooks@digium.com>

	* channels/chan_sip.c: SIP channel name uniqueness SIP channel
	  names were supposed to be unique by way of a name suffix derived
	  from the pointer to the channel's private data. Uniqueness was
	  preserved on 32-bit systems, but not on 64-bit systems. This
	  patch, as suggested by kpfleming, replaces this suffix with a
	  simple incremented unsigned int. (closes issue #15152) Reported
	  by: palbrecht Review: https://reviewboard.asterisk.org/r/420/

2009-11-02 18:08 +0000 [r226889]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Fix a bug where the recorded privacy
	  introduction file would not get removed if the caller hung up
	  while the called party had not yet answered. This was fixed by
	  introducing an argument to the 'n' option which, when enabled,
	  removes the introduction file under all scenarios. This was done
	  to preserve the behavior that has existed for quite some time.
	  (closes issue #14674) Reported by: ulogic Patches: bug14674.patch
	  uploaded by jpeeler (license 325)

2009-11-02 17:14 +0000 [r226811]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.redhat.asterisk: Don't allow two separate
	  instances of safe_asterisk when restarting from the init script.
	  (closes issue #14562) Reported by: davidw Patches: Initially
	  20091022__issue14562.diff.txt uploaded by tilghman (license 14)
	  Modified to 20091030__Issue14562_diff.txt uploaded by davidw
	  (license 780) Tested by: davidw

2009-11-02 15:31 +0000 [r226688-226736]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes crash on iterator_destroy on
	  uninitialized iterator (closes issue #16162) Reported by: krn

	* channels/chan_iax2.c: changes calltoken debug messages from
	  LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes
	  issue #16144) Reported by: aragon

2009-10-29 18:11 +0000 [r226531]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, doc/localchannel.txt: Add an option to
	  enabling passing music on hold start and stop requests through
	  instead of acting on them in chan_local. (closes issue #14709)
	  Reported by: dimas

2009-10-28 20:06 +0000 [r226377-226382]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Update documentation in sip.conf.sample.
	  Update the documentation in sip.conf.sample in order to make it
	  more clear that directmedia/canreinvite do not cause Asterisk to
	  ignore reINVITEs. It is only used to stop Asterisk from
	  generating a reINVITE, but does not stop it from accepting them
	  if necessary. (closes issue #15644) Reported by: lmadsen

	* doc/channelvariables.txt: Update CALLINGSUBADDR channel variable
	  documentation. (closes issue #15734) Reported by: alecdavis
	  Patches: channelvariables.tex.diff.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis

2009-10-28 18:02 +0000 [r226138-226304]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/linkedlists.h: Fix documentation (pointed out by
	  TheDavidFactor on #-dev)

	* main/manager.c: Manager output is not always NULL-terminated, so
	  force a NULL at the end of the filestream. (closes issue #15495)
	  Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
	  by tilghman (license 14) Tested by: pdf

2009-10-26 22:13 +0000 [r225957]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: detect
	  ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set
	  OSARCH to linux-gnu even if host_os is linux-gnueabi * When
	  checking if we are Linux, check OSARCH rather than host_os The
	  newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather
	  than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even
	  in such a case. OSARCH is tested for the value of 'linux-gnu' in
	  one or two places in the tree. This patch also fixes the check
	  libcap to check for $OSARCH rather than $host_os . See also:
	  http://wiki.debian.org/ArmEabiPort

2009-10-23 14:00 +0000 [r225581]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Don't force menuselect.makeopts to be rebuilt on every
	  build. For some reason the menuselect.makeopts file was listed as
	  PHONY in the Makefile, resulting in 'make' needing to rebuild it
	  for every build. This then resulted in the embedded module rules
	  being rebuilt on every build, which can be slow and is
	  unnecessary. This patch fixes the problem by properly allowing
	  'make' to know when the menuselect.makeopts file needs to be
	  rebuilt (defining the proper dependencies).

2009-10-22 21:51 +0000 [r225484]  Leif Madsen <lmadsen@digium.com>

	* doc/valgrind.txt, contrib/valgrind.supp (added): Clean valgrind
	  output by suppressing false errors. Update valgrind.txt
	  documentation and add valgrind.supp file in order to allow those
	  who are creating valgrind output to have less false errors in the
	  logfile. (closes issue #16007) Reported by: atis Patches:
	  valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp
	  uploaded by atis (license 242) Tested by: atis, amorsen

2009-10-21 20:58 +0000 [r225243]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: IAX2: VNAK loop caused by signaling frames
	  with no destination call number It is possible for the PBX thread
	  to queue up signaling frames before a destination call number is
	  received. This can result in signaling frames being sent out with
	  no destination call number. Since recent versions of Asterisk
	  require accurate destination callnumbers for all Full Frames,
	  this can cause a VNAK loop to occur. To resolve this no signaling
	  frames are sent until a destination callnumber is received, and
	  destination call numbers are now only required for iax_pvt
	  matching when the frame is an ACK. Review:
	  https://reviewboard.asterisk.org/r/413/

2009-10-21 16:44 +0000 [r225169-225171]  Russell Bryant <russell@digium.com>

	* main/translate.c: Revert 225169, as this doesn't account for the
	  possibility of a list of frames.

	* main/translate.c: Isolate the frame returned from
	  ast_translate().

2009-10-21 16:02 +0000 [r225103-225105]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, apps/app_meetme.c, include/asterisk/channel.h: Fix
	  documentation for ast_softhangup() and correct the misuse
	  thereof. (closes issue #16103) Reported by: majorbloodnok

	* apps/app_voicemail.c: Suffix is not needed for a match

2009-10-21 14:37 +0000 [r225032]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, channels/chan_sip.c,
	  configs/sip.conf.sample, channels/chan_iax2.c: IAX/SIP
	  shrinkcallerid option The shrinking of caller id removes '(', '
	  ', ')', non-trailing '.', and '-' from the string. This means
	  values such as 555.5555 and test-test result in 555555 and
	  testtest. There are instances, such as Skype integration, where a
	  specific value is passed via caller id that must be preserved
	  unmodified. This patch makes the shrinking of caller id optional
	  in chan_sip and chan_iax in order to support such cases. By
	  default this option is on to preserve previous expected behavior.
	  (closes issue #15940) Reported by: dimas Patches: v2-15940.patch
	  uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c
	  uploaded by dvossel (license 671) Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/408/

2009-10-21 02:59 +0000 [r224931]  Russell Bryant <russell@digium.com>

	* include/asterisk/translate.h, main/dsp.c, main/frame.c,
	  main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
	  include/asterisk/frame.h: Isolate frames returned from a DSP
	  instance or codec translator. The reasoning for these changes are
	  the same as what I wrote in the commit message for rev 222878.

2009-10-20 22:07 +0000 [r224855]  Tilghman Lesher <tlesher@digium.com>

	* main/audiohook.c: Pay attention to the return value of the
	  manipulate function. While this looks like an optimization, it
	  prevents a crash from occurring when used with certain audiohook
	  callbacks (diagnosed with SVN trunk, backported to 1.4 to keep
	  the source consistent across versions).

2009-10-20 17:46 +0000 [r224773]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Add support for relaying early media in the
	  features attended transfer option. (closes issue #14828) Reported
	  by: licedey

2009-10-19 23:44 +0000 [r224670]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp.c: Correct timestamp calculations when RTP sample rates
	  over 8kHz are used. While testing some endpoints that support
	  16kHz and 32kHz sample rates, some log messages were generated
	  due to calc_rxstamp() computing timestamps in a way that produced
	  odd results, so this patch sanitizes the result of the
	  computations.

2009-10-19 19:47 +0000 [r224565]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Do not attempt early media bridging (ie: direct
	  RTP setup) if options are enabled that should prevent it. (closes
	  issue #14763) Reported by: cupotka

2009-10-17 01:32 +0000 [r224330]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix stale caller id data from being
	  reported in AMI NewChannel event The problem here is that
	  chan_dahdi is designed in such a way to set certain values in the
	  dahdi_pvt only once. One of those such values is the configured
	  caller id data in chan_dahdi.conf. For PRI, the configured caller
	  id data could be overwritten during a call. Instead of saving the
	  data and restoring, it was decided that for all non-analog
	  channels it was simply best to not set the configured caller id
	  in the first place and also clear it at the end of the call.
	  (closes issue #15883) Reported by: jsmith

2009-10-16 20:25 +0000 [r224260]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Never released PRI channels when using
	  Busy() or Congestion() dialplan apps. When the Busy() or
	  Congestion() application is used towards ISDN (an ISDN progress
	  is sent), the responding ISDN Disconnect or Release may contain
	  the ISDN cause user busy or one of the congestion causes. In
	  chan_dahdi.c these causes will only set the needbusy or
	  needcongestion flags and not activate the softhangup procedure.
	  Unfortunately only the latter can interrupt the endless wait loop
	  of Busy()/Congestion(). Result: PRI channels staying in state
	  busy for the rest of asterisk life or until the other end times
	  out and forces the call to clear. (in issue 0014292) Reported by:
	  tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso
	  (license 564) (This patch is unrelated to the issue.)

2009-10-13 20:58 +0000 [r223955]  Jean Galarneau <jgalarneau@digium.com>

	* channels/chan_dahdi.c: Fix PRI timer T309 operation

2009-10-12 23:12 +0000 [r223804]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_dial.c: Ensure ringing continues for branched calls
	  after progress is received While waiting for an answer, don't
	  send progress for branched calls for which ringing was sent.
	  (closes issue #15028) Reported by: fnordian

2009-10-12 15:30 +0000 [r223692]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Remove automatic switching from T.38 to
	  voice mode in chan_sip. chan_sip has some code to automatically
	  switch from T.38 mode to voice mode when a voice frame is written
	  to the channel while it is in T.38 mode; this was intended to
	  handle the situation when a FAX transmission has ended and the
	  channel is not yet hung up, but is causing problems at the
	  beginning of FAX sessions as well when there are still voice
	  frames 'in flight' at the time the T.38 negotiation completes.
	  This patch removes the automatic switchover. (issue #16025)
	  Reported by: jamicque

2009-10-11 18:34 +0000 [r223485-223550]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Remove a duplicate ao2_iterator_destroy().

	* main/autoservice.c: Remove some unnecessary code.

	* main/autoservice.c: Don't use data outside of its scope. The
	  purpose of this code was to have a hangup frame put on the list
	  of deferred frames. However, the code that read the hangup frame
	  was outside of the scope of where the hangup frame was declared.

2009-10-09 18:20 +0000 [r223225]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Signal timeouts by returning AST_CONTROL_RINGING
	  when originating calls. (closes issue #15104) Reported by:
	  nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
	  (license 96) Tested by: nblasgen, mnicholson

2009-10-09 18:17 +0000 [r223213]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Fix potential memory leak in app_dial.c

2009-10-09 17:52 +0000 [r223142-223205]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes sip registration using authuser in
	  user.conf (closes issue #14954) Reported by: tornblad Tested by:
	  mmichelson, tornblad, dvossel

	* channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
	  (closes issue https://issues.asterisk.org/view.php?id=15949)
	  Reported by: ebroad Patches: authparsefix.patch uploaded by
	  ebroad (license 878) 15949_trunk.diff uploaded by dvossel
	  (license 671) Tested by: ebroad

2009-10-08 19:45 +0000 [r222878]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, main/frame.c, main/file.c,
	  include/asterisk/frame.h: Make filestream frame handling safer by
	  isolating frames before returning them. This patch is related to
	  a number of issues on the bug tracker that show crashes related
	  to freeing frames that came from a filestream. A number of fixes
	  have been made over time while trying to figure out these
	  problems, but there re still people seeing the crash. (Note that
	  some of these bug reports include information about other
	  problems. I am specifically addressing the filestream frame crash
	  here.) I'm still not clear on what the exact problem is. However,
	  what is _very_ clear is that we have seen quite a few problems
	  over time related to unexpected behavior when we try to use
	  embedded frames as an optimization. In some cases, this
	  optimization doesn't really provide much due to improvements made
	  in other areas. In this case, the patch modifies filestream
	  handling such that the embedded frame will not be returned.
	  ast_frisolate() is used to ensure that we end up with a
	  completely mallocd frame. In reality, though, we will not
	  actually have to malloc every time. For filestreams, the frame
	  will almost always be allocated and freed in the same thread.
	  That means that the thread local frame cache will be used. So,
	  going this route doesn't hurt. With this patch in place, some
	  people have reported success in not seeing the crash anymore.
	  (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
	  Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
	  (license 2) Tested by: aragon, russell (closes issue #15817)
	  Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
	  Reported by: marhbere Review:
	  https://reviewboard.asterisk.org/r/386/

2009-10-08 19:45 +0000 [r222877]  David Vossel <dvossel@digium.com>

	* main/netsock.c, include/asterisk/netsock.h: fixes an
	  ast_netsock_list memory leak. ABE-1998 Review:
	  https://reviewboard.asterisk.org/r/395/

2009-10-08 16:33 +0000 [r222691-222797]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn_config.c: Fix memory leak if chan_misdn config
	  parameter is repeated. Memory leak when the same config option is
	  set more than once in an misdn.conf section. Why must this be
	  considered? Templates! Defining a template with default port
	  options and later adding to or overriding some of them. Patches:
	  memleak-misdn.patch JIRA ABE-1998

	* channels/chan_misdn.c: chan_misdn.c:process_ast_dsp() memory leak
	  misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
	  does not occur. The translated frame "f2" when passing through
	  ast_dsp_process() is not freed whenever it is not used further in
	  process_ast_dsp(). Then in the end it is never ever freed.
	  Patches: translate.patch JIRA ABE-1993

2009-10-07 17:41 +0000 [r222542]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: crash on transfer handle_invite_replaces()
	  attempts to uplock a pvt's owner channel without first verifing
	  that it exists. (issue #16027)

2009-10-06 23:51 +0000 [r222393-222462]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Add missing unlock(s) in dahdi_read (two
	  cases in trunk) (closes issue #15683) Reported by: alecdavis

	* channels/chan_dahdi.c: Fix potential crash when entire span
	  request is received. The variable index used in this scenario for
	  accessing the dahdi_pvts was wrong and was most likely copied
	  from the several other places it is used correctly. (closes issue
	  #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
	  uploaded by tsearle (license 373)

2009-10-06  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27-rc2

2009-10-06 01:16 +0000 [r222152]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h,
	  res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c:
	  Fix ao2_iterator API to hold references to containers being
	  iterated. See Mantis issue for details of what prompted this
	  change. Additional notes: This patch changes the ao2_iterator API
	  in two ways: F_AO2I_DONTLOCK has become an enum instead of a
	  macro, with a name that fits our naming policy; also, it is now
	  necessary to call ao2_iterator_destroy() on any iterator that has
	  been created. Currently this only releases the reference to the
	  container being iterated, but in the future this could also
	  release other resources used by the iterator, if the iterator
	  implementation changes to use additional resources. (closes issue
	  #15987) Reported by: kpfleming Review:
	  https://reviewboard.asterisk.org/r/383/

2009-10-02 17:32 +0000 [r222026]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Removes unnecessary unlock, clarifies a
	  memcpy.

2009-10-02 16:58 +0000 [r221776-221970]  Tilghman Lesher <tlesher@digium.com>

	* main/astobj2.c: Ensure the result of the hash function is
	  positive. Negative array offsets suck.

	* main/asterisk.c, main/rtp.c, main/say.c: Fix a bunch of
	  off-by-one errors

2009-10-01 23:18 +0000 [r221769]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h:
	  Occasionally losing use of B channels in chan_misdn. I have not
	  been able to reproduce the problem of losing channels. However, I
	  have seen in the code a reentrancy problem that might give these
	  symptoms. The reentrancy patch does several things: 1) Guards B
	  channel and B channel structure allocation. 2) Makes the B
	  channel structure find routines more precise in locating records.
	  3) Never leave a B channel allocated if we received cause 44. The
	  last item may cause temporary outgoing call problems, but they
	  should clear when the line becomes idle. (closes issue #15490)
	  Reported by: slutec18 Patches:
	  issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
	  (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
	  Reported by: FabienToune Patches:
	  issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
	  (license 664) Tested by: FabienToune, rmudgett, slutec18

2009-10-01 15:24 +0000 [r221360-221588]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Use unsigned ints for portinuri flags.

	* channels/chan_sip.c: Make portinuri a bitfield.

	* channels/chan_sip.c, configs/sip.conf.sample: Fix SRV lookup and
	  Request-URI generation in chan_sip. This patch adds a new field
	  "portinuri" to the sip dialog struct and the sip peer struct.
	  That field is used during RURI generation to determine if the
	  port should be included in the RURI. It is also used in some
	  places to determine if an SRV lookup should occur. (closes issue
	  #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson
	  Review: https://reviewboard.asterisk.org/r/369/

2009-09-30 19:02 +0000 [r221303]  Matthias Nick <mnick@digium.com>

	* funcs/func_strings.c: changed the prototype definition of
	  csv_quote

2009-09-30 16:55 +0000 [r221200]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Avoid a potential NULL dereference. (closes issue
	  #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
	  uploaded by tilghman (license 14) Tested by: kobaz

2009-09-30 15:41 +0000 [r221153-221157]  Matthias Nick <mnick@digium.com>

	* configs/cdr_custom.conf.sample, funcs/func_strings.c: added a new
	  dialplan function 'CSV_QUOTE' and changed the
	  cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
	  Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
	  mnick

	* funcs/func_strings.c: check bounds - prevents for buffer overflow

2009-09-30 14:49 +0000 [r221086]  Terry Wilson <twilson@digium.com>

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
	  configs/sip.conf.sample: Change the SSRC by default when our
	  media stream changes Be default, change SSRC when doing an audio
	  stream changes Asterisk doesn't honor marker bit when reinvited
	  to already-bridged RTP streams,resulting in far-end stack
	  discarding packets with "old" timestamps that areactually part of
	  a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever
	  there is a reinvite, unless the 'constantssrc' is set to true in
	  sip.conf. The original issue reported to Digium support detailed
	  the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
	  Application Server Call comes in fromITSP, Asterisk dials the app
	  server which sends a re-invite back toAsterisk--not to negotiate
	  to send media directly to the ITSP, but to indicatethat it's
	  changing the stream it's sending to Asterisk. The app
	  servergenerates a new SSRC, sequence numbers, timestamps, and
	  sets the marker bit on the new stream. Asterisk passes through
	  the teimstamp of the new stream, butdoes not reset the SSRC,
	  sequence numbers, or set the marker bit. When the timestamp on
	  the new stream is older than the timestamp on the originalstream,
	  the ITSP (which doesn't know there has been any change) discards
	  the newframes because it thinks they are too old. This patch
	  addresses this by changing the SSRC on a stream update unless
	  constantssrc=true is set in sip.conf. Review:
	  https://reviewboard.asterisk.org/r/374/

2009-09-29 20:14 +0000 [r220907]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
	  spyee is masqueraded and chanspy_ds_chan_fixup() is called with
	  the channel locked. (closes issue #15965) Reported by: atis
	  Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
	  (license 96) Tested by: atis

2009-09-29 17:59 +0000 [r220873]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Reduce CPU usage related to building a peer
	  merely for devicestates. This fixes a 100% CPU problem in the SIP
	  driver, found by profiling the driver while the problem was
	  occurring. (closes issue #14309) Reported by: pkempgen Patches:
	  20090924__issue14309.diff.txt uploaded by tilghman (license 14)
	  Tested by: pkempgen, vrban

2009-09-28 19:09 +0000 [r220717]  Sean Bright <sean@malleable.com>

	* Makefile.rules: When selecting DONT_OPTIMIZE in menuselect,
	  explicitly pass -O0 to the compiler so we override any default
	  optimization levels for a particular install.

2009-09-24 19:39 +0000 [r220288]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_playback.c, main/pbx.c, apps/app_disa.c: Implicitly
	  sending a progress signal breaks some applications. Call
	  Progress() in your dialplan if you explicitly want progress to be
	  sent. (Reverts change 216430, closes issue #15957) Reported by:
	  Pavel Troller on the Asterisk-Dev mailing list
	  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html

2009-09-24 18:18 +0000 [r220099-220213]  Sean Bright <sean@malleable.com>

	* Makefile: Resolve parallel build warnings. Reported by Klaus
	  Darilion on the asterisk-dev mailing list.

	* Makefile, build_tools/mkpkgconfig: Remove the remaining bashisms
	  in the Makefile/mkpkgconfig

2009-09-24 08:33 +0000 [r220027]  Michiel van Baak <michiel@vanbaak.info>

	* build_tools/mkpkgconfig: mkpkgconfig does not need bash so make
	  it use /bin/sh This fixes building on all systems that don't have
	  bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys
	  on #asterisk-dev

2009-09-22 21:37 +0000 [r219816]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: When IMAP variables were changed during a
	  reload, Voicemail did not use the new values. This change
	  introduces a configuration version variable, which ensures that
	  connections with the old values are not reused but are allowed to
	  expire normally. (closes issue #15934) Reported by:
	  viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
	  tilghman (license 14) Tested by: viniciusfontes

2009-09-21 16:55 +0000 [r219720]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Reverting merge 219520. This change was not
	  necessary.

2009-09-20 17:52 +0000 [r219653]  Tilghman Lesher <tlesher@digium.com>

	* main/file.c: Really stop the stream, when ast_closestream() is
	  called. (closes issue #15129) Reported by: bmh Patches:
	  20090918__issue15129.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/372/

2009-09-19 02:51 +0000 [r219586]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Make sure the iax_pvt exists before
	  dereferencing it. This fixes the latest crash posted on issue
	  15609. (issue #15609)

2009-09-18 23:19 +0000 [r219450-219519]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: iax2 frame double free The iax frame's
	  retrans sched id was written over right before iax2_frame_free
	  was called. In iax2_frame_free that retrans id is used to delete
	  the sched item. By writing over the retrans field before the
	  sched item could be deleted, it was possible for a retransmit to
	  occur on a freed frame.

	* channels/chan_sip.c: via-header branches not updated correctly on
	  INVITE INVITE requests must always contain a new unique branch
	  id. When a new branch id is created for an INVITE, the dialog's
	  invite_branch variable must be updated so CANCEL requests use the
	  correct branch id. (closes issue #15262) Reported by: maniax
	  Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
	  (license 608) invite_new_branch_trunk.diff uploaded by dvossel
	  (license 671) Tested by: maniax, dvossel

2009-09-17 22:20 +0000 [r219320]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Send a 100 Trying response when we detect a
	  spiral. This was problematic during spiral tests at SIPit...
	  along with some other things as well.

2009-09-17 21:29 +0000 [r219303]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: INVITE w/Replaces deadlock fix This patch
	  cleans up the locking logic in chan_sip.c's
	  handle_invite_replaces() function as well as making use of
	  ast_do_masquerade() rather than forcing the masquerade on an
	  ast_read(). The code had several redundant unlocks that would
	  result in 'freed more times than we've locked!' errors. I cleaned
	  these up as well as moving all the unlock logic to the end of the
	  function. This patch should also resolve the issue people were
	  having with the replacecall channel never being unlocked with one
	  legged calls. (closes issue #15151) Reported by: irroot Patches:
	  invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
	  Tested by: irroot, dvossel Review:
	  https://reviewboard.asterisk.org/r/371/

2009-09-17  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.27-rc1

2009-09-17 14:58 +0000 [r219136]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, include/asterisk/cdr.h,
	  include/asterisk/channel.h: Prevent a potential race condition
	  and crash when hanging up a channel by removing the channel from
	  the channel list before begining channel tear down. This fix may
	  potentially cause problems with CDR backends that access the
	  channel a CDR is associated with via the channel list. This fix
	  makes the channel unavabile at the time when the CDR backend is
	  invoked. This has been documented in include/asterisk/cdr.h.
	  (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
	  Review: https://reviewboard.asterisk.org/r/362/

2009-09-16 23:21 +0000 [r219023]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c, configs/extensions.conf.sample: Properly deal with
	  quotes in the arguments of '#exec' includes. (closes issue
	  #15583) Reported by: pkempgen Patches:
	  20090726__issue15583.diff.txt uploaded by tilghman (license 14)
	  20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
	  169) Tested by: pkempgen

2009-09-16 18:00 +0000 [r218867]  David Brooks <dbrooks@digium.com>

	* main/pbx.c: Fixes CID pattern matching behavior to mirror that of
	  extension pattern matching. Pattern matching for extensions uses
	  a type of scoring system, giving values for specificity to each
	  character in the pattern. Unfortunately, this is done character
	  by character, in order. This does lead to some less specific
	  patterns being first in line for matching, but it will usually
	  get the job done. This patch merely brings CID matching to the
	  same level as extension matching. This patch does not attempt to
	  tackle the problem shared by extension matching. (closes issue
	  #14708) Reported by: klaus3000

2009-09-16 13:33 +0000 [r218798]  Russell Bryant <russell@digium.com>

	* contrib/firmware/iax/iaxy.bin (removed), UPGRADE.txt: Remove the
	  IAXy firmware from Asterisk. The firmware can now be found on
	  downloads.digium.com, where the rest of our binary downloads
	  live. This was the last part of our Asterisk tarballs that was
	  considered non-free by Debian. :-) (closes issue #15838) Reported
	  by: paravoid

2009-09-15 22:27 +0000 [r218730]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: If the user enters the same password as
	  before, don't signal an error when the change does nothing.
	  (closes issue #15492) Reported by: cbbs70a Patches:
	  20090713__issue15492.diff.txt uploaded by tilghman (license 14)

2009-09-15 16:29 +0000 [r218623]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix small memory leak in handle_init_event
	  by always destroying the pthread attr before returning.

2009-09-15 16:03 +0000 [r218578]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Send request contact header field with
	  response to registrer queries instead of the address of record.
	  (closes issue #14438) Reported by: ravindrad Patches:
	  regquerypatch uploaded by ravindrad (license 684) Tested by:
	  ravindrad

2009-09-15 16:01 +0000 [r218577]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_followme.c: Ensure FollowMe sets language in channels it
	  creates. Also, not in the original bug report, but related fields
	  are accountcode and musicclass, and the inheritance of
	  datastores. (closes issue #15372) Reported by: Romik Patches:
	  20090828__issue15372.diff.txt uploaded by tilghman (license 14)
	  Tested by: cervajs

2009-09-15 14:57 +0000 [r218497-218498]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: revert accidental commit

	* channels/chan_sip.c, sounds/Makefile: Use proper hostname for
	  downloading sound files.

2009-09-14 21:47 +0000 [r218401]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix handling of DAHDI_EVENT_REMOVED event
	  to prevent crash in do_monitor. After talking to rmudgett about
	  some of his recent iflist locking changes, it was determined that
	  the only place that would destroy a channel without being
	  explicitly to do so was in handle_init_event. The loop to walk
	  the interface list has been modified to wait to destroy the
	  channel until the dahdi_pvt of the channel to be destroyed is no
	  longer needed. (closes issue #15378) Reported by: samy

2009-09-14 19:16 +0000 [r218331]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, sounds/Makefile: Don't say "Please try
	  again" if we don't give the user another chance to try again.
	  (issue #15055, SWP-129) Reported by: jthurman

2009-09-14 14:53 +0000 [r218223]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_directed_pickup.c: Ensure we don't pickup ourselves when
	  doing pickup by exten. (closes issue #15100) Reported by:
	  lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan
	  (license 779)

2009-09-10 23:52 +0000 [r217917-217989]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Don't ring another channel, if there's not
	  enough time for a queue member to answer. (Fixes AST-228)

	* contrib/scripts/iax-friends.sql, channels/chan_sip.c,
	  channels/chan_iax2.c: Backport realtime fix to 1.4

2009-09-10 21:06 +0000 [r217806]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: IAX2 encryption regression The IAX2 Call
	  Token security patch inadvertently broke the use of encryption
	  due to the reorganization of code in the socket_process()
	  function. When encryption is used, an incoming full frame must
	  first be decrypted before the information elements can be parsed.
	  The security release mistakenly moved IE parsing before
	  decryption in order to process the new Call Token IE. To resolve
	  this, decryption of full frames is once again done before looking
	  into the frame. This involves searching for an existing callno,
	  checking the pvt to see if encryption is turned on, and
	  decrypting the packet before the internal fields of the full
	  frame are accessed. associated with AST-2009-006 (closes issue
	  #15834) Reported by: karesmakro Patches:
	  iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
	  Tested by: dvossel, karesmakro Review:
	  https://reviewboard.asterisk.org/r/355/

2009-09-10 19:52 +0000 [r217668-217735]  Olle Johansson <oej@edvina.net>

	* utils/Makefile: Reinstate muted that was removed by mistake.
	  muted doesn't compile any more on os/x, so I have to disable it
	  in order to testcompile other code...

	* utils/Makefile, channels/chan_sip.c: Remove harmful code that
	  causes endless loops. Remove code that causes loops in
	  registrations. We have agreed that the patch that this code was
	  part of was bad. I am ripping out the code that causes the issue.
	  putnopvut needs to check the rest of the patch, if it needs to be
	  changed as well. This solves the issue reported in #15540, but
	  needs more work before we close it (as described above).

2009-09-08 20:01 +0000 [r217156]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: When MOH is playing on the channel,
	  announcements sent through the conference are not heard. (closes
	  issue #14588) Reported by: voipas Patches:
	  20090716__issue14588__2.diff.txt uploaded by tilghman (license
	  14) Tested by: lmadsen, twisted, tilghman

2009-09-04 13:56 +0000 [r216432-216435]  Michiel van Baak <michiel@vanbaak.info>

	* main/utils.c, include/asterisk/lock.h: make asterisk compile
	  under devmode with DEBUG_THREADS enabled on OpenBSD

	* channels/chan_sip.c: make chan_sip compile under devmode again

2009-09-04 13:45 +0000 [r216430]  Olle Johansson <oej@edvina.net>

	* apps/app_playback.c, main/pbx.c, channels/chan_sip.c,
	  apps/app_disa.c, configs/sip.conf.sample: Make apps send PROGRESS
	  control frame for early media and fix too early media issue in
	  SIP The issue at hand is that some legacy (dying) PBX systems
	  send empty media frames on PRI links *before* any call progress.
	  The SIP channel receives these frames and by default signals 183
	  Session progress and starts sending media. This will cause phones
	  to play silence and ignore the later 180 ringing message. A bad
	  user experience. The fix is twofold: - We discovered that
	  asterisk apps that support early media ("noanswer") did not send
	  any PROGRESS frame to indicate early media. Fixed. - We introduce
	  a setting in chan_sip so that users can disable any relay of
	  media frames before the outbound channel actually indicates any
	  sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be
	  disabled for backward compatibility. In later versions of
	  Asterisk, this will be enabled. We don't assume that it will
	  change your Asterisk phone experience - only for the better. We
	  encourage third-party application developers to make sure that if
	  they have applications that wants to send early media, add a
	  PROGRESS control frame transmission to make sure that all channel
	  drivers actually will start sending early media. This has not
	  been the default in Asterisk previous to this patch, so if you
	  got inspiration from our code, you need to update accordingly.
	  Sorry for the trouble and thanks for your support. This code has
	  been running for a few months in a large scale installation (over
	  250 servers with PRI and/or BRI links to old PBX systems). That's
	  no proof that this is an excellent patch, but, well, it's tested
	  :-)

2009-09-04 13:16 +0000 [r216369]  Michiel van Baak <michiel@vanbaak.info>

	* main/astobj2.c: Make sure 'start' is always initialized. This is
	  the same as rev 216222 in trunk but 1.4 is affected as well

2009-09-04 10:48 +0000 [r216008-216263]  Russell Bryant <russell@digium.com>

	* doc/IAX2-security.txt (added), /: Merged revisions 216262 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009)
	  | 2 lines Add a plain text version of the IAX2 security document.
	  ........

	* /, UPGRADE.txt: Merged revisions 216080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009)
	  | 2 lines Add a note about IAX2 to UPGRADE.txt. ........

	* /, doc/IAX2-security.pdf (added): Merged revisions 216005 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009)
	  | 2 lines Add IAX2 security document related to AST-2009-006.
	  ........

2009-09-03 18:32 +0000 [r216000]  David Vossel <dvossel@digium.com>

	* channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
	  include/asterisk/acl.h, channels/iax2-parser.h,
	  include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
	  channels/chan_iax2.c: Merge code associated with AST-2009-006
	  (closes issue #12912) Reported by: rathaus Tested by: tilghman,
	  russell, dvossel, dbrooks

2009-09-02 21:41 +0000 [r215682]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Re-send non-100 provisional responses to
	  prevent cancellation From section 13.3.1.1 of RFC 3261: If the
	  UAS desires an extended period of time to answer the INVITE, it
	  will need to ask for an "extension" in order to prevent proxies
	  from canceling the transaction. A proxy has the option of
	  canceling a transaction when there is a gap of 3 minutes between
	  responses in a transaction. To prevent cancellation, the UAS MUST
	  send a non-100 provisional response at every minute, to handle
	  the possibility of lost provisional responses. (closes issue
	  #11157) Reported by: rjain Tested by: twilson Review:
	  https://reviewboard.asterisk.org/r/315/

2009-09-01 23:04 +0000 [r215270]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* apps/app_softhangup.c: Use strrchr() so SoftHangup will correctly
	  truncate multi-hyphen channel names In general channel names are
	  in the form Foo/Bar-Z, but the channel name could have multiple
	  hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
	  channel name at the last hyphen. (closes issue #15810) Reported
	  by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
	  dhubbard (license 733)

2009-08-31 16:16 +0000 [r214940]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Also unlock the "other" channel, when
	  returning, due to glare. (closes issue #15787) Reported by:
	  tim_ringenbach Patches: chan_local.diff uploaded by tim
	  ringenbach (license 540) Tested by: tim_ringenbach

2009-08-28 20:13 +0000 [r214357-214701]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Modify comment to be a bit more accurate. We have
	  kept this comment around long enough, that it's pretty clear that
	  we're keeping the code, because changing the code would require a
	  pretty fundamental architectural shift. We've also taken
	  criticism in some quarters, because it was believed that it was
	  referring to the code being nasty. No, the code isn't nasty, just
	  the operation itself is rather odd. Fixed for eternity (probably
	  not).

	* autoconf/libcurl.m4 (added), configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Use autoconf to
	  detect libcurl, as this enables cross-compilation checks,
	  something we didn't allow before. (closes issue #15714) Reported
	  by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded
	  by tilghman (license 14) Tested by: pprindeville

	* autoconf/ast_ext_lib.m4, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: One more build
	  system change, to make the descriptions look better, if we have
	  better information.

	* autoconf/ast_ext_lib.m4, configure,
	  include/asterisk/autoconfig.h.in: Make autoheader descriptions
	  render correctly in our autoconfig.h file. (Figured out while
	  working with issue #14906)

2009-08-26 16:36 +0000 [r214194]  David Vossel <dvossel@digium.com>

	* main/channel.c: ast_write() ignores ast_audiohook_write() results
	  In ast_write(), if a channel has a list of audiohooks, those
	  lists are written to and the resulting frame is what ast_write()
	  should continue with. The problem was the returned audiohook
	  frame was not being handled at all, and the original frame passed
	  into it did not contain the mixed audio, so essentially audio was
	  being lost. One result of this was chan_spy's whisper mode no
	  longer worked. To complicate the issue, frames passed into
	  ast_write may either be a single frame, or a list of frames. So,
	  as the list of frames is processed in the audiohook_write, the
	  returned frames had to be added to a new list. (closes issue
	  #15660) Reported by: corruptor Tested by: dvossel

2009-08-25 19:28 +0000 [r213899-214069]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: I should always compile before committing...

	* main/say.c: Fix pronunciation of German dates. (closes issue
	  #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
	  by Benjamin Kluck (license 803)

	* main/pbx.c: Improve error message by informing user exactly which
	  function is missing a parethesis. (closes issue #15242) Reported
	  by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
	  dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
	  loloski (license 68)

	* Makefile: Use the default runlevels for Debian derivatives,
	  instead of making up our own. (closes issue #14730) Reported by:
	  pkempgen

2009-08-21 20:23 +0000 [r213631]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Ensure that T.38 INVITEs generated by
	  Asterisk properly result in T.38 being enabled. (closes issue
	  #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by
	  mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni

2009-08-21 16:52 +0000 [r213559]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk.h: Permit DEBUG_FD_LEAKS to be used with C++
	  source files. (closes issue #15698) Reported by: slavon Patches:
	  20090817__issue15698.diff.txt uploaded by tilghman (license 14)
	  Tested by: slavon, tilghman

2009-08-21 16:03 +0000 [r213493]  Jason Parker <jparker@digium.com>

	* configs/queues.conf.sample: Clarify queues.conf comments to
	  specify that variables should be set in the dialplan. (closes
	  issue #15755) Reported by: trendboy

2009-08-20 20:33 +0000 [r213339]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_features.c: Fix a crash by checking the proper pointer
	  for validity before deferencing it. (closes issue #15751)
	  Reported by: atis Patches: ast_bridge_call_peer_cdr.patch
	  uploaded by atis (license 242)

2009-08-20 19:53 +0000 [r213283]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.exports (added): Make all the symbols for the
	  C-client callbacks global

2009-08-19 21:18 +0000 [r213103]  David Vossel <dvossel@digium.com>

	* apps/app_mixmonitor.c: Fixes memory leak caused by incorrectly
	  freeing mixmonitor (closes issue #15699) Reported by: edantie
	  Patches: mixmonitor.patch uploaded by edantie (license 862)

2009-08-18 20:26 +0000 [r212913]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/musiconhold-opsound.txt (added), CREDITS, /, UPGRADE.txt,
	  sounds/sounds.xml, build_tools/prep_tarball,
	  doc/musiconhold-fpm.txt (removed), doc/00README.1st,
	  sounds/Makefile: Convert this branch to Opsound music-on-hold.
	  For more details:
	  http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/

2009-08-18 16:36 +0000 [r212763]  Sean Bright <sean@malleable.com>

	* main/manager.c: Delay the creation of temporary files until we
	  have a valid manager command to handle. Without this patch,
	  asterisk creates a temporary file before determining if the
	  specified command is valid. If invalid, we weren't properly
	  cleaning up the file. (closes issue #15730) Reported by: zmehmood
	  Patches: M15730.diff uploaded by junky (license 177) Tested by:
	  zmehmood

2009-08-18 16:00 +0000 [r212727]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: Removed some deadwood and added some
	  doxygen comments.

2009-08-17 16:34 +0000 [r212498]  Jeff Peeler <jpeeler@digium.com>

	* channels/misdn_config.c: Fix segfault when reloading chan_misdn.
	  If more ports were specified than configured in misdn.conf a
	  reload would crash asterisk. The problem was the unconfigured
	  port was using data from the previously configured port. When the
	  data for an unconfigured port was freed a crash would result from
	  the double free. (closes issue #12113) Reported by: agupta
	  Patches: bug12113.patch uploaded by jpeeler (license 325)

2009-08-17 15:36 +0000 [r212430]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix uninitialized variable.

2009-08-12 23:04 +0000 [r211953]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: This patch adds additional checking when
	  generating queue log TRANSFER events. The additional checks
	  prevent generation of false TRANSFER events in certain
	  situations. (closes issue #14536) Reported by: aragon Patches:
	  queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
	  Tested by: aragon, mnicholson

2009-08-12 18:46 +0000 [r211807]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Backport fix so that outbound CANCEL
	  requests have same branch as challenged INVITEs. There already
	  was code present to be sure that a CANCEL will contain the same
	  branch-id as the INVITE it is cancelling. However, for INVITES
	  which are challenged downstream, this mechanism did not work
	  properly. Now this is taken care of. This is a backport of a fix
	  already present in all 1.6.X branches and in trunk. It also fixes
	  ABE-1907.

2009-08-10 19:48 +0000 [r211528-211583]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES: Conversion specifiers, not format
	  specifiers

	* main/indications.c, main/cli.c, pbx/pbx_loopback.c,
	  channels/chan_dahdi.c, res/res_smdi.c, pbx/pbx_spool.c,
	  channels/chan_skinny.c, pbx/pbx_ael.c, apps/app_dial.c,
	  main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c,
	  funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c,
	  apps/app_disa.c, channels/iax2-provision.c, funcs/func_cut.c,
	  pbx/dundi-parser.c, apps/app_talkdetect.c, channels/chan_misdn.c,
	  apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
	  pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
	  main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
	  doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
	  main/utils.c, apps/app_followme.c, utils/frame.c,
	  channels/misdn_config.c, main/cdr.c, main/channel.c,
	  channels/chan_phone.c, main/manager.c, apps/app_osplookup.c,
	  apps/app_setcallerid.c, res/res_agi.c, apps/app_rpt.c,
	  channels/chan_mgcp.c, apps/app_adsiprog.c, main/dnsmgr.c,
	  channels/chan_sip.c, apps/app_waitforsilence.c, agi/eagi-test.c,
	  main/acl.c, apps/app_queue.c, channels/chan_oss.c,
	  agi/eagi-sphinx-test.c, channels/chan_h323.c, pbx/pbx_dundi.c,
	  apps/app_sms.c, apps/app_verbose.c, apps/app_dahdibarge.c,
	  funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /,
	  res/res_features.c, apps/app_record.c, funcs/func_strings.c,
	  apps/app_random.c, apps/app_alarmreceiver.c,
	  channels/chan_iax2.c: AST-2009-005

2009-08-09 15:41 +0000 [r211274]  Tilghman Lesher <tlesher@digium.com>

	* main/astfd.c: Small oops. Clear the flags which have been
	  checked.

2009-08-07 20:11 +0000 [r211112]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: Resolve a deadlock involving app_chanspy and
	  masquerades. (ABE-1936)

2009-08-07 18:16 +0000 [r210913-211038]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: QUEUE_MEMBER_LIST _really_ wants the interface
	  name, not the membername. This is a partial revert of revision
	  82590, which was an attempted cleanup, but in reality, it broke
	  QUEUE_MEMBER_LIST, which has always been intended as a method by
	  which component interfaces could be queried from the queue.
	  Membername isn't useful here, because that field cannot be used
	  to obtain further information about the member. See the
	  documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
	  QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
	  member argument for further justification. (closes issue #15664)
	  Reported by: rain Patches: app_queue-queue_member_list.diff
	  uploaded by rain (license 327)

	* main/channel.c: Because channel information can be accessed
	  outside of the channel thread, we must lock the channel prior to
	  modifying it. (closes issue #15397) Reported by: caspy Patches:
	  20090714__issue15397.diff.txt uploaded by tilghman (license 14)
	  Tested by: caspy

2009-08-05 19:18 +0000 [r210575]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Dialplan starts execution before the
	  channel setup is complete. * Issue 15655: For the case where
	  dialing is complete for an incoming call, dahdi_new() was asked
	  to start the PBX and then the code set more channel variables. If
	  the dialplan hungup before these channel variables got set,
	  asterisk would likely crash. * Fixed potential for overlap
	  incoming call to erroneously set channel variables as global
	  dialplan variables if the ast_channel structure failed to get
	  allocated. * Added missing set of CALLINGSUBADDR in the dialing
	  is complete case. (closes issue #15655) Reported by: alecdavis

2009-08-05 18:46 +0000 [r210563]  Leif Madsen <lmadsen@digium.com>

	* doc/imapstorage.txt: Update imapstorage.txt documentation.
	  Updated the imapstorage.txt documentation to reflect that issues
	  with c-client versions older than 2007 seem to cause crashing
	  issues that are not seen with more recent versions. Documentation
	  has been updated to reflect this. (closes issue #14496) Reported
	  by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
	  uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
	  dbrooks

2009-08-04 14:51 +0000 [r210237]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Eliminate spurious compiler warnings from system
	  headers on *BSD platforms. Ensure that system headers located in
	  /usr/local/include are actually treated as system headers by the
	  compiler, and not as local headers which are subject to warnings
	  from the -Wundef compiler option and others. (closes issue
	  #15606) Reported by: mvanbaak

2009-08-03 16:15 +0000 [r210067]  David Brooks <dbrooks@digium.com>

	* channels/chan_dahdi.c: Fixes dialplan wildcard extension taking
	  precedence over call pickup code. Prior to this patch, a wildcard
	  extension in the dialplan (for example, _*.) would take
	  precedence over picking up a call in the channel's pickup group.
	  This patch simply moves the block of code handling pickup group
	  matching to above the extension matching code. (closes issue
	  #14735) Reported by: stevedavies Review:
	  https://reviewboard.asterisk.org/r/319/

2009-08-03 16:11 +0000 [r210064-210066]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_oss.c, apps/app_playback.c, main/asterisk.exports,
	  configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, main/strcompat.c, configure.ac,
	  funcs/func_cut.c: Reverting index() fix, applying a different
	  methodology, based upon developer discussions. (related to issue
	  #15639)

	* main/asterisk.exports, include/asterisk/compat.h: Helps if we
	  export the index() function. (Related to issue #15639)

	* configure, include/asterisk/autoconfig.h.in, main/strcompat.c,
	  configure.ac: Apparently, some platforms don't have the index()
	  function. (closes issue #15639) Reported by: nmav

2009-08-01 11:27 +0000 [r209838-209879]  Russell Bryant <russell@digium.com>

	* main/db1-ast/mpool/mpool.c: Resolve a valgrind warning about a
	  read from uninitialized memory. (issue #15396) Reported by:
	  aragon

	* apps/app_milliwatt.c: Modify how Playtones() is used in
	  Milliwatt() to resolve gain issue. When Milliwatt() was changed
	  internally to use Playtones() so that the proper tone was used,
	  it introduced a drop in gain in the output signal. So, use the
	  playtones API directly and specify a volume argument such that
	  the output matches the gain of the original Milliwatt() code.
	  (closes issue #15386) Reported by: rue_mohr Patches:
	  issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
	  rue_mohr

2009-08-01 00:52 +0000 [r209759]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c, utils/frame.c, main/Makefile,
	  channels/misdn/ie.c: Minor changes inspired by testing with
	  latest GCC. The latest GCC (what will become 4.5.x) has a few new
	  warnings, that in these cases found some either downright buggy
	  code, or at least seriously poorly designed code that could be
	  improved.

2009-07-28 00:12 +0000 [r209315]  Tilghman Lesher <tlesher@digium.com>

	* sounds/sounds.xml: Publish French extra sounds

2009-07-27 17:44 +0000 [r209131]  Mark Michelson <mmichelson@digium.com>

	* main/udptl.c, configs/udptl.conf.sample: Allow for UDPTL to use
	  only even-numbered ports if desired. There are some VoIP
	  providers out there that will not accept SDP offers with odd
	  numbered UDPTL ports. While it is my personal opinion that these
	  VoIP providers are misinterpreting RFC 2327, it really is not a
	  big deal to play along with their silly little games. Of course,
	  since restricting UDPTL ports to only even numbers reduces the
	  range of available ports by half, so the option to use only even
	  port numbers is off by default. A user can enable the behavior by
	  setting use_even_ports=yes in udptl.conf. (closes issue #15182)
	  Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson
	  (license 60) Tested by: CGMChris

2009-07-27 09:56 +0000 [r208990]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_crypto.c: backport rev 205532 from trunk: pthread_self
	  returns a pthread_t which is not an unsigned int on all pthread
	  implementations. Casting it to an unsigned int fixes compiler
	  warnings.

2009-07-27 01:18 +0000 [r208923]  Jeff Peeler <jpeeler@digium.com>

	* main/translate.c, channels/chan_iax2.c: Fix logic errors from
	  208746

2009-07-25 06:19 +0000 [r208746]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_skinny.c, main/translate.c, channels/chan_iax2.c:
	  Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial
	  changes, but I did not know of any other way to fix the
	  "dereferencing type-punned pointer will break strict-aliasing
	  rules" error without creating a tmp variable in chan_skinny.

2009-07-24 19:24 +0000 [r208622]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Don't impose an arbitrary limit on member lines
	  in queues.conf I know what some of you are thinking: "UGH! Mark,
	  why are you using ast_strdup and ast_free for the string when you
	  can just use ast_strdupa and let the memory free itself?! Have
	  the bats been chewing on your brain again?" Based on past
	  experiences, I don't like using ast_strdupa inside a loop. It's a
	  good way to potentially exhaust stack space. Also, since this
	  only happens when reloading queues, I don't think that heap
	  allocations and frees are going to be a huge problem. (closes
	  issue #15559) Reported by: amorsen

2009-07-24 18:38 +0000 [r208592]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: Do not log an ERROR if autoservice_stop()
	  returns -1. This does not indicate an error. A return of -1 just
	  means that the channel has been hung up. (reported in
	  #asterisk-dev)

2009-07-24 18:26 +0000 [r208386-208587]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Only send a BYE when hanging up a channel
	  that is up. For cases where Asterisk sends an INVITE and receives
	  a non 2XX final response, Asterisk would follow the INVITE
	  transaction by immediately sending a BYE, which was unnecessary.
	  (closes issue #14575) Reported by: chris-mac

	* channels/chan_sip.c: Fix a problem where a 491 response could be
	  sent out of dialog. This generalizes the fix for issue 13849. The
	  initial fix corrected the problem that Asterisk would reply with
	  a 491 if a reinvite were received from an endpoint and we had not
	  yet received an ACK from that endpoint for the initial INVITE it
	  had sent us. This expansion also allows Asterisk to appropriately
	  handle an INVITE with authorization credentials if Asterisk had
	  not received an ACK from the previous transaction in which
	  Asterisk had responded to an unauthorized INVITE with a 407.
	  (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
	  uploaded by mmichelson (license 60) Tested by: klaus3000

2009-07-23 19:19 +0000 [r208380]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Only set the priindication setting when
	  not performing a reload (closes issue #14696) Reported by:
	  fdecher

2009-07-23 16:29 +0000 [r208262-208312]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove inaccurate XXX comment.

	* channels/chan_sip.c: Properly handle 183 responses which do not
	  contain an SDP. (closes issue #15442) Reported by: ffloimair
	  Patches: 15442.patch uploaded by mmichelson (license 60) Tested
	  by: tkarl, ffloimair

2009-07-22 20:23 +0000 [r207945-208083]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.exports, include/asterisk/compat.h: Export symbols
	  for functions included in our compatibility headers. (closes
	  issue #15556) Reported by: smw1218

	* funcs/func_strings.c: Force an error if a blank is passed to
	  QUOTE (because the documentation states the argument is not
	  optional). This change makes URIENCODE and QUOTE behave
	  similarly, since the documentation states that the argument is
	  not optional, for both. (closes issue #15439) Reported by:
	  pkempgen Patches: 20090706__issue15439.diff.txt uploaded by
	  tilghman (license 14)

2009-07-21 20:16 +0000 [r207786-207827]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Wait for wink before dialing when using
	  E&M wink signaling There was already code for other signaling
	  types in dahdi_handle_event to handle dialing if a dial operation
	  dial string was present. Simply add SIG_EMWINK to the list.
	  (closes issue #14434) Reported by: araasch

	* channels/chan_dahdi.c: Revert r207573, this approach could
	  potentially block for an unacceptable amount of time.

2009-07-21 14:26 +0000 [r207714]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Document default timeout for AMI originations.
	  AST-224

2009-07-21 13:04 +0000 [r207647]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/lpc10/Makefile, main/db1-ast/Makefile, Makefile,
	  agi/Makefile, codecs/Makefile, utils/Makefile, main/Makefile,
	  codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
	  pbx/Makefile, res/Makefile, channels/Makefile: Ensure that
	  user-provided CFLAGS and LDFLAGS are honored. This commit changes
	  the build system so that user-provided flags (in ASTCFLAGS and
	  ASTLDFLAGS) are supplied to the compiler/linker *after* all flags
	  provided by the build system itself, so that the user can
	  effectively override the build system's flags if desired. In
	  addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either*
	  in the environment before running 'make', or as variable
	  assignments on the 'make' command line. As a result, the use of
	  COPTS and LDOPTS is no longer necessary, so they are no longer
	  documented, but are still supported so as not to break existing
	  build systems that supply them when building Asterisk.

2009-07-20 23:23 +0000 [r207573]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Wait for wink before dialing when using
	  E&M wink signaling This patch adds a new dahdi_wait function to
	  specifically wait for the wink event. If the wink is not
	  eventually received the channel is hung up. (closes issue #14434)
	  Reported by: araasch Patches: emwinkmod uploaded by araasch
	  (license 693)

2009-07-20 19:39 +0000 [r207423]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Answer video SDP offers properly when
	  videosupport is not enabled. Copied from Review board: In issue
	  12434, the reporter describes a situation in which audio and
	  video is offered on the call, but because videosupport is
	  disabled in sip.conf, Asterisk gives no response at all to the
	  video offer. According to RFC 3264, all media offers should have
	  a corresponding answer. For offers we do not intend to actually
	  reply to with meaningful values, we should still reply with the
	  port for the media stream set to 0. In this patch, we take note
	  of what types of media have been offered and save the information
	  on the sip_pvt. The SDP in the response will take into account
	  whether media was offered. If we are not otherwise going to
	  answer a media offer, we will insert an appropriate m= line with
	  the port set to 0. It is important to note that this patch is
	  pretty much a bandage being applied to a broken bone. The patch
	  *only* helps for situations where video is offered but
	  videosupport is disabled and when udptl_pt is disabled but T.38
	  is offered. Asterisk is not guaranteed to respond to every media
	  offer. Notable cases are when multiple streams of the same type
	  are offered. The 2 media stream limit is still present with this
	  patch, too. In trunk and the 1.6.X branches, things will be a bit
	  different since Asterisk also supports text in SDPs as well.
	  (closes issue #12434) Reported by: mnnojd Review:
	  https://reviewboard.asterisk.org/r/311 Review:
	  https://reviewboard.asterisk.org/r/313

2009-07-20 16:26 +0000 [r207360]  Russell Bryant <russell@digium.com>

	* main/channel.c: Only do the chan->fdno check in ast_read() in a
	  developer build. I changed this check to only happen in a
	  dev-mode build. I also added a comment explaining what is going
	  on. I also made it so that detection of this situation does not
	  affect ast_read() operation. (closes issue #14723) Reported by:
	  seadweller

2009-07-20  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26

2009-07-13  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc6

2009-07-13 15:12 +0000 [r206126]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Print CID match in "show dialplan". (closes issue
	  #14702) Reported by: klaus3000 Patches:
	  patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000
	  (license 65)

2009-07-10 17:39 +0000 [r205877]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Properly ACK 487 responses to canceled
	  INVITEs. From the review board request: The fix from review 298
	  has exposed a new bug in chan_sip. When we hang up an outgoing
	  call, we first will dump all the outstanding packets on the
	  sip_pvt using __sip_pretend_ack. Then, if we can, we send a
	  CANCEL. The problem with this is that since destroyed all the
	  outstanding packets on the dialog, we cannot match the incoming
	  487 response to our INVITE. Because we cannot match the response,
	  we do not send an ACK. To correct this, instead of using
	  __sip_pretend_ack, I have changed the code to loop through the
	  list of packets and call __sip_semi_ack on each one instead. This
	  causes us to stop retransmitting the requests, but we still have
	  them around in case we get responses for them. When discussing
	  this earlier today with Josh Colp, we both agreed that in the
	  majority of cases, this would be enough of a fix. However, we
	  also agreed that we should have a safety net in place in case we
	  never receive a response to our initial INVITE. To handle this, I
	  have modified __sip_autodestruct to behave similar to the way it
	  does in Asterisk 1.4. If there are outstanding packets on the
	  sip_pvt, the needdestroy flag is not set, and the last request on
	  the dialog was either a CANCEL or BYE, then we set the
	  needdestroy flag and reschedule destruction for 10 seconds in the
	  future. If, though, the needdestroy flag is set, then we use
	  __sip_pretend_ack to kill the remaining outstanding packets so
	  that the monitor thread can destroy the sip_pvt. I ran two
	  separate tests. First, I placed a call from my Aastra phone to my
	  Polycom phone. I hung up the Aastra before the Polycom answered.
	  I verified through sip debug output that Asterisk properly ACKed
	  the 487 received from the Polycom. For my second test, I set up a
	  SIPp UAS scenario so that it would send a 200 OK in response to a
	  CANCEL but would not send a 487 for the original INVITE. I
	  verified that after about 40 seconds, Asterisk properly cleans up
	  the outgoing sip_pvt for the call. Review:
	  https://reviewboard.asterisk.org/r/308

2009-07-10 16:23 +0000 [r205804]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP registration auth loop caused by stale
	  nonce If an endpoint sends two registration requests in a very
	  short period of time with the same nonce, both receive 401
	  responses from Asterisk, each with a different nonce (the second
	  401 containing the current nonce and the first one being stale).
	  If the endpoint responds to the first 401, it does not match the
	  current nonce so Asterisk sends a third 401 with a newly
	  generated nonce (which updates the current nonce)... Now if the
	  endpoint responds to the second 401, it does not match the
	  current nonce either and Asterisk sends a fourth 401 with a newly
	  generated nonce... This loop goes on and on. There appears to be
	  a simple fix for this. If the nonce from the request does not
	  match our nonce, but is a good response to a previous nonce,
	  instead of sending a 401 with a newly generated nonce, use the
	  current one instead. This breaks the loop as the nonce is not
	  updated until a response is received. Additional logic has been
	  added to make sure no nonce can be responded to twice though.
	  (closes issue #15102) Reported by: Jamuel Patches:
	  patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff
	  uploaded by dvossel (license 671) Tested by: Jamuel Review:
	  https://reviewboard.asterisk.org/r/289/

2009-07-10 15:51 +0000 [r205775]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Ensure that outbound NOTIFY requests are
	  properly routed through stateful proxies. With this change, we
	  make note of Record-Route headers present in any SUBSCRIBE
	  request that we receive so that our outbound NOTIFY requests will
	  have the proper Route headers in them. (closes issue #14725)
	  Reported by: ibc

2009-07-09 23:37 +0000 [r205728]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: No audio on calls from Asterisk to various
	  ISDN devices until DTMF sent by caller. Add missing clearing of
	  the dialing flag when the ISDN call is CONNECTED. (i.e. When
	  libpri generates the event PRI_EVENT_ANSWER.) (closes issue
	  #15420) Reported by: scottbmilne Patches:
	  bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
	  Tested by: scottbmilne, alecdavis (closes issue #15416) Reported
	  by: avinoash (closes issue #15389) Reported by: alecdavis This
	  patch should also fix the following issue: (issue #15205)
	  Reported by: vinsik

2009-07-09 16:18 +0000 [r205409-205599]  David Vossel <dvossel@digium.com>

	* include/asterisk/time.h: Changing ast_samp2tv to not use floating
	  point.

	* main/rtp.c, channels/chan_iax2.c, include/asterisk/frame.h: Fixes
	  8khz assumptions Many calculations assume 8khz is the codec rate.
	  This is not always the case. This patch only addresses chan_iax.c
	  and res_rtp_asterisk.c, but I am sure there are other areas that
	  make this assumption as well. Review:
	  https://reviewboard.asterisk.org/r/306/

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
	  include/asterisk/pbx.h: moving ast_devstate_to_extenstate to
	  pbx.c from devicestate.c ast_devstate_to_extenstate belongs in
	  pbx.c. This change fixes a compile time error with chan_vpb as
	  well.

2009-07-08 19:26 +0000 [r205349]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Prevent phantom calls to queue members. If a
	  caller were to hang up while a periodic announcement or position
	  were being said, the return value for those functions would
	  incorrectly indicate that the caller was still in the queue. With
	  these changes, the problem does not occur. (closes issue #14631)
	  Reported by: latinsud Patches: queue_announce_ghost_call2.diff
	  uploaded by latinsud (license 745) (with small modification from
	  me)

2009-07-08 18:19 +0000 [r205288]  Jason Parker <jparker@digium.com>

	* config.guess, config.sub: Update config.guess and config.sub from
	  the savannah.gnu.org git repo.

2009-07-08 16:53 +0000 [r205215]  David Vossel <dvossel@digium.com>

	* include/asterisk/time.h: ast_samp2tv needs floating point for
	  16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate
	  is 16000. The .5 is currently stripped off because we don't
	  calculate using floating points. This causes madness with 16khz
	  audio. (issue ABE-1899) Review:
	  https://reviewboard.asterisk.org/r/305/

2009-07-08 16:26 +0000 [r205188]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Add redirection warnings for the invalid language
	  codes previously removed.

2009-07-08 15:54 +0000 [r205149]  Russell Bryant <russell@digium.com>

	* res/res_crypto.c: Make OpenSSL usage thread-safe. OpenSSL is not
	  thread-safe by default. However, making it thread safe is very
	  easy. We just have to provide a couple of callbacks. One callback
	  returns a thread ID. The other handles locking. For more
	  information, start with the "Is OpenSSL thread-safe?" question on
	  the FAQ page of openssl.org.

2009-07-02 21:59 +0000 [r204834]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Removed confusing warning message "Got
	  Busy in Connected State" If an incoming mISDN call is answered
	  with the Answer application and a subsequent Dial gets a busy
	  endpoint then it is valid for that already connected channel to
	  get the busy indication. Asterisk will play the busy tones until
	  the dialplan plays something else or hangs up the call. (closes
	  issue #11974) Reported by: fvdb

2009-07-02 18:15 +0000 [r204755]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
	  include/asterisk/pbx.h: moving device state functions from pbx.h
	  to devicestate.h to sync with other branches

2009-07-02  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc5

2009-07-02 15:05 +0000 [r204681]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
	  include/asterisk/pbx.h: Improved mapping of extension states from
	  combined device states. This fixes a few issues with incorrect
	  extension states and adds a cli command, core show
	  device2extenstate, to display all possible state mappings.
	  (closes issue #15413) Reported by: legart Patches:
	  exten_helper.diff uploaded by dvossel (license 671) Tested by:
	  dvossel, legart, amilcar Review:
	  https://reviewboard.asterisk.org/r/301/

2009-06-30 20:23 +0000 [r204556]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c, UPGRADE.txt: More incorrect language codes, plus
	  ensuring that regionalizations use the specified language, and
	  not English for grammar. (closes issue #15022) Reported by:
	  greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by
	  tilghman (license 14)

2009-06-30 18:47 +0000 [r204474]  Jason Parker <jparker@digium.com>

	* main/say.c: Fix ast_say_counted_noun to correctly handle Polish.
	  Fix a comment typo in passing.

2009-06-30 18:23 +0000 [r204469]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c, UPGRADE.txt: "tw" is the language specification for
	  Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported
	  by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded
	  by tilghman (license 14) 20090617__issue15346__trunk.diff.txt
	  uploaded by tilghman (license 14)
	  20090617__issue15346__1.6.0.diff.txt uploaded by tilghman
	  (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by
	  tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt
	  uploaded by tilghman (license 14) Tested by: volivier

2009-06-29 22:45 +0000 [r204243-204300]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add error message so that it is clear why a
	  SIP peer was not processed when a DNS lookup fails on a host or
	  outboundproxy. (closes issue #13432) Reported by: p_lindheimer
	  Patches: outboundproxy.patch uploaded by p (license 558)

	* channels/chan_sip.c: Fix build oops.

	* channels/chan_sip.c: Fix a problem where chan_sip would ignore
	  "old" but valid responses. chan_sip has had a problem for quite a
	  long time that would manifest when Asterisk would send multiple
	  SIP responses on the same dialog before receiving a response. The
	  problem occurred because chan_sip only kept track of the highest
	  outgoing sequence number used on the dialog. If Asterisk sent two
	  requests out, and a response arrived for the first request sent,
	  then Asterisk would ignore the response. The result was that
	  Asterisk would continue retransmitting the requests and ignoring
	  the responses until the maximum number of retransmissions had
	  been reached. The fix here is to rearrange the code a bit so that
	  instead of simply comparing the sequence number of the response
	  to our latest outgoing sequence number, we walk our list of
	  outstanding packets and determine if there is a match. If there
	  is, we continue. If not, then we ignore the response. In doing
	  this, I found a few completely useless variables that I have now
	  removed. (closes issue #11231) Reported by: flefoll Review:
	  https://reviewboard.asterisk.org/r/298

2009-06-29 19:36 +0000 [r204170]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c, funcs/func_strings.c: Revision 189537 was
	  supposed to make 1.4 more correct. Instead, it broke func_odbc.
	  Reverting. (closes issue #15317, issue #14614)

2009-06-29 17:04 +0000 [r204067]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: segfault after SPINLOCK schedule delete
	  Using the SPINLOCK schedule delete macro can result in the
	  iax_pvt lock being given up. This makes it possible for the
	  iax_pvt to dissappear when we thought we held the mutex the
	  entire time. To resolve this, the iax_pvt's ref count is
	  incremented. (closes issue #15377) Reported by: aragon Patches:
	  iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested
	  by: aragon, dvossel

2009-06-29 15:04 +0000 [r204012]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Place unlock of mutex in an else block so
	  that it does not get unlocked twice. (closes issue #15400)
	  Reported by: aragon

2009-06-27 00:55 +0000 [r203908]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: The ISDN CPE side should not exclusively
	  pick B channels normally. Before this patch, Asterisk
	  unconditionally picked B channels exclusively on the CPE side and
	  normally allowed alternative B channels on the network side. Now
	  Asterisk does the opposite. Reasons for the CPE side to normally
	  not pick B channels exclusively: * For CPE point-to-multipoint
	  mode (i.e. phone side), the CPE side does not have enough
	  information to exclusively pick B channels. (There may be other
	  devices on the line.) * Q.931 gives preference to the network
	  side picking B channels. * Some telcos require the CPE side to
	  not pick B channels exclusively. (closes issue #14383) Reported
	  by: mbrancaleoni

2009-06-26 22:09 +0000 [r203848]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Make sure to recreate the dahdi pseudo
	  channel after dahdi restart (closes issue #14477) Reported by:
	  timking

2009-06-26 21:16 +0000 [r203785]  Russell Bryant <russell@digium.com>

	* main/file.c: Don't fast forward past the end of a message. This
	  is nice change for users of the voicemail application. If someone
	  gets a little carried away with fast forwarding through a
	  message, they can easily get to the end and accidentally exit the
	  voicemail application by hitting the fast forward key during the
	  following prompt. This adds some safety by not allowing a fast
	  forward past the end of a message. (closes issue #14554) Reported
	  by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
	  707) Tested by: lacoursj

2009-06-26 20:03 +0000 [r203719]  David Brooks <dbrooks@digium.com>

	* apps/app_voicemail.c: Fixing voicemail's error in checking max
	  silence vs min message length Max silence was represented in
	  milliseconds, yet vmminsecs (minmessage) was represented as
	  seconds. Also, the inequality was reversed. The warning, if
	  triggered, was "Max silence should be less than minmessage or you
	  may get empty messages", which should have been logged if max
	  silence was greater than minmessage, but the check was for less
	  than. Also, conforming if statement to coding guidelines. closes
	  issue #15331) Reported by: markd Review:
	  https://reviewboard.asterisk.org/r/293/

2009-06-25 21:13 +0000 [r203380]  Terry Wilson <twilson@digium.com>

	* main/cli.c: I didn't see that Mark already fixed the underlying
	  issue! Yay for removing useless code.

2009-06-25 21:02 +0000 [r203375]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Fix a case where CDR answer time could be
	  before the start time involving parking. (closes issue #13794)
	  Reported by: davidw Patches: 13794.patch uploaded by murf
	  (license 17) 13794.patch.160 uploaded by murf (license 17) Tested
	  by: murf, dbrooks

2009-06-25 20:09 +0000 [r203311]  Terry Wilson <twilson@digium.com>

	* main/cli.c: Don't try to free NULL

2009-06-25 18:52 +0000 [r203230]  Mark Michelson <mmichelson@digium.com>

	* main/astmm.c: Prevent false positives when freeing a NULL pointer
	  with MALLOC_DEBUG enabled.

2009-06-25 16:02 +0000 [r203115]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Resolve a crash related to a T.38 reinvite
	  race condition. This change resolves a crash observed locally
	  during some T.38 testing. A call was set up using a call file,
	  and when the T.38 reinvite came in, the channel state was still
	  AST_STATE_DOWN. The reason is explained by a comment in the code
	  that previously lived in the handling of AST_STATE_RINGING. This
	  change modifies the logic to handle the same race condition for
	  any channel state that is not UP. (closes ABE-1895)

2009-06-24 21:01 +0000 [r203036]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Improved chan_dahdi.conf pritimer error
	  checking. Valid format is: pritimer=timer_name,timer_value *
	  Fixed segfault if the ',' is missing. * Completely check the
	  range returned by pri_timer2idx() to prevent possible access
	  outside array bounds.

2009-06-24 18:28 +0000 [r202966]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Use the handy UNLINK macro instead of
	  hand-coding the same thing in-line.

2009-06-24  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc4 

2009-06-23 16:28 +0000 [r202671]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: MWI NOTIFY contains a wrong URI if Asterisk
	  listens to non-standard port and transport (closes issue #14659)
	  Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt
	  uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff
	  uploaded by dvossel (license 671) Tested by: dvossel, klaus3000
	  Review: https://reviewboard.asterisk.org/r/288/

2009-06-23 15:22 +0000 [r202572-202601]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix more memory leaks that may result if rtp
	  is not successfully allocated.

	* channels/chan_sip.c: Fix potential memory leak in chan_sip when
	  video rtp is not allocated properly.

2009-06-22 20:08 +0000 [r202414-202496]  Russell Bryant <russell@digium.com>

	* main/channel.c: Report CallerID change during a masquerade.
	  Reported by: markster

	* channels/chan_sip.c: Make Polycom subscription type override
	  check more explicit.

2009-06-22 14:44 +0000 [r202336-202342]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove an extra debug line left from
	  previous commit.

	* channels/chan_sip.c: Fix a situation in which Asterisk would not
	  stop retransmitting 487s. If a CANCEL were received by Asterisk,
	  we would send a 487 in response to the original INVITE and a 200
	  OK for the CANCEL. If there were a network hiccup which caused
	  the 200 OK and the 487 to be lost, then the UA communicating with
	  Asterisk may try to retransmit its CANCEL. Asterisk's response to
	  this used to be to try sending another 487 to the canceled INVITE
	  and another 200 OK to the CANCEL. The problem here is that the
	  originally-sent 487 was sent "reliably" meaning that it will be
	  retransmitted until it is received properly. So when we receive
	  the second CANCEL it is likely that the first batch of 487s we
	  sent is still going strong and reaches the UA. The result was
	  that the second set of 487s would be retransmitted constantly
	  until the maximum number of retries had been reached. The fix for
	  this is that if we receive a second CANCEL for an INVITE, then we
	  cancel the retransmission of the first set of 487s and start a
	  second set. This causes the dialog to be terminated reasonably.
	  (closes issue #14584) Reported by: klaus3000 Patches:
	  14584_v2.patch uploaded by mmichelson (license 60) Tested by:
	  klaus3000

	* channels/chan_sip.c: Fix a possible infinite loop in SDP parsing
	  during glare situation. There was a while loop in
	  get_ip_and_port_from_sdp which was controlled by a call to
	  get_sdp_iterate. The loop would exit either if what we were
	  searching for was found or if the return was NULL. The problem is
	  that get_sdp_iterate never returns NULL. This means that if what
	  we were searching for was not present, the loop would run
	  infinitely. This modification of the loop fixes the problem.
	  (closes issue #15213) Reported by: schmidts (closes issue #15349)
	  Reported by: samy (closes issue #14464) Reported by: pj (closes
	  issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
	  uploaded by mmichelson (license 60) Tested by: aragon

2009-06-20 17:51 +0000 [r202153]  Sean Bright <sean@malleable.com>

	* channels/chan_sip.c: Since we don't have sip_pvt_lock() in 1.4,
	  we need to use ast_mutex_* directly. (closes issue #15366)
	  Reported by: loloski

2009-06-19 21:21 +0000 [r202022]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Added deadlock protection to
	  try_suggested_sip_codec in chan_sip.c. Review:
	  https://reviewboard.asterisk.org/r/287/

2009-06-19 20:22 +0000 [r201993]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: timestamp was being converted to host order
	  as a short rather than a long (closes issue #15361) Reported by:
	  ffloimair Patches: ts_issue.diff uploaded by dvossel (license
	  671)

2009-06-19 00:40 +0000 [r201828]  Tilghman Lesher <tlesher@digium.com>

	* res/res_features.c: If the "h" extension fails, give it another
	  chance in main/pbx.c. If the "h" extension fails, give it another
	  chance in main/pbx.c, when it returns from the bridge code. Fixes
	  an issue where the "h" extension may occasionally not fire, when
	  a Dial is executed from a Macro. Debugged in #asterisk with user
	  tompaw.

2009-06-18  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc3

2009-06-18 15:24 +0000 [r201600]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix memory corruption and leakage related
	  reloads of non files mode MoH classes. For Music on Hold classes
	  that are not files mode, meaning that we are executing an
	  application that will feed us audio data, we use a thread to
	  monitor the external application and read audio from it. This
	  thread also makes use of the MoH class object. In the MoH class
	  destructor, we used pthread_cancel() to ask the thread to exit.
	  Unfortunately, the code did not wait to ensure that the thread
	  actually went away. What needed to be done is a pthread_join() to
	  ensure that the thread fully cleans up before we proceed. By
	  adding this one line, we resolve two significant problems: 1)
	  Since the thread was never joined, it never fully goes away. So,
	  on every reload of non-files mode MoH, an unused thread was
	  sticking around. 2) There was a race condition here where the
	  application monitoring thread could still try to access the MoH
	  class, even though the thread executing the MoH reload has
	  already destroyed it. (issue #15109) Reported by: jvandal (issue
	  #15123) Reported by: axisinternet (issue #15195) Reported by:
	  amorsen (issue AST-208)

2009-06-17 19:59 +0000 [r201450]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Change the datastore traversal in
	  ast_do_masquerade to use a safe list traversal. It is possible
	  for datastore fixup functions to remove the datastore from the
	  list and free it. In particular, the queue_transfer_fixup in
	  app_queue does this. While I don't yet know of this causing any
	  crashes, it certainly could. Found while discussing a separate
	  issue with Brian Degenhardt.

2009-06-17 19:28 +0000 [r201423]  David Vossel <dvossel@digium.com>

	* apps/app_mixmonitor.c: StopMixMonitor race condition (not giving
	  up file immediately) StopMixMonitor only indicates to the
	  MixMonitor thread to stop writing to the file. It does not
	  guarantee that the recording's file handle is available to the
	  dialplan immediately after execution. This results in a race
	  condition. To resolve this, the filestream pointer is placed in a
	  datastore on the channel. When StopMixMonitor is called, the
	  datastore is retrieved from the channel and the filestream is
	  closed immediately before returning to the dialplan.
	  Documentation indicating the use of StopMixMonitor to free files
	  has been updated as well. (closes issue #15259) Reported by:
	  travisghansen Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/283/

2009-06-17 18:45 +0000 [r201380]  David Brooks <dbrooks@digium.com>

	* channels/chan_sip.c: Checks for NULL sip_pvt pointer in
	  chan_sip.c->acf_channel_read() Zombie channels could be passed,
	  and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
	  checking for NULL pointer. (closes issue #15330) Reported by:
	  okrief Tested by: dbrooks

2009-06-17 12:03 +0000 [r200991-201261]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/linkedlists.h: Correct AST_LIST_APPEND_LIST
	  behavior when list to be appended is empty. When the list to be
	  appended is empty, and the list to be appended to is *not*,
	  AST_LIST_APPEND_LIST would actually cause the target list to
	  become broken, and no longer have a pointer to its last entry.
	  This patch fixes the problem. (reported by Stanislaw Pitucha on
	  the asterisk-dev mailing list)

	* apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
	  build_tools/cflags-devmode.xml, main/autoservice.c, main/frame.c,
	  apps/app_meetme.c, main/slinfactory.c,
	  include/asterisk/linkedlists.h, main/file.c,
	  include/asterisk/channel.h, include/asterisk/frame.h: Improve
	  support for media paths that can generate multiple frames at
	  once. There are various media paths in Asterisk (codec
	  translators and UDPTL, primarily) that can generate more than one
	  frame to be generated when the application calling them expects
	  only a single frame. This patch addresses a number of those
	  cases, at least the primary ones to solve the known problems. In
	  addition it removes the broken TRACE_FRAMES support, fixes a
	  number of bugs in various frame-related API functions, and cleans
	  up various code paths affected by these changes.
	  https://reviewboard.asterisk.org/r/175/

2009-06-16 13:25 +0000 [r200875]  Eliel C. Sardanons <eliels@gmail.com>

	* res/res_smdi.c: Show the interface name on error, if it is not
	  found. If the smdiport specified is not found, show the interface
	  name instead of '(null)'.

2009-06-15 15:21 +0000 [r200513]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add INFO to our allowed methods so that
	  endpoints know they may send it to us. AST-223

2009-06-12 19:06 +0000 [r200360]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Suppress a warning message and give a better
	  return code when generating inband ringing after a call is
	  answered. (closes issue #15158) Reported by: madkins Patches:
	  15158.patch uploaded by mmichelson (license 60) Tested by:
	  madkins

2009-06-11 22:20 +0000 [r200185]  Sean Bright <sean.bright@gmail.com>

	* Makefile: Backport fix for parallel build warnings from trunk
	  r199781.

2009-06-11 12:12 +0000 [r200037]  Leif Madsen <lmadsen@digium.com>

	* build_tools/make_version_h: Fix path for .flavor and .version.
	  (issue #14737) Reported by: davidw Patches: flavor.patch uploaded
	  by davidw (license 780) Tested by: davidw

2009-06-10 16:08 +0000 [r199856]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/utils.h: __WORDSIZE is not available on all
	  platforms, so use sizeof(void *) instead.

2009-06-09  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc2

2009-06-08 19:28 +0000 [r199626-199628]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/utils.h: Fix a typo in the stack size
	  calculation just introduced.

	* include/asterisk/utils.h: Increase the size of our thread stack
	  on 64 bit processors. We were setting the stack size for each
	  thread to 240KB regardless of architecture, which meant that in
	  some scenarios we actually had less available stack space on 64
	  bit processors (pointers use 8 bytes instead of 4). So now we
	  calculate the stack size we reserve based on the platform's
	  __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
	  bit -> 1008KB (that's right, we're ready for 128 bit processors)
	  Patch typed by me but written by several members of
	  #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
	  issue #14932) Reported by: jpiszcz Patches:
	  06052009_issue14932.patch uploaded by seanbright (license 71)
	  Tested by: seanbright

2009-06-05 21:19 +0000 [r199297]  David Vossel <dvossel@digium.com>

	* main/pbx.c: Fixes issue with hints giving unexpected results.
	  Hints with two or more devices that include ONHOLD gave
	  unexpected results. (closes issue #15057) Reported by:
	  p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
	  (license 671) pbx.c.1.4.patch uploaded by p (license 558)
	  devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
	  p_lindheimer, dvossel Review:
	  https://reviewboard.asterisk.org/r/254/

2009-06-04 19:00 +0000 [r199138]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Additional updates to AST-2009-001

2009-06-04 14:14 +0000 [r198957-199022]  Sean Bright <sean.bright@gmail.com>

	* main/asterisk.c, main/loader.c, include/asterisk.h: Safely handle
	  AMI connections/reload requests that occur during startup. During
	  asterisk startup, a lock on the list of modules is obtained by
	  the primary thread while each module is initialized. Issue 13778
	  pointed out a problem with this approach, however. Because the
	  AMI is loaded before other modules, it is possible for a module
	  reload to be issued by a connected client (via Action: Command),
	  causing a deadlock. The resolution for 13778 was to move
	  initialization of the manager to happen after the other modules
	  had already been lodaded. While this fixed this particular issue,
	  it caused a problem for users (like FreePBX) who call AMI scripts
	  via an #exec in a configuration file (See issue 15189). The
	  solution I have come up with is to defer any reload requests that
	  come in until after the server is fully booted. When a call comes
	  in to ast_module_reload (from wherever) before we are fully
	  booted, the request is added to a queue of pending requests. Once
	  we are done booting up, we then execute these deferred requests
	  in turn. Note that I have tried to make this a bit more
	  intelligent in that it will not queue up more than 1 request for
	  the same module to be reloaded, and if a general reload request
	  comes in ('module reload') the queue is flushed and we only issue
	  a single deferred reload for the entire system. As for how this
	  will impact existing installations - Before 13778, a reload
	  issued before module initialization was completed would result in
	  a deadlock. After 13778, you simply couldn't connect to the
	  manager during startup (which causes problems with
	  #exec-that-calls-AMI configuration files). I believe this is a
	  good general purpose solution that won't negatively impact
	  existing installations. (closes issue #15189) (closes issue
	  #13778) Reported by: p_lindheimer Patches:
	  06032009_15189_deferred_reloads.diff uploaded by seanbright
	  (license 71) Tested by: p_lindheimer, seanbright Review:
	  https://reviewboard.asterisk.org/r/272/

	* pbx/pbx_spool.c: Fix a possible crash in pbx_spool. We were
	  trying to reference members of a struct that had previously been
	  freed. This patch makes sure that we free the struct after it has
	  been removed from the spooler queue. (closes issue #15072)
	  Reported by: garlew Patches: spool.diff uploaded by garlew
	  (license 376)

2009-06-03 15:49 +0000 [r198891]  David Vossel <dvossel@digium.com>

	* main/channel.c, res/res_features.c, include/asterisk/channel.h:
	  Generic call forward api, ast_call_forward() The function
	  ast_call_forward() forwards a call to an extension specified in
	  an ast_channel's call_forward string. After an ast_channel is
	  called, if the channel's call_forward string is set this function
	  can be used to forward the call to a new channel and terminate
	  the original one. I have included this api call in both
	  channel.c's ast_request_and_dial() and res_feature.c's
	  feature_request_and_dial(). App_dial and app_queue already
	  contain call forward logic specific for their application and
	  options. (closes issue #13630) Reported by: festr Review:
	  https://reviewboard.asterisk.org/r/271/

2009-06-01 20:07 +0000 [r198665]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: If using the old deprecated format, a
	  reload would cause the class to disappear. (closes issue #14759)
	  Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt
	  uploaded by tilghman (license 14) Tested by: lmadsen

2009-05-30 19:36 +0000 [r198370]  Sean Bright <sean.bright@gmail.com>

	* res/res_jabber.c: Properly terminate AMI JabberSend response
	  messages. The response message (either Error or Success) needs an
	  extra trailing \r\n after the fields to inform the client that
	  the message is complete. (closes issue #14876) Reported by: srt
	  Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
	  (license 71) asterisk_14876.patch uploaded by srt (license 378)
	  trunk-14876-2.diff uploaded by phsultan (license 73)

2009-05-30 03:42 +0000 [r198311]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages
	  expired. (closes issue #14561) Reported by: cmoss28

2009-05-30 02:46 +0000 [r198251]  Sean Bright <sean.bright@gmail.com>

	* apps/app_dial.c: Treat an empty FORWARD_CONTEXT the same way we
	  treat a missing one. (closes issue #15056) Reported by:
	  p_lindheimer Patches: 05292009_bug15056.diff uploaded by
	  seanbright (license 71) Tested by: p_lindheimer

2009-05-29 18:53 +0000 [r198068]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, main/channel.c, res/res_features.c,
	  include/asterisk/cdr.h: Use AST_CDR_NOANSWER instead of
	  AST_CDR_NULL as the default CDR disposition. This change also
	  involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is
	  used on originated channels to distinguish: them from dialed
	  channels. (closes issue #12946) Reported by: meral Patches:
	  null-cdr2.diff uploaded by mnicholson (license 96) Tested by:
	  mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested
	  by: sum

2009-05-29 18:14 +0000 [r197998]  Sean Bright <sean.bright@gmail.com>

	* Makefile: Fix 'make config' target for Slackware. There was a
	  missing semi-colon after the echo statement in the Makefile that
	  was causing problems for some users. Fix suggested by reporter.
	  (closes issue #15225) Reported by: pdavis

2009-05-28 23:57 +0000 [r197895]  Leif Madsen <lmadsen@digium.com>

	* apps/app_mixmonitor.c: Update MixMonitor documentation. Updated
	  the MixMonitor documentation for the 'b' option so that it is
	  more obvious that you must not optimize awat the Local channel
	  when using this option. (issue #14829)

2009-05-28  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.26-rc1

2009-05-28 15:51 +0000 [r197620]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
	  not peer 'blah' is in trunk mode or not.

2009-05-28 15:27 +0000 [r197588]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
	  for media to arrive from an alternate source when responding to a
	  reinvite with 491. When we receive a SIP reinvite, it is possible
	  that we may not be able to process the reinvite immediately since
	  we have also sent a reinvite out ourselves. The problem is that
	  whoever sent us the reinvite may have also sent a reinvite out to
	  another party, and that reinvite may have succeeded. As a result,
	  even though we are not going to accept the reinvite we just
	  received, it is important for us to not have problems if we
	  suddenly start receiving RTP from a new source. The fix for this
	  is to grab the media source information from the SDP of the
	  reinvite that we receive. This information is passed to the RTP
	  layer so that it will know about the alternate source for media.
	  Review: https://reviewboard.asterisk.org/r/252

2009-05-28 15:21 +0000 [r197562]  Eliel C. Sardanons <eliels@gmail.com>

	* channels/chan_sip.c: Use the address we already know when
	  reloading a peer with nat=yes. If we already have an address for
	  a peer, and we are reloading the sip configuration, try to use
	  that address to contact the peer, instead of getting it from the
	  Contact. (closes issue #15194) Reported by: ibc Patches:
	  sip.patch uploaded by eliel (license 64) Tested by: manwe

2009-05-28 14:49 +0000 [r197537]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, include/asterisk/audiohook.h,
	  main/audiohook.c: Add flags to chanspy audiohook so that audio
	  stays in sync. There are two flags being added to the chanspy
	  audiohook here. One is the pre-existing
	  AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
	  the read and write slinfactories on the audiohook do not skew
	  beyond a certain tolerance. In addition, there is a new audiohook
	  flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
	  we do not allow for a slinfactory to build up a substantial
	  amount of audio before flushing it. For this particular issue,
	  this means that the person spying on the call will hear the
	  conversations in real time with very little delay in the audio.
	  (closes issue #13745) Reported by: geoffs Patches: 13745.patch
	  uploaded by mmichelson (license 60) Tested by: snblitz

2009-05-28 13:44 +0000 [r197466]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where the flag indicating the
	  presence of rport would get overwritten by the nat setting. The
	  presence of rport is now stored as a separate flag. Once the
	  dialog is setup and authenticated (or it passes through
	  unauthenticated) the proper nat flag is set. (closes issue
	  #13823) Reported by: dimas

2009-05-27 20:12 +0000 [r197264]  Sean Bright <sean.bright@gmail.com>

	* Makefile: Use bash explicitly when calling
	  build_tools/mkpkgconfig from the Makefile. Since we use bashisms
	  in build_tools/mkpkgconfig, we should call on bash explicitly
	  when running from the Makefile, otherwise we get errors during a
	  'make install.' (closes issue #15209) Reported by: seandarcy

2009-05-27 20:07 +0000 [r197259]  Olle Johansson <oej@edvina.net>

	* doc/asterisk-conf.txt: Typo fix

2009-05-27 19:09 +0000 [r197194]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_cut.c: Use a different determinator on whether to
	  print the delimiter, since leading fields may be blank. (closes
	  issue #15208) Reported by: ramonpeek Patch by me, though inspired
	  in part by a patch from ramonpeek

2009-05-27 16:49 +0000 [r197124]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, include/asterisk/channel.h: Fix broken attended
	  transfers The bridge was terminating immediately after the
	  attended transfer was completed. The problem was because upon
	  reentering ast_channel_bridge nexteventts was checked to see if
	  it was set and if so could possibly return AST_BRIDGE_COMPLETE.
	  (closes issue #15183) Reported by: andrebarbosa Tested by:
	  andrebarbosa, tootai, loloski

2009-05-27 13:54 +0000 [r197024]  Sean Bright <sean.bright@gmail.com>

	* apps/app_queue.c: Fix handling of the 'state_interface' option of
	  the 'queue add member' CLI command. This change relates to
	  r184980, which was a backport of the state interface changes to
	  app_queue from trunk. trunk and all of the 1.6.x branches are not
	  affected. 'queue add member' allows for specifying an interface
	  to use for device state when adding a queue member via CLI, but
	  the validation code was not properly updated to reflect this
	  optional argument. (closes issue #15198) Reported by: loloski
	  Patches: 05272009_app_queue.diff uploaded by seanbright (license
	  71) Tested by: loloski

2009-05-26 18:14 +0000 [r196826]  Russell Bryant <russell@digium.com>

	* res/res_convert.c: Resolve a file handle leak. The frames here
	  should have always been freed. However, out of luck, there was
	  never any memory leaked. However, after file streams became
	  reference counted, this code would leak the file stream for the
	  file being read. (closes issue #15181) Reported by: jkroon

2009-05-26 13:06 +0000 [r196657]  Joshua Colp <jcolp@digium.com>

	* contrib/scripts/safe_asterisk: Remove some bash specific stuff
	  from safe_asterisk. (closes issue #10812) Reported by: paravoid
	  Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license
	  46)

2009-05-22 13:54 +0000 [r196116]  Joshua Colp <jcolp@digium.com>

	* channels/chan_misdn.c: Fix a bug where using immediate with mISDN
	  caused a cause code of 16 to get sent back instead of 1 if the
	  's' extension did not exist. (closes issue #12286) Reported by:
	  lmamane

2009-05-21 19:04 +0000 [r195991]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Sign problem calculating timestamp for iax
	  frame leads to no audio on the receiving peer. There are rare
	  cases in which a frame's delivery timestamp is slightly less than
	  the iax2_pvt's offset. This causes the pvt's timestamp to be a
	  small negative number, but since the timestamp value is unsigned
	  it looks like a huge positive number. This patch checks for this
	  negative case and sets the ms to zero. A similar check is already
	  done right below this one in the 'else' statement. (closes issue
	  #15032) Reported by: guillecabeza Patches:
	  chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
	  380) Tested by: guillecabeza (closes issue #14216) Reported by:
	  Andrey Sofronov

2009-05-21 15:25 +0000 [r195881]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, res/res_features.c, include/asterisk/cdr.h: This
	  commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and
	  AST_CDR_FLAG_LOCKED from being updated in certain cases. This is
	  accomplished by adding two functions to update the answer time
	  and disposition of calls that checks for the proper lock flags.
	  These functions are used in the ast_bridge_call() function so
	  that ForkCDR(A) calls are respected. This patch also modifies the
	  way ast_bridge_call() chooses the cdr record to base the
	  bridged_cdr on. Previously the first unlocked cdr record would be
	  chosen, now instead the first cdr record is chosen and forked cdr
	  records are moved to the bridge_cdr. This allows the original cdr
	  record and any forked cdr records to be properly updated with
	  answer and end times. (closes issue #13797) Reported by: sh0t
	  Tested by: sh0t (closes issue #14744) Reported by: deepesh

2009-05-21  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.25

2009-05-13  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.4.25-rc1

2009-05-13 13:38 +0000 [r194208]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Fix RFC2833 issues with DTMF getting duplicated and
	  with duration wrapping over. (closes issue #14815) Reported by:
	  geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
	  Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
	  #14460) Reported by: moliveras Tested by: moliveras

2009-05-13 00:52 +0000 [r194137]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Fix logic for how to proceed with a single digit
	  extension. (closes issue #15091) Reported by: andrew Patches:
	  20090512__issue15091.diff.txt uploaded by tilghman (license 14)
	  Tested by: andrew

2009-05-12 22:15 +0000 [r194028]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: This change modifies app_queue to properly
	  generate CDR records in failure situations. This involves setting
	  a proper cdr disposition coresponding to the given failure
	  condition and ensuring the proper information is stored in the
	  cdr record. (closes issue #13691) Reported by: dferrer Tested by:
	  mnicholson (closes issue #13637) Reported by: atis Tested by:
	  atis

2009-05-12 20:39 +0000 [r193955]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Avoid initializing routines if the
	  authentication fails. Fixes a crash (RR) issue. (closes issue
	  #14508) Reported by: tiziano Patches:
	  20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
	  377)

2009-05-12 18:18 +0000 [r193880]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if
	  we are actually sending a SIP CANCEL. The problem was that the
	  hangup code was setting the invitestate too early. The result of
	  this was that we would always send a CANCEL request, even if it
	  was not an appropriate time to do so (e.g. we have not yet
	  received a provisional response for our INVITE). Note that this
	  same fix had been applied to trunk and the 1.6.X branches
	  starting with revision 155467. This is why you will see this
	  revision being blocked from those places. AST-216

2009-05-11 22:48 +0000 [r193755]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Move 300 bytes around on the stack, to make
	  more room for an extension buffer. This allows more concurrent
	  extensions to be copied for a single voicemail, without creating
	  a possibility of upsetting existing users, where a dialplan could
	  run out of stack space where it had run fine before.
	  Alternatively, we could have allocated off the heap, but that is
	  a larger change and would have increased the chance for
	  instability introduced by this change. This is really solved
	  starting in 1.6.0.11, as the use of an ast_str buffer allows an
	  unlimited number of extensions (up to available memory). We
	  additionally create a new warning message when the buffer length
	  is exceeded, permitting administrators to see an issue after the
	  fact, whereas previously the list was silently truncated. (closes
	  issue #14739) Reported by: p_lindheimer Patches:
	  20090417__bug14739.diff.txt uploaded by tilghman (license 14)
	  Tested by: p_lindheimer

2009-05-11 19:09 +0000 [r193613]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Sent wrong message to clear a call we
	  started if the other end has not responed yet. In the state
	  MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
	  yet), it is not allowed to clear the call with RELEASE_COMPLETE.
	  It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
	  allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
	  5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
	  JIRA ABE-1862

2009-05-11 17:35 +0000 [r193544]  Leif Madsen <lmadsen@digium.com>

	* funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI
	  documentation. (issue #15073) Reported by: pkempgen Patches:
	  20090511__issue15073.diff.txt uploaded by tilghman (license 14)

2009-05-08 21:01 +0000 [r193391]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Set the proper disposition on originated calls.
	  (closes issue #14167) Reported by: jpt Patches:
	  call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
	  Tested by: dlotina, rmartinez, mnicholson

2009-05-08 14:51 +0000 [r193262]  David Vossel <dvossel@digium.com>

	* channels/misdn_config.c: "misdn show config" segfaults asterisk,
	  if no MSN lists (closes issue #14976) Reported by: alecdavis
	  Patches: misdn_config.diff.txt uploaded by alecdavis (license
	  585) Tested by: alecdavis, FabienToune

2009-05-08 14:03 +0000 [r193193]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/logger.conf.sample, main/logger.c: Make absolute paths
	  for logger channels work properly (Note: This is not a new
	  feature, it was previously undocumented and broken.) The Asterisk
	  logger has a feature to support absolute pathnames for logger
	  channels, but the code implementing the feature was broken. This
	  has been fixed, and the absolute path feature is now documented
	  in the sample logger.conf.

2009-05-07 23:41 +0000 [r193119]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Fix Background within a Macro for FreePBX. If the
	  single digit DTMF is an extension in the specified context, then
	  go there and signal no DTMF. Otherwise, we should exit with that
	  DTMF. If we're in Macro, we'll exit and seek that DTMF as the
	  beginning of an extension in the Macro's calling context. If
	  we're not in Macro, then we'll simply seek that extension in the
	  calling context. Previously, someone complained about the
	  behavior as it related to the interior of a Gosub routine, and
	  the fix (#14011) inadvertently broke FreePBX (#14940). This
	  change should fix both of these situations, but with the possible
	  incompatibility that if a single digit extension does not exist
	  (but a longer extension COULD have matched), it would have
	  previously gone immediately to the "i" extension, but will now
	  need to wait for a timeout. (closes issue #14940) Reported by:
	  p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
	  tilghman (license 14) Tested by: p_lindheimer

2009-05-07 22:17 +0000 [r193050]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Give a more helpful message when an
	  incoming call's dialed extension does not match. Added the dialed
	  extension and context to the chan_misdn messages warning that the
	  dialed number cannot be matched in the dialplan.

2009-05-07 16:29 +0000 [r192932]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Eliminate repetition of fullcontact during
	  reconstruction. If the fullcontact field appears in both the
	  sippeers and the sipregs table, then during reconstruction of the
	  field, it will otherwise be doubled. (closes issue #14754)
	  Reported by: Alexei Gradinari Patches:
	  20090506__bug14754.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen

2009-05-06 22:15 +0000 [r192858]  Jeff Peeler <jpeeler@digium.com>

	* res/res_features.c: Make ParkedCall application stop execution of
	  the dialplan after hang up Just changed park_exec to always
	  return non-zero. I really wasn't entirely sure at first if this
	  was a bug. Decided it was since it would be surprising when not
	  using ParkedCall in the dialplan to hang up and have dialplan
	  execution continue. (closes issue #14555) Reported by:
	  francesco_r

2009-05-06 13:30 +0000 [r192633]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Update some old logic to stop both begin and
	  end DTMF frames from reaching the core if rfc2833 is not enabled.
	  (closes issue #15036) Reported by: dimas Patches: v1-15036.patch
	  uploaded by dimas (license 88)

2009-05-05 19:56 +0000 [r192524]  Sean Bright <sean.bright@gmail.com>

	* static-http/astman.js: Fix Javascript error when using astman.js
	  in Internet Explorer. Internet Explorer (tested with 7.0) does
	  not like trailing commas on constructs like object initializers,
	  so get rid of them to avoid some errors. (closes issue #15026)
	  Reported by: rajnishgiri Patches: bug15026.patch uploaded by
	  seanbright (license 71) Tested by: seanbright

2009-05-05 18:22 +0000 [r192429-192454]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Fix an incorrect assumption that certain
	  values on the channel will always exist when they may not. The
	  CDR code involved with bridges wrongly assumed that the currently
	  executing application and data values will always exist. It is
	  possible for this to be false when call forwarding is involved.
	  (closes issue #14984) Reported by: gincantalupo

	* apps/app_followme.c: Fix a bug where the followme application
	  would continue trying numbers after the caller hung up. (closes
	  issue #13624) Reported by: sgenyuk

2009-05-04 22:37 +0000 [r192213]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: global mohinterpret setting is ignored
	  mohinterpret and mohsuggest global variables were not copied over
	  during build_users and build_peers. (closes issue #14728)
	  Reported by: dimas Patches: v1-14728.patch uploaded by dimas
	  (license 88) Tested by: dimas, dvossel

2009-05-02 18:48 +0000 [r191628-191778]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a bug which resulted from the Hebrew
	  voicemail commit. This fixes a case where a certain message could
	  get played twice. (closes issue #13155) Reported by:
	  greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded
	  by greenfieldtech (license 369) Tested by: greenfieldtech

	* apps/app_chanspy.c: Kevin has informed me that thi sort of thing
	  is not necessary.

	* apps/app_chanspy.c: Move static buffers to outside for loops in
	  app_chanspy. Similar to seanbright's commit 191422, this moves
	  some static buffers to be defined outside of for loops since it
	  is undefined if memory will be re-used or if the stack will grow
	  with each iteration of the loop.

2009-05-01 20:00 +0000 [r191559]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or
	  23), not 1. (closes issue #14993) Reported by: BigJimmy Patches:
	  causepatch uploaded by BigJimmy (license 371)

2009-05-01 17:40 +0000 [r191488]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Fix DTMF not being sent to other side after a
	  partial feature match This fixes a regression from commit 176701.
	  The issue was that ast_generic_bridge never exited after the
	  feature digit timeout had elapsed, which prevented the queued
	  DTMF from being sent to the other side. This issue was reported
	  to me directly.

2009-05-01 15:42 +0000 [r191422]  Sean Bright <sean.bright@gmail.com>

	* apps/app_queue.c: Move the defintion of the a couple arrays out
	  of loops. According to Kevin, it is unspecified as to whether a
	  variable defined inside a block is allocated once by the compiler
	  or for each pass through the block (loops being the only
	  interesting case), so just define these before we get into our
	  loop to be sure.

2009-04-29 23:10 +0000 [r191220]  Tilghman Lesher <tlesher@digium.com>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to
	  compile with FDLEAK checking enabled.

2009-04-29 18:07 +0000 [r191096]  David Brooks <dbrooks@digium.com>

	* pbx/pbx_config.c: Patch to fix tab-completion crash on "remove
	  extension" This patch simply removes some old code back before
	  Asterisk used editline. This fixes the crash that occurred when
	  tab-completing "remove extension". (closes issue #14689) Reported
	  by: isaacgal

2009-04-29 15:23 +0000 [r191041]  Sean Bright <sean.bright@gmail.com>

	* apps/app_queue.c: Fix a crash in app_queue with very long member
	  lists. A user reported via #asterisk that with very long lists of
	  members, a crash occurs in ast_strdupa, so just use a single
	  buffer and ast_copy_string instead of stack allocating copys of
	  each interface name.

2009-04-27 19:29 +0000 [r190721]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent
	  line endings' when autoconf 2.63 is used Attempt to make
	  configure script regeneration 'safe' using autoconf 2.63, which
	  embeds a bare CR into the script, thus making Subversion complain
	  about inconsistent line endings This commit changes the MIME type
	  of the configure script to be 'binary' thus making Subversion no
	  longer inspect line endings, and as a bonus 'svn diff' will no
	  longer try to generate diff output for it, which is not generally
	  useful anyway.

2009-04-27 19:03 +0000 [r190661-190662]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c: Fix a typo from 190661.

	* res/res_smdi.c: Resolve a crash in res_smdi when used with
	  chan_dahdi. When chan_dahdi goes to get an SMDI message, it
	  provides no search criteria. It just grabs the next message that
	  arrives. This code was written with the SMDI dialplan functions
	  in mind, since that is now the preferred method of using SMDI.
	  However, this broke support of it being used from chan_dahdi.
	  (closes AST-212)

2009-04-23 21:07 +0000 [r190356]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove a bogus ast_channel_unlock().

2009-04-23 19:13 +0000 [r190286]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Fix a bug in chan_local glare hangup
	  detection. If both sides of a Local channel were hung up at
	  around the same time it was possible for one thread to destroy
	  the local private structure and have the other thread immediately
	  try to remove the already freed structure from the local channel
	  list.

2009-04-23 10:07 +0000 [r190187]  Olle Johansson <oej@edvina.net>

	* include/asterisk/lock.h: unistd.h is required for usleep() on
	  Darwin. It will not hurt to include it always on other platforms
	  either.

2009-04-22 21:35 +0000 [r190092]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Detect availability of
	  pthread_rwlock_timedwrlock() before using it. (closes issue
	  #14930) Reported by: tilghman Patches:
	  20090420__bug14930.diff.txt uploaded by tilghman (license 14)
	  Tested by: mvanbaak, tilghman

2009-04-22 19:20 +0000 [r189991]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/chan_h323.h: Make chan_h323 respect packetization
	  settings Previously, packetization settings were ignored and now
	  they are not. A new config option 'autoframing' has been added to
	  mirror the way chan_sip handles it. Turning on the autoframing
	  option (available both as a global option or per peer) overrides
	  the local settings with the remote packetization settings.
	  Testing was performed with varying packetization levels with the
	  following codecs: ulaw, alaw, gsm, and g729. (closes issue
	  #12415) Reported by: pj Patches:
	  2009012200_h323packetization.diff.txt uploaded by mvanbaak
	  (license 7), modified by me

2009-04-22 14:29 +0000 [r189849]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove
	  \r from downloaded file On some systems, sed does not recognize
	  \r in the pattern the way it was used here. Use tr instead
	  because this works the same across systems. (closes issue #14936)
	  Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded
	  by mvanbaak (license 7) Tested by: leobrown, mvanbaak

2009-04-21 15:52 +0000 [r189601-189664]  Doug Bailey <dbailey@digium.com>

	* utils/muted.c: Remove daemon call on systems that do not support
	  forking.

	* main/config.c, configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, configure.ac: Add check in configure
	  script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This
	  allows config.c to compile when linked against uclibc that does
	  not support these parameters

2009-04-20 22:02 +0000 [r189537]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for
	  func_odbc/ARRAY() for problems that occur with certain special
	  characters. In certain cases, due to the way Set() works in 1.4,
	  values may not get set properly. This is a workaround for 1.4
	  only that corrects for these issues, without making func_odbc
	  more difficult to use properly. (closes issue #14614) Reported
	  by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by
	  tilghman (license 14)
	  double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff
	  uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman

2009-04-20 21:10 +0000 [r189463-189465]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c: Update CDR appropriately when
	  AST_CAUSE_NO_ANSWER is set

	* apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL

2009-04-20 20:58 +0000 [r189462]  Sean Bright <sean.bright@gmail.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints
	  in AEL. AEL was not handling the case of a device hint containing
	  an @ symbol, which caused parking hints (e.g.
	  hint(park:exten@context)) to error out the parser. This patch
	  makes AEL treat the @ the same way it treats colon and ampersand
	  now, meaning the characters are included in verbatim. (closes
	  issue #14941) Reported by: bpgoldsb Patches: bug14941.patch
	  uploaded by seanbright (license 71) Tested by: bpgoldsb

2009-04-20 19:10 +0000 [r189391]  Doug Bailey <dbailey@digium.com>

	* main/manager.c, main/db1-ast/recno/rec_open.c,
	  channels/chan_iax2.c: Clean up problem with manager
	  implementation of mmap where it was not testing against
	  MAP_FAILED response. Got rid of shadowed variable used in
	  processign the mmap results. Change test of mmap results to
	  compare against MAP_FAILED

2009-04-20 14:04 +0000 [r189277]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Move the check for chan->fdno == -1 to after the
	  zombie/hangup check. Many users were finding that their hung up
	  channels were staying up and causing 100% CPU usage. (issue
	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
	  uploaded by mmichelson (license 60) Tested by: falves11, bamby

2009-04-18 01:27 +0000 [r189203]  David Vossel <dvossel@digium.com>

	* channels/chan_agent.c: Fixed autologoff in agents.conf not
	  working when agent logs in via AgentLogin app An agent logs in by
	  calling an extension that calls the AgentLogin app. In
	  agents.conf ackcall=always is set, so when they get a call they
	  have the choice to either acknowledge it or ignore it.
	  autologoff=10 is set as well, so if the agent ignores the call
	  over 10sec one may assume that the agent should be logged out
	  (and in this case hungup on as well), but this was not happening.
	  (closes issue #14091) Reported by: evandro Patches:
	  autologoff.diff uploaded by dvossel (license 671) Review:
	  http://reviewboard.digium.com/r/225/

2009-04-17 21:27 +0000 [r189134]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: Modifed/added some debug messages.
	  JIRA ABE-1835

2009-04-17 15:43 +0000 [r189009]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c: Make Busy() application set the CDR disposition to
	  BUSY. (closes issue #14306) Reported by: cristiandimache

2009-04-17 14:41 +0000 [r188937-188946]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where a value used to create the
	  channel name was bogus. This commit fixes the scenario where an
	  incoming call is authenticated using a peer entry. Previously the
	  channel name was created using either the username setting from
	  the sip.conf entry or the IP address that the call came from. Now
	  the channel name will be created using the peer name itself. This
	  commit will not change the way the channel name is generated for
	  users or friends. (closes issue #14256) Reported by: Nick_Lewis
	  Patches: chan_sip.c-chname.patch uploaded by Nick (license 657)
	  Tested by: Nick_Lewis, file

	* channels/chan_dahdi.c: Fix a situation where the DAHDI channel
	  private structure lock was not unlocked when it should have been.
	  (issue AST-210)

2009-04-16 21:41 +0000 [r188835]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Only update realtime, if global option
	  rtupdate != false (closes issue #14885) Reported by: deepesh
	  Patches: 20090413__bug14885.diff.txt uploaded by tilghman
	  (license 14) Tested by: deepesh

2009-04-16 21:37 +0000 [r188833]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is
	  enabled. Leave jitter setting alone. JIRA ABE-1835

2009-04-16 21:02 +0000 [r188773]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Umask should not be exported into global
	  namespace. (closes issue #14912) Reported by: jcapp

2009-04-15 22:08 +0000 [r188646]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c: National prefix inserted even when caller
	  ID not available When the caller ID is restricted, the expected
	  behavior is for the caller id to be blank. In chan_dahdi, the
	  national prefix is placed onto the callers number even if its
	  restricted (empty) causing the caller id to be the national
	  prefix rather than blank. (closes issue #13207) Reported by:
	  shawkris Patches: national_prefix.diff uploaded by dvossel
	  (license 671) Review: http://reviewboard.digium.com/r/220/

2009-04-15 20:04 +0000 [r188582]  Mark Michelson <mmichelson@digium.com>

	* main/file.c: Update ast_readvideo_callback to match
	  ast_readaudio_callback. This fixes potential refcount errors that
	  may occur on ast_filestreams. AST-208

2009-04-14 15:02 +0000 [r188287]  David Vossel <dvossel@digium.com>

	* main/audiohook.c: audio_audiohook_write_list() does not correctly
	  update sample size after ast_translate.
	  audio_audiohook_write_list() does not take into account that the
	  sample size may change after translation depending on if the
	  original frame is is 8khz or 16khz. While no 16kz codecs are
	  supported in 1.4 at the moment, this will save headaches in the
	  future if they ever are. the sample size is now updated after
	  translating to reflect this possibility. Thanks to jcolp and
	  mmichelson for helping me work this out. (issue AST-197)

2009-04-13 23:04 +0000 [r188149]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: If fileconfig limit exceeds our maximum, then set
	  the limit to the maximum. (Closes issue #14888) Reported by:
	  falves11

2009-04-10 22:16 +0000 [r187962]  Jeff Peeler <jpeeler@digium.com>

	* channels/Makefile: Fix module embedding for chan_h323. Include
	  libchanh323.a in the modules.link file so that all the symbols
	  can be resolved at link time. (closes issue #11966) Reported by:
	  dome

2009-04-10 19:26 +0000 [r187865]  Russell Bryant <russell@digium.com>

	* channels/chan_dahdi.c: Support "signaling" in addition to
	  "signalling". The sample configuration file has references to
	  both spellings.

2009-04-10 17:28 +0000 [r187763]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/realtime_pgsql.sql,
	  contrib/scripts/sip-friends.sql: Add lastms column to the
	  contributed table designs

2009-04-09 18:51 +0000 [r187484]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Handle a SIP race condition (reinvite before
	  an ACK) properly. RFC 5047 explains the proper course of action
	  to take if a reINVITE is received before the ACK from a previous
	  invite transaction. What we are to do is to treat the reINVITE as
	  if it were both an ACK and a reINVITE and process it normally.
	  Later, when we receive the ACK we had been expecting, we will
	  ignore it since its CSeq is less than the current iseqno of the
	  sip_pvt representing this dialog. (closes issue #13849) Reported
	  by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
	  (license 60) Tested by: mmichelson, klaus3000

2009-04-09 18:39 +0000 [r187209-187482]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Oops, typo

	* main/manager.c, include/asterisk/lock.h: Race condition between
	  ast_cli_command() and 'module unload' could cause a deadlock. Add
	  lock timeouts to avoid this potential deadlock. (closes issue
	  #14705) Reported by: jamessan Patches:
	  20090320__bug14705.diff.txt uploaded by tilghman (license 14)
	  Tested by: jamessan

	* channels/chan_sip.c, apps/app_sendtext.c: Permit zero-length text
	  messages in SIP. (Related to an issue posted to the -users list,
	  subject "AEL2, BASE64_DECODE and hexadecimal")

	* main/astfd.c (added): Oops, missed this file in the last commit.

	* main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
	  utils/Makefile, include/asterisk.h, main/Makefile, main/file.c:
	  Add debugging mode for diagnosing file descriptor leaks. (Related
	  to issue #14625)

	* main/manager.c: Backport resolution for file descriptor leak in
	  1.6.0 to 1.4. This fixes short reads in http manager sessions,
	  such as those done by the ast-gui branch. (Fixes AST-198)

2009-04-08 19:16 +0000 [r186832-187135]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Fix a crash due to too few arguments to
	  RetryDial. (closes issue #14852) Reported by: junky Patches:
	  retry_fix.diff uploaded by junky (license 177)

	* res/res_musiconhold.c: Fix a small logical error when loading moh
	  classes. We were unconditionally incrementing the number of
	  mohclasses registered. However, we should actually only increment
	  if the call to moh_register was successful. While this probably
	  has never caused problems, I noticed it and decided to fix it
	  anyway.

	* main/channel.c: Make a couple of changes with regards to a new
	  message printed in ast_read(). "ast_read() called with no
	  recorded file descriptor" is a new message added after a bug was
	  discovered. Unfortunately, it seems there are a bunch of places
	  that potentially make such calls to ast_read() and trigger this
	  error message to be displayed. This commit does two things to
	  help to make this message appear less. First, the message has
	  been downgraded to a debug level message if dev mode is not
	  enabled. The message means a lot more to developers than it does
	  to end users, and so developers should take an effort to be sure
	  to call ast_read only when a channel is ready to be read from.
	  However, since this doesn't actually cause an error in operation
	  and is not something a user can easily fix, we should not spam
	  their console with these messages. Second, the message has been
	  moved to after the check for any pending masquerades. ast_read()
	  being called with no recorded file descriptor should not
	  interfere with a masquerade taking place. This could be seen as a
	  simple way of resolving issue #14723. However, I still want to
	  try to clear out the existing ways of triggering this message,
	  since I feel that would be a better resolution for the issue.

	* formats/format_wav.c, formats/format_wav_gsm.c: Fix a few typos
	  of the word "frequency." (closes issue #14842) Reported by:
	  jvandal Patches: frequency-typo.diff uploaded by jvandal (license
	  413)

	* main/channel.c: Set the AST_FEATURE_WARNING_ACTIVE flag when a
	  p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
	  warning sounds will not be properly played to either party of the
	  bridge. (closes issue #14845) Reported by: adomjan

2009-04-07 22:16 +0000 [r186775]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_macro.c: Fix Macro documentation to match current (and
	  intended) behavior. (See -dev mailing list)

2009-04-07 20:43 +0000 [r186719]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Ensure that \r\n is printed after the ActionID in
	  an OriginateResponse. (closes issue #14847) Reported by: kobaz

2009-04-06 13:54 +0000 [r186565]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Revert commit 186445 because it causes the
	  build to fail when IMAP_STORAGE is used.

2009-04-03 20:19 +0000 [r186458]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: Fix a bug where DAHDI/Zaptel channels
	  would not properly switch formats when requested Don't offer
	  AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
	  provide a slight performance benefit, the translation core in
	  Asterisk has some flaws when a channel driver offers multiple raw
	  formats. this fix is much simpler than fixing the translation
	  core to solve that issue (although that will be done later).

2009-04-03 19:56 +0000 [r186415-186445]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Found a conflict in the last commit, due to
	  multiple targets

	* apps/app_voicemail.c, configs/voicemail.conf.sample: Distinguish
	  in a sent email between simple sends and forwards. (closes issue
	  #11678) Reported by: jamessan Patches:
	  20090330__bug11678.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman, lmadsen

2009-04-03 15:48 +0000 [r186320]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/crypto.h: Fix a problem with the crypto variable
	  definitions not actually being defined properly. (closes issue
	  #14804) Reported by: jvandal

2009-04-03 01:57 +0000 [r186229]  Russell Bryant <russell@digium.com>

	* cdr/cdr_radius.c: Fix a memory leak in cdr_radius. I came across
	  this while doing some testing of my ast_channel_ao2 branch. After
	  running a test overnight that generated over 5 million calls,
	  Asterisk had taken up about 1 GB of my system memory. So, I
	  re-ran the test with MALLOC_DEBUG turned on. However, it showed
	  no leaks in Asterisk during the test, even though Asterisk was
	  still consuming it somehow. Instead, I turned to valgrind, which
	  when run with --leak-check=full, told me exactly where the leak
	  came from, which was from allocations inside the radiusclient-ng
	  library. This explains why MALLOC_DEBUG did not report it. After
	  a bit of analysis, I found that we were leaking a little bit of
	  memory every time a CDR record was passed to cdr_radius. I don't
	  actually have a radius server set up to receive CDR records.
	  However, I always have my development systems compile and install
	  all modules. In addition to making sure there are not build
	  errors across modules, always loading modules helps find bugs
	  like this, too, so it is strongly recommend for all developers.

2009-04-02 21:55 +0000 [r186174]  Mark Michelson <mmichelson@digium.com>

	* configs/features.conf.sample: Fix instructions in one-step
	  parking comment to make more sense. Changed a capital K to a
	  lowercase k.

2009-04-02 17:21 +0000 [r186081]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: ensure that the buffer passed to
	  DAHDI_SET_BUFINFO is fully initialized

2009-04-02 17:09 +0000 [r186057-186059]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  186056 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009)
	  | 2 lines Fix for AST-2009-003 ........

	* channels/chan_sip.c: Avoid multiple warning messages in SIP, due
	  to this column not existing

2009-04-02 13:43 +0000 [r185952]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: the DAHDI_GETCONF, DAHDI_SETCONF and
	  DAHDI_GET_PARAMS ioctls were recently corrected to show that they
	  do, in fact, read data from userspace as part of their work. due
	  to this fix, valgrind now reports a number of cases where
	  chan_dahdi passed an uninitialized (or partially) buffer to these
	  ioctls, which could lead to unexpected behavior. this patch
	  corrects chan_dahdi to ensure that buffers passed to these ioctls
	  are always fully initialized.

2009-04-01 19:02 +0000 [r185845]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes issue with dropped calles due to
	  re-Invite glare and re-Invites never executing after a 491
	  Acknowledgement for 491 responses were never being processed
	  because it didn't match our pending invite's seqno. Since the ACK
	  was never processed, the 491 frame would continue to be
	  retransmitted until eventually the call was dropped due to max
	  retries. Now during a pending invite, if we receive another
	  invite, we send an 491 and hold on to that glare invite's seqno
	  in the "glareinvite" variable for that sip_pvt struct. When ACK's
	  are received, we first check to see if it is in response to our
	  pending invite, if not we check to see if it is in response to a
	  glare invite. In this case, it is in response to the glare invite
	  and must be dealt with or the call is dropped. I've changed the
	  wait time for resending the re-Invite after receving a 491
	  response to comply with RFC 3261. Before this patch the scheduled
	  re-Invite would only change a flag indicating that the re-Invite
	  should be sent out, now it actually sends it out as well. (closes
	  issue #12013) Reported by: alx Review:
	  http://reviewboard.digium.com/r/213/

2009-04-01 13:47 +0000 [r185771]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix a case where DTMF could bypass audiohooks.
	  This change fixes a situation where an audiohook that wants DTMF
	  would not actually get it. This is in the code path where we end
	  DTMF digit length emulation while handling a NULL frame.

2009-03-31 22:00 +0000 [r185468-185599]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix crash that would occur if an empty member
	  was specified in queues.conf. (closes issue #14796) Reported by:
	  pida

	* channels/chan_sip.c: Use AST_SCHED_DEL_SPINLOCK instead of
	  manually using the logic.

	* apps/app_voicemail.c: Fix Russian voicemail intro to say the word
	  "messages" properly. (closes issue #14736) Reported by: chappell
	  Patches: voicemail_no_messages.diff uploaded by chappell (license
	  8)

2009-03-31 16:37 +0000 [r185362]  David Brooks <dbrooks@digium.com>

	* channels/chan_gtalk.c: Fix incorrect parsing in chan_gtalk when
	  xmpp contains extra whitespaces To drill into the xmpp to find
	  the capabilities between channels, chan_gtalk calls iks_child()
	  and iks_next(). iks_child() and iks_next() are functions in the
	  iksemel xml parsing library that traverse xml nodes. The bug here
	  is that both iks_child() and iks_next() will return the next
	  iks_struct node *regardless* of type. chan_gtalk expects the next
	  node to be of type IKS_TAG, which in most cases, it is, but in
	  this case (a call being made from the Empathy IM client), there
	  exists iks_struct nodes which are not IKS_TAG data (they are
	  extraneous whitespaces), and chan_gtalk doesn't handle that case,
	  so capabilities don't match, and a call cannot be made.
	  iks_first_tag() and iks_next_tag(), on the other hand, will not
	  return the very next iks_struct, but will check to see if the
	  next iks_struct is of type IKS_TAG. If it isn't, it will be
	  skipped, and the next struct of type IKS_TAG it finds will be
	  returned. This assures that chan_gtalk will find the iks_struct
	  it is looking for. This fix simply changes all calls to
	  iks_child() and iks_next() to become calls to iks_first_tag() and
	  iks_next_tag(), which resolves the capability matching. The
	  following is a payload listing from Empathy, which, due to the
	  extraneous whitespace, will not be parsed correctly by iksemel:
	  <iq from='dbrooksjab@235-22-24-10/Telepathy'
	  to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
	  <session xmlns='http://www.google.com/session'
	  initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
	  id='1837267342'> <description
	  xmlns='http://www.google.com/session/phone'> <payload-type
	  clockrate='16000' name='speex' id='96'/> <payload-type
	  clockrate='8000' name='PCMA' id='8'/> <payload-type
	  clockrate='8000' name='PCMU' id='0'/> <payload-type
	  clockrate='90000' name='MPA' id='97'/> <payload-type
	  clockrate='16000' name='SIREN' id='98'/> <payload-type
	  clockrate='8000' name='telephone-event' id='99'/> </description>
	  </session> </iq>

2009-03-31 15:34 +0000 [r185298]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix some state_interface stuff that was in
	  trunk but not in the backport to 1.4. Issue #14359 was fixed
	  between the time that I posted the review of the backport of the
	  state interface change for 1.4. This merges the changes from that
	  issue back into 1.4. (closes issue #14359) Reported by:
	  francesco_r

2009-03-31 14:06 +0000 [r185196]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c: Fix crash when moving audiohooks between
	  channels. Handle the scenario where we are called to move
	  audiohooks between channels and the source channel does not
	  actually have any on it. (closes issue #14734) Reported by:
	  corruptor

2009-03-30 20:40 +0000 [r185120-185121]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn_config.c, configs/misdn.conf.sample: Update the
	  channel allocation method documentation.

	* channels/misdn/isdn_lib.c: Make chan_misdn BRI TE side normally
	  defer channel selection to the NT side. Channel allocation
	  collisions are not handled by chan_misdn very well. This patch
	  simply avoids the problem for BRI only. For PRI, allocation
	  collisions are still possible but less likely since there are
	  simply more channels available and each end could use a different
	  allocation strategy. misdn.conf options available:
	  te_choose_channel - Use to force the TE side to allocate
	  channels. method - Specify the channel allocation strategy.
	  (closes issue #13488) Reported by: Christian_Pinedo Patches:
	  isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes,
	  festr

2009-03-30 16:17 +0000 [r184980-185031]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix queue weight behavior so that calls in
	  low-weight queues are not inappropriately blocked. (This is
	  copied and pasted from the review request I made for this patch)
	  Asterisk has some odd behavior when queue weights are used. The
	  current logic used when potentially calling a queue member is: If
	  the member we are going to call is part of another queue and
	  _that other queue has any callers in it_ and has a higher weight
	  than the queue we are calling from, then don't try to contact
	  that member. The issue here is what I have marked with
	  underscores. If the higher-weighted queue has any callers in it
	  at all, then the queue member will be unreachable from the
	  lower-weighted queue. This has the potential to be really really
	  bad if using a queue strategy, such as leastrecent or
	  fewestcalls, with the potential to call the same member
	  repeatedly. The fix proposed by garychen on issue 13220 is very
	  simple and, as far as I can see, works well for this situation.
	  With this set of changes, the logic used becomes: If the member
	  we are going to call is part of another queue, the other queue
	  has a higher weight than the queue we are calling from, and the
	  higher weight queue has at least as many callers as available
	  members, then do not try to contact the queue member. If the
	  higher weighted queue has fewer callers than available members,
	  then there is no reason to deny the call to this member since the
	  other queue can afford to spare a member. Since the fix involved
	  writing a generic function for determining the number of
	  available members in the queue, I also modified the is_our_turn
	  function to make use of the new num_available_members function to
	  determine if it is our turn to try calling a member. There is one
	  small behavior change. Before writing this patch, if you had
	  autofill disabled, then if you were the head caller in a queue,
	  you would automatically be told that it was your turn to try
	  calling a member. This did not take into account whether there
	  were actually any queue members available to take the call. Now
	  we actually make sure there is at least one member available to
	  take the call if autofill is disabled. (closes issue #13220)
	  Reported by: garychen Review:
	  http://reviewboard.digium.com/r/202/

	* configs/queues.conf.sample, apps/app_queue.c: Backport state
	  interface changes to app_queue from trunk. After several issues
	  raised on the Asterisk bugtracker against the 1.4 branch were
	  determined to be fixable with the state interface change
	  available in the 1.6.X series, it finally came time to just suck
	  it up and backport the change. For a detailed explanation of what
	  this change entails, the original trunk commit for this feature
	  may be found here:
	  http://svn.digium.com/view/asterisk?view=revision&revision=97203
	  In addition, the details for the use of this change to fix the
	  problems stated in issue #12970 may be found in the review
	  request I made for this change. It is linked below. (closes issue
	  #12970) Reported by: edugs15 Review:
	  http://reviewboard.digium.com/r/116

2009-03-30 14:35 +0000 [r184947]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Improve our handling of T38 in the initial
	  INVITE from a device. We now answer with matching media streams
	  to what is requested. If an INVITE is received with both a T38
	  and RTP media stream this means we answer with both. For any
	  outgoing calls created as a result of this inbound one no T38 is
	  requested in the initial INVITE. Instead if we start receiving
	  udptl packets we trigger a reinvite on the outbound side. (closes
	  issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief,
	  file, afu Review: http://reviewboard.digium.com/r/208/

2009-03-29 05:51 +0000 [r184842]  Russell Bryant <russell@digium.com>

	* apps/app_followme.c: Ensure targs variable is fully initialized.
	  (closes issue #14758) Reported by: tim_ringenbach

2009-03-27 13:06 +0000 [r184565]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix an issue where nat=yes would not always
	  take effect for the RTP session on outgoing calls. If calls were
	  placed using an IP address or hostname the global nat setting was
	  copied over but was not set on the RTP session itself. This
	  caused the RTP stack to not perform symmetric RTP actions.
	  (closes issue #14546) Reported by: acunningham

2009-03-26 22:17 +0000 [r184447]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: use new, improved 8kHz prompts

2009-03-26 21:07 +0000 [r184388]  David Vossel <dvossel@digium.com>

	* apps/app_test.c: pri loop TestClient/TestServer fails: server
	  SEND DTMF 8 app_test was failing when sending the last DTMF
	  digit, 8, because of the 100ms pause issued after DTMF is sent.
	  During this pause the other side would hang up causing the test
	  to look like it failed. Now the other side waits a second before
	  hanging up. (closes issue #12442) Reported by: tzafrir

2009-03-25 14:12 +0000 [r184188]  Eliel C. Sardanons <eliels@gmail.com>

	* main/asterisk.c: Avoid destroying the CLI line when moving the
	  cursor backward and trying to autocomplete. When moving the
	  cursor backward and pressing TAB to autocomplete, a NULL is put
	  in the line and we are loosing what we have already wrote after
	  the actual cursor position. (closes issue #14373) Reported by:
	  eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
	  Tested by: lmadsen

2009-03-24 22:34 +0000 [r184078]  Mark Michelson <mmichelson@digium.com>

	* apps/app_senddtmf.c: Change NULL pointer check to be
	  ast_strlen_zero. The 'digit' variable is guaranteed to be
	  non-NULL, so the if statement could never evaluate true. Changing
	  to ast_strlen_zero makes the logic correct. This was found while
	  reviewing ast_channel_ao2 code review.

2009-03-24 15:25 +0000 [r183913]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample: Additionally note that the
	  operator option needs an 'o' extension. (Related to issue #14731)

2009-03-23 17:59 +0000 [r183700]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: Fix a memory leak in res_monitor.c The only
	  way that this leak would occur is if Monitor were started using
	  the Manager interface and no File: header were given. Discovered
	  while reviewing the ast_channel_ao2 review request.

2009-03-20 16:53 +0000 [r183559]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a crash in IAX2 registration handling
	  found during load testing with dvossel.

2009-03-19 23:37 +0000 [r183481]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c: Add missing datastore inherit (exists in all
	  other branches)

2009-03-19 19:40 +0000 [r183386]  David Vossel <dvossel@digium.com>

	* include/asterisk/features.h, apps/app_dial.c, res/res_features.c:
	  Cleaning up a few things in detect disconnect patch Initialized
	  ast_call_feature in detect_disconnect to avoid accessing
	  uninitialized memory. Cleaned up /param tags in features.h. No
	  longer send dynamic features in ast_feature_detect. issue #11583

2009-03-19 19:21 +0000 [r183319-183342]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Reordering, to change prior to unlocking

	* channels/chan_dahdi.c: Delay signalling progress until a PRI
	  channel really signals progress. (closes issue #13034) Reported
	  by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
	  tilghman (license 14) patch_trunk_183progress_klaus3000.txt
	  uploaded by klaus3000 (license 65) Tested by: klaus3000

2009-03-19 18:28 +0000 [r183291]  Jason Parker <jparker@digium.com>

	* main/asterisk.exports: Export some more required symbols.

2009-03-19 17:52 +0000 [r183145-183241]  Russell Bryant <russell@digium.com>

	* main/loader.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Remove the use of RTLD_NOLOAD, as it is not
	  behaving like expected.

	* main/asterisk.exports: Allow the AES API to work.

	* main/asterisk.exports: Add missing semicolon in exports script.

2009-03-19 16:15 +0000 [r183126]  David Vossel <dvossel@digium.com>

	* include/asterisk/features.h, apps/app_dial.c, res/res_features.c,
	  res/res_features.exports: Allow disconnect feature before a call
	  is bridged feature.conf has a disconnect option. By default this
	  option is set to '*', but it could be anything. If a user wishes
	  to disconnect a call before the other side answers, only '*' will
	  work, regardless if the disconnect option is set to something
	  else. This is because features are unavailable until bridging
	  takes place. The default disconnect option, '*', was hardcoded in
	  app_dial, which doesn't make any sense from a user perspective
	  since they may expect it to be something different. This patch
	  allows features to be detected from outside of the bridge, but
	  not operated on. In this case, the disconnect feature can be
	  detected before briding and handled outside of features.c.
	  (closes issue #11583) Reported by: sobomax Patches:
	  patch-apps__app_dial.c uploaded by sobomax (license 359)
	  11583.latest-patch uploaded by murf (license 17)
	  detect_disconnect.diff uploaded by dvossel (license 671) Tested
	  by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/

2009-03-19 16:13 +0000 [r183123]  Russell Bryant <russell@digium.com>

	* main/asterisk.exports: Allow the CallerID API to work again.

2009-03-19 16:04 +0000 [r183115]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix an issue where cancelled outgoing SIP
	  calls would erroneously report the device as "in use." A user was
	  having an issue where if an outgoing SIP call was canceled, the
	  SIP device would remain in use if we had not received any
	  response to the initial INVITE we sent out. The SIP device would
	  remain in use until the autocongestion timer was exhausted. I
	  tracked down the cause of this to be the section of code I am
	  removing here. I asked several people what the purpose of this
	  code was meant to be, but no one could give me any sort of answer
	  as to why this was here. The person who was having this issue has
	  been using this patch for several months and it has stopped the
	  problems they have had. AST-196

2009-03-18 20:02 +0000 [r182963-182965]  Jeff Peeler <jpeeler@digium.com>

	* configure, autoconf/ast_check_openh323.m4: fix typo which broke
	  configure

	* channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
	  configure, autoconf/ast_check_openh323.m4,
	  channels/h323/compat_h323.h, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
	  Plus library to be used in addition to the OpenH323 library
	  Chan_h323 can now be compiled against both the previously
	  supported versions of OpenH323 as well as the current H.323 Plus
	  (version 1.20.2). The configure script has been modified to look
	  in the default install location of h323 to hopefully help avoid
	  using the environment variables OPENH323DIR and PWLIBDIR. Also,
	  the CLI command "h323 show version" has been added which
	  indicates which version of h323 is in use. (closes issue 0011261)
	  Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
	  uploaded by jthurman (license 614)

2009-03-18 11:31 +0000 [r182882]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/callerid.h, channels/chan_dahdi.c,
	  main/callerid.c: fix another symbol namespace issue (reported by
	  Andrew on asterisk-dev)

2009-03-18 02:09 +0000 [r182810]  Russell Bryant <russell@digium.com>

	* main/poll.c, main/io.c, main/channel.c, main/manager.c,
	  channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c,
	  include/asterisk/poll-compat.h, channels/chan_alsa.c,
	  main/asterisk.c, apps/app_nbscat.c, main/Makefile,
	  include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
	  include/asterisk/io.h, include/asterisk/channel.h: Fix cases
	  where the internal poll() was not being used when it needed to
	  be. We have seen a number of problems caused by poll() not
	  working properly on Mac OSX. If you search around, you'll find a
	  number of references to using select() instead of poll() to work
	  around these issues. In Asterisk, we've had poll.c which
	  implements poll() using select() internally. However, we were
	  still getting reports of problems. vadim investigated a bit and
	  realized that at least on his system, even though we were
	  compiling in poll.o, the system poll() was still being used. So,
	  the primary purpose of this patch is to ensure that we're using
	  the internal poll() when we want it to be used. The changes are:
	  1) Remove logic for when internal poll should be used from the
	  Makefile. Instead, put it in the configure script. The logic in
	  the configure script is the same as it was in the Makefile.
	  Ideally, we would have a functionality test for the problem, but
	  that's not actually possible, since we would have to be able to
	  run an application on the _target_ system to test poll()
	  behavior. 2) Always include poll.o in the build, but it will be
	  empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
	  throughout the source tree to ast_poll(). I feel that it is good
	  practice to give the API call a new name when we are changing its
	  behavior and not using the system version directly in all cases.
	  So, normally, ast_poll() is just redefined to poll(). On systems
	  where AST_POLL_COMPAT is defined, ast_poll() is redefined to
	  ast_internal_poll(). 4) Change poll() in main/poll.c to be
	  ast_internal_poll(). It's worth noting that any code that still
	  uses poll() directly will work fine (if they worked fine before).
	  So, for example, out of tree modules that are using poll() will
	  not stop working or anything. However, for modules to work
	  properly on Mac OSX, ast_poll() needs to be used. (closes issue
	  #13404) Reported by: agalbraith Tested by: russell, vadim
	  http://reviewboard.digium.com/r/198/

2009-03-18 01:55 +0000 [r182802-182808]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c, main/asterisk.exports (added),
	  res/res_odbc.exports (added), res/res_speech.exports (added),
	  res/res_config_odbc.c, res/res_features.exports (added),
	  build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
	  res/res_indications.c, default.exports (added), makeopts.in,
	  res/res_jabber.exports (added), res/res_monitor.exports (added),
	  res/res_config_pgsql.c, res/res_snmp.c, main/Makefile,
	  res/res_smdi.exports (added), include/asterisk/astobj2.h,
	  res/res_crypto.c, res/res_agi.exports (added), Makefile.rules,
	  res/res_musiconhold.c: Improve the build system to *properly*
	  remove unnecessary symbols from the runtime global namespace.
	  Along the way, change the prefixes on some internal-only API
	  calls to use a common prefix. With these changes, for a module to
	  export symbols into the global namespace, it must have *both* the
	  AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
	  the linker to leave the symbols exposed in the module's .so file
	  (see res_odbc.exports for an example).

	* main/astobj2.c, main/asterisk.exports (removed),
	  res/res_odbc.exports (removed), main/channel.c,
	  res/res_config_odbc.c, res/res_features.exports (removed),
	  default.exports (removed), include/asterisk/frame.h,
	  res/res_jabber.exports (removed), res/res_config_pgsql.c,
	  main/Makefile, res/res_smdi.exports (removed),
	  include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
	  res/res_agi.exports (removed), res/res_speech.exports (removed),
	  include/asterisk/linkedlists.h, main/file.c,
	  build_tools/strip_nonapi (added), res/res_adsi.exports (removed),
	  res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
	  apps/app_chanspy.c, res/res_monitor.exports (removed),
	  main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
	  apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
	  res/res_musiconhold.c: revert commit that included extranous
	  changes

	* /: remove accidentally merged properties

	* main/astobj2.c, main/asterisk.exports (added),
	  res/res_odbc.exports (added), main/channel.c,
	  res/res_config_odbc.c, res/res_features.exports (added),
	  default.exports (added), include/asterisk/frame.h,
	  res/res_jabber.exports (added), res/res_config_pgsql.c,
	  main/Makefile, res/res_smdi.exports (added),
	  include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
	  res/res_agi.exports (added), res/res_speech.exports (added),
	  include/asterisk/linkedlists.h, main/file.c,
	  build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
	  res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
	  apps/app_chanspy.c, res/res_monitor.exports (added),
	  main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
	  /, apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
	  res/res_musiconhold.c: Improve the build system to *properly*
	  remove unnecessary symbols from the runtime global namespace.
	  Along the way, change the prefixes on some internal-only API
	  calls to use a common prefix. With these changes, for a module to
	  export symbols into the global namespace, it must have *both* the
	  AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
	  the linker to leave the symbols exposed in the module's .so file
	  (see res_odbc.exports for an example).

2009-03-17 20:13 +0000 [r182652]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c, apps/app_flash.c: Allow dahdichanname to
	  work as advertised. (closes issue #14056) Reported by: dsedivec
	  Patches: load_from_zapata_conf.patch uploaded by dsedivec
	  (license 638)

2009-03-17 05:50 +0000 [r182449]  Tilghman Lesher <tlesher@digium.com>

	* main/db.c: Fix race in astdb The underlying db1 implementation
	  does not fully isolate the pages retrieved from astdb, so the
	  lock protecting accesses needs to be extended until the copy from
	  the shared memory structure is done. (closes issue #14682)
	  Reported by: makoto

2009-03-16  Leif Madsen <lmadsen@digium.com>

	* Released 1.4.24

2009-03-06  Leif Madsen <lmadsen@digium.com>

	* Released 1.4.24-rc1

2009-03-06 18:23 +0000 [r180567]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Make compilation succeed in dev-mode when
	  IMAP storage is enabled.

2009-03-06 17:19 +0000 [r180532]  David Vossel <dvossel@digium.com>

	* main/enum.c: Fix handling of backreferences for ENUM lookups
	  enum.c did not handle regex backtraces correctly. The '\1' in the
	  regex is a backreference that requires a pattern match to be
	  inserted. The way the code used to work is that it would find the
	  backreference and insert the entire input string minus the '+'.
	  This is incorrect. The regexec() function takes in a variable
	  called pmatch which is an array of structs containing the start
	  and end indexes for each backreference substring. The original
	  code actually passed the pmatch array pointer into regexec but
	  never did anything with it. Now when a backtrace is found, the
	  backtrace number is looked up in the pmatch array and the correct
	  substring is inserted. (closes issue #14576) Reported by:
	  chris-mac Review: http://reviewboard.digium.com/r/187/

2009-03-05 23:26 +0000 [r180380-180464]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: [IMAP] Fix message retrieval issues when
	  identical mailbox names were defined in separate contexts. There
	  was a fix put in a while back so that an X-Asterisk-VM-Context
	  message header was added to stored IMAP voicemails. This would
	  allow for us to differentiate if the same mailbox name was used
	  in multiple contexts. The problem still left was that not all
	  places where messages were retrieved actually attempted to use
	  this header for information when retrieving messages. This commit
	  fixes that so that MWI and message retrieval from VoiceMailMain
	  work as expected. (closes issue #13853) Reported by: vicks1
	  Patches: 13853_v2.patch uploaded by mmichelson (license 60)
	  Tested by: lmadsen

	* apps/app_voicemail.c, configs/voicemail.conf.sample: Fix broken
	  mailbox parsing when searchcontexts option is enabled. When using
	  the searchcontexts option in voicemail.conf, the code made the
	  assumption that all mailbox names defined were unique across all
	  contexts. However, the code did nothing to actually enforce this
	  assumption, nor did it do anything to alert a user that he may
	  have created an ambiguity in his voicemail.conf file by defining
	  the same mailbox name in multiple contexts. With this change, we
	  now will issue a nice long warning if searchcontexts is on and we
	  encounter the same mailbox name in multiple contexts and ignore
	  any duplicates after the first box. Whether searchcontexts is
	  enabled or not, if we come across a duplicate mailbox in the same
	  context, then we will issue a warning and ignore the duplicated
	  mailbox. I have also added a small note to voicemail.conf.sample
	  in the explanation for searchcontexts explaining that you cannot
	  define the same mailbox in multiple contexts if you have enabled
	  the option. (closes issue #14599) Reported by: lmadsen Patches:
	  14599.patch uploaded by mmichelson (license 60) (with slight
	  modification) Tested by: lmadsen

2009-03-05 18:22 +0000 [r180372]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp.c, main/frame.c, include/asterisk/frame.h: Fix problems
	  when RTP packet frame size is changed During some code analysis,
	  I found that calling ast_rtp_codec_setpref() on an ast_rtp
	  session does not work as expected; it does not adjust the
	  smoother that may on the RTP session, in fact it summarily drops
	  it, even if it has data in it, even if the current format's
	  framing size has not changed. This is not good. This patch
	  changes this behavior, so that if the packetization size for the
	  current format changes, any existing smoother is safely updated
	  to use the new size, and if no smoother was present, one is
	  created. A new API call for smoothers,
	  ast_smoother_reconfigure(), was required to implement these
	  changes. Review: http://reviewboard.digium.com/r/184/

2009-03-04 19:22 +0000 [r180194]  Joshua Colp <jcolp@digium.com>

	* main/callerid.c: Look for the number in a callerid string
	  starting from the end. This way a value using <> can exist in the
	  name portion. (issue #AST-194)

2009-03-03 23:01 +0000 [r180010]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c: Make sure we still support zapchan in
	  users.conf, in addition to dahdichan.

2009-03-03 22:48 +0000 [r180006]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Clarify some
	  documentation of queues.conf.sample It had always been possible
	  to explicitly specify a "blank" value for a sound file in
	  queues.conf and have no sound played back. The problem with this
	  is that it would result in some ugly CLI warnings from file.c.
	  This commit introduces a check when playing a file in app_queue
	  to see if the name of the file is zero-length and return early if
	  that is the case. Also, the ability to specify the blank sound
	  files in queues.conf is now mentioned more clearly in
	  queues.conf.sample (closes issue #14227) Reported by: caspy

2009-03-03 18:27 +0000 [r179840]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Do not assume that the bridge_cdr is still
	  attached to the channel when the 'h' exten is finished executing.
	  It is possible for a masquerade operation to occur when the 'h'
	  exten is operating. This operation moves the CDR records around
	  causing the bridge_cdr to no longer exist on the channel where it
	  is expected to. We can not safely modify it afterwards because of
	  this, so don't even try. (closes issue #14564) Reported by: meric

2009-03-03 18:11 +0000 [r179807]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
	  utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y,
	  main/ast_expr2f.c: These changes allow AEL to better check ${}
	  constructs within $[...], that are concatenated with text. I
	  modified and added rules in ast_expr2.fl to better handle the
	  concatenations. I added some default routines to ast_expr2.y so
	  the standalone would compile. It also looks like I haven't run
	  this thru bison since 2.1, so it's good to get this updated. The
	  Makefile has comments added now for check_expr2 and check_expr to
	  explain what they are for, and how to run them. The testexpr2s
	  stuff has been removed, in favor of check_expr2. expr2.testinput
	  has been updated to include the two expressions that inspired
	  these changes (from mcnobody on #asterisk this morning) The
	  regression has been run and all looks well.

2009-03-03 16:45 +0000 [r179741]  Russell Bryant <russell@digium.com>

	* main/channel.c: Ensure chan->fdno always gets reset to -1 after
	  handling a channel fd event. Since setting fdno to -1 had to be
	  moved, a couple of other code paths that do process an fd event
	  return early and do not pass through the code path where it was
	  moved to. So, set it to -1 in a few other places, too.

2009-03-03 14:38 +0000 [r179671]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Move where fdno is set to the default value to
	  *after* the read callback of the channel driver is called. We
	  have to do this as the underlying channel driver may need the
	  fdno value to determine what to read.

2009-03-03 13:53 +0000 [r179608]  Russell Bryant <russell@digium.com>

	* main/channel.c: Make it easier to detect an improper call to
	  ast_read(). When you call ast_waitfor() on a channel, the index
	  into the channel fds array that holds the file descriptor that
	  poll() determines has input available is stored in fdno. This
	  patch clears out this value after a call to ast_read() and also
	  reports errors if ast_read() is called without an fdno set. From
	  a discussion on the asterisk-dev list.

2009-03-02 23:54 +0000 [r179536]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Fix bridging regression from commit 176701 This
	  fixes a bad regression where the bridge would exit after an
	  attended transfer was made. The problem was due to nexteventts
	  getting set after the masquerade which caused the bridge to
	  return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
	  tim_ringenbach

2009-03-02 23:34 +0000 [r179532]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Move ast_waitfor() down to avoid the results
	  of the API call becoming stale. This call to ast_waitfor() was
	  being done way too soon in this section of code. Specifically,
	  there was code in between the call to waitfor and the code that
	  uses the result that puts the channel in autoservice. By putting
	  the channel in autoservice, the previous results of ast_waitfor()
	  become meaningless, as the autoservice thread will do it's own
	  ast_waitfor() and ast_read() on the channel. So, when we came
	  back out of autoservice and eventually hit the block of code that
	  calls ast_read() on the channel, there may not actually be any
	  input on the channel available. Even though the previous call to
	  ast_waitfor() in app_meetme said there was input, the autoservice
	  thread has since serviced the channel for some period of time.
	  This bug manifested itself while dvossel was doing some testing
	  of MeetMe in Asterisk trunk. He was using the timerfd timing
	  module. When the code hit ast_read() erroneously, it determined
	  that it must have been called because of input on the timer fd,
	  as chan->fdno was set to AST_TIMING_FD, since that was the cause
	  of the last legitimate call to ast_read() done by autoservice. In
	  this test, an IAX2 channel was calling into the MeetMe
	  conference. It was _much_ more likely to be seen with an IAX2
	  channel because of the way audio is handled. Every audio frame
	  that comes in results in a call to ast_queue_frame(), which then
	  uses ast_timer_enable_continuous() to notify the channel thread
	  that a frame is waiting to be handled. So, the chances of
	  ast_waitfor() indicating that a channel needs servicing due to a
	  timer event on an IAX2 event is very high. Finally, it is
	  interesting to note that if a different timing interface was
	  being used, this bug would probably not be noticed. When
	  ast_read() is called and erroneously thinks that there is a timer
	  event to handle, it calls the ast_timer_ack() function. The
	  pthread and dahdi timing modules handle the ack() function being
	  called when there is no event by simply ignoring it. In the case
	  of the timerfd module, it results in a read() on the timer fd
	  that will block forever, as there is no data to read. This caused
	  Asterisk to lock up very quickly. Thanks to dvossel and
	  mmichelson for the fun debugging session. :-)

2009-03-02 23:09 +0000 [r179468]  Tilghman Lesher <tlesher@digium.com>

	* main/app.c: When ending a recording with silence detection,
	  remember to reduce the duration. The end of the recording is
	  correspondingly trimmed, but the duration was not trimmed by the
	  number of seconds trimmed, so the saved duration was necessarily
	  longer than the actual soundfile duration. (closes issue #14406)
	  Reported by: sasargen Patches: 20090226__bug14406.diff.txt
	  uploaded by tilghman (license 14) Tested by: sasargen

2009-03-02 22:58 +0000 [r179461]  Russell Bryant <russell@digium.com>

	* main/channel.c: Ensure that only one thread is calling
	  ast_settimeout() on a channel at a time. For example, with an
	  IAX2 channel, you can have both the channel thread and the
	  chan_iax2 processing threads calling this function, and doing so
	  twice at the same time is a bad thing. (Found in a debugging
	  session with dvossel and mmichelson)

2009-03-02 20:14 +0000 [r179395]  Jason Parker <jparker@digium.com>

	* main/editline/configure, main/editline/np/unvis.c,
	  main/editline/sys.h, main/editline/configure.in: Remove several
	  silly warnings in editline. One about a broken preprocessor
	  directive, and another about strlcpy/strlcat. (closes issue
	  #14264) Reported by: dimas

2009-02-27 19:03 +0000 [r179056]  Jason Parker <jparker@digium.com>

	* doc/channelvariables.txt: Update documentation for DIALEDTIME and
	  ANSWEREDTIME variables. (closes issue #14566) Reported by:
	  klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
	  klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
	  klaus3000 (license 65)

2009-02-26 21:27 +0000 [r178956]  Steve Murphy <murf@digium.com>

	* configs/features.conf.sample, res/res_features.c: This change
	  moves the default feature digit timeout to 1000 ms from the
	  previous default of 500. As per bug 14515, a dev discussion
	  arrived at a "mediated concensus" of a default feature digit
	  timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for
	  distracted phone users in phone booths; kpfleming put his foot
	  down at 1.0 sec. Users who found the previous default max delay
	  of 250 msec perfect, are welcome to override the new default.
	  Notice that I said that 250 msec was the default; wait a minute,
	  you might say, the config file said it was 500 msec!; well,
	  because of the bug fix for 14515, we found that 500 msec was
	  actually enforcing a max of 250. The bug fix would restore 500
	  msec, but we felt even that was a bit tight for most users...
	  2000 msec was pushed earlier by mmichelson, so that reduces to
	  1000 msec after the bug fix. Enjoy!

2009-02-26 17:24 +0000 [r178838]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: IAX2 prune realtime fix Now prune_users()
	  and prune_peers() are called instead of reload_config() to prune
	  all users/peers that are realtime. These functions remove all
	  users/peers with the rtfriend and delme flags set.
	  iax2_prune_realtime() also lacked the code to properly delete a
	  single friend. For example. if iax2 prune realtime <friend> was
	  called, only the peer instance would be removed. The user would
	  still remain. (closes issue #14479) Reported by: mousepad99
	  Review: http://reviewboard.digium.com/r/176/

2009-02-26 17:09 +0000 [r178640-178804]  Steve Murphy <murf@digium.com>

	* res/res_features.c: This patch prevents the feature detection
	  timeout from being cut in half. Because the ast_channel_bridge()
	  call will return 0 and pass a frame pointer for both DTMF_BEGIN
	  and DTMF_END, the feature_timer field in hte config struct is
	  getting decremented twice, which effectively cuts the
	  digittimeout in half. I added conditions to the if statement to
	  only let DTMF_END frames to flow thru, which solved the problem.
	  Also, when the frame pointer is null, let control flow thru--
	  this usually happens on timeouts. I added a comment to the code
	  to explain what's going on and why. Many thanks to sodom for
	  reporting this problem. Personnally, it always seemed like
	  something was wrong with the featuredigittimeout, but I never
	  could quite decide what... and was too busy to investigate. This
	  bug forced the issue, and now we know. Sodom had other issues in
	  14515, but I couldn't reproduce them. If he still has problems,
	  and wants to get them solved, he is welcome to reopen 14515.
	  (closes issue #14515) Reported by: sodom Patches: 14515.patch
	  uploaded by murf (license 17) Tested by: murf, sodom

	* main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the
	  fixes nec. to make 1.4 asterisk dialplan expressions ($[...])
	  8-bit transparent While I was updating ast_expr2.fl, I missed one
	  rule that would allow 8-bit chars to be caught in tokens; and in
	  so doing, it absorbs the ${ sequence and messes up the checking
	  of raw exprs by AEL. Trunk already has these changes. (closes
	  issue #14543) Reported by: klaus3000 Patches: patch.14543
	  uploaded by murf (license 17) Tested by: murf

2009-02-25 12:43 +0000 [r178508]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Update the copyright year for the main page of
	  the doxygen documentation.

2009-02-24 23:25 +0000 [r178445]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample: Add section about the #exec
	  command in configuration files. (closes issue #14540) Reported
	  by: jtodd Patch by: jtodd, with additional notes by tilghman
	  (license 14)

2009-02-24 20:36 +0000 [r178373]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset
	  to 0 properly. (issue #14460) Reported by: moliveras Tested by:
	  russell

2009-02-24 17:02 +0000 [r178266]  Terry Wilson <twilson@digium.com>

	* apps/app_dahdiras.c, res/res_musiconhold.c: Change include order
	  to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets
	  defined before linux/types.h is included, the __s32 type doesn't
	  get defined

2009-02-24 15:16 +0000 [r178205]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Skip check for extension when subscribing
	  for MWI. Since the remote side is not actually subscribing to a
	  specific extension when subscribing for MWI just skip the check
	  to see if the extension exists. They can't use it to specify the
	  mailbox either since we require configuration of that in sip.conf
	  (closes issue #14531) Reported by: festr

2009-02-23 23:09 +0000 [r178141]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Fix infinite DTMF when a BEGIN is received without an
	  END. This commit is related to rev 175124 of 1.4 where a previous
	  attempt was made to fix this problem. The problem with the
	  previous patch was that the inserted code needed to go _before_
	  setting the lastrxts to the current timestamp. Because those were
	  the same, the dtmfcount variable was never decremented, and so
	  the END was never sent. In passing, I removed the dtmfsamples
	  variable which was completed unused. I also removed a redundant
	  setting of the lastrxts variable. (closes issue #14460) Reported
	  by: moliveras

2009-02-20 22:59 +0000 [r177701-177786]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Don't print the CR-NL combination when we aren't
	  outputting to the manager. An embedded CR-NL in a CLI command
	  screws up several AMI parsers that don't expect to see that
	  combination in the middle of output. (Closes issue #14305)
	  Reported by: martins Patch by: tilghman

	* include/asterisk/threadstorage.h: This exception does not appear
	  to still be true for Solaris 10, and OpenSolaris definitely needs
	  it to be removed. Fixed for snuff-home on -dev channel.

2009-02-20 20:17 +0000 [r177696]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
	  undefined audio codecs in chan_iax2 During iax2 call negotiation,
	  supported codecs are passed in an Information Element containing
	  a 2 byte field where each bit correlates to a specific codec. In
	  1.4 only audio codec bits 0-12 are defined, leaving bits 13-15
	  undefined. By default all bits are enabled unless specified
	  otherwise. Since its a 2 byte field and 13-15 are not defined,
	  these bits are never turned off. In trunk, bits 13-15 are
	  defined, which means 1.4 is advertising support for codecs it
	  does not have when talking to trunk. I fixed this by adding
	  #define for undefined audio codec bits. These bits are then
	  removed from iax2's full bandwidth capabilities. (closes issue
	  #14283) Reported by: jcovert

2009-02-19 22:51 +0000 [r177540]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch
	  fixes a problem with 8-bit input to the ast_expr2 scanner. The
	  real culprit was the --full argument to flex in the Makefile!
	  This causes a 7-bit scanner to be generated. I reviewed the rules
	  and found one rule where I needed to specifically include 8-bit
	  chars for a token. I tested against the text supplied by ibercom,
	  and all looks very well. This has been there a surprisingly long
	  time! (closes issue #14498) Reported by: ibercom Patches:
	  14498.patch uploaded by murf (license 17) Tested by: murf

2009-02-19 22:26 +0000 [r177536]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Fix up potential crashes, by reducing the
	  sharing between interactive and non-interactive threads. (closes
	  issue #14253) Reported by: Skavin Patches:
	  20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Skavin

2009-02-19 18:58 +0000 [r177450]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Force a MWI notification after subscribe
	  request. Reported by the Resiprocate dev team. Thanks!

2009-02-19 16:37 +0000 [r177383]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: If we are able to create a speech
	  structure unset the ERROR variable in case it was previously set.
	  (issue #LUMENVOX-13)

2009-02-18 22:43 +0000 [r177225]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression
	  of sorts that was introduced in rev 24425. It basically fixes
	  AST-190/ABE-1782. What was wrong: the user has 6000 extensions in
	  one context; and then 6000 contexts, one per extension. The
	  parser could only handle about 4893 of the 6000 extens in the
	  single context. This was due to the regression I mentioned. To
	  get rid of shift/reduce conflicts, Luigi set up right-recursive
	  lists for globals, context elements, switch lists, and
	  statements. Right recursive lists got rid of the warnings, but
	  instead, they use up a tremendous amount of stack space when the
	  lists are long. I saw this a few years back, and resolved not to
	  fix it until someone complained. That day has arrived! After the
	  changes were made, I ran the regression test suite, and there
	  were no problems. I took the test case the user provided, and
	  added 100,000 extensions to the single context, that already had
	  6,000 extens in it. (I'll see your 6, and raise you 100!) It
	  takes a few minutes to read it all in, check it and generate code
	  for it, but no problems. So, I think I can say that
	  fundamentally, there are no longer any limits on the number of
	  items you can place in contexts, statement blocks, switches, or
	  globals, beyond your virt mem constraints.

2009-02-18 20:06 +0000 [r177160]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx,
	  autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h,
	  channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
	  channels/h323/ast_ptlib.h (added), configure,
	  channels/h323/compat_h323.h, configure.ac,
	  channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323
	  to build against PTLib as well as the older PWLib Several changes
	  in PTLib have occurred requiring build time detection. Changes
	  accounted for include the library name change, config option
	  change, install location change, and a boolean type change which
	  is handled by ast_ptlib.h. Also, the sed check has been modified
	  to properly work with autoconf >= 2.62. (closes issue #14224)
	  Reported by: bergolth Patches: asterisk-autoconf-sed.patch
	  uploaded by bergolth (license 661) asterisk-pwlib-v3.patch
	  uploaded by bergolth (license 661) Tested by: jpeeler

2009-02-18 18:30 +0000 [r177096]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/config.h: Document the return value of the
	  update method (as requested on -dev list)

2009-02-18 17:41 +0000 [r176945-177039]  Doug Bailey <dbailey@digium.com>

	* main/utils.c: Merged revisions 177035 manually from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
	  dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
	  Fixed error where a check for an zero length, terminated string
	  was needed. ........

	* main/utils.c: Need to take into account the \0 terminator of the
	  old string to determine the amount available.

2009-02-18 00:34 +0000 [r176810]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
	  with G723. This commit brings in the changes that were living out
	  on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch.
	  codec_dahdi.c now always uses signed linear as the simple codec
	  so that a soft g729 codec will not end up being preferred to the
	  hardware codec. There are also changes to allow codec_dahdi.c to
	  feed packets to the hardware in the native sample size of the
	  codec. This solves problems with choppy audio when using G723.

2009-02-17 21:54 +0000 [r176701]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, res/res_features.c, include/asterisk/channel.h:
	  Modify bridging to properly evaluate DTMF after first warning is
	  played The main problem is currently if the Dial flag L is used
	  with a warning sound, DTMF is not evaluated after the first
	  warning sound. To fix this, a flag has been added in
	  ast_generic_bridge for playing the warning which ensures that if
	  a scheduled warning is missed, multiple warrnings are not played
	  back (due to a feature evaluation or waiting for digits).
	  ast_channel_bridge was modified to store the nexteventts in the
	  ast_bridge_config structure as that information was lost every
	  time ast_channel_bridge was reentered, causing a hangup due to
	  incorrect time calculations. (closes issue #14315) Reported by:
	  tim_ringenbach Reviewed on reviewboard:
	  http://reviewboard.digium.com/r/163/

2009-02-17 21:21 +0000 [r176426-176661]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Backport change to 1.4: Prior to
	  masquerade, move the group definitions to the channel performing
	  the masq, so that the group count lingers past the bridge.
	  (closes issue #14275) Reported by: kowalma Patches:
	  20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: kowalma

	* channels/chan_sip.c: After a 'sip reload', qualifies for realtime
	  peers weren't immediately restarted, instead waiting until the
	  next registration. We're now caching the qualify across a
	  reload/restart and starting the qualify immediately upon loading
	  the peer. (closes issue #14196) Reported by: pdf Patches:
	  20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
	  Tested by: pdf

2009-02-16 23:30 +0000 [r176354]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not
	  being relayed correctly during bridging This should have been
	  committed with rev176247, but I missed it. srcupdate frames no
	  longer break out of the native bridge, but are not being sent to
	  the other call leg either. This fixs that. issue #13749

2009-02-16 21:41 +0000 [r176254]  Kevin P. Fleming <kpfleming@digium.com>

	* main/utils.c: correct a logic error in the last stringfields
	  commit... don't mark additional space as allocated if the string
	  was built using already-allocated space

2009-02-16 21:39 +0000 [r176249-176252]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c: Remove unused variable and make dev-mode
	  compilation happy

	* apps/app_meetme.c: Open the DAHDI pseudo device and set it to be
	  nonblocking atomically Apparently on FreeBSD, attempting to set
	  the O_NONBLOCKING flag separately from opening the file was
	  causing an "inappropriate ioctl for device" error. While I cannot
	  fathom why this would be happening, I certainly am not opposed to
	  making the code a bit more compact/efficient if it also fixes a
	  bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
	  uploaded by ys (license 281) Tested by: ys

2009-02-16 21:28 +0000 [r176247]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE
	  breaking out of native bridge In iax2, when a
	  AST_CONTROL_SRCUPDATE is received it breaks from the native
	  bridge, but since there is no code path to handle srcupdate it
	  just goes to be beginning of the loop. This was causing packet
	  storms of srcupdate frames between servers. Now srcupdate frames
	  do not break the native bridge for processing. (closes issue
	  #13749) Reported by: adiemus

2009-02-16 21:10 +0000 [r176216]  Kevin P. Fleming <kpfleming@digium.com>

	* main/utils.c: fix a flaw in the ast_string_field_build() family
	  of API calls; these functions made no attempt to reuse the space
	  already allocated to a field, so every time the field was written
	  it would allocate new space, leading to what appeared to be a
	  memory leak.

2009-02-16 15:33 +0000 [r176029]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't have the Via header stored as a
	  stringfield as it can change often during the lifetime of a
	  dialog. This issue crept up with subscriptions on the AA50. When
	  an outgoing NOTIFY is sent a new branch value is created and the
	  Via header is changed to reflect it. Since this was a stringfield
	  a new spot in the pool was used for the value while the old was
	  left untouched/unused. If the current pool was full a new pool
	  was created. This would cause memory usage to increase steadily.
	  (issue #AA50-2332)

2009-02-15 23:37 +0000 [r175921]  Michiel van Baak <michiel@vanbaak.info>

	* main/pbx.c, channels/chan_sip.c, main/devicestate.c,
	  include/asterisk/manager.h: fix mis-spelling of the word
	  registered. Reported by De_Mon on #asterisk-dev.

2009-02-15 20:33 +0000 [r175777-175825]  Olle Johansson <oej@edvina.net>

	* formats/format_ilbc.c: format_ilbc does not depend on codec
	  libraries and can therefore always be made. My mistake. Ursäkta!

	* formats/format_ilbc.c: Disable format_ilbc.so by default, like
	  codec_ilbc.so

	* channels/chan_sip.c: Make sure that the debug line is not printed
	  on debug level 0

2009-02-13 21:53 +0000 [r175698]  Jason Parker <jparker@digium.com>

	* include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather,
	  Zaptel is actually Zaptel. (in case you're confused, DAHDI is
	  still DAHDI)

2009-02-13 19:47 +0000 [r175407-175590]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a potential crash situation when using
	  IMAP voicemail If calling into VoiceMailMain when using IMAP
	  storage, it was possible to crash Asterisk by hanging up the
	  phone when prompted for a voicemail mailbox. This patch fixes the
	  issue. While it may appear that this patch is superficial, it
	  allows code execution to continue to the failure case just below
	  the IMAP_STORAGE code block where this patch has been applied
	  (closes issue #14473) Reported by: dwpaul Patches:
	  voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
	  689)

	* main/file.c: Fix a place where filestreams were not refcounted
	  properly This section was already present in trunk and other
	  branches, but did not exist in 1.4. (closes issue #14395)
	  Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut
	  (license 60) Tested by: ZX81

2009-02-12 21:19 +0000 [r175311]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c: Fix crashes when receiving certain T.38 packets.
	  Also, increase the maximum size of T.38 packets and warn users
	  when they try to set the limits above those maximums. (closes
	  issue #13050) Reported by: schern Patches:
	  20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: schern

2009-02-12 20:34 +0000 [r175187-175294]  Jeff Peeler <jpeeler@digium.com>

	* res/res_features.c: Fix ParkedCall event information for From
	  field in the case of a blind transfer If the parker information
	  can not be obtained from the peer, try and see if the
	  BLINDTRANSFER channel variable has been set. Previously, a blind
	  transfer to the ParkAndAnnounce app would return nothing for the
	  From. Closes AST-189

	* res/res_features.c: Fix crash in event of failed attempt to
	  transfer to parking The peer may not necessarily exist, such as
	  in the case of a transfer to ParkAndAnnounce. In this case don't
	  try to play a sound to it.

2009-02-12 16:51 +0000 [r175124]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Don't send DTMF for infinite time if we do not
	  receive an END event. I thought that this was going to end up
	  being a pretty gnarly fix, but it turns out that there was
	  actually already a configuration option in rtp.conf, dtmftimeout,
	  that was intended to handle this situation. However, in between
	  Asterisk 1.2 and Asterisk 1.4, the code that processed the option
	  got lost. So, this commit brings it back to life. The default
	  timeout is 3 seconds. However, it is worth noting that having
	  this be configurable at all is not really the recommended
	  behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the
	  time period of extending the tone is necessary to avoid that a
	  tone "gets stuck". Regardless of the algorithm used, the tone
	  SHOULD NOT be extended by more than three packet interarrival
	  times. A slight extension of tone durations and shortening of
	  pauses is generally harmless. Three seconds will pretty much
	  _always_ be far more than three packet interarrival times.
	  However, that behavior is not required, so I'm going to leave it
	  with our legacy behavior for now. Code from
	  svn/asterisk/team/russell/issue_14460 (closes issue #14460)
	  Reported by: moliveras

2009-02-12 10:16 +0000 [r175029]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Set the initiator attribute to lowercase
	  in our replies when receiving calls. This attribute contains a
	  JID that identifies the initiator of the GoogleTalk voice
	  session. The GoogleTalk client discards Asterisk's replies if the
	  initiator attribute contains uppercase characters. (closes issue
	  #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
	  by jcovert (license 551) Tested by: jcovert

2009-02-12 00:19 +0000 [r174997]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy
	  commit by russell via SMS.

2009-02-12 00:01 +0000 [r174985-174986]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Clear out the current event after forcing the end of
	  a digit

	* main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END
	  event is ever received

2009-02-11 20:54 +0000 [r174885]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, apps/app_macro.c: Restore a behavior that was
	  recently changed, when we fixed issue #13962 and issue #13363
	  (related to issue #6176). When a hangup occurs during a Macro
	  execution in earlier 1.4, the h extension would execute within
	  the Macro context, whereas it was always supposed to execute only
	  within the main context (where Macro was called). So this fix
	  checks for an "h" extension in the deepest macro context where a
	  hangup occurred; if it exists, that "h" extension executes,
	  otherwise the main context "h" is executed. (closes issue #14122)
	  Reported by: wetwired Patches: 20090210__bug14122.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: andrew

2009-02-10 18:50 +0000 [r174644]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Go off hold when we get an empty reinvite
	  telling us to. (closes issue #14448) Reported by: frawd Patches:
	  hold_invite_nosdp.patch uploaded by frawd (license 610)

2009-02-10 17:52 +0000 [r174583]  Matthew Nicholson <mnicholson@digium.com>

	* main/jitterbuf.c: Improve behavior of jitterbuffer when
	  maxjitterbuffer is set. This change improves the way the
	  jitterbuffer handles maxjitterbuffer and dramatically reduces the
	  number of frames dropped when maxjitterbuffer is exceeded. In the
	  previous jitterbuffer, when maxjitterbuffer was exceeded, all new
	  frames were dropped until the jitterbuffer is empty. This change
	  modifies the code to only drop frames until maxjitterbuffer is no
	  longer exceeded. Also, previously when maxjitterbuffer was
	  exceeded, dropped frames were not tracked causing stats for
	  dropped frames to be incorrect, this change also addresses that
	  problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
	  by mnicholson (license 96) Tested by: mnicholson Review:
	  http://reviewboard.digium.com/r/144/

2009-02-10 02:27 +0000 [r174369]  Steve Murphy <murf@digium.com>

	* apps/app_rpt.c: This patch solves some compiler complaints in
	  both 32 and 64-bit environments.

2009-02-09 17:11 +0000 [r174282]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Don't do an SRV lookup if a port is
	  specified RFC 3263 says to do A record lookups on a hostname if a
	  port has been specified, so that's what we're going to do. See
	  section 4.2. (closes issue #14419) Reported by: klaus3000
	  Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by
	  klaus3000 (license 65)

2009-02-09 14:48 +0000 [r174218]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Don't overwrite our pointer to the music
	  class when music on hold stops. We will use this if it starts
	  again to see if we can resume the music where it left off.
	  (closes issue #14407) Reported by: mostyn

2009-02-07 16:15 +0000 [r174148]  Russell Bryant <russell@digium.com>

	* res/snmp/agent.c: Fix a race condition that could cause a crash.

2009-02-06 23:36 +0000 [r174082]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_sip.c: check ast_strlen_zero() before calling
	  ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp()
	  The reporter didn't actually upload a properly-formed patch,
	  instead a modified chan_sip.c file was uploaded. I created a
	  patch to determine the changes, then modified the suggested
	  changes to create a proper fix. The summary above is a complete
	  description of the changes. (closes issue #13547) Reported by:
	  tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license
	  258) Tested by: tecnoxarxa

2009-02-06 17:15 +0000 [r173967-173968]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove a debug message I put in by accident.

	* channels/chan_sip.c: Some clients do not put the call-id for
	  replaces at the beginning, so support it being anywhere in the
	  string. (closes issue #14350) Reported by: fhackenberger

2009-02-06 16:20 +0000 [r173917]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Limit the addition of the Contact header in
	  SIP responses according to various SIP RFCs. (closes issue
	  #13602) Reported by: hjourdain Tested by: mnicholson

2009-02-06 15:43 +0000 [r173900]  Tilghman Lesher <tlesher@digium.com>

	* utils/muted.c: Backport OS X fix from trunk (AGAIN, closes issue
	  #14360)

2009-02-05 23:19 +0000 [r173770]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix logic regarding when to perform an SRV
	  lookup for outgoing REGISTER requests With this fix, we only will
	  perform an SRV lookup at the following times: * The first time we
	  register with a remote registrar * If we send a REGISTER but do
	  not receive a response * If the sendto() function returns an
	  error While I wrote the patch that fixes this issue, a huge
	  amount of credit is due to Brett Bryant, who wrote the initial
	  patch on which I based this one. (closes issue #12312) Reported
	  by: jrast Patches: 12312.patch uploaded by putnopvut (license 60)
	  Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/

2009-02-05 20:47 +0000 [r173696]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Add new configuration option to make shared
	  IMAP mailboxes function as expected. The new option is
	  "imapvmshareid" which is an ID to tag multiple mailboxes using
	  the same IMAP storage location to function as one mailbox. This
	  allows all messages to be retrieved for any user in the group.
	  The patch alters the 'X-Asterisk-VM-Extension' header that is
	  responsible for matching voicemails for a given user. (closes
	  issue #13673) Reported by: howardwilkinson

2009-02-05 20:29 +0000 [r173392-173692]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix situations where queue members could be
	  autopaused unexpectedly Specifically, this patch prevents us from
	  autopausing members when we receive a busy or congestion frame
	  from them. (closes issue #14376) Reported by: fiddur Patches:
	  14376.patch uploaded by putnopvut (license 60) Tested by: fiddur

	* apps/app_mixmonitor.c: Add some missing cleanup to app_mixmonitor

	* apps/app_mixmonitor.c: Fix a problem where a channel pointer
	  becomes invalid due to masquerading or hanging up. app_mixmonitor
	  runs its own thread to monitor the channel's activity and write
	  the mixed audio to a file. Since this thread runs independently
	  of the channel, it is possible that the mixmonitor thread's
	  channel pointer will point to freed memory when the channel
	  either is masqueraded or hangs up (technically, both cases are
	  hangups, but we need to handle the cases slightly differently).
	  The solution for this is to employ a datastore, which has the
	  nice benefit of allowing us to hook into channel masquerades and
	  hangups and update our pointer as necessary. If this looks
	  familiar, this same technique is employed in app_chanspy.
	  app_chanspy is a bit more involved since it does a lot more
	  operations on the channel that is being spied upon.
	  app_mixmonitor does have an extra touch that app_chanspy doesn't
	  have, though. Since there is a thread race between the channel's
	  thread and the mixmonitor thread on a hangup, we em- ploy a
	  condition-and-boolean combination to ensure that the channel
	  thread finishes with our structure before the mixmonitor thread
	  attempts to free it. No crashes! (closes issue #14374) Reported
	  by: aragon Patches: 14374.patch uploaded by putnopvut (license
	  60) Tested by: aragon, putnopvut

	* apps/app_chanspy.c: Revert my previous change because it was
	  stupid

	* apps/app_chanspy.c: Add a missing unlock. Extremely unlikely to
	  ever matter, but it's needed.

2009-02-03 23:35 +0000 [r173248]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
	  off calls. Fixes issue with IAX2 transfers not taking place. As
	  it was, a call that was being transfered would never be handed
	  off correctly to the call ends because of how call numbers were
	  stored in a hash table. The hash table, "iax_peercallno_pvt",
	  storing all the current call numbers did not take into account
	  the complications associated with transferring a call, so a
	  separate hash table was required. This second hash table
	  "iax_transfercallno_pvt" handles calls being transfered, once the
	  call transfer is complete the call is removed from the transfer
	  hash table and added to the peer hash table resuming normal
	  operations. Addition functions were created to handle storing,
	  removing, and comparing items in the iax_transfercallno_pvt
	  table. (issue #13468) Review:
	  http://reviewboard.digium.com/r/140/

2009-02-03 21:57 +0000 [r173211]  Jeff Peeler <jpeeler@digium.com>

	* res/res_features.c: Parking attempts made to one end of a bridge
	  no longer will hang up due to a parking failure. Parking attempts
	  made using either one-touch, or doing either a blind or assisted
	  transfer to the parking extension now keep up the bridge instead
	  of hanging up the attempted parked party. Normal causes for the
	  parking attempt to fail includes the specific specified extension
	  (via PARKINGEXTEN) not being available or if all the parking
	  spaces are currently in use. To avoid having to reverse a
	  masquerade park_space_reserve was made to provide foresight if a
	  parking attempt will succeed and if so reserve the parking space.
	  (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
	  http://reviewboard.digium.com/r/133/

2009-02-03 00:15 +0000 [r173070]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample: Add warning to standard config,
	  that globals may be overridden by other dialplan configuration
	  files. (closes issue #14388) Reported by: macli

2009-02-02 23:48 +0000 [r173066]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: Fix a feature inheritance bug I added after
	  code review

2009-02-02 20:28 +0000 [r172962]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
	  channels/chan_dahdi.c * Added doxygen comments to the major dahdi
	  structures. * Fixed PRI using an incorrect string value if the
	  extension delimiter is not present in the Dial() function. *
	  Fixed some uninitialized string variables on FXS ports.
	  configs/chan_dahdi.conf.sample * Updated some documentation.
	  These changes are already in trunk -r172400

2009-01-31 00:15 +0000 [r172517-172639]  Terry Wilson <twilson@digium.com>

	* configs/features.conf.sample, res/res_features.c: Rename new
	  parkedcallparking option to parkedcallreparking Since this option
	  actually already existed in 1.6.0+, use the same name so as not
	  to confuse people when they upgrade

	* configs/features.conf.sample, apps/app_dial.c,
	  main/global_datastores.c, res/res_features.c,
	  doc/channelvariables.txt, include/asterisk/global_datastores.h,
	  CHANGES: Fix feature inheritance with builtin features When using
	  builtin features like parking and transfers, the AST_FEATURE_*
	  flags would not be set correctly for all instances when either
	  performing a builtin attended transfer, or parking a call and
	  getting the timeout callback. Also, there was no way on a
	  per-call basis to specify what features someone should have on
	  picking up a parked call (since that doesn't involve the Dial()
	  command). There was a global option for setting whether or not
	  all users who pickup a parked call should have
	  AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
	  PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
	  variable which can be set either in the dialplan or with setvar
	  in channels that support it. This variable can be set to any
	  combination of 't', 'k', 'w', and 'h' (case insensitive matching
	  of the equivalent dial options), to set what features should be
	  activated on this channel. The patch moves the setting of the
	  features datastores into the bridging code instead of app_dial to
	  help facilitate this. 2) adds global options parkedcallparking,
	  parkedcallhangup, and parkedcallrecording to be similar to the
	  parkedcalltransfers option for globally setting features. 3) has
	  builtin_atxfer call builtin_parkcall if being transfered to the
	  parking extension since tracking everything through multiple
	  masquerades, etc. is difficult and error-prone 4) attempts to fix
	  all cases of return calls from parking and completed builtin
	  transfers not having the correct permissions (closes issue
	  #14274) Reported by: aragon Patches:
	  fix_feature_inheritence.diff.txt uploaded by otherwiseguy
	  (license 396) Tested by: aragon, otherwiseguy Review
	  http://reviewboard.digium.com/r/138/

2009-01-29 22:54 +0000 [r172438]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, apps/app_nbscat.c, autoconf/ast_func_fork.m4,
	  apps/app_festival.c, build_tools/menuselect-deps.in, configure,
	  apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c,
	  apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c:
	  Lose the CAP_NET_ADMIN at every fork, instead of at startup.
	  Otherwise, if Asterisk runs as a non-root user and the
	  administrator does a 'restart now', Asterisk loses the ability to
	  set QOS on packets. (closes issue #14004) Reported by: nemo
	  Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
	  (license 14) Tested by: Corydon76

2009-01-29 08:48 +0000 [r172169]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure that we always add the hangupcause
	  headers. In some cases, the owner was disconnected before we
	  checked for the cause. This patch implements a temporary storage
	  in the pvt and use that instead. The code is based on ideas from
	  code from Adomjan in issue #13385 (Add support for Reason:
	  header) Thanks to Klaus Darillion for testing! (closes issue
	  #14294) related to issue #13385 Reported by: klaus3000 and
	  adomjan Patches: bug14294b.diff uploaded by oej (license 306)
	  Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by
	  adomjan (license 487) Tested by: oej, klaus3000

2009-01-28 18:51 +0000 [r172030]  Steve Murphy <murf@digium.com>

	* apps/app_channelredirect.c, main/pbx.c, main/manager.c,
	  res/res_features.c, include/asterisk/channel.h: This patch fixes
	  h-exten running misbehavior in manager-redirected situations.
	  What it does: 1. A new Flag value is defined in
	  include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which
	  used as a messenge to the bridge hangup exten code not to run the
	  h-exten there (nor publish the bridge cdr there). It will done at
	  the pbx-loop level instead. 2. In the manager Redirect code, I
	  set this flag on the channel if the channel has a non-null pbx
	  pointer. I did the same for the second (chan2) channel, which
	  gets run if name2 is set... and the first succeeds. 3. I restored
	  the ending of the cdr for the pbx loop h-exten running code.
	  Don't know why it was removed in the first place. 4. The first
	  attempt at the fix for this bug was to place code directly in the
	  async_goto routine, which was called from a large number of
	  places, and could affect a large number of cases, so I tested
	  that fix against a fair number of transfer scenarios, both with
	  and without the patch. In the process, I saw that putting the fix
	  in async_goto seemed not to affect any of the blind or attended
	  scenarios, but still, I was was highly concerned that some other
	  scenarios I had not tested might be negatively impacted, so I
	  refined the patch to its current scope, and jmls tested both. In
	  the process, tho, I saw that blind xfers in one situation, when
	  the one-touch blind-xfer feature is used by the peer, we got
	  strange h-exten behavior. So, I inserted code to swap CDRs and to
	  set the HANGUP_DONT field, to get uniform behavior. 5. I added
	  code to the bridge to obey the HANGUP_DONT flag, skipping both
	  publishing the bridge CDR, and running the h-exten; they will be
	  done at the pbx-loop (higher) level instead. 6. I removed all the
	  debug logs from the patch before committing. 7. I moved the
	  AUTOLOOP set/reset in the h-exten code in res_features so it's
	  only done if the h-exten is going to be run. A very minor
	  performance improvement, but technically correct. (closes issue
	  #14241) Reported by: jmls Patches:
	  14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by
	  murf (license 17) Tested by: murf, jmls

2009-01-28 17:25 +0000 [r171963]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Clarify log message (suggested by
	  manxpower on #asterisk-dev)

2009-01-28 13:07 +0000 [r171837]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Add a better explanation of the
	  difference between the device namespace and the dialplan for
	  newbies.

2009-01-27 21:55 +0000 [r171621-171689]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fix devicestate problems for "always-on"
	  agent channels A revision to chan_agent attempted to "inherit"
	  the device state of the underlying channel in order to report the
	  device state of an agent channel more accurately. The problem
	  with the logic here is that it makes no sense to use this for
	  always-on agents. If the agent is logged in, then to the
	  underlying channel, the agent will always appear to be "in use,"
	  no matter if the agent is on a call or not. The reason is that to
	  the underlying channel, the channel is currently in use on a call
	  to the AgentLogin application. The most common cause that I found
	  for this issue to occur was for a SIP channel to be the
	  underlying channel type for an Agent channel. If the SIP phone
	  re-registers, then the registration will cause the device state
	  core to query the device state of the SIP channel. Since the SIP
	  channel is in use, the Agent channel would also inherit this
	  status. Once the agent channel was set to "in use" there was no
	  way that the device state could change on that channel unless the
	  agent logged out. The solution for this problem is a bit
	  different in 1.4 than it is in the other branches. In 1.4, there
	  will be a one-line fix to make sure that only callback agents
	  will inherit device state from their underlying channel type. For
	  the other branches of Asterisk, since callback support has been
	  removed, there is also no need for device state inheritance in
	  chan_agent, so I will simply be removing it from the code. In
	  addition, the 1.4 source is getting a new comment to help the
	  next person who edits chan_agent.c. I'm adding a comment that a
	  agent_pvt's loginchan field may be used to determine if the agent
	  is a callback agent or not. (closes issue #14173) Reported by:
	  nathan Patches: 14173.patch uploaded by putnopvut (license 60)
	  Tested by: nathan, aramirez

	* main/slinfactory.c: Prevent a crash from occurring when a jitter
	  buffer interpolated frame is removed from a slinfactory
	  slinfactory used the "samples" field of an ast_frame in order to
	  determine the amount of data contained within the frame. In
	  certain cases, such as jitter buffer interpolated frames, the
	  frame would have a non-zero value for "samples" but have NULL
	  "data" This caused a problem when a memcpy call in
	  ast_slinfactory_read would attempt to access invalid memory. The
	  solution in use here is to never feed frames into the slinfactory
	  if they have NULL "data" (closes issue #13116) Reported by:
	  aragon Patches: 13116.diff uploaded by putnopvut (license 60)

2009-01-27 14:33 +0000 [r171527]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Use the same branch tag in CANCEL as in
	  INVITE Originally putnopvut implemented some changes in revision
	  142079 that according to the bug report seemed to have worked
	  then, but somehow fails now. I guess code, as humans, get old and
	  forget stuff. Anyway, this bug caused CANCEL not to work with
	  picky systems. Thanks Fredrik for pointing out where the bug in
	  the SIP messaging was. (closes issue #14346) Reported by: oej
	  Patches: bug14346.diff uploaded by oej (license 306) Tested by:
	  oej

2009-01-26 21:31 +0000 [r171452]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Resolve some synchronization issues in
	  chan_iax2 scheduler handling. The important changes here are
	  related to the synchronization between threads adding items into
	  the scheduler and the scheduler handling thread. By adjusting the
	  lock and condition handling, we ensure that the scheduler thread
	  sleeps no longer and no less than it is supposed to. We also
	  ensure that it does not wake up more often than it has to. There
	  is no bug report associated with this. It is just something that
	  I found while putting scheduler thread handling into a reusable
	  form (review 129). Review: http://reviewboard.digium.com/r/131/

2009-01-26 12:51 +0000 [r171264]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't retransmit 401 on REGISTER requests
	  when alwaysauthreject=yes (closes issue #14284) Reported by:
	  klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt
	  uploaded by klaus3000 (license 65) Tested by: klaus3000

2009-01-25 23:44 +0000 [r171120-171187]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_oss.c: Correctly track the hookstate (closes issue
	  #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt
	  uploaded by Corydon76 (license 14)

	* res/res_agi.c: Err, yeah.

	* res/res_agi.c: Add thread to kill zombies, when child processes
	  don't die immediately on SIGHUP. (closes issue #13968) Reported
	  by: eldadran Patches: 20090114__bug13968.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: eldadran

2009-01-25 13:33 +0000 [r170979]  Sean Bright <sean.bright@gmail.com>

	* apps/app_page.c: Resolve a logic error that was causing Page() to
	  crash when more than one channel was specified. (closes issue
	  #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
	  uploaded by seanbright (license 71) Tested by: kc0bvu

2009-01-24 13:55 +0000 [r170836]  Tilghman Lesher <tlesher@digium.com>

	* configs/res_odbc.conf.sample: Remove superfluous implementation
	  note (closes issue #14319)

2009-01-23 20:55 +0000 [r170671-170719]  Mark Michelson <mmichelson@digium.com>

	* configs/res_odbc.conf.sample: Add notes to the idlecheck
	  explanation in res_odbc.conf.sample (closes issue #14319)
	  Reported by: klaus3000 Patches:
	  patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000
	  (license 65)

	* contrib/i18n.testsuite.conf: Update contrib/i18n.testsuite.conf
	  to not use deprecated syntax * Convert Wait,1 to Wait(1) *
	  Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
	  priorities beyond the first Also added test for Chinese numbers,
	  too. (closes issue #14320) Reported by: dant Patches:
	  i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
	  670)

2009-01-23 20:16 +0000 [r170648]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: When a channel is answered make sure any
	  indications currently playing stop. Usually the phone would do
	  this but if the channel was already answered then they are being
	  generated by Asterisk and we darn well need to stop them. (closes
	  issue #14249) Reported by: RadicAlish

2009-01-23  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.4.23.1 released.

	* channels/chan_iax2.c: Regression fix for AST-2009-001 security
	fix.

2009-01-21  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.4.23 released.

2009-01-20 18:49 -0500 [r169581]  Terry Wilson <twilson@digium.com>

	* One-touch parking was calling back the wrong channel on timeout

2009-01-20 13:40 -0500 [r169485]  Terry Wilson <twilson@digium.com>

	* Don't play audio to the channel if we've masqueraded  (closes 
	  issue #14066) Reported by: bluefox Tested by: otherwiseguy, 
	  bluefox

2009-01-16  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.23-rc4 released.

2009-01-16 00:19 +0000 [r168745]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
	  in this case) fails a consistency check because it can't find a
	  matching extension. The problem was a missing instruction to end
	  the range notation in the code where it converts the pattern into
	  a regex and uses the regex code to determine the match. I tested
	  using the AEL code the user supplied, and now, the consistency
	  check passes. (closes issue #14141) Reported by: dimas

2009-01-15 18:43 +0000 [r168721]  Olle Johansson <oej@edvina.net>

	* configs/extconfig.conf.sample: Meetme actually has realtime but
	  wasn't documented

2009-01-15 18:22 +0000 [r168716]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: Convert call to park_call_full to
	  masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
	  return value, we need to use masqueraded parking, otherwise we
	  will try to call ast_hangup() in __pbx_run() and in
	  do_parking_thread() and then promptly crash. (closes issue
	  #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
	  issue #14228) Reported by: kobaz Tested by: otherwiseguy

2009-01-15 01:20 +0000 [r168633]  Tilghman Lesher <tlesher@digium.com>

	* /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
	  security fix AST-2009-001 (Closes issue #14238)

2009-01-15 00:11 +0000 [r168628]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix some crashes from bad datastore handling in
	  app_queue.c * The queue_transfer_fixup function was searching for
	  and removing the datastore from the incorrect channel, so this
	  was fixed. * Most datastore operations regarding the
	  queue_transfer datastore were being done without the channel
	  locked, so proper channel locking was added, too. (closes issue
	  #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
	  putnopvut (license 60) Tested by: ZX81, festr

2009-01-14 21:48 +0000 [r168622]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
	  process ID values. The allocated process IDs could overflow their
	  respective NT and TE fields. Affects outgoing calls.

2009-01-14 20:52 +0000 [r168614]  Sean Bright <sean.bright@gmail.com>

	* contrib/scripts/autosupport: Update autosupport script to supply
	  info for both Zaptel and DAHDI in 1.4 and be sure to run
	  dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
	  #14132) Reported by: dsedivec Patches:
	  asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
	  asterisk-trunk-autosupport.patch uploaded by dsedivec (license
	  638)

2009-01-14 19:34 +0000 [r168608]  Steve Murphy <murf@digium.com>

	* apps/app_page.c: app_page was failing to compile in dev-mode on
	  my gcc-4.2.4 system. This change gets rid of the warning.

2009-01-14 19:02 +0000 [r168603]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c: Don't read into a buffer without first checking if
	  a value is beyond the end. (closes issue #13600) Reported by:
	  atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
	  (license 14) Tested by: atis

2009-01-14 16:19 +0000 [r168598]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fix a logic error I found while searching
	  through chan_agent.c I found that the allow_multiple_logins
	  function would never return 0 due to an incorrect comparison
	  being used when traversing the list of agents. While I was
	  modifying this function, I also did a little bit of coding
	  guidelines cleanup, too.

2009-01-14 01:27 +0000 [r168593]  Terry Wilson <twilson@digium.com>

	* apps/app_page.c: Don't overflow when paging more than 128
	  extensions The number of available slots for calls in app_page
	  was hardcoded to 128. Proper bounds checking was not in place to
	  enforce this limit, so if more than 128 extensions were passed to
	  the Page() app, Asterisk would crash. This patch instead
	  dynamically allocates memory for the ast_dial structures and
	  removes the (non-functional) arbitrary limit. This issue would
	  have special importance to anyone who is dynamically creating the
	  argument passed to the Page application and allowing more than
	  128 extensions to be added by an outside user via some external
	  interface. The patch posted by a_villacis was slightly modified
	  for some coding guidelines and other cleanups. Thanks,
	  a_villacis! (closes issue #14217) Reported by: a_villacis
	  Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
	  uploaded by a (license 660) Tested by: otherwiseguy

2009-01-13 19:13 +0000 [r168561]  Russell Bryant <russell@digium.com>

	* main/indications.c, main/channel.c, apps/app_read.c,
	  channels/chan_misdn.c, funcs/func_channel.c,
	  include/asterisk/indications.h, apps/app_disa.c, main/app.c,
	  res/snmp/agent.c, include/asterisk/channel.h,
	  res/res_indications.c: Revert unnecessary indications API change
	  from rev 122314

2009-01-13 18:34 +0000 [r168551]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't pass a value with a side effect to a
	  macro (closes issue #14176) Reported by: paraeco Patches:
	  chan_sip.c.diff uploaded by paraeco (license 658)

2009-01-13 17:48 +0000 [r168546]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_logic.c: If either conditional is NULL, don't try
	  copying it. (closes issue #14226) Reported by: caspy Patches:
	  20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)

2009-01-12 21:42 +0000 [r168507-168516]  Jeff Peeler <jpeeler@digium.com>

	* res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
	  the app CDR field for AGI commands that are not executing an
	  application via "exec".

	* channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
	  Tested by: denisgalvao This gits rid of the notion of an
	  owning_app allowing the request and hangup to be initiated by
	  different threads. Originating from an active agent channel
	  requires this. The implementation primarily changes __login_exec
	  to wait on a condition variable rather than a lock. Review:
	  http://reviewboard.digium.com/r/35/

2009-01-12 14:58 +0000 [r168482]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: I am reverting the fix made in revision
	  168128 (and its upward merges) after being contacted by Olle
	  Johansson and being shown how this fix is incorrect. Thanks to
	  Olle for clearing this up for me.

2009-01-12 14:57 +0000 [r168480]  Russell Bryant <russell@digium.com>

	* configs/indications.conf.sample: s/ringdance/ringcadence/ for
	  Bulgaria

2009-01-10 20:47 +0000 [r168267-168382]  Kevin P. Fleming <kpfleming@digium.com>

	* README: small commit to test new server

	* README: small commit to test new server

	* sounds/Makefile: update to use new sound file packages that
	  include license files

2009-01-09 22:14 +0000 [r168198]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Make this compile for mvanbaak

2009-01-09 21:28 +0000 [r168191]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
	  Miscellaneous doxygen comments added.

2009-01-09 20:08 +0000 [r168128]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add check_via calls to more request handlers
	  INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
	  checking the topmost Via to determine where to send the response.
	  Adding check_via calls to those request handlers solves this.
	  (closes issue #13071) Reported by: baron Patches: check_via.patch
	  uploaded by baron (license 531) Tested by: baron

2009-01-08 22:08 +0000 [r167840]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Don't truncate database results at 255 chars.
	  (closes issue #14069) Reported by: evandro Patches:
	  20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)

2009-01-08 17:24 +0000 [r167620-167714]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: remove an unnecessary argument to
	  queue_request()

	* channels/chan_sip.c: When a SIP request or response arrives for a
	  dialog with an associated Asterisk channel, and the lock on that
	  channel cannot be obtained because it is held by another thread,
	  instead of dropping the request/response, queue it for later
	  processing when the channel lock becomes available.
	  http://reviewboard.digium.com/r/117/

2009-01-07 22:35 +0000 [r167432-167566]  Russell Bryant <russell@digium.com>

	* main/file.c: Fix the last couple of places where free() was
	  improperly used directly.

	* main/file.c: Don't fclose() the file early, the filestream
	  destructor will handle it.

	* main/file.c: Only try to close the file if one was actually
	  opened

	* main/file.c: Don't use free() directly. This caused a crash since
	  ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
	  #asterisk-dev

	* main/indications.c: Treat an empty string the same way as a NULL
	  country argument. In passing, simplify the handling of returning
	  a default tone zone.

2009-01-06 21:35 +0000 [r167299]  Mark Michelson <mmichelson@digium.com>

	* main/db.c: Use the correct variable when creating the format
	  string (closes issue #14177) Reported by: nic_bellamy Patches:
	  asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
	  (license 299)

2009-01-06 20:48 +0000 [r167260]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
	  Jan 2009) | 2 lines Security fix AST-2009-001. ........

2009-01-05 16:51 +0000 [r167179]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: A couple of changes to T.38 SDP attribute
	  handling There are some boolean attributes for T.38 such as
	  T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
	  T38FaxTranscodingJBIG. By simply being present, we should treat
	  these as a "true" value. The current code, however, was requiring
	  a 1 or 0 as the value of the attribute in order to parse it. This
	  is due to the fact that there are some T.38 endpoints and
	  gateways that also transmit this information incorrectly. This
	  patch follows the "be liberal in what you accept and strict in
	  what you send" philosophy by accepting both the correctly- and
	  incorrectly-formatted attributes, but only sending information as
	  it is supposed to be sent. It was also discovered that a
	  particular type of T.38 gateway sends some non-standard T.38 SDP
	  attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
	  it used T38MaxDatagram and T38FaxMaxRate respectively. We now
	  will properly accept these attributes as well. Note that there
	  are a lot of patches cited in the below commit message template.
	  This is because the person who submitted these patches is an
	  awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
	  issue #13976) Reported by: linulin Patches:
	  chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
	  (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
	  by arcivanov (license 648)
	  chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
	  (license 648) Tested by: arcivanov

2009-01-01 00:01 +0000 [r166953-167095]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_alsa.c: Repeat attempts to write when we receive
	  -EAGAIN from the driver, as detailed in the ALSA sample code (see
	  http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
	  Reported by: Jerry Geis (via the -users list) Fixed by: me
	  (license 14)

	* channels/chan_local.c: Also inherit the musiconhold class.
	  (Closes #14153) Reported by: Jerry Geis, via the users list.
	  Patch by: me (license 14)

2008-12-28 15:13 +0000 [r166772]  Russell Bryant <russell@digium.com>

	* channels/misdn_config.c: Use strncat() instead of an sprintf() in
	  which source and target buffers overlap
	  http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html

2008-12-23 15:35 +0000 [r166592]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, channels/chan_iax2.c: Compile, even if both
	  DAHDI and Zaptel are not installed. (Closes issue #14120)

2008-12-23 15:16 +0000 [r166568]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Fix a crash resulting from a datastore with
	  inheritance but no duplicate callback The fix for this is to
	  simply set the newly created datastore's data pointer to NULL if
	  it is inherited but has no duplicate callback. (closes issue
	  #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
	  putnopvut (license 60) Tested by: francesco_r

2008-12-23 04:05 +0000 [r166509]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Use the integer form of condition for integer
	  comparisons. (closes issue #14127) Reported by: andrew

2008-12-22 20:56 +0000 [r166380]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c: Fix a deadlock relating to channel locks
	  and autoservice It has been discovered that if a channel is
	  locked prior to a call to ast_autoservice_stop, then it is likely
	  that a deadlock will occur. The reason is that the call to
	  ast_autoservice_stop has a check built into it to be sure that
	  the thread running autoservice is not currently trying to
	  manipulate the channel we are about to pull out of autoservice.
	  The autoservice thread, however, cannot advance beyond where it
	  currently is, though, because it is trying to acquire the lock of
	  the channel for which autoservice is attempting to be stopped.
	  The gist of all this is that a channel MUST NOT be locked when
	  attempting to stop autoservice on the channel. In this particular
	  case, the channel was locked by a call to ast_read. A call to
	  ast_exists_extension led to autoservice being started and stopped
	  due to the existence of dialplan switches. It may be that there
	  are future commits which handle the same symptoms but in a
	  different location, but based on my looks through the code, it is
	  very rare to see a construct such as this one. (closes issue
	  #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
	  by putnopvut (license 60) Tested by: rtrauntvein Review:
	  http://reviewboard.digium.com/r/107/

2008-12-22 17:22 +0000 [r166262-166297]  Russell Bryant <russell@digium.com>

	* main/utils.c: Fix up timeout handling in ast_carefulwrite().

	* include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
	  count handling of MoH classes using astobj2 to resolve crashes.
	  (closes issue #13566) Reported by: igorcarneiro Tested by:
	  russell Review: http://reviewboard.digium.com/r/106/

2008-12-19 23:34 +0000 [r166157]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, funcs/func_audiohookinherit.c (added),
	  channels/chan_sip.c, include/asterisk/audiohook.h,
	  main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
	  Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
	  13538.patch uploaded by putnopvut (license 60) Tested by:
	  putnopvut

2008-12-19 22:30 +0000 [r166093]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
	  apps/app_queue.c: This merges the masqpark branch into 1.4 These
	  changes eliminate the need for (and use of) the KEEPALIVE return
	  code in res_features.c; There are other places that use this
	  result code for similar purposes at a higher level, these appear
	  to be left alone in 1.4, but attacked in trunk. The reason these
	  changes are being made in 1.4, is that parking ends a channel's
	  life, in some situations, and the code in the bridge (and some
	  other places), was not checking the result code properly, and
	  dereferencing the channel pointer, which could lead to memory
	  corruption and crashes. Calling the masq_park function eliminates
	  this danger in higher levels. A series of previous commits have
	  replaced some parking calls with masq_park, but this patch puts
	  them ALL to rest, (except one, purposely left alone because a
	  masquerade is done anyway), and gets rid of the code that tests
	  the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
	  bug 13820 inspired this work, this patch does not solve all the
	  problems mentioned there. I have tested this patch (again) to
	  make sure I have not introduced regressions. Crashes that
	  occurred when a parked party hung up while the parking party was
	  listening to the numbers of the parking stall being assigned, is
	  eliminated. These are the cases where parking code may be
	  activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
	  parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
	  xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
	  manager. The interesting testing cases for parking are: I. A
	  calls B, A parks B a. B hangs up while A is getting the numbers
	  announced. b. B hangs up after A gets the announcement, but
	  before the parking time expires c. B waits, time expires, A is
	  redialed, A answers, B and A are connected, after which, B hangs
	  up. d. C picks up B while still in parking lot. II. A calls B, B
	  parks A a. A hangs up while B is getting the numbers announced.
	  b. A hangs up after B gets the announcement, but before the
	  parking time expires c. A waits, time expires, B is redialed, B
	  answers, A and B are connected, after which, A hangs up. d. C
	  picks up A while still in parking lot. Testing this throroughly
	  involves acting all the permutations of I and II, in situations
	  1,2,3, and 4. Since I added a few more changes (ALL references to
	  KEEPALIVE in the bridge code eliimated (I missed one earlier), I
	  retested most of the above cases, and no crashes. H-extension
	  weirdness. Current h-extension execution is not completely
	  correct for several of the cases. For the case where A calls B,
	  and A parks B, the 'h' exten is run on A's channel as soon as the
	  park is accomplished. This is expected behavior. But when A calls
	  B, and B parks A, this will be current behavior: After B parks A,
	  B is hung up by the system, and the 'h' (hangup) exten gets run,
	  but the channel mentioned will be a derivative of A's... Thus, if
	  A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
	  channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
	  will be those relating to Channel A. And, in the case where A is
	  reconnected to B after the park time expires, when both parties
	  hang up after the joyful reunion, no h-exten will be run at all.
	  In the case where C picks up A from the parking lot, when either
	  A or C hang up, the h-exten will be run for the C channel. CDR's
	  are a separate issue, and not addressed here. As to WHY this
	  strange behavior occurs, the answer lies in the procedure
	  followed to accomplish handing over the channel to the parking
	  manager thread. This procedure is called masquerading. In the
	  process, a duplicate copy of the channel is created, and most of
	  the active data is given to the new copy. The original channel
	  gets its name changed to XXX<ZOMBIE> and keeps the PBX
	  information for the sake of the original thread (preserving its
	  role as a call originator, if it had this role to begin with),
	  while the new channel is without this info and becomes a call
	  target (a "peer"). In this case, the parking lot manager thread
	  is handed the new (masqueraded) channel. It will not run an
	  h-exten on the channel if it hangs up while in the parking lot.
	  The h exten will be run on the original channel instead, in the
	  original thread, after the bridge completes. See bug 13820 for
	  our intentions as to how to clean up the h exten behavior.
	  Review: http://reviewboard.digium.com/r/29/

2008-12-19 19:48 +0000 [r165991]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
	  apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
	  apps/app_dahdiscan.c, codecs/codec_dahdi.c,
	  res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
	  #13480) Reported by: tzafrir Replace a bunch of if defined checks
	  for Zaptel/DAHDI through several new defines in dahdi_compat.h.
	  This removes a lot of code duplication. Example from bug: #ifdef
	  HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
	  open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
	  open(DAHDI_FILE_PSEUDO, O_RDRW);

2008-12-19 15:03 +0000 [r165796-165889]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: Ensure that the chanspy datastore is fully
	  initialized. This patch resolved some random crash issues
	  observed by a user on a BSD system (closes issue #14111) Reported
	  by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)

	* main/utils.c: Make ast_carefulwrite() be more careful. This patch
	  handles some additional cases that could result in partial writes
	  to the file description. This was done to address complaints
	  about partial writes on AMI. (issue #13546) (more changes needed
	  to address potential problems in 1.6) Reported by: srt Tested by:
	  russell Review: http://reviewboard.digium.com/r/99/

2008-12-18 21:14 +0000 [r165767]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Add mutexes around accesses to the IMAP
	  library interface. This prevents certain crashes, especially when
	  shared mailboxes are used. (closes issue #13653) Reported by:
	  howardwilkinson Patches:
	  asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
	  howardwilkinson (license 590) Tested by: jpeeler

2008-12-18 18:52 +0000 [r165661]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Set the process group ID on the MOH
	  process so that all children will get killed (closes issue
	  #14099) Reported by: caspy Patches:
	  res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
	  645)

2008-12-18 17:11 +0000 [r165537-165591]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Only care about a compatible codec for early bridging
	  if we are actually bridging to another channel. If we are not we
	  actually want to bring the audio back to us. (closes issue
	  #13545) Reported by: davidw

	* apps/app_followme.c: Do not crash if we are not passed in a
	  followme id. (closes issue #14106) Reported by: ys Patches:
	  app_followme.c.2.diff uploaded by ys (license 281)

2008-12-17  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.23-rc3 released.

2008-12-17 21:14 +0000 [r165317]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_macro.c: Reverse the fix from issue #6176 and add proper
	  handling for that issue. (Closes issue #13962, closes issue
	  #13363) Fixed by myself (license 14)

2008-12-17 20:51 +0000 [r164977-165255]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
	  apps/app_queue.c: Fix some memory leaks found while looking at
	  how realtime configs are handled. Also cleaned up some coding
	  guidelines violations in app_realtime.c, mostly related to
	  spacing

	* channels/chan_sip.c: After looking through SIP registration code
	  most of the day, this is one of the few things I could find that
	  was just plain wrong. Even though it probably isn't possible for
	  it to happen, it seems weird to have code that checks if a
	  pointer is NULL and then immediately dereferences that pointer if
	  it was NULL.

2008-12-16 21:38 +0000 [r164672-164881]  Russell Bryant <russell@digium.com>

	* main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
	  report that a thread is exiting while holding a lock. If the last
	  lock attempt was a trylock, and it failed, it will still be in
	  the list of locks so that it can be reported. (closes issue
	  #13219) Reported by: pj

	* apps/app_macro.c: Do not dereference the channel if
	  AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
	  while looking at the code for app_macro. This return code means
	  that another thread has assumed ownership of the channel and it
	  can no longer be touched. (I hate this return code with a
	  passion, by the way.)

	* main/manager.c: Add "restart gracefully" to the AMI blacklist of
	  CLI commands. "module unload" was already identified as a command
	  that can not be used from the AMI. "restart gracefully"
	  effectively unloads all modules, and will run in to the same
	  problems. (closes issue #13894) Reported by: kernelsensei

	* include/asterisk/threadstorage.h, main/threadstorage.c: Fix
	  memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
	  One issue was that the ast_mutex_* API was being used within the
	  context of the thread local data destructors. We would go off and
	  allocate more thread local data while the pthread lib was in the
	  middle of destroying it all. This led to a memory leak. Another
	  issue was an invalid argument being provided to the the
	  object_add API call. (closes issue #13678) Reported by: ys Tested
	  by: Russell

	* channels/chan_sip.c: Fix a memory leak related to the use of the
	  "setvar" configuration option. The problem was that these
	  variables were being appended to the list of vars on the sip_pvt
	  every time a re-registration or re-subscription came in. Since
	  it's just a waste of memory to put them there unless the request
	  was an INVITE, then the fix is to check the request type before
	  copying the vars. (closes issue #14037) Reported by: marvinek
	  Tested by: russell

2008-12-16 15:15 +0000 [r164634]  Steve Murphy <murf@digium.com>

	* main/pbx.c: I added a sentence to clarify why - and ' ' are
	  ignored in patterns as per bug 14076. Leif says he'll put some
	  stuff about it in the extensions.conf sample, etc.

2008-12-16 14:28 +0000 [r164605]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Don't try to change working directory if a
	  directory was not configured. (closes issue #14089) Reported by:
	  caspy

2008-12-15 19:53 +0000 [r164416-164422]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/pbx.h: Add the deadlock note to
	  ast_spawn_extension as well

	* include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
	  autoservice and pbx doxygen regarding a potential deadlock
	  scenario so that it is avoided in the future

2008-12-15 18:11 +0000 [r164204-164350]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Do not try to unlock a non-existant channel
	  if the transfer fails. (closes issue #13800) Reported by: dwagner
	  Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
	  (license 608)

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/channel.h: Use autoconf logic to determine
	  whether the system has timersub or not. Do not blindly assume
	  Solaris does not. (closes issue #13838) Reported by: ano

	* apps/app_dial.c: Can we try not to assign an unsigned int to -1?
	  (closes issue #14074) Reported by: wetwired

2008-12-15 14:31 +0000 [r164201]  Russell Bryant <russell@digium.com>

	* main/channel.c, res/res_features.c: Handle a case where a call
	  can be bridged to a channel that is still ringing. The issue that
	  was reported was about a case where a RINGING channel got
	  redirected to an extension to pick up a call from parking. Once
	  the parked call got taken out of parking, it heard silence until
	  the other side answered. Ideally, the caller that was parked
	  would get a ringing indication. This patch fixes this case so
	  that the caller receives ringback once it comes out of parking
	  until the other side answers. The fixes are: - Make sure we
	  remember that a channel was an outgoing channel when doing a
	  masquerade. This prevents an erroneous ast_answer() call on the
	  channel, which causes a bogus 200 OK to be sent in the case of
	  SIP. - Add some additional comments to explain related parts of
	  code. - Update the handling of the ast_channel visible_indication
	  field. Storing values that are not stateful is pointless. Control
	  frames that are events or commands should be ignored. - When a
	  bridge first starts, check to see if the peer channel needs to be
	  given ringing indication because the calling side is still
	  ringing. - Rework ast_indicate_data() a bit for the sake of
	  readability. (closes issue #13747) Reported by: davidw Tested by:
	  russell Review: http://reviewboard.digium.com/r/90/

2008-12-13 23:22 +0000 [r164082]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c: Change the default calldurationlimit from the
	  special value 0 to -1, so we can better detect an exceptional
	  case. This follows on to the changes made in revision 156386.
	  Related to issue #13851. (closes issue #13974) Reported by:
	  paradise Patches: 20081208__bug13974.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: file, blitzrage, ZX81

2008-12-12 22:20 +0000 [r163785]  Russell Bryant <russell@digium.com>

	* /: Set the reviewboard:url property on 1.4, as well

2008-12-12 22:03 +0000 [r163761]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
	  immediately exiting Asterisk, but also add a pointer inside
	  editline to look back to asterisk.c, so others don't spend as
	  much time as I did looking (in the wrong place) for the
	  appropriate function. Reported by: ZX81, via the #asterisk-users
	  channel Fixed by: me (license 14)

2008-12-12 14:40 +0000 [r163448-163511]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
	  so that it is actually 32 bits on 64-bit machines, as well.
	  (inspired by issue #13879)

	* main/channel.c, main/autoservice.c, include/asterisk/channel.h:
	  Resolve issues that could cause DTMF to be processed out of
	  order. These changes come from team/russell/issue_12658 1) Change
	  autoservice to put digits on the head of the channel's frame
	  readq instead of the tail. If there were frames on the readq that
	  autoservice had not yet read, the previous code would have
	  resulted in out of order processing. This required a new API call
	  to queue a frame to the head of the queue instead of the tail. 2)
	  Change up the processing of DTMF in ast_read(). Some of the
	  problems were the result of having two sources of pending DTMF
	  frames. There was the dtmfq and the more generic readq. Both were
	  used for pending DTMF in various scenarios. Simplifying things to
	  only use the frame readq avoids some of the problems. 3) Fix a
	  bug where a DTMF END frame could get passed through when it
	  shouldn't have. If code set END_DTMF_ONLY in the middle of digit
	  emulation, and a digit arrived before emulation was complete,
	  digits would get processed out of order. (closes issue #12658)
	  Reported by: dimas Tested by: russell, file Review:
	  http://reviewboard.digium.com/r/85/

2008-12-11 23:35 +0000 [r163383]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
	  on certain shells, the terminal is messed up. By intercepting
	  those events with a signal handler in the remote console, we can
	  avoid those issues. (closes issue #13464) Reported by: tzafrir
	  Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
	  (license 14) Tested by: blitzrage

2008-12-11 22:44 +0000 [r163316]  Matt Nicholson <mnicholson@digium.com>

	* pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
	  (closes issue #13819) Reported by: adomjan Patches:
	  pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
	  dundi_clearecache3.diff uploaded by mnicholson (license 96)
	  Tested by: adomjan

2008-12-11 21:46 +0000 [r163092-163253]  Russell Bryant <russell@digium.com>

	* funcs/func_strings.c, funcs/func_cut.c: Fix some observed
	  slowdowns in dialplan processing. The change is to remove
	  autoservice usage from dialplan functions that do not need it
	  because they do not perform operations that potentially block.
	  (closes issue #13940) Reported by: tbelder

	* res/res_features.c: Fix an issue that made it so you could only
	  have a single caller executing a custom feature at a time. This
	  was especially problematic when custom features ran for any
	  appreciable amount of time. The fix turned out to be quite
	  simple. The dynamic features are now stored in a read/write list
	  instead of a list using a mutex. (closes issue #13478) Reported
	  by: neutrino88 Fix suggested by file

2008-12-11 16:51 +0000 [r163088]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Don't wait forever, if there's a specified
	  recording timeout. (closes issue #13885) Reported by: bamby
	  Patches: res_agi.c.patch uploaded by bamby (license 430)

2008-12-11 16:46 +0000 [r163080-163084]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Revert this cast to long. Using time_t here
	  causes build failures on a FreeBSD 32-bit build.

	* apps/app_queue.c: Fix a potential crash due to unsafe datastore
	  handling. This patch also contains a conversion from using long
	  to time_t for representing times for a queue, as well as some
	  whitespace fixes. (closes issue #14060) Reported by: nivek
	  Patches: datastore_fixup.patch.corrected uploaded by nivek
	  (license 636) with slight modification from me Tested by: nivek

2008-12-10 22:52 +0000 [r162874-162926]  Jeff Peeler <jpeeler@digium.com>

	* res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
	  check. Pointed out by mmichelson, thanks!

	* res/res_musiconhold.c: (closes issue #13229) Reported by:
	  clegall_proformatique Ensure that moh_generate does not return
	  prematurely before local_ast_moh_stop is called. Also, the sleep
	  in mp3_spawn now only occurs for http locations since it seems to
	  have been added originally only for failing media streams.

2008-12-10 19:01 +0000 [r162738-162804]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix subscription based MWI up a bit. We only
	  want to put sip: at the beginning of the URI if it is not already
	  there and revert code to ignore destination check if subscribing
	  for MWI. (closes issue #12560) Reported by: vsauer Patches:
	  patch001.diff uploaded by ramonpeek (license 266)

	* channels/chan_sip.c: When a SIP peer unregisters set the expiry
	  time back to 0 so that the 200 OK contains an expires of 0.
	  (closes issue #13599) Reported by: hjourdain Patches:
	  chan_sip.c.diff uploaded by hjourdain (license 583)

2008-12-10 16:45 +0000 [r162671]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
	  Reported by: wetwired Tested by: murf I checked, and I added a
	  mod to the trunk version of Asterisk that would make it 8-bit
	  transparent on 27 Nov 2007, but I made no such updates to 1.4. My
	  best guess is that 1.4 was released, and it was not appropriate
	  to commit an enhancement. But I'm going to add the same fixes to
	  1.4 now, for the following reasons: 1. wetwired is correct; 1.4
	  is **mostly** 8-bit transparent now. This is because the lexical
	  token forming rules use . in most 'word' state continuances. It's
	  just the beginning of a 'word' that is picky. 2. Accepting 8-bit
	  chars in some places and not others leads to bug reports like
	  this.

2008-12-10 16:44 +0000 [r162659-162670]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/stringfields.h: Update to stringfield handling
	  so that side-effects on parameters are not evaluated multiple
	  times. An example where this caused a problem was in chan_sip.c,
	  with the line ast_string_field_set(p, fromdomain, ++fromdomain);
	  This patch was originally uploaded to issue #13783 by jamessan.
	  While the issue was closed for other reasons, this patch is valid
	  and fixes a separate problem, and is thus being committed.

	* channels/chan_sip.c: Revert fix for issue 13570. It has caused
	  more problems than it helped to fix. (closes issue #13783)
	  Reported by: navkumar (closes issue #14025) Reported by: ffs

	* doc/misdn.txt: Add missing documentation to misdn.txt (closes
	  issue #14052) Reported by: festr Patches: misdn.txt.patch
	  uploaded by festr (license 443)

2008-12-10 16:05 +0000 [r162653]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Increment the sequence number on the end packets for
	  RFC2833. After reading the RFC some more and doing some testing I
	  agree with this change. (closes issue #12983) Reported by: vt
	  Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
	  520)

2008-12-09 23:08 +0000 [r162463]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Oops, should be "tz", not "zonetag".

2008-12-09 22:17 +0000 [r162413]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
	  the test_for_thread_safety() function completely. The test is not
	  valid. Besides, if we actually suspected that recursive mutexes
	  were not working, we would get a ton of LOG_ERROR messages when
	  DEBUG_THREADS is turned on. (inspired by a discussion on the
	  asterisk-dev list)

2008-12-09 21:53 +0000 [r162348]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: We appear to have documented tz= in the
	  [general] section of voicemail.conf, without actually having
	  implemented it. Oops. (Reported by Olivier on the -users list)

2008-12-09 21:14 +0000 [r162341]  Joshua Colp <jcolp@digium.com>

	* apps/app_directed_pickup.c: Add 'down' as a valid state for
	  directed call pickup. This creeps up when we receive session
	  progress when dialing a device and not ringing. (closes issue
	  #14005) Reported by: ddl

2008-12-09 20:57 +0000 [r162286]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Fix an issue where callers on an incoming call
	  on an SLA trunk would not hear ringback. We need to make sure
	  that we don't start writing audio to the trunk channel until
	  we're actually ready to answer it. Otherwise, the channel driver
	  will treat it as inband progress, even though all they are
	  getting is silence. (closes issue #12471) Reported by: mthomasslo

2008-12-09 20:44 +0000 [r162273]  Joshua Colp <jcolp@digium.com>

	* apps/app_festival.c: Fix double declaration of 'x' on the PPC
	  platform. (closes issue #14038) Reported by: ffloimair

2008-12-09 20:28 +0000 [r162265]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: If we fail to start a thread for the pbx to run in,
	  we need to be sure to decrease the number of active calls on the
	  system. This fix may relate to ABE-1713, but it is not certain
	  yet.

2008-12-09 20:20 +0000 [r162264]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
	  seanbright on #asterisk-dev, I have added a default rule, and an
	  option to suppress the default rule from being generated in the
	  flex output, for the sake of those OS's where they didn't tweak
	  flex's ECHO macro, and the compiler doesn't like it. The
	  regressions are OK with this.

2008-12-09 19:47 +0000 [r162188-162204]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Make sure that the timestamp for DTMF is not the same
	  as the previous voice frame and do not send audio when
	  transmitting DTMF as this confuses some equipment. (closes issue
	  #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
	  (license 11) Tested by: ip-rob, bujones

	* main/rtp.c: Take video into account when early bridging RTP.
	  (closes issue #13535) Reported by: davidw

2008-12-09 18:13 +0000 [r162136]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
	  and ast_copy_string and messed up the standalone stuff. Fixed.

2008-12-09 17:07 +0000 [r162071]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_phone.c: For some reason, after a distclean, gcc
	  started returning 'value computed is not used'. Fixing (for
	  --enable-dev-mode).

2008-12-09 16:46 +0000 [r162014]  Russell Bryant <russell@digium.com>

	* apps/app_disa.c: Allow DISA to handle extensions that start with
	  #. (closes issue #13330) Reported by: jcovert

2008-12-09 16:31 +0000 [r162013]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
	  pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
	  Patches: 14019.diff uploaded by murf (license 17) Tested by:
	  ckjohnsonme, murf This crash was the result of a few small errors
	  that would combine in 64-bit land to result in a crash. 32-bit
	  land might have seen these combine to mysteriously drop the args
	  to an application call, in certain circumstances. Also, in trying
	  to find this bug, I spotted a situation in the flex input, where,
	  in passing back a 'word' to the parser, it would allocate a
	  buffer larger than necessary. I changed the usage in such
	  situations, so that strdup was not used, but rather, an
	  ast_malloc, followed by ast_copy_string. I removed a field from
	  the pval struct, in u2, that was never getting used, and set in
	  one spot in the code. I believe it was an artifact of a previous
	  fix to make switch cases work invisibly with extens. And, for
	  goto's I removed a '!' from before a strcmp, that has been there
	  since the initial merging of AEL2, that might prevent the proper
	  target of a goto from being found. This was pretty harmless on
	  its own, as it would just louse up a consistency check for users.
	  Many thanks to ckjohnsonme for providing a simplified and
	  complete set of information about the bug, that helped
	  considerably in finding and fixing the problem. Now, to get
	  aelparse up and running again in trunk, and out of its "horribly
	  broken" state, so I can run the regression suite!

2008-12-09 14:52 +0000 [r161948]  Russell Bryant <russell@digium.com>

	* main/app.c: Fix a problem with GROUP() settings on a masquerade.
	  The previous code carried over group settings from the old
	  channel to the new one. However, it did nothing with the group
	  settings that were already on the new channel. This patch removes
	  all group settings that already existed on the new channel. I
	  have a more complicated version of this patch which addresses
	  only the most blatant problem with this, which is that a channel
	  can end up with multiple group settings in the same category.
	  However, I could not think of a use case for keeping any of the
	  group settings from the old channel, so I went this route for
	  now. (closes AST-152)

2008-12-08 17:52 +0000 [r161725]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make the usereqphone option work again.
	  (closes issue #13474) Reported by: mmaguire Patches:
	  20080912_bug13474.diff uploaded by mmaguire (license 571)

2008-12-05 21:02 +0000 [r161426]  Sean Bright <sean.bright@gmail.com>

	* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
	  161421 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
	  2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
	  int). (closes issue #14006) Reported by: alphaque Patches:
	  astobj2.h-patch uploaded by alphaque (license 259) (Slightly
	  modified by seanbright) ........

2008-12-05 16:51 +0000 [r161354]  Dwayne M. Hubbard <dhubbard@digium.com>

	* utils/smsq.c: kill a warning

2008-12-05 14:12 +0000 [r161287]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Fix a NULL format string warning found by buildbot.

2008-12-04 18:30 +0000 [r161013]  Jeff Peeler <jpeeler@digium.com>

	* main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
	  jpeeler This mirrors a check that was present in ast_rtp_read to
	  also be in ast_rtp_raw_write to not schedule sending the receiver
	  report if the remote RTCP endpoint address isn't present in the
	  RTCP structure. Closes AST-142.

2008-12-04 16:44 +0000 [r160943]  Mark Michelson <mmichelson@digium.com>

	* main/callerid.c: Fix a callerid parsing issue. If someone
	  formatted callerid like the following: "name <number>" (including
	  the quotation marks), then the parts would be parsed as name:
	  "name number: number This is because the closing quotation mark
	  was not discovered since the number and everything after was
	  parsed out of the string earlier. Now, there is a check to see if
	  the closing quote occurs after the number, so that we can know if
	  we should strip off the opening quote on the name. Closes AST-158

2008-12-03 21:54 +0000 [r160770]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Some compilers warn on null format strings;
	  some don't (caught by buildbot)

2008-12-03 21:38 +0000 [r160764]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c: Only show this warning when we want to
	  show it. (closes issue #13982) Reported by: coolmig Patches:
	  chan_agent.c.patch uploaded by coolmig (license 621)

2008-12-03 20:41 +0000 [r160703]  Steve Murphy <murf@digium.com>

	* funcs/func_callerid.c: (closes issue #13597) Reported by:
	  john8675309 Patches: patch.13597 uploaded by murf (license 17)
	  Tested by: murf, john8675309 This patch causes the setcid func to
	  update the CDR clid after setting the channel field.

2008-12-03 17:55 +0000 [r160480-160570]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: During bridge code, the channel bridge may
	  return a retry code, if a transfer was initiated but not yet
	  completed. If the bridge is immediately retried, then we may send
	  a storm of TXREQ packets, even though the first set is sent
	  reliably (retransmitted). Fixes AST-137.

	* pbx/pbx_spool.c: If an entry is added to the directory during a
	  scan when another entry expires, then that new entry will not be
	  processed promptly, but must wait for either a future entry to
	  start or a current entry's retry to occur. If no other entries
	  exist in the directory (other than the new entries) when a bunch
	  expire, then the new entries must wait until another new entry is
	  added to be processed. This was a rather weird race condition,
	  really. Fixes AST-147.

	* pbx/pbx_spool.c: Don't start scanning the directory until all
	  modules are loaded, because some required modules (channels,
	  apps, functions) may not yet be in memory yet. Fixes AST-149.

	* channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
	  list: "I guess that having only ip-phones in mind is not a good
	  approach. Since it is possible to have a sip proxy connected to
	  asterisk we could receive a 407 (unauthorized) or 483 (too many
	  hops) as response and dialog ending would not be a good
	  behavior." So modified.

2008-12-02 23:58 +0000 [r160390-160411]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: Channel is masqueraded, don't keep alive

	* res/res_features.c: A situation like A calls B, A builtin_atxfers
	  B to C, C parks B would lead to a crash. Thanks to file for
	  telling me how to fix it! (closes issue #13854) Reported by: Adam
	  Lee Tested by: otherwiseguy

2008-12-02 17:42 +0000 [r160297]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: When the text does not match exactly (e.g.
	  RTP/SAVP), then the %n conversion fails, and the resulting
	  integer is garbage. Thus, we must initialize the integer and
	  check it afterwards for success. (closes issue #14000) Reported
	  by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
	  folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
	  by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
	  uploaded by folke (license 626)

2008-12-02 01:16 +0000 [r160266]  Terry Wilson <twilson@digium.com>

	* include/asterisk/astmm.h: make compile with dev mode and malloc
	  debug

2008-12-02 00:25 +0000 [r160207]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/stringfields.h, apps/app_voicemail.c,
	  main/pbx.c, main/frame.c: Ensure that Asterisk builds with
	  --enable-dev-mode, even on the latest gcc and glibc.

2008-12-01  Tilghman Lesher <tilghman@digium.com>

	* Released 1.4.23-rc2

2008-12-01 17:27 +0000 [r160003]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
	  iax2_setoption(), as well, since they both have the potential to
	  send control frames in the middle of call setup. We have to wait
	  until we have received a message back from the remote end before
	  we try to send any more frames. Otherwise, the remote end will
	  consider it invalid, and we'll get stuck in an INVAL/VNAK storm.

2008-12-01 16:08 +0000 [r159976]  Michiel van Baak <michiel@vanbaak.info>

	* main/manager.c: Get rid of the useless format string and argument
	  in the Bogus/ manager channelname. Noted by kpfleming and name
	  Bogus/manager suggested by eliel

2008-12-01 14:52 +0000 [r159900]  Russell Bryant <russell@digium.com>

	* .cleancount: Force a "make clean" to avoid a bizarre build issue
	  ...

2008-12-01 14:05 +0000 [r159897]  Michiel van Baak <michiel@vanbaak.info>

	* main/manager.c: make manager compile on OpenBSD. The last (10th)
	  argument to ast_channel_alloc here should be a pointer and NULL
	  is not really a pointer.

2008-11-29 16:58 +0000 [r159808]  Kevin P. Fleming <kpfleming@digium.com>

	* main/enum.c, utils/frame.c, configure, res/res_agi.c,
	  include/asterisk/module.h, main/logger.c, main/dns.c,
	  include/asterisk/threadstorage.h, include/asterisk/utils.h,
	  include/asterisk/devicestate.h, channels/chan_sip.c,
	  include/asterisk/dundi.h, main/jitterbuf.c,
	  channels/chan_agent.c, configure.ac, utils/astman.c,
	  include/asterisk/cli.h, include/asterisk/channel.h,
	  include/jitterbuf.h, include/asterisk/manager.h,
	  main/ast_expr2.c, Makefile, include/asterisk/logger.h,
	  include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
	  include/asterisk/linkedlists.h, include/asterisk/lock.h,
	  include/asterisk/strings.h, makeopts.in,
	  include/asterisk/stringfields.h, utils/check_expr.c,
	  channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
	  update dev-mode compiler flags to match the ones used by default
	  on Ubuntu Intrepid, so all developers will see the same warnings
	  and errors since this branch already had some printf format
	  attributes, enable checking for them and tag functions that
	  didn't have them format attributes in a consistent way

2008-11-26 20:21 +0000 [r159476-159571]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_oss.c, channels/busy.h (removed),
	  channels/ring_tone.h (added), channels/chan_alsa.c,
	  channels/ringtone.h (removed), channels/busy_tone.h (added),
	  channels/Makefile: rename these files so as to avoid conflicts
	  when users update their working copies and have unversioned files
	  already in place

	* channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
	  Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
	  (added), channels/Makefile: simplify (and slightly bug-fix) the
	  recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
	  and channels/ringtone.h to the repository instead of generating
	  them repeatedtly; most users do not change the settings to build
	  them, but the Makefile rules are still there if they wish to do
	  so ensure that 'make clean' removes dependency files for .i files
	  that are created in COMPILE_DOUBLE mode

2008-11-25 22:41 +0000 [r159316]  Steve Murphy <murf@digium.com>

	* main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
	  by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
	  Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
	  fix. the change to cdr.c allows no-answer to percolate up into
	  CDR's, and feels like the right place to locate this fix; if BUSY
	  is done here, no-answer should be, too.

2008-11-25 21:56 +0000 [r159246-159269]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
	  (closes issue #13919) Reported by: barthpbx Patches:
	  chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
	  barthpbx

	* /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
	  Nov 2008) | 7 lines Regression fix for last security fix. Set the
	  iseqno correctly. (closes issue #13918) Reported by: ffloimair
	  Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
	  (license 14) Tested by: ffloimair ........

2008-11-25 17:34 +0000 [r159158]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
	  go along with ao2_lock() and ao2_unlock()

2008-11-25 16:23 +0000 [r159096]  Terry Wilson <twilson@digium.com>

	* apps/app_festival.c: Add missing variable declaration in the PPC
	  code

2008-11-25 04:50 +0000 [r159025]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: System call ioperm is non-portable, so check for
	  its existence in autoconf. (Closes issue #13863)

2008-11-22 00:04 +0000 [r158629]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
	  issue #13786) Reported by: tzafrir When compiling against Zaptel
	  dahdi_compat will now only define all the DAHDI defines if the
	  Zaptel define is present. Also, there is no such thing as
	  DAHDI_PRI.

2008-11-21 23:14 +0000 [r158603]  Steve Murphy <murf@digium.com>

	* res/res_features.c: In reference to the fix made for 13871, I was
	  merging the fix into 1.6.0 and realized I missed the code in the
	  h-exten block, and didn't catch it because my test case had the
	  h-exten commented out. So, this corrects the code I missed, as a
	  preventative against another crash report. Tested with the
	  h-exten defined, all is well.

2008-11-21 23:07 +0000 [r158600]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: The passed extension may not be the same in the list
	  as the current entry, because we strip spaces when copying the
	  extension into the structure. Therefore, use the copied item to
	  place the item into the list. (found by lmadsen on -dev, fixed by
	  me)

2008-11-21 22:05 +0000 [r158539]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h: When compiling with
	  DEBUG_THREADS, report the real file/func/line for
	  ao2_lock/ao2_unlock

2008-11-21 21:19 +0000 [r158483]  Steve Murphy <murf@digium.com>

	* res/res_features.c: (closes issue #13871) Reported by: mdu113
	  This one is totally my fault. The code doesn't even create a
	  bridge if the channel CDR has POST_DISABLED. I didn't check for
	  that at the end of the bridge. Fixed with a few small insertions.
	  Tested. Looks good. No cdr generated, no crash, no unnecc. data
	  objects created either.

2008-11-21 15:24 +0000 [r158053-158306]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This change had somehow gotten reverted due to
	  a completely unrelated commit. Thanks to Theo Belder on the
	  Asterisk-dev list for pointing this out.

	* include/asterisk/file.h, main/frame.c, main/file.c,
	  include/asterisk/frame.h: There was an issue when attempting to
	  reference an embedded frame in a freed ast_filestream. This patch
	  makes use of the ao2 functions to make sure that we do not free
	  an ast_filestream structure until the embedded ast_frame has been
	  "freed" as well. (closes issue #13496) Reported by: fst-onge
	  Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
	  60) Tested by: putnopvut Closes AST-89

	* channels/chan_sip.c: We don't handle 4XX responses to BYE well.
	  According to section 15 of RFC 3261, we should terminate a dialog
	  if we receive a 481 or 408 in response to our BYE. Since I am
	  aware of at least one phone manufacturer who may sometimes send a
	  404 as well, I am being liberal and saying that any 4XX response
	  to a BYE should result in a terminated dialog. (closes issue
	  #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
	  putnopvut (license 60) Closes AST-129

	* apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
	  cause on the calling channel in the case that ast_call() fails.
	  For incoming SIP channels, this was causing us to send a 603
	  instead of a 486 when the call-limit was reached on the
	  destination channel. (closes issue #13867) Reported by: still_nsk
	  Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
	  blitzrage

2008-11-20 01:46 +0000 [r158010]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Merged revision 157977 from
	  https://origsvn.digium.com/svn/asterisk/team/group/issue8824
	  ........ Fixes JIRA ABE-1726 The dial extension could be empty if
	  you are using MISDN_KEYPAD to control ISDN provider features.

2008-11-19 21:34 +0000 [r157859]  Kevin P. Fleming <kpfleming@digium.com>

	* main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
	  channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
	  main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
	  codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
	  channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
	  res/snmp, res/Makefile: the gcc optimizer frequently finds broken
	  code (use of uninitalized variables, unreachable code, etc.),
	  which is good. however, developers usually compile with the
	  optimizer turned off, because if they need to debug the resulting
	  code, optimized code makes that process very difficult. this
	  means that we get code changes committed that weren't adequately
	  checked over for these sorts of problems. with this build system
	  change, if (and only if) --enable-dev-mode was used and
	  DONT_OPTIMIZE is turned on, when a source file is compiled it
	  will actually be preprocessed (into a .i or .ii file), then
	  compiled once with optimization (with the result sent to
	  /dev/null) and again without optimization (but only if the first
	  compile succeeded, of course). while making these changes, i did
	  some cleanup work in Makefile.rules to move commonly-used
	  combinations of flag variables into their own variables, to make
	  the file easier to read and maintain

2008-11-18 22:47 +0000 [r157503]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add some missing invite state changes
	  necessary in the sip_write function. Not setting the invite state
	  correctly on the call was resulting in the Record application
	  leaving empty files. I also have updated the doxygen comment next
	  to the declaration of the INV_EARLY_MEDIA constant to reflect
	  that we also use this state when we *send* a 18X response to an
	  INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
	  sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
	  (license 162) Tested by: putnopvut

2008-11-18 19:13 +0000 [r157365]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
	  This fix is the result of a bug fix in ast_app_separate_args
	  r124395. If an argument does not exist it should always be set to
	  a null string rather than a null pointer.

2008-11-18 18:25 +0000 [r157305]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, channels/chan_local.c, res/res_features.c,
	  include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
	  the end_bridge_callback of app_dial and app_followme which would
	  occur at the end of an attended transfer. The error occurred
	  because we initially stored a pointer to an ast_channel which
	  then was hung up due to a masquerade. This commit adds a "fixup"
	  callback to the bridge_config structure to allow for
	  end_bridge_callback_data to be changed in the case that a new
	  channel pointer is needed for the end_bridge_callback.

2008-11-15 19:31 +0000 [r157104-157163]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, Makefile.rules: when an individual directory dist-clean
	  is run, run clean in that directory first, and when running
	  top-level dist-clean, do not run subdirectory clean operations
	  twice

	* Makefile.moddir_rules: dist-clean should remove dependency
	  information files as well

	* contrib/asterisk-ng-doxygen: major update to doxygen
	  configuration file: 1) update to doxygen 1.5.x style file, as
	  used in trunk 2) tell doxygen where are header files are, so
	  include-file processing can be done 3) make all macros that are
	  used to define variables/functions be expanded, so that doxygen
	  will properly document the resulting variable/function 4) make
	  all macros that are used to provide the contents of a variable
	  (structure) be expanded, so that doxygen will be able to document
	  the resulting fields 5) suppress compiler attributes
	  (__attribute__(xxx)) from being seen by doxygen, so it will
	  properly match up function definition and usage (for an example
	  of th effect of this, look at the doxygen docs for ast_log() from
	  before and afte this commit)

2008-11-14 15:18 +0000 [r156816]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: If the prompt to reenter a voicemail
	  password timed out, it resulted in the password not being saved,
	  even if the input matched what you gave when first prompted to
	  enter a new password. This is because the return value of
	  ast_readstring was checked, but not checked properly. This bug
	  was discovered by Jared Smith during an Asterisk training course.
	  Thanks for reporting it!

2008-11-14 00:41 +0000 [r156688-156755]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_while.c: ast_waitfordigit() requires that the channel be
	  up, for no good logical reason. This prevents While/EndWhile from
	  working within the "h" extension. Reported by: jgalarneau (for
	  ABE C.2) Fixed by: me

	* main/manager.c: Provide more space for all the data which can
	  appear in an originating channel name. (closes issue #13398)
	  Reported by: bamby Patches: manager.c.diff uploaded by bamby
	  (license 430)

2008-11-13 11:58 +0000 [r156485-156510]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, autoconf/ast_gcc_attribute.m4: revert this change...
	  non-functional changes don't belong here

	* configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
	  error... no functional change

2008-11-12 21:18 +0000 [r156386]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c: When using call limits under 1 second, infinite
	  call lengths are allowed, instead. (closes issue #13851) Reported
	  by: ruddy

2008-11-12 19:36 +0000 [r156297]  Steve Murphy <murf@digium.com>

	* main/pbx.c: It turns out that the 0x0XX00 codes being returned
	  for N, X, and Z are off by one, as per conversation with jsmith
	  on #asterisk-dev; he was teaching a class and disconcerted that
	  this published rule was not being followed, with patterns _NXX,
	  _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
	  have been. This change, tested on these 3 patterns now picks the
	  proper one. However, this change may surprise users who set up
	  dialplans based on previous behavior, which has been there for
	  what, 2 and half years or so now.

2008-11-12 19:26 +0000 [r156294]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: If the SLA thread is not started, then reload
	  causes a memory leak. (closes issue #13889) Reported by: eliel
	  Patches: app_meetme.c.patch uploaded by eliel (license 64)

2008-11-12 19:10 +0000 [r156289]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: For whatever reason, gcc only warned me about
	  the possible use of an uninitialized variable when compiling
	  1.6.1.

2008-11-12 18:39 +0000 [r156229]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Revert revision 132506, since it
	  occasionally caused IAX2 HANGUP packets not to be sent, and
	  instead, schedule a task to destroy the iax2 pvt structure 10
	  seconds later. This allows the IAX2 HANGUP packet to be queued,
	  transmitted, and ACKed before the pvt is destroyed. (closes issue
	  #13645) Reported by: dzajro Patches:
	  20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/

2008-11-12 17:53 +0000 [r156178]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: (closes issue #13173) Reported by: pep This
	  change adds an announce_thread responsible for playing
	  announcements to an existing conference. This allows all
	  announcing to be immediately stopped if necessary but more
	  importantly allows other threads that need to play something to
	  not block. There are multiple benefits to this, but the actual
	  bug is for solving the scenario for a channel to be unusable
	  after hang up for the entire duration of the parting
	  announcement. The parting announcement can be extremely long
	  depending on what the user recorded upon joining the conference.
	  Reviewed by Russell on Review Board:
	  http://reviewboard.digium.com/r/25/

2008-11-12 17:38 +0000 [r156167]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: When doing some tests, I was having a crash at
	  the end of every call if an attended transfer occurred during the
	  call. I traced the cause to the CDR on one of the channels being
	  NULL. murf suggested a check in the end bridge callback to be
	  sure the CDR is non-NULL before proceeding, so that's what I'm
	  adding.

2008-11-12 17:29 +0000 [r156164]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Move the sanity check that makes sure "always
	  fork" is not set along with the console option to be after the
	  code that reads options from asterisk.conf. This resolves a
	  situation where Asterisk can start taking up 100% when
	  misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
	  me log in to his system to figure out what was causing the 100%
	  CPU problem.)

2008-11-10 21:07 +0000 [r155861]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Channel drivers assume that when their
	  indicate callback is invoked, that the channel on which the
	  callback was called is locked. This patch corrects an instance in
	  chan_agent where a channel's indicate callback is called directly
	  without first locking the channel. This was leading to some
	  observed locking issues in chan_local, but considering that all
	  channel drivers operate under the same expectations, the generic
	  fix in chan_agent is the right way to go. AST-126

2008-11-10 20:49 +0000 [r155803]  Tilghman Lesher <tlesher@digium.com>

	* doc/valgrind.txt: I got tired of saying this in every single
	  bugnote referring to this file.

2008-11-09 01:08 +0000 [r155553]  Sean Bright <sean.bright@gmail.com>

	* apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
	  apps/app_followme.c: Use static functions here instead of nested
	  ones. This requires a small change to the ast_bridge_config
	  struct as well. To understand the reason for this change, see the
	  following post:
	  http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html

2008-11-07 22:27 +0000 [r155398]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Clarify error message. (closes issue #13809)
	  Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: denke

2008-11-06 19:45 +0000 [r155011]  Mark Michelson <mmichelson@digium.com>

	* configs/voicemail.conf.sample: The documentation listed the
	  ability to set 'maxmsg' per context. The truth is that you can
	  only set this in the general section or per mailbox. Thus I am
	  updating the sample config file to be more accurate. Thanks to
	  sasargen on IRC for bringing up this issue.

2008-11-05 16:44 +0000 [r154724]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: The logic of a strcasecmp call was
	  reversed (closes issue #13841) Reported by: clegall_proformatique

2008-11-05 16:06 +0000 [r154685]  Steve Murphy <murf@digium.com>

	* main/channel.c: This fix was prompted by communication from user,
	  who was seeing thousands of error logs... looks like EAGAIN. Made
	  such uninteresting.

2008-11-04 20:49 +0000 [r154365]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: On busy systems, it's possible for the
	  values checked within a single line of code to change, unless the
	  structure is locked to ensure a consistent state. (closes issue
	  #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: kowalma

2008-11-04 19:01 +0000 [r154266]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
	  the wrong state when it receives the indication
	  AST_CONTROL_RINGING.

2008-11-04 18:58 +0000 [r154060-154263]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_h323.c: Make the monitor thread non-detached, so it
	  can be joined (suggested by Russell on -dev list).

	* apps/app_voicemail.c: Attempting to expunge a mailbox when the
	  mailstream is NULL will crash Asterisk. (Closes issue #13829)
	  Reported by: jaroth Patch by: me (modified jaroth's patch)

	* main/rtp.c: Remove the potential for a division by zero error.
	  (Closes issue #13810)

2008-11-03 13:01 +0000 [r153823]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
	  main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
	  res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
	  this branch on the first pass

2008-11-02 19:51 +0000 [r153651]  Russell Bryant <russell@digium.com>

	* include/asterisk/features.h: features.h depends on linkedlists.h,
	  so include it

2008-11-01 18:22 +0000 [r153337]  Kevin P. Fleming <kpfleming@digium.com>

	* utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
	  funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
	  channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
	  pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
	  pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
	  formats/format_wav.c, apps/app_festival.c,
	  main/db1-ast/hash/hash_page.c, main/translate.c,
	  res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
	  utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
	  pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
	  apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
	  utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
	  apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
	  apps/app_authenticate.c, formats/format_wav_gsm.c,
	  res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
	  potential problems found by gcc 4.3.x, primarily bare strings
	  being passed to printf()-like functions and ignored results from
	  read()/write() and friends

2008-10-31 22:36 +0000 [r153270]  Terry Wilson <twilson@digium.com>

	* res/res_features.c, apps/app_followme.c: Add end_bridge_callback
	  for app_follome and add AUTOLOOP flag to res_features

2008-10-31  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.4.23-rc1 released.

2008-10-31 16:30 +0000 [r153114]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Turn off qualify on uncached realtime peers.
	  (Closes issue #13383)

2008-10-31 15:45 +0000 [r153095]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
	  Recent CDR fixes moved execution of the 'h' exten into the
	  bridging code, so variables that were set after ast_bridge_call
	  was called would not show up in the 'h' exten. Added a callback
	  function to handle setting variables, etc. from w/in the bridging
	  code. Calls back into a nested function within the function
	  calling ast_bridge_call (closes issue #13793) Reported by:
	  greenfieldtech

2008-10-30 20:58 +0000 [r152992]  Sean Bright <sean.bright@gmail.com>

	* bootstrap.sh: The -I argument to aclocal needs a space before the
	  include directory name.

2008-10-30 20:33 +0000 [r152922-152958]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_h323.c: Cannot join detached threads. See
	  http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
	  (Closes issue #13400)

	* channels/chan_local.c: Unlock before returning, when extension
	  doesn't exist. (closes issue #13807) Reported by: eliel Patches:
	  chan_local.c.patch uploaded by eliel (license 64)

2008-10-30 16:53 +0000 [r152811]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cdr.c: instead of comparing the string pointer to 0, let's
	  compare the value that was actually parsed out of the string
	  (found by sparse)

2008-10-29 05:23 +0000 [r152539]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
	  issue #13795) Reported by: andrew53 Patches:
	  chan_sip_sizeof.patch uploaded by andrew53 (license 519)

2008-10-29 05:19 +0000 [r152535-152538]  Steve Murphy <murf@digium.com>

	* configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
	  A little documentation cross-ref between features and dial and
	  queue... I wasted some time (stupidly) trying to get the
	  one-touch parking stuff working, because it didn't occur to me
	  that I had to also have the corresponding options in the dial
	  command! Duh! (In all this time, I never set this up before!) So,
	  to keep some poor fool from suffering the same fate, I made the
	  features.conf.sample file mention the corresponding opts in
	  dial/queue; and the docs for dial/app specifically mention the
	  corresponding decls in the feature.conf file. I hope this doesn't
	  spoil some vast, eternal plan...

	* apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
	  include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
	  avoid this crash is not to try to find the channel by name in the
	  list, which is slow and resource consuming, but rather to pay
	  attention to the result codes from the ast_bridge_call, to which
	  I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
	  returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
	  then don't touch the channel pointer. If you get
	  AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
	  don't touch the peer pointer. Updated the several places where
	  the results from a bridge were not being properly obeyed, and
	  fixed some code I had introduced so that the results of the
	  bridge were not overridden (in trunk). All the places that
	  previously tested for AST_PBX_NO_HANGUP_PEER now have to check
	  for both AST_PBX_NO_HANGUP_PEER and
	  AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
	  parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
	  while A is getting the parking slot announcement, immediately
	  after being put on hold. 2. A calls B; B answers; A parks B; B
	  hangs up after A has been hung up, but before the park times out.
	  3. A calls B; B answers; B parks A; A hangs up while B is getting
	  the parking slot announcement, immediately after being put on
	  hold. 4. A calls B; B answers; B parks A; A hangs up after B has
	  been hung up, but before the park times out. No crash. I also ran
	  the scenarios above against valgrind, and accesses looked good.

2008-10-28 22:32 +0000 [r152368-152463]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Quoting in the wrong direction (Fixes
	  AST-107)

	* apps/app_dial.c: Reset all DIAL variables back to blank, in case
	  Dial is called multiple times per call (which could otherwise
	  lead to inconsistent status reports). (closes issue #13216)
	  Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: ruddy

2008-10-27 23:28 +0000 [r152286]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Buffer policy setting for half is not
	  needed.

2008-10-27 21:32 +0000 [r152215]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Inherit ALL elements of CallerID across a
	  local channel. (closes issue #13368) Reported by: Peter Schlaile
	  Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
	  (license 14)

2008-10-26 20:23 +0000 [r152059]  Sean Bright <sean.bright@gmail.com>

	* funcs/func_strings.c: Since passing \0 as the second argument to
	  strchr is valid (and will match the trailing \0 of a string) we
	  need to check that first, otherwise we end up with incorrect
	  results. Fix suggested by reporter. (closes issue #13787)
	  Reported by: meitinger

2008-10-25 10:59 +0000 [r151905]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Move AMI initialization to occur after loading
	  modules. This prevents a deadlock when someone tries to initiate
	  a module reload from the AMI just as Asterisk is starting.
	  (closes issue #13778) Reported by: hotsblanc Fix suggested by
	  hotsblanc

2008-10-23 16:04 +0000 [r151763]  Terry Wilson <twilson@digium.com>

	* configs/features.conf.sample, res/res_features.c, CHANGES:
	  Backport fix from 1.6.0 that allows you to set
	  parkedcalltransfers=no|caller|callee|both, but default to both
	  which would be the equivalent of the existing behaviour. The
	  problem was that if someone parked a call, the callee and caller
	  would both get assigned the builtin transfer feature, which would
	  not only be potentially giving someone the ability to transfer
	  themselves when they shouldn't have it, but would also dissallow
	  reinviting the media off of the call. (closes issue #12854)
	  Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
	  otherwiseguy Tested by: davidw, otherwiseguy

2008-10-20 04:57 +0000 [r151240-151241]  Kevin P. Fleming <kpfleming@digium.com>

	* autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
	  configure.ac: rename this macro to properly reflect what it does

	* autoconf/ast_check_pwlib.m4 (added), autoconf (added),
	  autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
	  (added), configure, autoconf/ast_gcc_attribute.m4 (added),
	  bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
	  autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
	  autoconf/ast_c_compile_check.m4 (added),
	  autoconf/ast_c_define_check.m4 (added),
	  autoconf/ast_prog_egrep.m4 (added),
	  autoconf/ast_check_openh323.m4 (added),
	  autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
	  (added), acinclude.m4 (removed): break up acinclude.m4 into
	  individual files, which will make it easier to maintain, easier
	  to add new macros (less patching) and will ease maintenance of
	  these macros across Asterisk branches

2008-10-19 19:51 +0000 [r151100-151167]  BJ Weschke <bweschke@btwtech.com>

	* main/asterisk.c: As per kpfleming's comments to the prior commit,
	  I'm reverting some of the changes here. A comment was made in bug
	  #13726 "3. The same mistake as in (2) is done in a few other
	  places in the code that check for: #if defined(HAVE_ZAPTEL) ||
	  defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
	  of main/asterisk.c, this is not incorrect because without
	  HAVE_ZAPTEL defined, we're missing the include for ioctl and the
	  namespace that defines DAHDI_TIMERCONFIG which is still required
	  when using Zaptel with the 1.4 branch.

	* main/asterisk.c: Fix the 1.4 branch compile again broken with
	  r150557 when using with Zaptel and not DAHDI (closes issue
	  #13740) reported by: jmls patch by: bweschke

2008-10-18 01:42 +0000 [r150816]  BJ Weschke <bweschke@btwtech.com>

	* main/manager.c: Using the GetVar handler in AMI is potentially
	  dangerous (insta-crash [tm]) when you use a dialplan function
	  that requires a channel and then you don't provide one or provide
	  an invalid one in the Channel: parameter. We'll handle this
	  situation exactly the same way it was handled in pbx.c back on
	  r61766. We'll create a bogus channel for the function call and
	  destroy it when we're done. If we have trouble allocating the
	  bogus channel then we're not going to try executing the function
	  call at all and run the risk of crashing. (closes issue #13715)
	  reported by: makoto patch by: bweschke

2008-10-17 17:18 +0000 [r150637]  Steve Murphy <murf@digium.com>

	* res/res_features.c: Interesting crash. In this case, you exit the
	  bridge with peer completely GONE. I moved the channel find call
	  up to cover the whole peer CDR reset code segment. This appears
	  to solve the crash without changing the logic at all.

2008-10-17 15:31 +0000 [r150557]  Jason Parker <jparker@digium.com>

	* main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
	  configure, configure.ac: Correctly allow chan_dahdi to compile
	  against older versions of Zaptel. Don't always define
	  HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
	  cleanup to make things clear. (closes issue #13726) Reported by:
	  tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)

2008-10-16 23:40 +0000 [r150298-150304]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Reverting changes from commits 150298 and 150301
	  since I was mistakenly under the assumption that dialplan
	  functions *always* required that a channel be present. I need to
	  go home earlier, I think :)

	* main/manager.c: And don't forget to return on the error condition

	* main/manager.c: Don't try to call a dialplan function's read
	  callback from the manager's GetVar handler if an invalid channel
	  has been specified. Several dialplan functions, including CHANNEL
	  and SIP_HEADER, do not check for NULL-ness of the channel being
	  passed in. (closes issue #13715) Reported by: makoto

2008-10-16 15:56 +0000 [r150124]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Fix memory leak found by customer

2008-10-16 15:26 +0000 [r150056]  Steve Murphy <murf@digium.com>

	* cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
	  RYM-150398 and AST-103 in internal Digium bug trackers. These
	  fixes address a really subtle memory corruption problem that
	  would happen in machines heavily loaded in production
	  environments. The corruption would always take the form of the
	  STMT object getting nulled out and one of the unixODBC calls
	  would crash trying to access statement->connection. It isn't
	  fully proven yet, but the server has now been running 2.5 days
	  without appreciable memory growth, or any gain of %cpu, and no
	  crashes. Whether this is the problem or not on that server, these
	  fixes are still warranted. As it turns out, **I** introduced
	  these errors unwittingly, when I corrected another crash earlier.
	  I had formed the build_query routine, and failed to remove
	  mutex_unlock calls in 3 places in the transplanted code. These
	  unlocks would only happen in error situations, but unlocking the
	  mutex early set the code up for a catastrophic failure, it
	  appears. It would happen only once every 100K-200K or more calls,
	  under heavy load... but that is enough. If another crash occurs,
	  with the same MO, I'll come back and remove my confession from
	  the log, and we'll keep searching, but the fact that we have
	  Asterisk dying from an asynchronous wiping of the STMT object,
	  only on some connection error, and that the server has lived for
	  2.5 days on this code without a crash, sure make it look like
	  this was the problem! Also, in several points, Statement handles
	  are set to NULL after SQLFreeHandle. This was mainly for
	  insurance, to guarantee a crash. As it turns out, the code does
	  not appear to be attempting to use these freed pointers. Asterisk
	  owes a debt of gratitude to Federico Alves and Frediano Ziglio
	  for their untiring efforts in finding this bug, among others.

2008-10-15 21:34 +0000 [r149683-149840]  BJ Weschke <bweschke@btwtech.com>

	* CHANGES: Another documentation fix. (closes issue #13708)

	* configs/agents.conf.sample: An update to the
	  documentation/example of agents.conf.sample with the correct
	  parameter for this feature as defined in chan_agent.c (closes
	  issue #13709)

2008-10-15 10:30 +0000 [r149452]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix some problems when parsing SIP messages
	  that have the maximum number of headers or body lines that we
	  support

2008-10-14 23:43 +0000 [r149130-149266]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Change this warning to an error message.
	  Suggestion comes from Sean Bright. Thanks Sean!

	* channels/chan_sip.c: Call register_peer_exten even in the case
	  that the peer's IP/port does not change. (closes issue #13309)
	  Reported by: dimas Patches: v2-13309.patch uploaded by dimas
	  (license 88)

	* include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
	  period for sync-triggered audiohooks so that if packetization of
	  audio is close (but not equal) we don't end up flushing the
	  audiohooks over small inconsistencies in synchronization. Related
	  to issue #13005, and solves the issue for most people who were
	  experiencing the problem. However, a small number of people are
	  still experiencing the problem on long calls, so I am not closing
	  the issue yet

	* apps/app_queue.c: Update the queue with the correct number of
	  calls and whether the call was completed within the service level
	  when a transfer takes place. This way, we do not "break" the
	  leastrecent and fewestcalls strategies by not logging a call
	  until after the transferred call has ended. (closes issue #13395)
	  Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
	  by Marquis (license 32)

	* channels/chan_sip.c: Don't allow reserved characters to be used
	  in register lines in sip.conf. (closes issue #13570) Reported by:
	  putnopvut

2008-10-14 20:09 +0000 [r149061]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_waitforsilence.c: Check correct values in the return of
	  ast_waitfor(); also, get rid of a possible memory leak. (closes
	  issue #13658) Reported by: explidous Patch by: me

2008-10-14 19:05 +0000 [r148990]  Leif Madsen <lmadsen@digium.com>

	* CHANGES: Add in some missing updates to the CHANGES file for
	  sip.conf (closes issue #13100) Reported and patch by:
	  gknispel_proformatique

2008-10-14 19:03 +0000 [r148916-148987]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Some compilers warn, some don't. Fixing.

	* apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
	  even when UTF-8 characters are used in headers like 'Subject' and
	  'To'. Closes AST-107.

2008-10-14 17:33 +0000 [r148912]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Deadlock prevention in chan_local. (closes
	  issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
	  by putnopvut (license 60) Tested by: tacvbo

2008-10-14 10:30 +0000 [r148611-148736]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: on Ubuntu (at least), recent versions of ld in binutils
	  delete all debugging symbols when -x is supplied; since the
	  reasons why -x is being passed are lost in the mists of time,
	  remove it so debugging will work properly

	* main/translate.c: it would be nice if this message printing code
	  had actually been tested before it was committed...

2008-10-10 16:25 +0000 [r147997-148257]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: User not notified of temporary greeting, if
	  ODBC storage is in use. (closes issue #13659) Reported by:
	  moliveras Patches: 20081009__bug13659.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: moliveras

	* apps/app_voicemail.c: When blank, callerid name and number should
	  display "unknown caller" in voicemail emails. (Closes issue
	  #13643)

2008-10-09 18:56 +0000 [r147941]  Jeff Peeler <jpeeler@digium.com>

	* res/res_features.c: (closes issue #13139) Reported by: krisk84
	  Tested by: krisk84 This change prevents a call that is placed in
	  the parkinglot to be picked up before the PBX is finished. If
	  another extension dials the parking extension before the PBX
	  thread has completed at minimum warnings will occur about the PBX
	  not properly being terminated. At worst, a crash could occur.

2008-10-08 22:22 +0000 [r147681]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: when parsing a text configuration option,
	  ensure that the buffer on the stack is actually large enough to
	  hold the legal values of that option, and also ensure that
	  sscanf() knows to stop parsing if it would overrun the buffer
	  (without these changes, specifying "buffers=...,immediate" would
	  overflow the buffer on the stack, and could not have worked as
	  expected)

2008-10-08 14:51 +0000 [r147517]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: If we receive DTMF make sure that the
	  state of the speech structure goes back to being not ready.
	  (issue #LUMENVOX-8)

2008-10-07 23:14 +0000 [r147429-147430]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: revert this change until i can understand
	  why it results in locking order changes

	* channels/chan_dahdi.c: don't start a PBX on incoming PRI call
	  channels until after we're done setting channel variables and
	  other things on the channel, otherwise the channel might go away
	  (if the dialplan hangs up quickly) before we are done, which
	  results in a spectacular crash

2008-10-07 16:48 +0000 [r147193]  Sean Bright <sean.bright@gmail.com>

	* apps/app_voicemail.c: Make 'imapsecret' an alias to
	  'imappassword' in voicemail.conf.

2008-10-06 20:52 +0000 [r146711-146799]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_callerid.c, apps/app_speech_utils.c,
	  funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c,
	  channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
	  funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c:
	  Dialplan functions should not actually return 0, unless they have
	  modified the workspace. To signal an error (and no change to the
	  workspace), -1 should be returned instead. (closes issue #13340)
	  Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt
	  uploaded by Corydon76 (license 14)

	* channels/chan_local.c: Check whether an extension exists in the
	  _call method, rather than the _alloc method, because we need to
	  evaluate the callerid (since that data affects whether an
	  extension exists). (closes issue #13343) Reported by: efutch
	  Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
	  (license 14) Tested by: efutch

2008-10-06 15:57 +0000 [r146643]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: ensure that the private structure for
	  pseudo channels is created without 'leaking' configuration data
	  from other configured channels (closes issue #13555) Reported by:
	  jeffg Patches: issue_13555.patch uploaded by kpfleming (license
	  421) Tested by: jeffg

2008-10-05 21:17 +0000 [r146448]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Fix silly formatting.

2008-10-03 22:51 +0000 [r146244]  Sean Bright <sean.bright@gmail.com>

	* apps/app_rpt.c: Change some preprocessor macros to struct
	  definitions so that we get app_rpt to build with DAHDI. (closes
	  issue #13576) Reported by: blitzrage

2008-10-03 20:44 +0000 [r146129]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/features.h, res/res_features.c, res/res_agi.c:
	  (closes issue #13425) Reported by: mdu113 Tested by: mdu113
	  Similar to r143204, masquerade the channel in the case of Park
	  being called from AGI.

2008-10-03 17:12 +0000 [r146026]  Steve Murphy <murf@digium.com>

	* res/res_features.c: (closes issue #13579) Reported by: dwagner
	  (closes issue #13584) Reported by: dwagner Tested by: murf,
	  putnopvut The thought occurred to me that the res= from the
	  extension spawn was ending up being returned from the bridge.
	  "Thou shalt not poison the return value". Made the change and it
	  appears to allow blind xfers to work as normal. If I'm wrong,
	  reopen the bugs. But it looks good to me! Many thanks to
	  putnopvut for helping me reproduce this!

2008-10-02 16:39 +0000 [r145751-145839]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Backport support for some of the keyword
	  modifications used in 1.6 (while warning that some options aren't
	  really supported) and add some warning messages. Some credit to
	  oej, who was complaining in #asterisk-dev.

	* res/res_odbc.c: Some sanity checks that may have led to prior
	  crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
	  of incorrectly-used constants.

2008-10-01 17:18 +0000 [r145479]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql
	  script to create the setinterfacevar column. (closes issue
	  #13549) Reported by: fiddur

2008-10-01  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.22 released.

2008-09-09  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.22-rc5 released.

2008-09-09 15:40 +0000 [r142063]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Ensure that the stored CDR reference is still
	  valid after the bridge before poking at it. Also, keep the
	  channel locked while messing with this CDR. (fixes crashes
	  reported in issue #13409)

2008-09-08 21:10 +0000 [r141809]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
	  the remote tag of an endpoint once a dialog has been confirmed.
	  Up until that point, it is possible and legal for the far-end to
	  send provisional responses with a different To: tag each time.
	  With this patch applied, these provisional messages will not
	  cause a matching problem. (closes issue #11536) Reported by: ibc
	  Patches: 11536v2.patch uploaded by putnopvut (license 60)

2008-09-08 21:02 +0000 [r141806]  Russell Bryant <russell@digium.com>

	* main/pbx.c: When doing an async goto, detect if the channel is
	  already in the middle of a masquerade. This can happen when
	  chan_local is trying to optimize itself out. If this happens,
	  fail the async goto instead of bursting into flames. (closes
	  issue #13435) Reported by: geoff2010

2008-09-08  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.22-rc4 released.

2008-09-08 20:15 +0000 [r141741]  Jason Parker <jparker@digium.com>

	* Makefile, redhat (removed): Remove RPM package targets from
	  Makefile (and all associated parts). This has never worked in
	  1.4, and we decided that it makes no sense to be done here. There
	  are many distros out there that already have "proper" spec files
	  that can be (re)used. Closes issue #13113 Closes issue #10950
	  Closes issue #10952

2008-09-08 16:26 +0000 [r141678]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Actually use Zaptel CFLAGS if using
	  Zaptel instead of DAHDI This fixes building against Zaptel when
	  using a custom path

2008-09-06 20:13 +0000 [r141565]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This fix comes from Joshua Colp The
	  Brilliant, who, given the trace, came up with a solution. This
	  will most likely will close 13235 and 13409. I'll wait till
	  Monday to verify, and then close these bugs.

2008-09-06 15:23 +0000 [r141503]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Reverting behavior change (AGI should not exit
	  non-zero on SUCCESS) (closes issue #13434) Reported by:
	  francesco_r

2008-09-05 21:10 +0000 [r141217-141366]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Agent's should not try to call a channel's
	  indicate callback if the channel has been hung up. It will likely
	  crash otherwise ABE-1159

	* apps/app_voicemail.c: Since greetings are not stored in IMAP, we
	  should not be DISPOSE'ing of them the same way we do with other
	  messages. (closes issue #13414) Reported by: mthomasslo Patches:
	  13414v2.patch uploaded by putnopvut (license 60) Tested by:
	  mthomasslo

	* channels/chan_sip.c: Commit 140417 had a logic flaw in it which
	  caused port 5060 to always be used when dialing a peer if no
	  explicit port was specified. This broke the behavior of
	  implicitly using the port from which the peer registered if no
	  port is specified. This commit fixes the logic flaw. (closes
	  issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
	  by putnopvut (license 60) Tested by: mdu113

2008-09-05 14:15 +0000 [r141094-141156]  Steve Murphy <murf@digium.com>

	* main/channel.c: A small change to prevent double-posting of
	  CDR's; thanks to Daniel Ferrer for bringing it to our attention

	* pbx/ael/ael-test/ref.ael-vtest25 (added),
	  pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
	  pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
	  #13357) Reported by: pj Tested by: murf (closes issue #13416)
	  Reported by: yarns Tested by: murf If you find this message
	  overly verbose, relax, it's probably not meant for you. This
	  message is meant for probably only two people in the whole world:
	  me, or the poor schnook that has to maintain this code because
	  I'm either dead or unavailable at the moment. This fix solves two
	  reports, both having to do with embedding a function call in a
	  ${} construct. It was tricky because the funccall syntax has
	  parenthesis () in it. And up till now, the 'word' token in the
	  flex stuff didn't allow that, because it would tend to steal the
	  LP and RP tokens. To be truthful, the "word" token was the
	  trickiest, most unstable thing in the whole lexer. I was lucky it
	  made this long without complaints. I had to choose every
	  character in the pattern with extreme care, and I knew that
	  someday I'd have to revisit it. Well, the day has come. So, my
	  brilliant idea (and I'm being modest), was to use the surrounding
	  ${} construct to make a state machine and capture everything in
	  it, no matter what it contains. But, I have to now treat the word
	  token like I did with comments, in that I turn the whole thing
	  into a state-machine sort of spec, with new contexts
	  "curlystate", "wordstate", and "brackstate". Wait a minute,
	  "brackstate"? Yes, well, it didn't take very many regression
	  tests to point out if I do this for ${} constructs, I also have
	  to do it with the $[] constructs, too. I had to create a separate
	  pcbstack2 and pcbstack3 because these constructs can occur inside
	  macro argument lists, and when we have two state machines
	  operating on the same structures we'd get problems otherwise. I
	  guess I could have stopped at pcbstack2 and had the brackstate
	  stuff share it, but it doesn't hurt to be safe. So, the pcbpush
	  and pcbpop routines also now have versions for "2" and "3". I had
	  to add the {KEYWORD} construct to the initial pattern for "word",
	  because previously word would match stuff like "default7",
	  because it was a longer match than the keyword "default". But,
	  not any more, because the word pattern only matches only one or
	  two characters now, and it will always lose. So, I made it the
	  winner again by making an optional match on any of the keywords
	  before it's normal pattern. I added another regression test to
	  make sure we don't lose this in future edits, and had to fix just
	  one regression, where it no longer reports a 'cascaded' error,
	  which I guess is a plus. I've given some thought as to whether to
	  apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
	  decided to put it in 1.4 because one of the bug reports was
	  against 1.4; and it is unexpected that AEL cannot handle this
	  situation. It actually reduced the amount of useless "cascade"
	  error messages that appeared in the regressions (by one line,
	  ehhem). There is a possible side-effect in that it does now do
	  more careful checking of what's in those ${} constructs, as far
	  as matching parens, and brackets are concerned. Some users may
	  find a an insidious problem and correct it this way. This should
	  be exceedingly rare, I hope.

2008-09-04 17:00 +0000 [r141028]  Jeff Peeler <jpeeler@digium.com>

	* res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
	  multiple parking problems: Crash when executing a park on an
	  extension dialed by AGI due to not returning the proper return
	  code. Crash when using a builtin feature that was a subset of a
	  enabled dynamic feature. Crash due to always hanging up the peer
	  despite the fact that the peer was supposed to be parked.

2008-09-03  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.22-rc3 released.

2008-09-03 14:29 +0000 [r140850]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
	  storage. (closes issue #13387) Reported by: moliveras Patches:
	  13387.patch uploaded by putnopvut (license 60) Tested by:
	  putnopvut, moliveras

2008-09-03 13:24 +0000 [r140816]  Russell Bryant <russell@digium.com>

	* main/poll.c: Don't freak out if the poll emulation receives NULL
	  for the pollfds array (closes issue #13307) Reported by: jcovert

2008-09-02 23:47 +0000 [r140751]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: After adding the context checking to
	  app_voicemail for IMAP storage, I left out a crucial place to
	  copy the context to the vm_state structure. This is the
	  correction.

2008-09-02 23:36 +0000 [r140670-140747]  Steve Murphy <murf@digium.com>

	* main/cdr.c: I am turning the warnings generated in ast_cdr_free
	  and post_cdr into verbose level 2 messages. Really, they matter
	  little to end users. You either get the CDR's you wanted, or you
	  don't, and it is a bug.

	* main/channel.c: After reconsidering, with respect to 13409,
	  ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
	  which generates lots of useless warnings that will undoubtably
	  generate complaints.

	* main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
	  tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
	  tomaso (license 564) I basically spent the day, verifying that
	  this patch solves the problem, and doesn't hurt in non-problem
	  cases. Why valgrind did not plainly reveal this leak absolutely
	  mystifies and stuns me. Many, many thanks to tomaso for finding
	  and providing the fix.

2008-09-02 18:14 +0000 [r140605]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_iax2.c: Make sure to use the correct length of the
	  mohinterpret and mohsuggest buffers when copying configuration
	  values. (closes issue #13336) Reported by:
	  decryptus_proformatique Patches:
	  chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
	  by decryptus (license 555)

2008-08-29 17:34 +0000 [r140417-140488]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
	  working on the ao2_containers branch, I noticed something a bit
	  strange. In all cases where we provide a callback function to
	  ao2_container_alloc, the callback function would only return 0 or
	  CMP_MATCH. After inspecting the ao2_callback() code carefully, I
	  found that if you're only looking for one specific item, then you
	  should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
	  continue traversing the current bucket until the end searching
	  for more matches. In cases like chan_iax2 where in 1.4, all the
	  peers are shoved into a single bucket, this makes for potentially
	  terrible performance since the entire bucket will be traversed
	  even if the peer is one of the first ones come across in the
	  bucket. All the changes I have made were for cases where the
	  callback function defined was passed to ao2_container_alloc so
	  that calls to ao2_find could find a unique instance of whatever
	  object was being stored in the container.

	* apps/app_voicemail.c: Add context checking when retrieving a
	  vm_state. This was causing a problem for people who had
	  identically named mailboxes in separate voicemail contexts. This
	  commit affects IMAP storage only. (closes issue #13194) Reported
	  by: moliveras Patches: 13194.patch uploaded by putnopvut (license
	  60) Tested by: putnopvut, moliveras

	* channels/chan_sip.c: Fix SIP's parsing so that if a port is
	  specified in a string to Dial(), it is not ignored. (closes issue
	  #13355) Reported by: acunningham Patches: 13355v2.patch uploaded
	  by putnopvut (license 60) Tested by: acunningham

2008-08-27 19:49 +0000 [r140299]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
	  when in pedantic mode. The problem was that the wrong tags would
	  be compared depending on the direction of the call. (closes issue
	  #13353) Reported by: flefoll Patches:
	  chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
	  (license 244)

2008-08-26 16:49 +0000 [r140115]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: add HAVE_PRI if define around
	  dahdi_close_pri_fd

2008-08-26 16:07 +0000 [r140060]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
	  This code used the return value of a completely unrelated
	  function to determine whether the scheduler should be run or not.
	  This would have caused the scheduler to not run in cases where it
	  should have. Also, leave a note about another scheduler issue
	  that needs to be addressed at some point.

2008-08-26 15:57 +0000 [r140056]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
	  Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
	  by: tzafrir, jpeeler This patch fixes closing open file
	  descriptors in the case of an error.

2008-08-26 15:27 +0000 [r140051]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a race condition with the IAX scheduler
	  thread. A lock and condition are used here to allow newly
	  scheduled tasks to wake up the scheduler just in case the new
	  task needs to run sooner than the current wakeup time when the
	  thread is sleeping. However, there was a race condition such that
	  a newly scheduled task would not properly wake up the scheduler
	  or affect the wake up period. The order of execution would have
	  been: 1) Scheduler thread determines wake up time of N ms. 2)
	  Another thread schedules a task and signals the condition, with
	  an execution time of < N ms. 3) Scheduler thread locks and goes
	  to sleep for N ms. By moving the sleep time determination to
	  inside the critical section, this possibility is avoided.

2008-08-26 15:22 +0000 [r140050]  Terry Wilson <twilson@digium.com>

	* Makefile: sounds/Makefile installs sounds using the "new"
	  language directory structure, but languageprefix needs to be set
	  = yes for sounds in subdirectories (digits/1, etc.) to play as
	  the correct language. Fix the generation of asterisk.conf to
	  include languageprefix=yes

2008-08-26 14:09 +0000 [r140029]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: correct a file location in an error
	  message

2008-08-25 21:47 +0000 [r139927]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: Fix a typo I made. Lesson learned, apply the
	  patch if one exists.

2008-08-25 21:31 +0000 [r139909]  Sean Bright <sean.bright@gmail.com>

	* build_tools/get_moduleinfo, build_tools/get_makeopts: Some
	  versions of awk (nawk, for example) don't like empty regular
	  expressions so be slightly more verbose. (closes issue #13374)
	  Reported by: dougm Patches: 13374.diff uploaded by seanbright
	  (license 71) Tested by: dougm

2008-08-25 20:46 +0000 [r139869]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely

2008-08-25 15:52 +0000 [r139769]  Mark Michelson <mmichelson@digium.com>

	* main/config.c: Fix the logic in config_text_file_save so that if
	  an UpdateConfig manager action is issued and the file specified
	  in DstFileName does not yet exist, an error is not returned.
	  (closes issue #13341) Reported by: vadim Patches: 13341.patch
	  uploaded by putnopvut (license 60) (with small modification from
	  seanbright)

2008-08-25 15:33 +0000 [r139764]  Steve Murphy <murf@digium.com>

	* main/pbx.c, res/res_features.c: This patch reverts the changes
	  made via 139347, and 139635, as users are seeing adverse
	  difference. I will un-close 13251. Back to the drawing board/
	  concept/ beginning/ whatever!

2008-08-22 22:24 +0000 [r139635]  Steve Murphy <murf@digium.com>

	* res/res_features.c: I found some problems with the code I
	  committed earlier, when I merged them into trunk, so I'm coming
	  back to clean up. And, in the process, I found an error in the
	  code I added to trunk and 1.6.x, that I'll fix using this patch
	  also.

2008-08-22 21:36 +0000 [r139621]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: (closes issue #13359) Reported by: Laureano
	  Patches: originate_channel_check.patch uploaded by Laureano
	  (license 265)

2008-08-22 19:45 +0000 [r139456-139553]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/threadstorage.h: Fix compilation when
	  DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
	  by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
	  (license 35)

	* main/frame.c: Remove show_frame_stats_deprecated since it is not
	  used anywhere and causes build errors if building under dev-mode
	  with TRACE_FRAMES selected in menuselect. (closes issue #13362)
	  Reported by: snuffy

	* channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!

	* channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
	  from incorrect locking order between iax2_pvt and ast_channel
	  structures. AST-13

2008-08-21 23:39 +0000 [r139387]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fixes loop that could possibly never exit
	  in the event of a channel never being able to be opened or
	  specify after a restart. (closes issue #11017)

2008-08-21 23:03 +0000 [r139347]  Steve Murphy <murf@digium.com>

	* main/pbx.c, res/res_features.c: (closes issue #13251) Reported
	  by: sergee Tested by: murf THis is a bold move for a static
	  release fix, but I wouldn't have made it if I didn't feel
	  confident (at least a *bit* confident) that it wouldn't mess
	  everyone up. The reasoning goes something like this: 1. We simply
	  cannot do anything with CDR's at the current point (in pbx.c,
	  after the __ast_pbx_run loop). It's way too late to have any
	  affect on the CDRs. The CDR is already posted and gone, and the
	  remnants have been cleared. 2. I was very much afraid that moving
	  the running of the 'h' extension down into the bridge code (where
	  it would be now practical to do it), would result in a lot more
	  calls to the 'h' exten, so I implemented it as another exten
	  under another name, but found, to my pleasant surprise, that
	  there was a 1:1 correspondence to the running of the 'h' exten in
	  the pbx_run loop, and the new spot at the end of the bridge. So,
	  I ifdef'd out the current 'h' loop, and moved it into the bridge
	  code. The only difference I can see is the stuff about the
	  AST_PBX_KEEPALIVE, and hopefully, if this is still an important
	  decision point, I can replicate it if there are complaints. To be
	  perfectly honest, the KEEPALIVE situation is not totally clear to
	  me, and how it relates to a post-bridge situation is less clear.
	  I suspect the users will point out everything in total clarity if
	  this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
	  into the channel before running the 'h' exten, which makes it
	  possible for users to edit the cdr before it goes out the door.
	  And, of course, with the endbeforehexten config var set, the
	  users can also get at the billsec/duration vals. After the h
	  exten finishes, the cdr is swapped back and processing continues
	  as normal. Please, all who deal with CDR's, please test this
	  version of Asterisk, and file bug reports as appropriate!

2008-08-21 10:11 +0000 [r139283]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
	  too.

2008-08-20 22:14 +0000 [r139213]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
	  issue here is that if you call ChanSpy and specify a spy group,
	  and sit in the application long enough looping through the
	  channel list, you will eventually run out of stack space and the
	  application with exit with a seg fault. The backtrace was always
	  inside of a harmless snprintf() call, so it was tricky to track
	  down. However, it turned out that the call to snprintf() was just
	  the biggest stack consumer in this code path, so it would always
	  be the first one to hit the boundary. (closes issue #13338)
	  Reported by: ruddy

2008-08-20 19:52 +0000 [r139151]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Fix bug where the samples were not accurate
	  when in G723 mode, which would cause the timestamp field of the
	  RTP header to be invalid.

2008-08-20 19:35 +0000 [r139145]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Backport support
	  for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
	  doing so just makes it difficult for people with channels that
	  are in alarm when Asterisk starts up to get them going once the
	  alarm is cleared (closes issue #12160) Reported by: tzafrir
	  Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
	  (license 46) Tested by: tzafrir

2008-08-20 17:14 +0000 [r139074]  Steve Murphy <murf@digium.com>

	* main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
	  murf The specialized reset routine is tromping on the flags field
	  of the CDR. I made a change to not reset the DISABLED bit. This
	  should get rid of this problem.

2008-08-20 15:37 +0000 [r139015]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: sip_read should properly handle a NULL
	  return from sip_rtp_read. (closes issue #13257) Reported by:
	  travishein

2008-08-19 23:22 +0000 [r138949]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
	  compatability define for Zaptel

2008-08-19 23:17 +0000 [r138942]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Reset agent_pvt variables back to the
	  values in agents.conf (from what the corresponding channel
	  variables were set to) when the agent logs out. (closes issue
	  #13098) Reported by: davidw Patches:
	  20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
	  bbryant (license 36) Tested by: davidw

2008-08-19 22:56 +0000 [r138938]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Add configuration option to
	  chan_dahdi.conf to allow buffering policy and number of buffers
	  to be configured per channel. Syntax: buffers=<num of
	  buffers>,<policy> Where the number of buffers is some
	  non-negative integer and the policy is either "full", "half", or
	  "immediate".

2008-08-19 18:50 +0000 [r138685-138886]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Add a lock and unlock prior to the
	  destruction of the chanspy_ds lock to ensure that no other
	  threads still have it locked. While this should not happen under
	  normal circumstances, it appears that if the spyer and spyee hang
	  up at nearly the same time, the following may occur. 1.
	  ast_channel_free is called on the spyee's channel. 2. The chanspy
	  datastore is removed from the spyee's channel in
	  ast_channel_free. 3. In the spyer's thread, the spyer attempts to
	  remove and destroy the datastore from the spyee channel, but the
	  datastore has already been removed in step 2, so the spyer
	  continues in the code. 4. The spyee's thread continues and calls
	  the datastore's destroy callback, chanspy_ds_destroy. This
	  involves locking the chanspy_ds. 5. Now the spyer attempts to
	  destroy the chanspy_ds lock. The problem is that in step 4, the
	  spyee has locked this lock, meaning that the spyer is attempting
	  to destroy a lock which is currently locked by another thread.
	  The backtrace provided in issue #12969 supports the idea that
	  this is possible (and has even occurred). This commit does not
	  close the issue, but should help in preventing one type of crash
	  associated with the use of app_chanspy.

	* apps/app_queue.c: Change the inequalities used in app_queue with
	  regards to timeouts from being strict to non-strict for more
	  accuracy. (closes issue #13239) Reported by: atis Patches:
	  app_queue_timeouts_v2.patch uploaded by atis (license 242)

2008-08-18 16:57 +0000 [r138663]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_dahdi.c: look for transcoder in proper place based
	  on build against Zaptel or DAHDI

2008-08-18 11:57 +0000 [r138569]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_dahdi.c: You know what's awesome? Code that
	  compiles... ;)

2008-08-18 02:05 +0000 [r138516]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: fix compilation warnings

2008-08-16 01:12 +0000 [r138309-138360]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: fixes use count to properly decrement if
	  an active dahdi channel is destroyed allowing module to be
	  unloaded

	* channels/chan_dahdi.c: add forgotten locks around ss_thread_count
	  in ss_thread for dahdi restart

2008-08-15 22:33 +0000 [r138258]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: More fixes for
	  realtime peers. (closes issue #12921) Reported by: Nuitari
	  Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
	  (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
	  (license 14) Tested by: Corydon76

2008-08-15 21:28 +0000 [r138119-138238]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: initialize condition variable
	  ss_thread_complete using ast_cond_init

	* channels/chan_dahdi.c: declared static mutexes using
	  AST_MUTEX_DEFINE_STATIC macro

	* channels/chan_dahdi.c: Fixes the dahdi restart functionality.
	  Dahdi restart allows one to restart all DAHDI channels, even if
	  they are currently in use. This is different from unloading and
	  then loading the module since unloading requires the use count to
	  be zero. Reloading the module is different in that the signalling
	  is not changed from what it was originally configured. Also, this
	  fixes not closing all the file descriptors for D-channels upon
	  module unload (which would prevent loading the module
	  afterwards). (closes issue #11017)

2008-08-15 15:07 +0000 [r138027]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Ensure that when a hangup occurs in
	  autoservice, that a hangup frame gets properly deferred to be
	  read from the channel owner when it gets taken out of
	  autoservice. (closes issue #12874) Reported by: dimas Patches:
	  v1-12874.patch uploaded by dimas (license 88)

2008-08-15 14:51 +0000 [r137847-138023]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c: Additional check for more string specifiers
	  than arguments. (closes issue #13299) Reported by: adomjan
	  Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
	  (license 14) func_strings.c-sprintf.patch uploaded by adomjan
	  (license 487) Tested by: adomjan

	* channels/chan_dahdi.c: Oops, wrong direction

	* channels/chan_dahdi.c: When creating the secondary subchannel
	  name, it is necessary to compare to the existing channel name
	  without the "Zap/" or "DAHDI/" prefix, since our test string is
	  also without that prefix. (closes issue #13027) Reported by:
	  dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
	  (license 525) (Slightly modified by me, to compensate for both
	  names)

2008-08-14 14:05 +0000 [r137731]  Russell Bryant <russell@digium.com>

	* configs/sip.conf.sample: Comments in this config file were
	  aligned only if your tab size was set to 8. So, convert tabs to
	  spaces so that things should be aligned regardless of what tab
	  size you use in your editor.

2008-08-14 02:03 +0000 [r137677-137679]  Kevin P. Fleming <kpfleming@digium.com>

	* Zaptel-to-DAHDI.txt: forgot one module name that changed

	* include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
	  build_tools/menuselect-deps.in, configure, configure.ac,
	  codecs/codec_dahdi.c: add support for Zaptel versions that
	  contain the new transcoder interface

2008-08-13 21:35 +0000 [r137580]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Register DAHDISendKeypadFacility
	  application if dahdi_chan_mode is set to DAHDI + Zap. Mark
	  ZapSendKeypadFacility application as deprecated on usage.

2008-08-13 20:46 +0000 [r137527-137530]  Kevin P. Fleming <kpfleming@digium.com>

	* Zaptel-to-DAHDI.txt (added): add document describing what users
	  will need to be aware of when upgrading to this version and using
	  DAHDI

	* apps/app_meetme.c: remove some more chan_zap references

	* doc/asterisk-conf.txt, channels/chan_dahdi.c: document
	  dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
	  read its configuration from zapata.conf if dahdichanname has been
	  set to 'no'

2008-08-13 14:33 +0000 [r137348-137405]  Sean Bright <sean.bright@gmail.com>

	* doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds

	* cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
	  and have it try to store CDR(userfield) if it is set. The new
	  behavior is to check for the userfield column on module load, and
	  if it exists, we will store CDR(userfield) when CDRs are written.
	  A similar patch already went into trunk and 1.6.0. (closes issue
	  #13290) Reported by: falves11

2008-08-11 13:33 +0000 [r137188]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_meetme.c: convert this module to be able to handle DAHDI
	  or Zaptel (reported on asterisk-users, don't know how this got
	  missed before)

2008-08-11 00:20 +0000 [r137138]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Deallocate database connection handle on
	  disconnect, as we allocate another one on connect. (closes issue
	  #13271) Reported by: dveiga

2008-08-09 17:11 +0000 [r136999]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
	  to 0 if not found instead of empty. pointed out by tzafrir on
	  #asterisk-dev

2008-08-09 15:25 +0000 [r136946]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
	  revisions 136945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
	  | 2 lines Regression fixes for Solaris ........

2008-08-08 00:15 +0000 [r136726]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
	  include/asterisk/ael_structs.h: (closes issue #13236) Reported
	  by: korihor Wow, this one was a challenge! I regrouped and ran a
	  new strategy for setting the ~~MACRO~~ value; I set it once per
	  extension, up near the top. It is only set if there is a switch
	  in the extension. So, I had to put in a chunk of code to detect a
	  switch in the pval tree. I moved the code to insert the set of
	  ~~exten~~ up to the beginning of the gen_prios routine, instead
	  of down in the switch code. I learned that I have to push the
	  detection of the switches down into the code, so everywhere I
	  create a new exten in gen_prios, I make sure to pass onto it the
	  values of the mother_exten first, and the exten next. I had to
	  add a couple fields to the exten struct to accomplish this, in
	  the ael_structs.h file. The checked field makes it so we don't
	  repeat the switch search if it's been done. I also updated the
	  regressions.

2008-08-07 18:25 +0000 [r136560]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/menuselect-deps.in, configure, configure.ac: change
	  the required dependency for codec_dahdi to only be satisfied by
	  DAHDI and not Zaptel, as the new transcoder interface is only in
	  DAHDI

2008-08-07 18:14 +0000 [r136544]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Updated codec_dahdi to use the new
	  transcoder interface in the first DAHDI release. Codec dahdi no
	  longer functions with the transcoder interface in zaptel at this
	  time (which the last zaptel release was 1.4.11). NOTE: Still
	  needs an update to the configure script to make sure that
	  codec_dahdi is only built if the new transcoder interface is
	  present in the drivers. (Issue: DAHDI-42)

2008-08-07 16:50 +0000 [r136488]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Update persistent state on all exit conditions.
	  (closes issue #12916) Reported by: sgenyuk Patches:
	  app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
	  by: sgenyuk, aragon

2008-08-07 16:30 +0000 [r136404-136484]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/doxyref.h: add a raw list of all libraries that
	  any part of Asterisk links directly to

	* apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
	  incorrectly ignores the casting away of 'const' for pointers when
	  the developer knows it is safe to do so

	* Makefile: remove config.cache during distclean, in case the user
	  is using autoconf caching

2008-08-07 01:31 +0000 [r136304-136348]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Also, parse
	  useincomingcalleridonzaptransfer (and add appropriate deprecation
	  warnings).

	* channels/chan_dahdi.c: For backwards compatibility with previous
	  1.4 versions which used "zapchan" in users.conf, ensure that we
	  still support it.

2008-08-06 21:18 +0000 [r136241]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  configs/misdn.conf.sample: * The allowed_bearers setting in
	  misdn.conf misspelled one of its options: digital_restricted. *
	  Fixed some other spelling errors and typos.

2008-08-06 20:42 +0000 [r136238]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: We only need to unregister the QueueStatus
	  manager command once on an unload

2008-08-06 20:14 +0000 [r136190]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
	  not a directory path. (closes issue #13007) Reported by:
	  klaus3000

2008-08-06 18:58 +0000 [r136168]  Russell Bryant <russell@digium.com>

	* Makefile: Remove the use of --no-print-directory when compiling
	  subdirectories. This allows vim :make functionality to work
	  properly when errors have occurred in the build. Without printing
	  the directories, vim did not know how to find the file that the
	  error occurred in. If the extra bit of build noise annoys anyone,
	  just let me know, and I'll make this optional.

2008-08-06 15:58 +0000 [r136062]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, channels/chan_skinny.c: Since adding the
	  AST_CONTROL_SRCUPDATE frame type, there are places where
	  ast_rtp_new_source may be called where the tech_pvt of a channel
	  may not yet have an rtp structure allocated. This caused a crash
	  in chan_skinny, which was fixed earlier, but now the same crash
	  has been reported against chan_h323 as well. It seems that the
	  best solution is to modify ast_rtp_new_source to not attempt to
	  set the marker bit if the rtp structure passed in is NULL. This
	  change to ast_rtp_new_source also allows the removal of what is
	  now a redundant pointer check from chan_skinny. (closes issue
	  #13247) Reported by: pj

2008-08-06 03:53 +0000 [r135899-135949]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Fix a longstanding bug in channel walking logic,
	  and fix the explanation to make sense. (Closes issue #13124)

	* main/translate.c: Since powerof() can return an error condition,
	  it's foolhardy not to detect and deal with that condition.
	  (Related to issue #13240)

	* include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
	  Bugfix for debugging code 2) Reduce compiler warnings for another
	  section of debugging code (Closes issue #13237)

2008-08-06 00:29 +0000 [r135841-135850]  Mark Michelson <mmichelson@digium.com>

	* /: Remove properties that should not be here

	* apps/app_skel.c: Revert inadvertent changes to app_skel that
	  occurred when I was testing for a memory leak

	* include/asterisk/abstract_jb.h, main/channel.c, /,
	  apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
	  Merging the issue11259 branch. The purpose of this branch was to
	  take into account "burps" which could cause jitterbuffers to
	  misbehave. One such example is if the L option to Dial() were
	  used to inject audio into a bridged conversation at regular
	  intervals. Since the audio here was not passed through the
	  jitterbuffer, it would cause a gap in the jitterbuffer's
	  timestamps which would cause a frames to be dropped for a brief
	  period. Now ast_generic_bridge will empty and reset the
	  jitterbuffer each time it is called. This causes injected audio
	  to be handled properly. ast_generic_bridge also will empty and
	  reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
	  frame since the change in audio source could negatively affect
	  the jitterbuffer. All of this was made possible by adding a new
	  public API call to the abstract_jb called ast_jb_empty_and_reset.
	  (closes issue #11259) Reported by: plack Tested by: putnopvut

2008-08-05 23:13 +0000 [r135799]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
	  include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
	  Tested by: murf I discovered that also, in the previous bug fixes
	  and changes, the cdr.conf 'unanswered' option is not being
	  obeyed, so I fixed this. And, yes, there are two 'answer' times
	  involved in this scenario, and I would agree with you, that the
	  first answer time is the time that should appear in the CDR. (the
	  second 'answer' time is the time that the bridge was begun). I
	  made the necessary adjustments, recording the first answer time
	  into the peer cdr, and then using that to override the bridge
	  cdr's value. To get the 'unanswered' CDRs to appear, I purposely
	  output them, using the dial cmd to mark them as DIALED (with a
	  new flag), and outputting them if they bear that flag, and you
	  are in the right mode. I also corrected one small mention of the
	  Zap device to equally consider the dahdi device. I heavily tested
	  10-sec-wait macros in dial, and without the macro call; I tested
	  hangups while the macro was running vs. letting the macro
	  complete and the bridge form. Looks OK. Removed all the
	  instrumentation and debug.

2008-08-05 21:34 +0000 [r135747]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
	  meaning of the flag IAX_HASCALLERID was perverted. This change
	  reverts IAX2 to the original meaning, which was, that the
	  callerid set on the client should be overridden on the server,
	  even if that means the resulting callerid is blank. In other
	  words, if you set "callerid=" in the IAX config, then the
	  callerid should be overridden to blank, even if set on the
	  client. Note that there's a distinction, even on realtime,
	  between the field not existing (NULL in databases) and the field
	  existing, but set to blank (override callerid to blank).

2008-08-05 13:25 +0000 [r135597]  Sean Bright <sean.bright@gmail.com>

	* main/cli.c: Use PATH_MAX for filenames

2008-08-04 20:15 +0000 [r135536]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample: fix a config sample typo

2008-08-04 17:07 +0000 [r135479-135482]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script

	* apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
	  Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
	  eliel (license 64)

2008-08-04 16:26 +0000 [r135473]  Russell Bryant <russell@digium.com>

	* configs/chan_dahdi.conf.sample: Add a minor clarification to the
	  documentation of mohinterpret and mohsuggest

2008-08-01 11:43 +0000 [r135055-135058]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_ices.c: make app_ices compile on OpenBSD.

	* channels/chan_skinny.c: fix some potential deadlocks in
	  chan_skinny (closes issue #13215) Reported by: qwell Patches:
	  2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
	  Tested by: mvanbaak

2008-07-31 22:18 +0000 [r134983]  Kevin P. Fleming <kpfleming@digium.com>

	* main/http.c: accomodate users who seem to lack a sense of humor
	  :-)

2008-07-31 21:53 +0000 [r134976]  Tilghman Lesher <tlesher@digium.com>

	* sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
	  callfiles and manager, to allow video calls to be set up from
	  callfiles and AMI. (closes issue #9531) Reported by: Geisj
	  Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
	  (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: Corydon76

2008-07-31 19:37 +0000 [r134915]  Russell Bryant <russell@digium.com>

	* apps/app_ices.c: Get app_ices working again (closes issue #12981)
	  Reported by: dlogan Patches:
	  20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
	  (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
	  bbryant (license 36) Tested by: bbryant

2008-07-31 19:23 +0000 [r134883]  Steve Murphy <murf@digium.com>

	* res/res_features.c: (closes issue #11849) Reported by: greyvoip
	  Tested by: murf OK, a few days of debugging, a bunch of
	  instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
	  5 solid notebook pages of notes later, I have made the small
	  tweek necc. to get the start time right on the second CDR when: A
	  Calls B B answ. A hits Xfer button on sip phone, A dials C and
	  hits the OK button, A hangs up C answers ringing phone B and C
	  converse B and/or C hangs up But does not harm the scenario
	  where: A Calls B B answ. B hits xfer button on sip phone, B dials
	  C and hits the OK button, B hangs up C answers ringing phone A
	  and C converse A and/or C hangs up The difference in start times
	  on the second CDR is because of a Masquerade on the B channel
	  when the xfer number is sent. It ends up replacing the CDR on the
	  B channel with a duplicate, which ends up getting tossed out. We
	  keep a pointer to the first CDR, and update *that* after the
	  bridge closes. But, only if the CDR has changed. I hope this
	  change is specific enough not to muck up any current CDR-based
	  apps. In my defence, I assert that the previous information was
	  wrong, and this change fixes it, and possibly other similar
	  scenarios. I wonder if I should be doing the same thing for the
	  channel, as I did for the peer, but I can't think of a scenario
	  this might affect. I leave it, then, as an exersize for the
	  users, to find the scenario where the chan's CDR changes and
	  loses the proper start time.

2008-07-31 16:45 +0000 [r134814]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.c: In case we have some processing threads
	  that free more frames than they allocate, do not let the frame
	  cache grow forever. (closes issue #13160) Reported by: tavius
	  Tested by: tavius, russell

2008-07-31 15:56 +0000 [r134758]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Add more timeout checks into app_queue,
	  specifically targeting areas where an unknown and potentially
	  long time has just elapsed. Also added a check to try_calling()
	  to return early if the timeout has elapsed instead of potentially
	  setting a negative timeout for the call (thus making it have *no*
	  timeout at all). (closes issue #13186) Reported by:
	  miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
	  (license 60) Tested by: miquel_cabrespina

2008-07-30 22:39 +0000 [r134704]  Tilghman Lesher <tlesher@digium.com>

	* main/sched.c, include/asterisk/sched.h: Oops, wrong define

2008-07-30 22:02 +0000 [r134652]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
	  issue #13051) Reported by: pj This patch substitutes commas in
	  the expr supplied to the if () statement, as in if ( expr ) ...
	  This solves both the bugs above, and makes the source symmetric
	  with switch statements, which were earlier reported to need this
	  sort of treatment. I tested this using the examples, both for the
	  compiler and at run time. Looks good.

2008-07-30 21:38 +0000 [r134649]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Qwell pointed out, via IRC, that the
	  previous fix only worked when explicitly set. When nothing is
	  set, and the option is implied, it breaks, because configure sets
	  the prefix to 'NONE'. Fixing.

2008-07-30 20:37 +0000 [r134540-134595]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
	  stack size to fix a crash when compiled with LOW_MEMORY. (closes
	  issue #13154) Reported by: edantie

	* funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
	  that used this function leaked an allocation the size of a
	  pointer. (reported by jmls in #asterisk-dev)

2008-07-30 19:47 +0000 [r134480-134536]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Only override sysconfdir and mandir when
	  prefix=/usr (closes issue #13093) Reported by: pabelanger

	* res/res_agi.c: launch_netscript sometimes returns -1, which fails
	  to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
	  set. (closes issue #13199) Reported by: smw1218

2008-07-30 18:31 +0000 [r134475]  Mark Michelson <mmichelson@digium.com>

	* main/app.c: Fix a spot where a function could return without
	  bringing a channel out of autoservice.

2008-07-30 15:29 +0000 [r134254-134352]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: use the proper method for building version.h

	* include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
	  channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
	  apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
	  build against the now-typedef-free dahdi/user.h

2008-07-29 15:54 +0000 [r134223]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Merging the imap_consistency branch. The
	  main aim of this branch was to make the IMAP code function in the
	  same manner as the ODBC code does, eliminating the need for so
	  many IMAP-specific code chunks. The focal point of all of this
	  work was to make the various macros (e.g. RETRIEVE, DISPOSE)
	  functionally equivalent. While doing the above work, I also fixed
	  a few bugs that I came across in my testing. Among these were 1.
	  Fixed message forwarding. This was completely broken when using
	  IMAP. 2. Fixed the inability to save new messages as old and vice
	  versa. 3. Fixed the "delete" options in voicemail.conf when using
	  IMAP storage. Even though a few bugs were fixed and the code is a
	  lot more consistent, the one thing that was *not* improved in
	  this branch was performance. The merge of this to trunk may not
	  come immediately due to the amount of work it will probably
	  involve. (closes issue #12764) Reported by: balsamcn

2008-07-28 21:50 +0000 [r134161]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Detect when sox fails to raise the volume,
	  because sox can't read the file. (closes issue #12939) Reported
	  by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: rickbradley

2008-07-26 15:31 +0000 [r133980]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
	  section to the docs in 1.4, as well, so that we can work on
	  having an accurate list for each version of Asterisk that is
	  supported

2008-07-25 18:00 +0000 [r133649-133709]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
	  #13161)

	* main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
	  some errant device states by making the devicestate API more
	  strict in terms of the device argument (only without the unique
	  identifier appended). (closes issue #12771) Reported by: davidw
	  Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
	  (license 14) Tested by: davidw, jvandal, murf

2008-07-25 15:00 +0000 [r133578]  Russell Bryant <russell@digium.com>

	* /, LICENSE: Merged revisions 133577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
	  | 2 lines Fix the IAX2 URI for calling Digium ........

2008-07-25 14:40 +0000 [r133572]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: We need to make sure to null-terminate the
	  "name" portion of SIP URI parameters so that there are no bogus
	  comparisons. Thanks to bbryant for pointing this out.

2008-07-24 21:17 +0000 [r133361-133488]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
	  issue #12707)

	* /: Blocked revisions 133360 via svnmerge ........ r133360 |
	  tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
	  This part was not correctly patched for AST-2008-010. ........

2008-07-23 21:49 +0000 [r133295]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c: inbandrelease is gone - it's now
	  inbanddisconnect

2008-07-23 21:05 +0000 [r133226-133237]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/stringfields.h, main/utils.c: revert an
	  optimization that broke ABI... thanks russell!

	* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
	  apps/app_dahdibarge.c, channels/chan_dahdi.c,
	  apps/app_dahdiras.c: make some more changes to the dahdi/zap
	  channel name support stuff to ensure allthe globals are 'const',
	  and clean up mmichelson's changes to app_chanspy to simplify the
	  code

2008-07-23 19:39 +0000 [r132974-133169]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
	  channels/chan_dahdi.c: As suggested by seanbright, the
	  PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
	  compile time, since dahdi_chan_name is determined at load time.
	  Also changed the next_unique_id_to_use to have the static
	  qualifier. Also added the dahdi_chan_name_len variable so that
	  strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
	  the suggestion.

	* apps/app_chanspy.c: Zap/pseudo is ten characters, but
	  DAHDI/pseudo is twelve. The strncmp call in next_channel should
	  account for this.

	* apps/app_chanspy.c: Update the "last" channel in next_channel in
	  app_chanspy so that the same pseudo channel isn't constantly
	  returned. related to issue #13124

	* channels/chan_dahdi.c: Small cleanup. Move the declaration of the
	  DAHDI_SPANINFO variable to the block where it is used. This
	  allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
	  Tzafrir for pointing this out on #asterisk-dev

	* channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
	  is not defined.

2008-07-23 15:52 +0000 [r132872-132942]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: ensure that after a channel is created, if
	  it happened to be in 'channel alarm' state, when that alarm
	  clears we won't generate a spurious 'alarm cleared' message
	  (closes issue #12160) Reported by: tzafrir

	* include/asterisk/stringfields.h, main/utils.c: minor optimization
	  for stringfields: when a field is being set to a larger value
	  than it currently contains and it happens to be the most recent
	  field allocated from the currentl pool, it is possible to 'grow'
	  it without having to waste the space it is currently using (or
	  potentially even allocate a new pool)

2008-07-23 11:37 +0000 [r132826]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: another Fix because of r119585, this
	  commit has broken high frequented BRI Ports, there was a
	  possibility that a channel, that was marked as in_use would be
	  reused later, the corresponding port could got stuck then. So it
	  is recommended to upgrade for chan_misdn users.

2008-07-22 22:14 +0000 [r132790]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
	  within Asterisk. Prior to this change, a spiraled INVITE would
	  cause a 482 Loop Detected to be sent to the caller. With this
	  change, if a potential loop is detected, the Request-URI is
	  inspected to see if it has changed from what was originally
	  received. If pedantic mode is on, then this inspection is fully
	  RFC 3261 compliant. If pedantic mode is not on, then a string
	  comparison is used to test the equality of the two R-URIs. This
	  has been tested by using OpenSER to rewrite the R-URI and send
	  the INVITE back to Asterisk. (closes issue #7403) Reported by:
	  stephen_dredge

2008-07-22 22:11 +0000 [r132784-132787]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/options.h, main/asterisk.c,
	  apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
	  apps/app_dahdiras.c: fix up namespace pollution for
	  dahdi_chan_mode enum correct registration of AMI actions in
	  chan_dahdi; in zap-only mode, only register the Zap flavors of
	  the actions (and use Zap prefixes for headers and acks), but in
	  dahdi+zap mode, register both Zap and DAHDI flavors of actions

	* Makefile.rules: add rules to create preprocessor output... useful
	  for debugging macros

2008-07-22 21:19 +0000 [r132713]  Tilghman Lesher <tlesher@digium.com>

	* configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
	  revisions 132711 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
	  | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........

2008-07-22 21:17 +0000 [r132704-132712]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: ensure that if any alarms exist at channel
	  creation time, they are handled identically to if they occurred
	  later, so that later alarm clearing will work properly and 'make
	  sense' (closes issue #12160) Reported by: tzafrir

	* configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
	  able to print a 'pretty' description of what it is doing

2008-07-22 20:10 +0000 [r132645]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
	  common question on the #asterisk iRC channel and on mailing lists
	  seems to be in regards to an error message when retransmit fails.
	  This is frequently misunderstood as a failure of Asterisk, not a
	  failure of the network to reach the other party. This document
	  tries to assist the Asterisk user in sorting out these issues by
	  explaining the logic and pointing at some possible causes.
	  Hopefully, we will get other questions now :-)

2008-07-22 19:57 +0000 [r132571-132642]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: correct wording in comment

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: use renamed
	  libpri API call for controlling this feature (was improperly
	  named before)

	* channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
	  on BRI spans, and don't attempt to use channel 24 as a D-channel
	  on spans of unexpected sizes

2008-07-21 20:51 +0000 [r132506-132507]  Brett Bryant <bbryant@digium.com>

	* apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
	  properly when it isn't supported on a channel (yet _another_
	  useful patch by eliel). (issue #13081) Reported by: eliel
	  Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
	  by: eliel

	* channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
	  not being removed when a call was rejected (from the calling box,
	  not the box that denied the registration). Related to revisions
	  132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
	  tested 1.4 with a backport from those revisions, so I didn't see
	  this problem (oops).

2008-07-19 16:45 +0000 [r132311]  Kevin P. Fleming <kpfleming@digium.com>

	* LICENSE: grant a license exception to allow distribution of
	  Asterisk binaries that use the UW IMAP Toolkit (which is licensed
	  under a non-GPL-compatible license)

2008-07-18 19:06 +0000 [r131970-132112]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
	  Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
	  uploaded by CharlesWang (license 444)

	* main/config.c: Textual clarification (closes issue #13106)
	  Reported by: flefoll Patches:
	  config.c.br14.120173.patch-unknown-directive uploaded by flefoll
	  (license 244)

	* include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
	  the destroy, to allow a scheduled job to continue, if it's
	  waiting on the mutex which the destroy thread has.

	* main/sched.c: Oops

	* main/sched.c, include/asterisk/sched.h: Preserve ABI
	  compatibility with last change

	* main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
	  Make the ast_assert call within ast_sched_del report something
	  useful.

2008-07-18 16:15 +0000 [r131921]  Kevin P. Fleming <kpfleming@digium.com>

	* main/dlfcn.c (removed), main/loader.c, main/Makefile,
	  include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
	  compatibility stuff, because no platforms that Asterisk currently
	  runs on it use it, and it doesn't build anyway

2008-07-18 15:34 +0000 [r131915]  Brett Bryant <bbryant@digium.com>

	* res/res_features.c: Fix a bug in blind transfers where the
	  BLINDTRANSFER variable isn't always set to the other end of the
	  blind transfer. (closes issue #12586)

2008-07-17 20:35 +0000 [r131790]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
	  as it appears that it was in error. This should fix talk call
	  progress on analog lines. (closes issue #12178) Reported by:
	  michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
	  Corydon76 (license 14)

2008-07-16 22:17 +0000 [r131491]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
	  peers to register with case-insensitive names (user_cmp_cb and
	  peer_cmp_cb are now both case-sensitive). (closes issue #13091)

2008-07-16 21:46 +0000 [r131480]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Apparently, in certain cases, a callno is
	  already destroyed when iax2_destroy is called.

2008-07-16 20:47 +0000 [r131421]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Always ensure that the channel's tech_pvt
	  reference is NULL after calling the destroy callback. (closes
	  issue #13060) Reported by: jpgrayson Patches:
	  chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
	  492)

2008-07-16 20:23 +0000 [r131299-131369]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Move the init_queue call back to where it used
	  to be (changed Sept 12 last year). It was moved then to prevent a
	  memory leak. Since then, the same memory leak recurred and was
	  fixed in a better way. Now it has been found that the placement
	  of this init_queue call can cause problems if a realtime queue
	  has values changed to an empty string. The problem is that the
	  default value for that queue parameter would not be set. (closes
	  issue #13084) Reported by: elbriga

	* apps/app_queue.c: Apparently, "thread safety" is important,
	  whatever that means. :P (Thanks Russell!)

	* apps/app_queue.c: Make absolutely certain that the transfer
	  datastore is removed from the calling channel once the caller is
	  finished in the queue. This could have weird con- sequences when
	  dialing local queue members when multiple transfers occur on a
	  single call. Also fixed a memory leak that would occur when an
	  attended transfer occurred from a queue member. (closes issue
	  #13047) Reported by: festr

2008-07-16 17:53 +0000 [r131242]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
	  problem was that, esoteric as it is, because the hangerupper
	  context immediately preceded the std-priv-extent macro, that the
	  checking code accidentally would fall from traversing hangerupper
	  into the std-priv-exten macro, where it would hit the hangerupper
	  in the 'includes', and proceed into an infinite recursion. A
	  small fix to traverse into the statements of the context instead
	  of the context solves this issue. I also added some commented out
	  printfs for debug, which were pretty handy in the face of a dorky
	  gdb. This was a problem around since the package was first
	  written; but evidently pretty rare in turning up in the field.

2008-07-15 17:47 +0000 [r131012]  Michiel van Baak <michiel@vanbaak.info>

	* main/cdr.c: remove 4 lines of redundant code. (closes issue
	  #13080) Reported by: gknispel_proformatique Patches:
	  trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)

2008-07-15 17:19 +0000 [r130889-130959]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, channels/chan_sip.c: astman_send_error does not
	  need a newline appended -- the API takes care of that for us.
	  (closes issue #13068) Reported by: gknispel_proformatique
	  Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
	  (license 261) asterisk_trunk_astman_send.patch uploaded by
	  gknispel (license 261)

	* channels/chan_iax2.c: Override the callerid in all cases when the
	  callerid is set in the user, not just when a remote callerid is
	  set. Also, if not set in the user, allow the remote CallerID to
	  pass through. (closes issue #12875) Reported by: dimas Patches:
	  20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)

2008-07-14 17:50 +0000 [r130792]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
	  app_dial to be sure there are no audiohooks present on the
	  channels involved. This fixed a one-way audio situation I had in
	  my test setup. I couldn't find any open issues that suggested
	  one-way audio with regards to mixmonitor (or other audiohook)
	  usage, though.

2008-07-14 17:10 +0000 [r130735]  Michiel van Baak <michiel@vanbaak.info>

	* main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
	  dnsmgr is not enabled. Previously this command would
	  automagically appear and disappear. This was confusing. (closes
	  issue #12796) Reported by: chappell Patches:
	  dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
	  russell, chappell, mvanbaak

2008-07-14 10:38 +0000 [r130634]  Russell Bryant <russell@digium.com>

	* main/audiohook.c: Bump up the debug level for a message.

2008-07-13 22:48 +0000 [r130573]  Michiel van Baak <michiel@vanbaak.info>

	* main/manager.c: fix memory leak when originate from manager
	  cannot create a thread (closes issue #13069) Reported by:
	  gknispel_proformatique Patches:
	  asterisk_trunk_action_originate.patch uploaded by gknispel
	  (license 261) Tested by: gknispel_proformatique, mvanbaak

2008-07-13 17:56 +0000 [r130514]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Reverting 2 changesets, as it breaks
	  incoming IAX2 calls (Related to issue #12963) Reported by:
	  mvanbaak

2008-07-12 10:25 +0000 [r130373]  Michiel van Baak <michiel@vanbaak.info>

	* pbx/pbx_ael.c: in 1.4 the functions still have | as argument
	  seperator. This commit fixes the use of RAND in the ael random
	  function. (closes issue #13061) Reported by: danpwi

2008-07-11 22:23 +0000 [r130298-130317]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: forcibly remove the modules that are changing names

	* include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
	  Makefile, main/channel.c, apps/app_dahdibarge.c,
	  channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
	  apps/app_dahdiras.c, main/file.c,
	  contrib/utils/zones2indications.c, include/asterisk/channel.h,
	  channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
	  work, with lots more to come... this tree is not yet ready for
	  users to be easily upgrading or switching, but it needs to be :-)

2008-07-11 20:03 +0000 [r130173-130236]  Mark Michelson <mmichelson@digium.com>

	* main/audiohook.c: Remove redundant logic

	* main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
	  this change has not been proven to fix any specific issue, it is
	  incorrect and could cause unforeseen problems.

2008-07-11 18:51 +0000 [r130102-130169]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Ensure that a destination callno of 0 will
	  not match for frames that do not start a dialog (new, lagrq, and
	  ping). (closes issue #12963) Reported by: russellb Patches:
	  chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)

	* channels/chan_agent.c: Pass the devicestate from an underlying
	  channel up through the Agent channel. This should make the Agent
	  always report the correct device state, even when the underlying
	  channel is used for other purposes. (closes issue #12773)
	  Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
	  by Corydon76 (license 14) Tested by: davidw

2008-07-11 16:08 +0000 [r130039-130042]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/configuration.txt, configs/extensions.conf.sample,
	  configs/sla.conf.sample, configs/zapata.conf.sample (removed),
	  contrib/scripts/autosupport, README,
	  configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
	  include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
	  configs/extensions.ael.sample, configs/smdi.conf.sample: new
	  installations should be using DAHDI instead of Zaptel, so the
	  sample config file is now chan_dahdi.conf instead of zapata.conf
	  also, convert remaining references to zapata.conf in various
	  places

	* configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: add support for a
	  configuration parameter for 'inband audio during RELEASE', which
	  is currently mandatory in libpri-1.4.4 but will become
	  configurable in libpri-1.4.5 later today (related to issue
	  #13042)

2008-07-11 14:18 +0000 [r129970]  Russell Bryant <russell@digium.com>

	* include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
	  ...

2008-07-11 14:14 +0000 [r129907-129967]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astmm.c: simplify calculation

	* main/astmm.c: fix a flaw found while experimenting with structure
	  alignment and padding; low-fence checking would not work properly
	  on 64-bit platforms, because the compiler was putting 4 bytes of
	  padding between the fence field and the allocation memory block
	  added a very obvious runtime warning if this condition reoccurs,
	  so the developer who broke it can be chastised into fixing it :-)

	* sounds/Makefile: don't attempt to set user/group ownership of
	  extracted sound files (reported on asterisk-users) (closes issue
	  #13059)

2008-07-10 21:57 +0000 [r129741-129803]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Correctly deal with duplicate NEW frames
	  (due to retransmission). Also, fixup the destination call number
	  matching to be more strict and reliable. (closes issue #12963)
	  Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
	  uploaded by jpgrayson (license 492) Tested by: jpgrayson,
	  Corydon76

	* res/res_config_odbc.c: Oops

2008-07-10 16:03 +0000 [r129567]  Russell Bryant <russell@digium.com>

	* sample.call: Note that pbx_spool.so is the module used for call
	  files (inspired by a question in #asterisk)

2008-07-10 13:57 +0000 [r129505]  Sean Bright <sean.bright@gmail.com>

	* main/editline: Update svn:ignore

2008-07-09 19:32 +0000 [r129436]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c: Fix a problem where inbound rfc2833 audio would be
	  sent to the core instead of being P2P bridged. When the core
	  regenerated the rfc2833 packet for the outbound leg, the SSRC
	  would be different than the RTP audio on the call leg causing
	  DTMF detection issues on the far end. (closes issue #12955)
	  Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
	  tsearle (license 373) Tested by: tonyredstone

2008-07-09 13:41 +0000 [r129343]  Sean Bright <sean.bright@gmail.com>

	* main/editline/makelist (removed), main/editline/makelist.in
	  (added), main/editline/configure, main/editline/Makefile.in,
	  main/editline/configure.in: Look for the system installed awk
	  instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
	  #asterisk-dev.

2008-07-08 21:31 +0000 [r129158-129208]  Mark Michelson <mmichelson@digium.com>

	* doc/imapstorage.txt: Update documentation to have the correct
	  option name

	* apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
	  timeouts to IMAP voicemail in 1.4 since it should solve bugs
	  people are experiencing. Specifically, there are times where
	  communication with the IMAP server causes system calls to block
	  forever. If this should happen when querying the mailbox so that
	  chan_sip's do_monitor thread can send MWI to a phone, it means
	  that SIP calls cannot be processed any more. The timeout options
	  are outlined in doc/imapstorage.txt. Defaults for the timeouts
	  are sixty seconds. (closes issue #12987) Reported by: mthomasslo

2008-07-08 20:27 +0000 [r129047-129149]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
	  Cause SIP to return a 480 instead of a 404 when a sip peer
	  exists, but is not registered. (closes issue #12885) Reported by:
	  ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
	  (license 14) Tested by: ibc

	* channels/chan_iax2.c: Timestamp decoding for video mini-frames is
	  bogus, because the timestamp only includes 15 bits, unlike voice
	  frames, which contain a 16-bit timestamp. (closes issue #13013)
	  Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
	  uploaded by jpgrayson (license 492)

2008-07-08 09:52 +0000 [r128912-128950]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't hangup the call if we can't resolve
	  the Contact if there's a proxy route set for the call. ---- This
	  comment was added a while ago and today it hit me badly. /* OEJ:
	  Possible issue that may need a check: If we have a proxy route
	  between us and the device, should we care about resolving the
	  contact or should we just send it? */

	* channels/chan_sip.c: Fix issues where repeated messages where
	  ignored, but retransmitted reliably instead of unreliably.
	  Reported by: johan Patches: 12746.txt uploaded by oej (license
	  306) Tested by: johan (issue #12746)

2008-07-08 00:01 +0000 [r128812-128856]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Check for non-NULL before stripping
	  characters. (closes issue #12954) Reported by: bfsworks Patches:
	  20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: deti

	* apps/app_voicemail.c: Stop using deprecated method, as requested
	  by Kevin.

2008-07-07 22:41 +0000 [r128795]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix handling of when a pvt disappears.
	  Properly return the pvt locked and don't hold the pvt lock while
	  destroying the ast_channel. (closes issue #13014) Reported by:
	  jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
	  jpgrayson (license 492)

2008-07-07 20:47 +0000 [r128737]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_iax2.c: Remove spurious trailing whitespace from
	  log messages and fix a spelling error in a log message. (closes
	  issue #13017) Reported by: jpgrayson Patches:
	  chan_iax2_space_after_newline.patch uploaded by jpgrayson
	  (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
	  (license 492)

2008-07-07 17:02 +0000 [r128639]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_iax2.c: By using the iaxdynamicthreadcount to
	  identify a thread, it was possible for thread identifiers to be
	  duplicated. By using a globally-unique monotonically- increasing
	  integer, this is now avoided. (closes issue #13009) Reported by:
	  jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
	  jpgrayson (license 492)

2008-07-07 16:51 +0000 [r128637]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: use tzafrir's patch to fix this problem
	  properly... i made the previous set of changes without thoroughly
	  testing them, doh! (closes issue #12911) Reported by: tzafrir
	  Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
	  (license 46) Tested by: tzafrir

2008-07-04 16:11 +0000 [r127973-128029]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_config.c: Move the free down one

	* main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
	  'dialplan remove extension' logic, so that it a) works with
	  cidmatch, and b) completes contexts correctly when the extension
	  is ambiguous. (closes issue #12980) Reported by: licedey Patches:
	  20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76

2008-07-03 22:20 +0000 [r127754-127895]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/Makefile: remove this, it has been moved to the main
	  Makefile

	* Makefile, main/editline/np/vis.c: a couple of small
	  Solaris-related fixes (closes issue #11885) Reported by: snuffy,
	  asgaroth

	* configure, main/Makefile, configure.ac, acinclude.m4: ensure that
	  DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
	  needed improve AST_EXT_LIB_CHECK to accept (and remember)
	  additional CFLAGS data like it does in trunk already (closes
	  issue #12911) Reported by: tzafrir

2008-07-03 00:16 +0000 [r127663]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
	  channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
	  The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
	  by: murf Tested by: murf, deeperror (closes issue #12907)
	  Reported by: falves11 Tested by: murf, falves11 (closes issue
	  #11849) Reported by: greyvoip As to 11849, I think these changes
	  fix the core problems brought up in that bug, but perhaps not the
	  more global problems created by the limitations of CDR's
	  themselves not being oriented around transfers. Reopen if necc,
	  but bug reports are not the best medium for enhancement
	  discussions. We need to start a second-generation CDR
	  standardization effort to cover transfers. (closes issue #11093)
	  Reported by: rossbeer Tested by: greyvoip, murf

2008-07-02 20:47 +0000 [r127560]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fix thread-safety of some of the
	  pbx_builtin_getvar_helper calls

2008-07-02 19:47 +0000 [r127501]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Merged revisions 127466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
	  tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
	  Solaris fix (closes issue #12949) Reported by: snuffy Patches:
	  bug_12949.diff uploaded by snuffy (license 35) ........

2008-07-01 23:36 +0000 [r127244]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Add error message to failed open(2) calls
	  inside the copy() function of app_voicemail. This idea came as
	  part of my work in helping to resolve issue #12764.

2008-07-01 20:25 +0000 [r126999-127133]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
	  slow search for matching callno in chan_iax2 (but allow it to be
	  reenabled for debugging)

	* channels/chan_iax2.c: Oops

	* channels/chan_iax2.c: Change around how we schedule pings and
	  lagrqs, and fix a reason why the jobs were not getting properly
	  cancelled. (closes issue #12903) Reported by: stevedavies
	  Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
	  (license 14) Tested by: stevedavies

	* channels/chan_iax2.c: Suppress annoying warning by finding the
	  remaining cases where the callno is not in the hash.

2008-07-01 14:59 +0000 [r126735-126902]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Use domain part of SIP uri in register=
	  configuration as fromdomain. Reported by: one47 Patches:
	  sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
	  issue #12474)

	* channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
	  by oej and IgorG. Reported by: IgorG Patches:
	  bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
	  IgorG, oej (closes issue #12299)

	* configs/sip.conf.sample: Clear up documentation on "domain="
	  setting in sip.conf Reported by: davidw (closes issue #12413)

	* channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
	  requests (to confirm that the dialog exist). Don't bother
	  checking the request URI. (closes issue #11264) Reported by: ibc

	* channels/chan_sip.c: Fix bad XML for hold notification. Reported
	  by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
	  (closes issue #12942)

2008-06-30 23:11 +0000 [r126680]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Load the proper channel configuration file
	  based on which driver was detected.

2008-06-30 22:30 +0000 [r126674]  Tilghman Lesher <tlesher@digium.com>

	* configs/zapata.conf.sample: Add note about other names for
	  EuroISDN

2008-06-30 16:05 +0000 [r126573]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
	  version of the recently added DEADLOCK_AVOIDANCE() macro. This
	  caused the lock to not actually be released, and as a result, not
	  avoid deadlocks at all. This resolves the issues reported in the
	  last while about Asterisk locking up all over the place (and most
	  commonly, in chan_iax2). (closes issue #12927) (closes issue
	  #12940) (closes issue #12925) (potentially closes others ...)

2008-06-30 12:50 +0000 [r126516]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Send all responses to an INVITE reliably, so
	  that we retransmit if we don't get an ACK and also fail if we
	  don't get the very same precious ACK. Based on patch by tsearle,
	  with my own additions. (closes issue #12951) Reported by: tsearle
	  Patches: busy_retransmit.patch uploaded by tsearle (license 373)

2008-06-29 18:05 +0000 [r126395]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx/Makefile: ignore warnings for prototypes in GTK headers

2008-06-27 22:01 +0000 [r125740-126056]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
	  to re-register. (closes issue #12863) Reported by: ricvil

	* include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
	  to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
	  HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
	  tonezone_compat.diff uploaded by opticron (license 267)

	* main/utils.c, include/asterisk/lock.h: In this debugging
	  function, copy to a buffer instead of using potentially unsafe
	  pointers.

	* channels/chan_local.c: Add proper deadlock avoidance. (closes
	  issue #12914) Reported by: ozan Patches:
	  20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: ozan

2008-06-26 23:03 +0000 [r125587]  Jason Parker <jparker@digium.com>

	* main/utils.c: Make sure to unlock the lock_info lock (huh?).
	  Possible deadlock?

2008-06-26 22:52 +0000 [r125476-125585]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Add the interface of a queue member to the
	  output of the "queue show" command so that it can easily be
	  associated with a queue member's name. This helps so that the
	  appropriate queue member can be removed or paused since the
	  interface is required, not the member's name. (closes issue
	  #12783) Reported by: davevg Patches: app_queue.diff uploaded by
	  davevg (license 209) with small mod from me

	* apps/app_queue.c: Backport of attended transfer queue_log patch
	  from trunk. This patch allows for attended transfers to be logged
	  in the queue_log the same way that blind transfers have always
	  been. It was decided by popular opinion on the asterisk-dev
	  mailing list that this should be backported to 1.4. Thanks to
	  everyone who gave an opinion.

	* apps/app_queue.c: Prior to this patch, the "queue show" command
	  used cached information for realtime queues instead of giving
	  up-to-date info. Now realtime is queried for the latest and
	  greatest in queue info. (closes issue #12858) Reported by: bcnit
	  Patches: queue_show.patch uploaded by putnopvut (license 60)

2008-06-26 16:32 +0000 [r125384]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add support for peer realm based auth (a few
	  missing lines, the rest is well documented but never worked)

2008-06-26 15:30 +0000 [r125327]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c: ensure that (whenever possible) if we
	  generate a log message because an ioctl() call to DAHDI/Zaptel
	  failed, that we include the reason it failed by including the
	  stringified error number (issue AST-80)

2008-06-26 11:01 +0000 [r125218-125276]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c: Check for rtcp structure before trying to delete
	  schedule. (closes issue #12872) Reported by: destiny6628 Patches:
	  20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: destiny6628

	* configs/agents.conf.sample: Document ackcall=always. (closes
	  issue #12852) Reported by: davidw

2008-06-25 22:21 +0000 [r125132]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_rpt.c, include/asterisk/dahdi_compat.h,
	  channels/chan_dahdi.c, configure,
	  include/asterisk/tonezone_compat.h (added), configure.ac: allow
	  tonezone to live in a different place than DAHDI/Zaptel, since
	  dahdi-tools and dahdi-linux are now separate packages and can be
	  installed in different places don't include tonezone.h in
	  dahdi_compat.h, because only a couple of modules need it get
	  app_rpt building again after the DAHDI changes (closes issue
	  #12911) Reported by: tzafrir

2008-06-25 00:46 +0000 [r124908-124965]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Pvt deadlock causes some channels to get
	  stuck in Reserved status. (closes issue #12621) Reported by:
	  fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: fabianoheringer

	* apps/app_voicemail.c: Occasionally control characters find their
	  way into CallerID. These need to be stripped prior to placing
	  CallerID in the headers of an email. (closes issue #12759)
	  Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: RobH

	* channels/chan_sip.c: Don't access the pvt structure if unable to
	  acquire the lock. (closes issue #12162) Reported by: norman
	  Patches: 12162-lockfail.diff uploaded by qwell (license 4)

2008-06-23 21:22 +0000 [r124743]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: emit a warning if the old IAX2 call
	  searching code finds a call when the new code did not... so that
	  we can get rid of the old code in 2-3 months

2008-06-22 02:54 +0000 [r124540]  Steve Murphy <murf@digium.com>

	* apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
	  Sorry, my testing did not contain the simple case of forkCDR(v),
	  I am much embarrassed to admit. If I had, I would have more
	  solidly initialized the opts element for varset.

2008-06-20 23:12 +0000 [r124395-124450]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_rpt.c: usleep with a value over 1,000,000 is
	  nonportable. Changing to use sleep() instead. (closes issue
	  #12814) Reported by: pputman Patches: app_rtp_sleep.patch
	  uploaded by pputman (license 81)

	* main/app.c: If the last character in a string to be parsed is the
	  delimiter, then we should count that final empty string as an
	  additional argument.

2008-06-20 21:14 +0000 [r124372]  Jeff Gehlbach <jeffg@opennms.org>

	* doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
	  digium-mib.txt and asterisk-mib.txt to placate smilint - bug
	  12905

2008-06-20 20:16 +0000 [r124182-124315]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: When using a Local channel, started by a
	  call file, with a destination of an AGI script, the AGI script
	  does not always get notified of a hangup if the underlying
	  channel hangs up early. (closes issue #11833) Reported by: IgorG
	  Patches: local_hangup-v1.diff uploaded by IgorG (license 20)

	* channels/chan_dahdi.c: It's possible for a hangup to be received,
	  even just after the initial cid spill. (closes issue #12453)
	  Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
	  uploaded by Corydon76 (license 14)

2008-06-19 20:28 +0000 [r124112]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix IMAP forwarding so that messages are
	  sent to the proper mailbox. (closes issue #12897) Reported by:
	  jaroth Patches: destination_forward.patch uploaded by jaroth
	  (license 50)

2008-06-19 19:55 +0000 [r124066]  Brett Bryant <bbryant@digium.com>

	* main/utils.c: Merge revision 124064 from trunk. Add errors that
	  report any locks held by threads when they are being closed.

2008-06-19 16:58 +0000 [r123710-123930]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Change informative messages to use the _multiple
	  variant when multiple formats are possible. (Closes issue #12848)
	  Reported by klaus3000

	* build_tools/strip_nonapi: Only process 40 arguments (20 files) at
	  once with xargs, because some older shells may force xargs to
	  separate on an odd boundary. (Closes issue #12883) Reported by
	  Nik Soggia

	* configs/sip.conf.sample: Correct description of notifyringing
	  option. (Closes issue #12890) Reported by gminet

	* main/asterisk.c: The RDTSC instruction was introduced on the
	  Pentium line of microprocessors, and is not compatible with
	  certain 586 clones, like Cyrix. Hence, asking for i386
	  compatibility was always incorrect. See
	  http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
	  by tecnoxarxa

	* main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
	  support for saying numbers in Hebrew. (closes issue #11662)
	  Reported by: greenfieldtech Patches: say.c.patch-12042008
	  uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
	  uploaded by greenfieldtech (with signficant changes to the
	  spreadsheet by me)

	* pbx/pbx_spool.c: Set the variables top-down, so that if a script
	  sets a variable more than once, the last one will take
	  precedence. (closes issue #12673) Reported by: phber Patches:
	  20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)

2008-06-17 20:26 +0000 [r123485]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Make chan_sip build under dev mode with
	  compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this

2008-06-17 18:56 +0000 [r123391]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Fix 3 more places where not locking the
	  structure could cause the wrong lock to be unlocked. (Closes
	  issue #12795)

2008-06-17 18:09 +0000 [r123274-123333]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
	  between the Cseq number and SIP request method in the Cseq:
	  header. Asterisk did not handle this properly, but with this
	  patch, all is well. (closes issue #12834) Reported by: tobias_e
	  Patches: 12834.patch uploaded by putnopvut (license 60) Tested
	  by: tobias_e

	* apps/app_queue.c: davidw pointed out that the holdtime
	  calculation used by app_queue does not use "boxcar" filtering as
	  the comments say. The term "boxcar" means that the number of
	  samples used to calculate stays constant, with new samples
	  replacing the oldest ones. The queue holdtime calculation uses
	  all holdtime samples collected since the queue was loaded, so the
	  comment has been changed to be accurate. (closes issue #12781)
	  Reported by: davidw

2008-06-17 15:48 +0000 [r123271]  Russell Bryant <russell@digium.com>

	* main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
	  by seanbright. When a container got destroyed, the underlying
	  bucket list entry for each object that was in the container at
	  that time did not get free'd.

2008-06-16 19:50 +0000 [r123110-123113]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c, channels/chan_dahdi.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
	  other channel drivers

	* channels/chan_sip.c: People expect that if "hasvoicemail" is set
	  in users.conf, even if "mailbox" isn't set, that SIP will detect
	  a mailbox. (closes issue #12855) Reported by: PLL Patches:
	  20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: PLL

2008-06-16 12:31 +0000 [r122869-122919]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only compare the first 15 characters so that
	  even if the charset is specified we still accept it as SDP.
	  (closes issue #12803) Reported by: lanzaandrea Patches:
	  chan_sip.c.diff uploaded by lanzaandrea (license 496)

	* channels/chan_sip.c: Don't send a BYE on a dialog that is already
	  gone during a REFER. (closes issue #12865) Reported by: flefoll
	  Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
	  flefoll (license 244)

2008-06-13 21:44 +0000 [r122713]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c: Short circuit the loop in autoservice_run if
	  there are no channels to poll. If we continued, then the result
	  would be calling poll() with a NULL pollfd array. While this is
	  fine with POSIX's poll(2) system call, those who use Asterisk's
	  internal poll mechanism (Darwin systems) would have a failed
	  assertion occur when poll is called. (related to issue #10342)

2008-06-13 18:57 +0000 [r122663]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
	  dahdi compatability header from assuming either dahdi or zaptel
	  is installed (may not have either)

2008-06-13 17:45 +0000 [r122617]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c: Remove extra option from previous solution
	  attempt

2008-06-13 17:36 +0000 [r122613]  Jeff Peeler <jpeeler@digium.com>

	* configure, configure.ac: (closes issue #12846) Reported by:
	  Netview Tested by: jpeeler Use correct location to search for
	  tonezone.

2008-06-13 16:29 +0000 [r122589]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, res/res_features.c: This should fix the behavior
	  of the 'T' dial feature being passed incorrectly to the
	  transferee when builtin_atxfers are used. Also, doing a
	  builtin_atxfer to parking was broken and is fixed here as well.
	  (closes issue #11898) Reported by: sergee Tested by: otherwiseguy

2008-06-12 19:08 +0000 [r122314]  Jeff Peeler <jpeeler@digium.com>

	* main/indications.c, include/asterisk/dahdi_compat.h (added),
	  main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
	  configure, apps/app_zapscan.c (removed), apps/app_zapras.c
	  (removed), main/app.c, include/asterisk/options.h,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
	  channels/chan_zap.c (removed), apps/app_page.c,
	  include/asterisk/indications.h, apps/app_dahdiras.c (added),
	  configure.ac, apps/app_disa.c, include/asterisk/channel.h,
	  apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
	  (removed), channels/chan_misdn.c, apps/app_flash.c,
	  build_tools/menuselect-deps.in, funcs/func_channel.c,
	  main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
	  codecs/codec_dahdi.c (added), res/res_indications.c,
	  pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
	  main/asterisk.c, apps/app_dahdibarge.c (added),
	  apps/app_meetme.c, include/asterisk/autoconfig.h.in,
	  apps/app_dahdiscan.c (added), acinclude.m4,
	  res/res_musiconhold.c, codecs/codec_zap.c (removed),
	  channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
	  usage favored, but all Zap stuff should continue working. Release
	  announcement to follow.

2008-06-12 18:50 +0000 [r122311]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Properly play a holdtime message if the
	  announce-holdtime option is set to "once." (closes issue #12842)
	  Reported by: ramonpeek Patches: patch001.diff uploaded by
	  ramonpeek (license 266)

2008-06-12 18:22 +0000 [r122259]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some race conditions that cause
	  ast_assert() to report that chan_iax2 tried to remove an entry
	  that wasn't in the scheduler

2008-06-12 15:46 +0000 [r122208]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_parkandannounce.c, res/res_features.c: (closes issue
	  #12193) Reported by: davidw Patch by: Corydon76, modified by me
	  to work properly with ParkAndAnnounce app

2008-06-12 15:18 +0000 [r122130-122137]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: Flipflop the sections for two options, since
	  the section for 'X' (exit context) may otherwise absorb
	  keypresses meant for 's' (admin/user menu). (closes issue #12836)
	  Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: blitzrage

	* main/channel.c: Occasionally, the alertpipe loses its nonblocking
	  status, so detect and correct that situation before it causes a
	  deadlock. (Reported and tested by ctooley via #asterisk-dev)

2008-06-12 14:51 +0000 [r122127]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
	  code added to ast_cdr_busy, _failed, and _noanswer was redundant.
	  Didn't spot it until I was resolving conflicts in trunk. Ugh.
	  Redundant code removed. It wasn't harmful. Just dumb.

2008-06-12  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.21 released.

2008-06-06  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.21-rc2 released.

2008-06-05 18:03 +0000 [r120731-120735]  Russell Bryant <russell@digium.com>

	* UPGRADE-1.2.txt: fix filename

	* UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
	  from Asterisk 1.2, for handy reference.

2008-06-05 16:56 +0000 [r120675]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Ignore appended resource when comparing JIDs.

2008-06-05 16:38 +0000 [r120671]  Russell Bryant <russell@digium.com>

	* doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
	  forwarding station isn't very useful for most people, so pull in
	  the changes that allow searching for SMDI messages based on other
	  components of the SMDI message. Also, update the SMDI
	  documentation.

2008-06-04 22:05 +0000 [r120513]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make sure that the string we set will survive
	  the unref of the queue member. Thanks to Russell, who pointed
	  this out.

2008-06-04 18:35 +0000 [r120425]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: If we fail to setup the PRI request channel,
	  don't continue, exit with an error. (closes issue #11989)
	  Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
	  uploaded by Corydon76 (license 14)

2008-06-04 16:26 +0000 [r120371]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: Make the "dialplan remove include" CLI command
	  actually work. Also, tweak some formatting, and make the success
	  message a little bit more clear. (closes AST-52)

2008-06-04 14:11 +0000 [r120285]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Tab completion when removing a member should
	  give the member's interface, not the name, since the interface is
	  what is expected for the command. (closes issue #12783) Reported
	  by: davevg

2008-06-04 13:31 +0000 [r120282]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
	  for when the dialplan is done reloading. (closes issue #12716)
	  Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
	  chappell (license 8)

2008-06-03 22:41 +0000 [r120226]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_loopback.c: Due to incorrect use of the
	  AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
	  any translation on the extension number before searching for it
	  in the target context. (closes issue #12473) Reported by:
	  chappell Patches: pbx_loopback.c.diff uploaded by chappell
	  (license 8)

2008-06-03 22:15 +0000 [r120173]  Jeff Peeler <jpeeler@digium.com>

	* main/config.c: (closes issue #11594) Reported by: yem Tested by:
	  yem This change decreases the buffer size allocated on the stack
	  substantially in config_text_file_load when LOW_MEMORY is turned
	  on. This change combined with the fix from revision 117462
	  (making mkintf not copy the zt_chan_conf structure) was enough to
	  prevent the crash.

2008-06-03 21:34 +0000 [r120168]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another place where peer->callno could
	  change at a very bad time, and also fix a place where a peer was
	  used after the reference was released. (inspired by rev 120001)

2008-06-03  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.21-rc1 released.

2008-06-03 18:23 +0000 [r120001-120061]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: When listing the manager users, managers in
	  users.conf are not shown, even though they are allowed to
	  connect. (closes issue #12594) Reported by: bkruse Patches:
	  12594-managerusers-2.diff uploaded by qwell (license 4) Tested
	  by: bkruse

	* channels/chan_iax2.c: Save the callno when we're poking, because
	  our peer structure could change during deadlock avoidance (and
	  thus we unlock the wrong callno, causing a cascade failure).
	  (closes issue #12717) Reported by: gewfie Patches:
	  20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: gewfie

2008-06-03 15:26 +0000 [r119929-119966]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
	  Hadn't updated this for the changes I made to preserve ${EXTEN}
	  in switches, which affected several tests because it adds extra
	  priorities, and at least one needed to be updated because of the
	  removal of the empty extension warning message.

	* pbx/pbx_ael.c: as per
	  http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
	  which is a message from Philipp Kempgen, requesting that the
	  WARNING that an extension is empty be reduced to a NOTICE or
	  less, as empty extensions are syntactically possible, and no big
	  deal. With which I agree, and have removed that WARNING message
	  entirely. I think it is not necessary to see this message. It
	  didn't state that a NoOp() was inserted automatically on your
	  behalf, and really, as users, who cares? Why freak out dialplan
	  writers with unnecessary warnings? The details of the
	  machinations a compiler goes thru to produce working assembly
	  code is of little interest to most programmers-- we will follow
	  the unix principal of doing our work silently.

2008-06-03 14:46 +0000 [r119926]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Treat ECONNREFUSED as an error that will
	  stop further retransmissions. (issue #AST-58, patch from
	  Switchvox)

2008-06-02 20:08 +0000 [r119742-119838]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Revert a change made for issue #12479. This
	  change caused a regression such that a dial string such as
	  (IAX2/foo) did not automatically fall back to dialing the 's'
	  extension anymore. (closes issue #12770) Reported by: dagmoller

	* main/manager.c: Improve CLI command blacklist checking for the
	  command manager action. Previously, it did not handle case or
	  whitespace properly. This made it possible for blacklisted
	  commands to get executed anyway. (closes issue #12765)

2008-06-02 14:32 +0000 [r119740]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
	  account with any configured XMPP client (in jabber.conf). The
	  actual connection is made when a call comes in Asterisk. Fix the
	  ast_aji_get_client function that was not able to retrieve an XMPP
	  client from its JID. (closes issue #12085) Reported by: junky
	  Tested by: phsultan

2008-06-02 12:30 +0000 [r119687]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
	  sent because it comes up too soon, make sure to reschedule so it
	  gets sent later.

2008-06-02 09:29 +0000 [r119585-119636]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
	  enabled

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
	  counter for unhandled_bmsg Print, this prevents the logs to be
	  flooded to fast and save CPU in this error scenario. Added
	  'last_used' element to bc structure, when a bchannel changes from
	  used to free this exact time will be marked in last_used. When a
	  new channel is requested the find_free_chan function will check
	  if the new empty channel was used within the last second, if yes
	  it will search for the next channel, if no it will return this
	  channel. This simple mechanism has prooven to prevent race
	  conditions where the NT and TE tried to allocate the exact same
	  channel at the same time (RELEASE cause: 44).

2008-06-02 01:06 +0000 [r119530-119533]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Change a debug message to an actual debug
	  message

	* apps/app_dial.c: Fix another typo in documentation

2008-06-01 20:47 +0000 [r119478]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_dial.c: small typo fix 'retires' => 'retries'

2008-05-30 21:17 +0000 [r119404]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: When joinempty=strict, it only failed on join
	  if there were busy members. If all members were logged out OR
	  paused, then it (incorrectly) let callers join the queue. (closes
	  issue #12451) Reported by: davidw

2008-05-30 19:46 +0000 [r119354]  Joshua Colp <jcolp@digium.com>

	* main/autoservice.c: Fix a bug I found while testing for another
	  issue.

2008-05-30 16:44 +0000 [r119301]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
	  check the $VERSION variable. The rc/init.d scripts, and
	  safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
	  (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
	  (closes issue #12687) Reported by: loloski Patches:
	  20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
	  mvanbaak (license 7) Tested by: loloski, mvanbaak

2008-05-30 12:55 +0000 [r119076-119238]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
	  May 2008) | 7 lines - Instead of only enforcing destination call
	  number checking on an ACK, check all full frames except for PING
	  and LAGRQ, which may be sent by older versions too quickly to
	  contain the destination call number. (As suggested by Tim Panton
	  on the asterisk-dev list) - Merge changes from
	  team/russell/iax2-frame-race, which prevents PING and LAGRQ from
	  being sent before the destination call number is known. ........

	* main/autoservice.c: Fix a race condition in channel autoservice.
	  There was still a small window of opportunity for a DTMF frame,
	  or some other deferred frame type, to come in and get dropped.
	  (closes issue #12656) (closes issue #12656) Reported by: dimas
	  Patches: v3-12656.patch uploaded by dimas (license 88) -- with
	  some modifications by me

	* include/asterisk/audiohook.h: Oddly enough, all of the contents
	  of audiohook.h were in there twice. I have removed the second
	  copy.

2008-05-29 20:24 +0000 [r119071]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Call waiting tone occurs too often, because
	  it's getting serviced by both subchannels. (closes issue #11354)
	  Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
	  by Corydon76 (license 14)

2008-05-29 19:04 +0000 [r118956-119012]  Russell Bryant <russell@digium.com>

	* apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
	  use ast_safe_sleep() instead of calling the wait application
	  (thanks to tilghman for pointing these out!)

	* /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
	  May 2008) | 7 lines Merge changes from
	  team/russell/iax2-another-fix-to-the-fix As described in the
	  following post to the asterisk-dev mailing list, only enforce
	  destination call numbers when processing an ACK.
	  http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
	  (closes issue #12631) ........

	* apps/app_milliwatt.c: - Mark app_milliwatt dependent on
	  res_indications (thanks to jsmith) - fix a typo in a log message
	  (thanks to qwell)

	* apps/app_milliwatt.c: Change milliwatt to use the proper tone by
	  default (1004 Hz) instead of 1000 Hz. An option is there to use
	  1000 Hz for anyone that might want it.

2008-05-29 17:33 +0000 [r118953-118954]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Define also when not DEBUG_THREADS

	* channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
	  channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
	  include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
	  code that ensures that when we do deadlock avoidance, we don't
	  lose the information about how a lock was originally acquired.

2008-05-29 00:25 +0000 [r118858]  Steve Murphy <murf@digium.com>

	* main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
	  issue #11721) (closes issue #12726) Reported by: arkadia Tested
	  by: murf These changes: 1. revert the changes made via bug 10668;
	  I should have known that such changes, even tho they made sense
	  at the time, seemed like an omission, etc, were actually integral
	  to the CDR system via forkCDR. It makes sense to me now that
	  forkCDR didn't natively end any CDR's, but rather depended on
	  natively closing them all at hangup time via traversing and
	  closing them all, whether locked or not. I still don't completely
	  understand the benefits of setvar and answer operating on locked
	  cdrs, but I've seen enough to revert those changes also, and stop
	  messing up users who depended on that behavior. bug 12726 found
	  reverting the changes fixed his changes, and after a long review
	  and working on forkCDR, I can see why. 2. Apply the suggested
	  enhancements proposed in 10668, but in a completely compatible
	  way. ForkCDR will behave exactly as before, but now has new
	  options that will allow some actions to be taken that will
	  slightly modify the outcome and side-effects of forkCDR. Based on
	  conversations I've had with various people, these small tweaks
	  will allow some users to get the behavior they need. For
	  instance, users executing forkCDR in an AGI script will find the
	  answer time set, and DISPOSITION set, a situation not covered
	  when the routines were first written. 3. A small problem in the
	  cdr serializer would output answer and end times even when they
	  were not set. This is now fixed.

2008-05-28 16:10 +0000 [r118716]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
	  Fixes a bug in chan_iax that uses send_command to poke a peer
	  while a channel is unlocked in some cases, and because it can
	  cause seemingly random failures could be related to some bugs in
	  the tracker...

2008-05-28 14:23 +0000 [r118558-118646]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
	  option to use the source IP address of RTP as the destination IP
	  address of UDPTL when a specific option is enabled. If the remote
	  side is properly configured (ports forwarded) then UDPTL will
	  flow. (closes issue #10417) Reported by: cstadlmann

	* channels/chan_sip.c: Fix an issue where codec preferences were
	  not set on dialogs that were not authenticated via a user or peer
	  and allow framing to work without rtpmap in the SDP. (closes
	  issue #12501) Reported by: slimey

2008-05-27 19:15 +0000 [r118551]  Tilghman Lesher <tlesher@digium.com>

	* main/cli.c: When showing an error message for a command, don't
	  shorten the command output, as it tends to confuse the user (it's
	  fine for suggesting other commands, however). Reported by:
	  seanbright (on #asterisk-dev) Fixed by: me

2008-05-27 19:07 +0000 [r118509]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Russell noted to me that in the case that
	  separate threads use their own addressing system, the fix I made
	  for issue 12376 does not guarantee uniqueness to the datastores'
	  uids. Though I know of no system that works this way, I am going
	  to change this right now to prevent trying to track down some
	  future bug that may occur and cause untold hours of debugging
	  time to track down. The change involves using a global counter
	  which increases with each new chanspy_ds which is created. This
	  guarantees uniqueness.

2008-05-27 18:58 +0000 [r118465]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: NULL character should terminate only commands
	  back to the core, not log messages to the console. (closes issue
	  #12731) Reported by: seanbright Patches:
	  20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: seanbright

2008-05-27 17:17 +0000 [r118416]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_voicemail.c: small update to the g() option of
	  app_voicemail to note that gain changes only work on zap channels
	  right now. issue #12578 shows it's not clear right now.

2008-05-27 16:38 +0000 [r118365]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Add a unique id to the datastore allocated in
	  app_chanspy since it is possible that multiple spies may be
	  listening to the same channel. (closes issue #12376) Reported by:
	  DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
	  (license 60) Tested by: destiny6628 (closes issue #12243)
	  Reported by: atis

2008-05-27 15:45 +0000 [r118358]  Tilghman Lesher <tlesher@digium.com>

	* configs/queues.conf.sample: Add a note that pbx_config.so is
	  needed for Local channels. (Closes issue #12671)

2008-05-25 16:02 +0000 [r118251]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Realtime flag affects construction in
	  multiple ways, so consulting whether rtcachefriends was set was
	  done too soon (needed to be done inside build_peer, not just as a
	  flag to build_peer). Also, fullcontact needed to be
	  reconstructed, because realtime separates the embedded ';' into
	  multiple fields. (closes issue #12722) Reported by: barthpbx
	  Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
	  (license 14) Tested by: barthpbx (Much of the discussion happened
	  on #asterisk-dev for diagnosing this issue)

2008-05-23 21:21 +0000 [r118163]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Fix a few things I missed to ensure
	  zt_chan_conf structure is not modified in mkintf

2008-05-23 13:18 +0000 [r118052-118055]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/utils.h: Add format type checking for recently
	  de-inlined function

	* doc/cli.txt (added), doc/00README.1st: Add information on using
	  the Asterisk console, including tab command line completion.
	  (Closes issue #12681)

2008-05-23 12:30 +0000 [r118048]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, main/utils.c: Don't declare a function
	  that takes variable arguments as inline, because it's not valid,
	  and on some compilers, will emit a warning.
	  http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
	  issue #12289) Reported by: francesco_r Patches by Tilghman, final
	  patch by me

2008-05-22 18:53 +0000 [r117809-117899]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Also remove preamble from asynchronous events
	  (reported by jsmith on #asterisk-dev)

	* funcs/func_realtime.c: Take into account the length of delimiters
	  when calculating result string length. (closes issue #12696)
	  Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
	  uploaded by adomjan (license 487)

2008-05-21 20:11 +0000 [r117582]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
	  is not modified in mkintf.

2008-05-21 19:38 +0000 [r117574]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Apply the autoframing setting to dialogs
	  that do not get matched against a user or peer.

2008-05-21 18:44 +0000 [r117519-117523]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_spool.c: Revert accidental commit of the last change

	* main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
	  output also when -rx is not being used (Related to issue #12702)

2008-05-21 18:28 +0000 [r117479-117514]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Don't filter the magic character in the network
	  verboser. It gets filtered once it reaches the client. (related
	  to issue #12702, pointed out by tilghman)

	* main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
	  marker in front of every message from ast_verbose() being sent to
	  remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
	  marker. (related to issue #12702)

	* main/asterisk.c: Don't display the verbose marker for calls to
	  ast_verbose() that do not include a VERBOSE_PREFIX in front of
	  the message. (closes issue #12702) Reported by: johnlange Patched
	  by me

2008-05-21 16:58 +0000 [r117462]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_zap.c: Pass a pointer for the conf parameter to the
	  function mkintf rather than the whole zt_chan_conf structure.

2008-05-20  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.20 released.

2008-05-14  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.20-rc3 released.

2008-05-14 12:51 +0000 [r116230]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Accept text messages even with Content-Type:
	  text/plain;charset=Södermanländska

2008-05-13 23:47 +0000 [r116088]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, include/asterisk/lock.h: A change to the way
	  channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
	  After debugging a deadlock, it was noticed that when
	  DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
	  of channel locks is obscured by the fact that all channel locks
	  appear to happen in the function ast_channel_lock(). This code
	  change redefines ast_channel_lock to be a macro which maps to
	  __ast_channel_lock(), which then relays the proper file name,
	  line number, and function name information to the core lock
	  functions so that this information will be displayed in the case
	  that there is some sort of locking error or core show locks is
	  issued.

2008-05-13 21:17 +0000 [r115990-116038]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c: Fix a deadlock involving channel
	  autoservice and chan_local that was debugged and fixed by
	  mmichelson and me. We observed a system that had a bunch of
	  threads stuck in ast_autoservice_stop(). The reason these threads
	  were waiting around is because this function waits to ensure that
	  the channel list in the autoservice thread gets rebuilt before
	  the stop() function returns. However, the autoservice thread was
	  also locked, so the autoservice channel list was never getting
	  rebuilt. The autoservice thread was stuck waiting for the channel
	  lock on a local channel. However, the local channel was locked by
	  a thread that was stuck in the autoservice stop function. It
	  turned out that the issue came down to the local_queue_frame()
	  function in chan_local. This function assumed that one of the
	  channels passed in as an argument was locked when called.
	  However, that was not always the case. There were multiple cases
	  in which this channel was not locked when the function was
	  called. We fixed up chan_local to indicate to this function
	  whether this channel was locked or not. The previous assumption
	  had caused local_queue_frame() to improperly return with the
	  channel locked, where it would then never get unlocked. (closes
	  issue #12584) (related to issue #12603)

	* main/autoservice.c: Fix an issue that I noticed in autoservice
	  while mmichelson and I were debugging a different problem. I
	  noticed that it was theoretically possible for two threads to
	  attempt to start the autoservice thread at the same time. This
	  change makes the process of starting the autoservice thread,
	  thread-safe.

2008-05-13 20:28 +0000 [r115944]  Joshua Colp <jcolp@digium.com>

	* channels/chan_alsa.c: Use the right flag to open the audio in
	  non-blocking. (closes issue #12616) Reported by:
	  nicklewisdigiumuser

2008-05-13 18:36 +0000 [r115884]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: If the socket dies (read returns 0=EOF), return
	  immediately. (Closes issue #12637)

2008-05-12 17:51 +0000 [r115735]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: If a thread holds no locks, do not print any
	  information on the thread when issuing a core show locks command.
	  This will help to de-clutter output somewhat. Russell said it
	  would be fine to place this improvement in the 1.4 branch, so
	  that's why it's going here too.

2008-05-09 16:34 +0000 [r115579]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Improve res_ninit and res_ndestroy autoconf logic on the Darwin
	  platform.

2008-05-08 19:19 +0000 [r115545-115568]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove debug output.

	* /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
	  May 2008) | 25 lines Fix a race condition that bbryant just found
	  while doing some IAX2 testing. He was running Asterisk trunk
	  running IAX2 calls through a few Asterisk boxes, however, the
	  audio was extremely choppy. We looked at a packet trace and saw a
	  storm of INVAL and VNAK frames being sent from one box to
	  another. It turned out that what had happened was that one box
	  tried to send a CONTROL frame before the 3 way handshake had
	  completed. So, that frame did not include the destination call
	  number, because it didn't have it yet. Part of our recent work
	  for security issues included an additional check to ensure that
	  frames that are supposed to include the destination call number
	  have the correct one. This caused the frame to be rejected with
	  an INVAL. The frame would get retransmitted for forever, rejected
	  every time ... This race condition exists in all versions that
	  got the security changes, in theory. However, it is really only
	  likely that this would cause a problem in Asterisk trunk. There
	  was a control frame being sent (SRCUPDATE) at the _very_
	  beginning of the call, which does not exist in 1.2 or 1.4.
	  However, I am fixing all versions that could potentially be
	  affected by the introduced race condition. These changes are what
	  bbryant and I came up with to fix the issue. Instead of simply
	  dropping control frames that get sent before the handshake is
	  complete, the code attempts to wait a little while, since in most
	  cases, the handshake will complete very quickly. If it doesn't
	  complete after yielding for a little while, then the frame gets
	  dropped. ........

	* channels/chan_sip.c: Don't give up on attempting an outbound
	  registration if we receive a 408 Timeout. (closes issue #12323)

	* contrib/scripts/postgres_cdr.sql (removed): remove
	  postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
	  well (closes issue #9676)

	* contrib/init.d/rc.debian.asterisk: Don't exit the script if
	  Asterisk is not running. (closes issue #12611)

	* main/pbx.c: Don't use a channel before checking for channel
	  allocation failure. (closes issue #12609) Reported by: edantie

	* contrib/init.d/rc.debian.asterisk: Use the same method for
	  executing Asterisk as the rest of the script. (closes issue
	  #12611) Reported by: b_plessis

2008-05-07  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.20-rc2 released.

2008-05-07 18:17 +0000 [r115512-115517]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Track peer references when stored in the
	  sip_pvt struct as the peer related to a qualify ping or a
	  subscription. This fixes some realtime related crashes. (closes
	  issue #12588) (closes issue #12555)

2008-05-06 19:55 +0000 [r115418-115422]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
	  7 lines read requires an argument on some non-bash shells (closes
	  issue #12593) Reported by: bkruse Patches:
	  getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
	  ........

	* res/res_musiconhold.c: Switch to using ast_random() rather than
	  just rand(). This does not fix the bug reported, but I believe it
	  is correct. (from issue #12446) Patches: bug_12446.diff uploaded
	  by snuffy (license 35)

2008-05-06 19:31 +0000 [r115415]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Don't print the terminating NUL. (Closes issue
	  #12589)

2008-05-06 13:54 +0000 [r115341]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Add in missing argument.

2008-05-05 22:50 +0000 [r115333]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, main/logger.c: Separate verbose output from CLI
	  output, by using a preamble. (closes issue #12402) Reported by:
	  Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
	  uploaded by Corydon76 (license 14)
	  20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
	  Corydon76 (license 14)

2008-05-05 22:10 +0000 [r115327]  Joshua Colp <jcolp@digium.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac: Make sure that either the main speex library
	  contains preprocess functions or that speexdsp does. If both fail
	  then speex stuff can not be built.

2008-05-05 21:41 +0000 [r115320]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Don't consider a caller "handled" until the
	  caller is bridged with a queue member. There was too much of an
	  opportunity for the member to hang up (either during a delay,
	  announcement, or overly long agi) between the time that he
	  answered the phone and the time when he actually was bridged with
	  the caller. The consequence of this was that if the member hung
	  up in that interval, then proper abandonment details would not be
	  noted in the queue log if the caller were to hang up at any point
	  after the member hangup. (closes issue #12561) Reported by:
	  ablackthorn

2008-05-05 20:17 +0000 [r115308-115312]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Reverse order, such that user configs override default
	  selections

	* include/asterisk/res_odbc.h: Err, the documentation on the return
	  value of ast_odbc_backslash_is_escape is exactly backwards.

2008-05-05 19:49 +0000 [r115297-115304]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Avoid putting opaque="" in Digest
	  authentication. This patch came from switchvox. It fixes
	  authentication with Primus in Canada, and has been in use for a
	  very long time without causing problems with any other providers.
	  (closes issue AST-36)

2008-05-05 03:22 +0000 [r115285]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
	  out if Asterisk is already running. (closes issue #12525)
	  Reported by: explidous Patches: 20080428__bug12525.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: mvanbaak

2008-05-04 02:09 +0000 [r115276-115282]  Joshua Colp <jcolp@digium.com>

	* configure, acinclude.m4: Expand the test function for GCC
	  attributes so that more complex attributes are properly
	  recognized.

	* include/asterisk/compiler.h: For my next trick I will make these
	  work with what our autoconf header file gives us.

	* configure, acinclude.m4: Treat warnings as errors when checking
	  if a GCC attribute exists. We have to do this as GCC will just
	  ignore the attribute and pop up a warning, it won't actually fail
	  to compile.

2008-05-02 20:25 +0000 [r115257]  Brett Bryant <bbryant@digium.com>

	* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, CHANGES: Add new "pri show version" command to show
	  the libpri version for support reasons.

2008-05-02 14:28 +0000 [r115196]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/sched.h: Clarify a comment that was, well, just
	  wrong. It turns out that ignoring the way that macros expand.
	  Instead, I have clarified in the comment why the macro will work
	  even if the scheduler id for the task to be deleted changes
	  during the execution of the macro.

2008-05-01 23:20 +0000 [r115017-115102]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h: Change the comment of deprecated to
	  an actual compiler deprecation

	* main/utils.c: '#' is another reserved character for URIs that
	  also needs to be escaped. (closes issue #10543) Reported by:
	  blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
	  Corydon76 (license 14)

2008-05-01  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.20-rc1 released.

2008-04-30 16:30 +0000 [r114891]  Russell Bryant <russell@digium.com>

	* include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
	  Merge changes from team/russell/iax2_find_callno and
	  iax2_find_callno_1.4 These changes address a critical performance
	  issue introduced in the latest release. The fix for the latest
	  security issue included a change that made Asterisk randomly
	  choose call numbers to make them more difficult to guess by
	  attackers. However, due to some inefficient (this is by far, an
	  understatement) code, when Asterisk chose high call numbers,
	  chan_iax2 became unusable after just a small number of calls. On
	  a small embedded platform, it would not be able to handle a
	  single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
	  more than about 16 IAX2 channels. Ouch. These changes address
	  some performance issues of the find_callno() function that have
	  bothered me for a very long time. On every incoming media frame,
	  it iterated through every possible call number trying to find a
	  matching active call. This involved a mutex lock and unlock for
	  each call number checked. So, if the random call number chosen
	  was 20000, then every media frame would cause 20000 locks and
	  unlocks. Previously, this problem was not as obvious since
	  Asterisk always chose the lowest call number it could. A second
	  container for IAX2 pvt structs has been added. It is an astobj2
	  hash table. When we know the remote side's call number, the pvt
	  goes into the hash table with a hash value of the remote side's
	  call number. Then, lookups for incoming media frames are a very
	  fast hash lookup instead of an absolutely insane array traversal.
	  In a quick test, I was able to get more than 3600% more IAX2
	  channels on my machine with these changes.

2008-04-30 16:23 +0000 [r114890]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
	  in Huntsville together with Mark M (putnopvut) (closes issue
	  #12363) Reported by: jvandal Tested by: putnopvut, oej

2008-04-30 14:46 +0000 [r114875-114880]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
	  for indexing through the iaxs/iaxsl arrays so that the size of
	  the arrays can be adjusted in one place, and change the size of
	  the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
	  defined

	* Makefile.rules: pay attention to *all* header files for
	  dependency tracking, not just the local ones (inspired by r578 of
	  asterisk-addons by tilghman)

2008-04-29 19:40 +0000 [r114848]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
	  variables instead of the channel's macrocontext and macroexten
	  fields. This is needed because if macros are daisy-chained, the
	  incorrect context and extension are placed on the new channel. I
	  also added locking to the channel prior to accessing these
	  variables as noted in trunk's janitor project file. (closes issue
	  #12549) Reported by: darren1713 Patches:
	  app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
	  (with modifications from me) Tested by: putnopvut

2008-04-29 17:08 +0000 [r114829]  Jason Parker <jparker@digium.com>

	* res/res_config_pgsql.c: Change warning message to debug, since
	  there are cases where 0 results is perfectly fine.

2008-04-29 12:53 +0000 [r114823]  Kevin P. Fleming <kpfleming@digium.com>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
	  2008) | 2 lines stop script from appending source code if run
	  multiple times ........

2008-04-28 04:47 +0000 [r114708]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
	  embedded, they take on a different name, without the ".so"
	  extension. Specifically check for this name, when we're checking
	  if a module is loaded. (Closes issue #12534)

2008-04-27 01:26 +0000 [r114695]  Sean Bright <sean.bright@gmail.com>

	* configure, configure.ac: When we don't explicitly pass a path to
	  the --with-tds configure option, we may end up finding tds.h in
	  /usr/local/include instead of /usr/include. If this happens, the
	  grep that looks for the version (from tdsver.h) will fail and
	  we'll have some problems during the build.

2008-04-26 13:15 +0000 [r114689]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/vmail.cgi: Clicking forward without selecting a
	  message leaves an errant .lock file. (closes issue #12528)
	  Reported by: pukepail Patches: patch.diff uploaded by pukepail
	  (license 431)

2008-04-25 21:54 +0000 [r114673]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use consistent logic for checking to see if
	  a call number has been chosen yet. Also, remove some redundant
	  logic I recently added in a fix.

2008-04-25 19:32 +0000 [r114662]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Move the unlock of the spyee channel to
	  outside the start_spying() function so that the channel is not
	  unlocked twice when using whisper mode.

2008-04-25 15:53 +0000 [r114649]  Tilghman Lesher <tlesher@digium.com>

	* configs/zapata.conf.sample, configs/iax.conf.sample,
	  configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
	  documentation files that actually exist. (closes issue #12516)
	  Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
	  by linuxmaniac (license 472)

2008-04-24 21:35 +0000 [r114624-114632]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Re-invite RTP during a masquerade so that,
	  for instance, an AMI redirect of two channels which are natively
	  bridged will preserve audio on both channels. This prevents a
	  problem with Asterisk not re-inviting due to one of the channels
	  having being a zombie. (closes issue #12513) Reported by:
	  mneuhauser Patches:
	  asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
	  mneuhauser (license 425)

	* apps/app_queue.c: Output of channel variables when
	  eventwhencalled=vars was set was being truncated two characters.
	  This patch corrects the problem. (closes issue #12493) Reported
	  by: davidw

	* channels/chan_local.c: Resolve a deadlock in chan_local by
	  releasing the channel lock temporarily. (closes issue #11712)
	  Reported by: callguy Patches: 11712.patch uploaded by putnopvut
	  (license 60) Tested by: acunningham

2008-04-24 19:53 +0000 [r114621]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Ensure that when we set the accountcode,
	  it actually shows up in the CDR. (Fix for AMI Originate) (Closes
	  issue #12007)

2008-04-24 15:55 +0000 [r114608]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a silly mistake in a change I made
	  yesterday that caused chan_iax2 to blow up very quickly. (issue
	  #12515)

2008-04-24 14:55 +0000 [r114603]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Only have one max-forwards header in
	  outbound REFERs. Discovered in the Asterisk SIP Masterclass in
	  Orlando. Thanks Joe!

2008-04-23 22:18 +0000 [r114597-114600]  Russell Bryant <russell@digium.com>

	* main/http.c: Improve some broken cookie parsing code. Previously,
	  manager login over HTTP would only work if the mansession_id
	  cookie was first. Now, the code builds a list of all of the
	  cookies in the Cookie header. This fixes a problem observed by
	  users of the Asterisk GUI. (closes AST-20)

	* apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
	  the correct next channel to not always work. Also, remove setting
	  the amount of time to wait for a digit from 5 seconds back down
	  to 1/10 of a second. I believe this was so the beep didn't get
	  played over and over really fast, but a while back I put in
	  another fix for that issue. (closes issue #12498) Reported by:
	  jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
	  jsmith (license 15)

2008-04-23 18:28 +0000 [r114594]  Jason Parker <jparker@digium.com>

	* res/res_musiconhold.c: Fix reload/unload for res_musiconhold
	  module. (closes issue #11575) Reported by: sunder Patches:
	  M11575_14_rev3.diff uploaded by junky (license 177)
	  bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)

2008-04-23 17:55 +0000 [r114587-114591]  Russell Bryant <russell@digium.com>

	* main/manager.c, include/asterisk/manager.h: Store the manager
	  session ID explicitly as 4 byte ID instead of a ulong. The
	  mansession_id cookie is coded to be limited to 8 characters of
	  hex, and this could break logins from 64-bit machines in some
	  cases. (inspired by AST-20)

	* channels/chan_iax2.c: Fix find_callno_locked() to actually return
	  the callno locked in some more cases.

2008-04-23 16:51 +0000 [r114584]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add 502 support for both directions, not
	  only one... (see r114571)

2008-04-23 14:54 +0000 [r114579]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Instead of stopping dialplan execution when SayNumber
	  attempts to say a large number that it can not print out a
	  message informing the user and continue on. (closes issue #12502)
	  Reported by: bcnit

2008-04-22 23:51 +0000 [r114571]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Treat a 502 just like a 503, when it comes
	  to processing a response code

2008-04-22 22:15 +0000 [r114522-114558]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: When we receive a full frame that is
	  supposed to contain our call number, ensure that it has the
	  correct one. (closes issue #10078) (AST-2008-006)

	* main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
	  thought I was going to be able to leave 1.4 alone, but that was
	  not the case. I ran into some problems with G.722 in 1.4, so I
	  have merged in all of the fixes in this area that I have made in
	  trunk/1.6.0, and things are happy again.

	* res/res_musiconhold.c: Trivial change to read the number of
	  samples from a frame before calling ast_write()

	* res/res_features.c: After a parked call times out, allow the call
	  back to the parker to time out. (closes issue #10890)

	* channels/chan_iax2.c: If the dial string passed to the call
	  channel callback does not indicate an extension, then consider
	  the extension on the channel before falling back to the default.
	  (closes issue #12479) Reported by: darren1713 Patches:
	  exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
	  116)

	* channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
	  team/russell/issue_9520 These changes make sure that the
	  reference count for sip_peer objects properly reflects the fact
	  that the peer is sitting in the scheduler for a scheduled
	  callback for qualifying peers or for expiring registrations.
	  Without this, it was possible for these callbacks to happen at
	  the same time that the peer was being destroyed. This was
	  especially likely to happen with realtime peers, and for people
	  making use of the realtime prune CLI command. (closes issue
	  #9520) Reported by: kryptolus Committed patch by me

2008-04-21 14:39 +0000 [r114322]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only drop audio if we receive it without a
	  progress indication. We allow other frames through such as DTMF
	  because they may be needed to complete the call. (closes issue
	  #12440) Reported by: aragon

2008-04-19 13:57 +0000 [r114297-114299]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_playback.c: Ensure that help text terminates with a
	  newline

	* res/res_musiconhold.c: MOH usage information needs a terminating
	  newline, or else "asterisk -rx 'help moh reload'" will hang.
	  Reported via -dev list, fixed by me.

2008-04-18 21:48 +0000 [r114275-114284]  Russell Bryant <russell@digium.com>

	* main/manager.c: Don't destroy a manager session if poll() returns
	  an error of EAGAIN.

	* Makefile: ensure directories are created before we try to install
	  stuff into them

	* Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
	  bininstall

2008-04-18 17:44 +0000 [r114257]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c, main/callerid.c: Clearing up error messages
	  so they make a bit more sense. Also removing a redundant error
	  message. Issue AST-15

2008-04-18 15:24 +0000 [r114248]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
	  (closes issue #12476) Reported by: davidw Patch by me

2008-04-18 13:33 +0000 [r114245]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: Only complete the SIP channel name once for
	  'sip show channel <channel>'

2008-04-18 06:49 +0000 [r114242]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_setcallerid.c: For consistency sake, ensure that the
	  values that ${CALLINGPRES} returns are valid as an input to
	  SetCallingPres. (Closes issue #12472)

2008-04-17 22:15 +0000 [r114230]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Remove redundant safety net. The check for
	  the autoservice channel list state accomplishes the same goal in
	  a better way. (issue #12470) Reported By: atis

2008-04-17 21:03 +0000 [r114207-114226]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Declaration of the peer channel in this scope
	  was making it so the peer variable defined in the outer scope was
	  never set properly, therefore making iterating through the
	  channel list always restart from the beginning. This bug would
	  have affected anyone who called chanspy without specifying a
	  first argument. (closes issue #12461) Reported by: stever28

	* main/frame.c, include/asterisk/dsp.h: Add prototype for
	  ast_dsp_frame_freed. I'm not sure how this was compiling
	  before...

	* main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
	  possible for a reference to a frame which was part of a freed DSP
	  to still be referenced, leading to memory corruption and eventual
	  crashes. This code change ensures that the dsp is freed when we
	  are finished with the frame. This change is very similar to a
	  change Russell made with translators back a month or so ago.
	  (closes issue #11999) Reported by: destiny6628 Patches:
	  11999.patch uploaded by putnopvut (license 60) Tested by:
	  destiny6628, victoryure

2008-04-17 16:23 +0000 [r114204]  Russell Bryant <russell@digium.com>

	* Makefile: Fix the bininstall target to install from subdirs, as
	  well. (closes issue AST-8, patch from bmd at switchvox)

2008-04-17 13:42 +0000 [r114198]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Use keepalives effectively in order diagnose
	  bug #12432.

2008-04-17 12:56 +0000 [r114195]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Add special case for when the agi cannot be
	  executed, to comply with the documentation that we return failure
	  in that case. (closes issue #12462) Reported by: fmueller
	  Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
	  (license 14) Tested by: fmueller

2008-04-17 10:51 +0000 [r114191]  Sean Bright <sean.bright@gmail.com>

	* apps/app_chanspy.c: Make sure we have enough room for the
	  recording's filename.

2008-04-16 20:46 +0000 [r114184]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
	  initializing the structure to all zeroes in case it contains
	  fields that we don't write values into (which it does as of
	  Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian

2008-04-16 19:59 +0000 [r114180]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_vpb.cc: Backport revisions for latest vpb drivers
	  to 1.4 (Closes issue #12457)

2008-04-16 17:30 +0000 [r114173]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Fix "fallthrough" behavior here, so config
	  options in a previously configured user don't override settings
	  in general. (closes issue #12458) Reported by: tzafrir Patches:
	  chanzap_users_sections.diff uploaded by tzafrir (license 46)

2008-04-16 14:10 +0000 [r114167]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Include the proper headers for using mkdir on
	  FreeBSD. (closes issue #12430) Reported by: ys Patches:
	  app_meetme.c.diff uploaded by ys (license 281)

2008-04-15 20:26 +0000 [r114148]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Handle subscribe queues in all situations...
	  Thanks to festr_ on irc for telling me about this bug.

2008-04-15 17:17 +0000 [r114120-114138]  Jason Parker <jparker@digium.com>

	* contrib/scripts/autosupport: Update Digium autosupport script,
	  for more useful information. (closes issue #12452) Reported by:
	  angler Patches: autosupport.diff uploaded by angler (license 106)

	* apps/app_queue.c: Allow autofill to work in the general section
	  of queues.conf. Additionally, don't try to (re)set options when
	  they have empty values in realtime (all unset columns would have
	  an empty value). (closes issue #12445) Reported by: atis Patches:
	  12445-autofill.diff uploaded by qwell (license 4)

	* channels/chan_h323.c: The call_token on the pvt can occasionally
	  be NULL, causing a crash. If it is NULL, we can skip this
	  channel, since it can't the one we're looking for. (closes issue
	  #9299) Reported by: vazir

2008-04-14 17:41 +0000 [r114106-114117]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Increase the retry count when attempting to show
	  channels. This apparently cleared an issue someone was seeing
	  when attempting to show channels when the load was high. (closes
	  issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
	  by russell (license 2) Tested by: falves11

	* apps/app_dial.c, apps/app_queue.c: If the datastore has been
	  moved to another channel due to a masquerade, then freeing the
	  datastore here causes an eventual double free when the new
	  channel hangs up. We should only free the datastore if we were
	  able to successfully remove it from the channel we are
	  referencing (i.e. the datastore was not moved). (closes issue
	  #12359) Reported by: pguido

	* main/channel.c: Save a local copy of the generate callback prior
	  to unlocking the channel in case the generate callback goes NULL
	  on us after the channel is unlocked. Thanks to Russell for
	  pointing this need out to me.

2008-04-14 14:52 +0000 [r114100-114103]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is possible for the remote side to say
	  they want T38 but not give any capabilities. (closes issue
	  #12414) Reported by: MVF

	* main/rtp.c: Don't change the SSRC when a new source comes into
	  play, this might happen quite often and depending on the remote
	  side... they might not like this. (closes issue #12353) Reported
	  by: dimas

2008-04-11 22:32 +0000 [r114083]  Terry Wilson <twilson@digium.com>

	* channels/chan_iax2.c: Several places in the code called
	  find_callno() (which releases the lock on the pvt structure) and
	  then immediately locked the call and did things with it.
	  Unfortunately, the call can disappear between the find_callno and
	  the lock, causing Bad Stuff(tm) to happen. Added
	  find_callno_locked() function to return the callno withtout
	  unlocking for instances that it is needed. (issue #12400)
	  Reported by: ztel

2008-04-11 21:35 +0000 [r114072]  Jason Parker <jparker@digium.com>

	* main/pbx.c: It's possible that a channel can have an async goto
	  on the successful execution of an application as well. Closes
	  issue #12172.

2008-04-11 15:44 +0000 [r114045-114063]  Mark Michelson <mmichelson@digium.com>

	* res/res_features.c: Fix a race condition that may happen between
	  a sip hangup and a "core show channel" command. This patch adds
	  locking to prevent the resulting crash. (closes issue #12155)
	  Reported by: tsearle Patches: show_channels_crash2.patch uploaded
	  by tsearle (license 373) Tested by: tsearle

	* main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
	  LOW_MEMORY is enabled.

	* channels/chan_sip.c: Be sure that we're not about to set
	  bridgepvt NULL prior to dereferencing it. (closes issue #11775)
	  Reported by: fujin

2008-04-10 17:26 +0000 [r114035]  Jason Parker <jparker@digium.com>

	* main/file.c: Only try to prefix language if we are not using an
	  absolute path (suffix it otherwise).
	  en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
	  issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
	  uploaded by qwell (license 4) Tested by: kuj, qwell

2008-04-10 15:58 +0000 [r114021-114032]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Forgot the 1.4 branch for russian language
	  fix. (closes issue #12404) Reported by: IgorG Patches:
	  voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)

	* apps/app_meetme.c: Create the directory where name recordings
	  will go if it does not exist. (closes issue #12311) Reported by:
	  rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)

	* channels/chan_sip.c: Don't add custom URI options if they don't
	  exist OR they are empty. (closes issue #12407) Reported by:
	  homesick Patches: uri_options-1.4.diff uploaded by homesick
	  (license 91)

2008-04-09 20:54 +0000 [r113927]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: We need to set the persistant_route [sic]
	  parameter for the sip_pvt during the initial INVITE, no matter if
	  we're building the route set from an INVITE request or response.
	  (closes issue #12391) Reported by: benjaminbohlmann Tested by:
	  benjaminbohlmann

2008-04-09 18:57 +0000 [r113874]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
	  not exist in cdr.conf, then an unload/load sequence is needed to
	  correct the problem. Track whether the load succeeded with a
	  variable, so we can fix this with a simple reload event, instead.

2008-04-09 16:50 +0000 [r113784]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: If we receive an AUTHREQ from the remote
	  server and we are unable to reply (for example they have a secret
	  configured, but we do not) then queue a hangup frame on the
	  Asterisk channel. This will cause the channel to hangup and a
	  HANGUP to be sent via IAX2 to the remote side which is the proper
	  thing to do in this scenario. (closes issue #12385) Reported by:
	  viraptor

2008-04-09 14:40 +0000 [r113681]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
	  a reinvite), then we should not send a BYE. (closes issue #12392)
	  Reported by: fnordian Patches: chan_sip.patch uploaded by
	  fnordian (license 110) with small modification from me

2008-04-09 01:34 +0000 [r113596]  Terry Wilson <twilson@digium.com>

	* channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
	  happy

2008-04-08 19:07 +0000 [r113507]  Mark Michelson <mmichelson@digium.com>

	* apps/app_parkandannounce.c: Fix potential buffer overflow that
	  could happen if more than 100 announce files were specified when
	  calling ParkAndAnnounce. This overflow is not exploitable
	  remotely and so there is no need for a security advisory. (closes
	  issue #12386) Reported by: davidw

2008-04-08 18:48 +0000 [r113402-113504]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Add a little more that is required for
	  previously added devices.

	* channels/chan_skinny.c: Add support for several new(ish) devices
	  - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
	  for providing me the required information.

	* main/asterisk.c: Work around some silliness caused by
	  sys/capability.h - this should fix compile errors a number of
	  users have been experiencing.

2008-04-08 16:51 +0000 [r113348-113399]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/astgenkey.8: Add security note on astgenkey's
	  manpage. (closes issue #12373) Reported by: lmamane Patches:
	  20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)

	* channels/chan_sip.c: Move check for still-bridged channels out a
	  little further, to avoid possible deadlocks. (Closes issue
	  #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: callguy

2008-04-08 15:03 +0000 [r113296]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/slinfactory.h, main/slinfactory.c,
	  main/audiohook.c: If audio suddenly gets fed into one side of a
	  channel after a lapse of frames flush the other factory so that
	  old audio does not remain in the factory causing the sync code to
	  not execute. (closes issue #12296) Reported by: jvandal

2008-04-07 21:34 +0000 [r113240]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
	  fixes a for loop (in realtime_peer) to check all the
	  ast_variables the loop was intending to test rather than just the
	  first one. The change exposed the problem of calling memcpy on a
	  NULL pointer, in this case the passed in sockaddr_in struct which
	  is now checked.

2008-04-07 18:00 +0000 [r113118]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Allow
	  playback with noanswer (and add earlyrtp option). (closes issue
	  #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
	  (license 30) Tested by: pj, qwell, DEA, wedhorn

2008-04-07 17:51 +0000 [r113117]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c: Force ast_mktime() to check for DST, since
	  strptime(3) does not. (Closes issue #12374)

2008-04-07 16:08 +0000 [r113065]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: This fix prevents a deadlock that was experienced
	  in chan_local. There was deadlock prevention in place in
	  chan_local, but it would not work in a specific case because the
	  channel was recursively locked. By unlocking the channel prior to
	  calling the generator's generate callback in
	  ast_read_generator_actions(), we prevent the recursive locking,
	  and therefore the deadlock. (closes issue #12307) Reported by:
	  callguy Patches: 12307.patch uploaded by putnopvut (license 60)
	  Tested by: callguy

2008-04-07 15:16 +0000 [r113012]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
	  Reported by: vinsik Tested by: tecnoxarxa This one line change
	  makes an if inside a for loop (in realtime_peer) check all the
	  ast_variables the loop was intending to test rather than just the
	  first one.

2008-04-04 19:26 +0000 [r112766-112820]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Free newly allocated channel before
	  returning

	* channels/chan_gtalk.c: Prevent call connections when codecs don't
	  match. (closes issue #10604) Reported by: keepitcool Patches:
	  branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
	  by: phsultan

2008-04-04 00:52 +0000 [r112709-112711]  Joshua Colp <jcolp@digium.com>

	* main/Makefile: Pass in the path to Zaptel for systems that
	  install Zaptel headers in a separate location.

	* main/asterisk.c: One thing at a time... let's get 1.4 building.

2008-04-03 23:57 +0000 [r112689]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/asterisk.c: add a Zaptel timer check to verify the timer is
	  responding when Zaptel support is compiled into Asterisk and
	  Zaptel drivers are loaded. This will help people not waste their
	  valuable time debugging side effects.

2008-04-03 14:32 +0000 [r112393-112599]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: Fix the testing of the "res" variable so
	  that it is more logically correct and makes the correct warning
	  and debug messages print. (closes issue #12361) Reported by:
	  one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
	  (license 23)

	* main/manager.c: Fix a race condition in the manager. It is
	  possible that a new manager event could be appended during a
	  brief time when the manager is not waiting for input. If an event
	  comes during this period, we need to set an indicator that there
	  is an event pending so that the manager doesn't attempt to wait
	  forever for an event that already happened. (closes issue #12354)
	  Reported by: bamby Patches: manager_race_condition.diff uploaded
	  by bamby (license 430) (comments added by me)

	* apps/app_queue.c: Ensure that there is no timeout if none is
	  specified. (closes issue #12349) Reported by: johnlange

2008-04-01  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.19 released.

2008-03-28  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.19-rc4 released.

2008-03-28 16:19 +0000 [r111658]  Jason Parker <jparker@digium.com>

	* formats/format_wav_gsm.c: The file size of WAV49 does not need to
	  be an even number. (closes issue #12128) Reported by: mdu113
	  Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
	  Tested by: qwell, mdu113

2008-03-28 14:35 +0000 [r111442-111605]  Tilghman Lesher <tlesher@digium.com>

	* doc/valgrind.txt: Update debugging text, since Valgrind
	  eliminated the --log-file-exactly option. (Closes issue #12320)

	* main/acl.c: For FreeBSD, at least, the ifa_addr element could be
	  NULL. (closes issue #12300) Reported by: festr Patches:
	  acl.c.patch uploaded by festr (license 443)

2008-03-27 13:03 +0000 [r111341-111391]  Steve Murphy <murf@digium.com>

	* apps/app_playback.c, main/pbx.c: These small documentation
	  updates made in response to a query in asterisk-users, where a
	  user was using Playback, but needed the features of Background,
	  and had no idea that Background existed, or that it might provide
	  the features he needed. I thought the best way to avert these
	  kinds of queries was to provide "See Also" references in all
	  three of "Background", "Playback", "WaitExten". Perhaps a project
	  to do this with all related apps is in order.

	* pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
	  #12302) Reported by: pj Tested by: murf These changes will set a
	  channel variable ~~EXTEN~~ just before generating code for a
	  switch, with the value of ${EXTEN}. The exten is marked as having
	  a switch, and ever after that, till the end of the exten, we
	  substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
	  arguments; (and the ${EXTEN: also). The reason for this, is that
	  because switches are coded using separate extensions to provide
	  pattern matching, and jumping to/from these switch extensions
	  messes up the ${EXTEN} value, which blows the minds of users.

2008-03-27 00:25 +0000 [r111245-111280]  Jason Parker <jparker@digium.com>

	* main/frame.c: Put this flag back so we don't change the API.

	* main/frame.c: Remove excessive smoother optimization that was
	  causing audio glitches (small "pops") after (about 200ms later)
	  an "incorrectly" sized frame was received. While it would be very
	  nice to keep this as optimized as possible, it makes no sense for
	  the smoother to be dropping random bits of audio like this. Isn't
	  that the whole point of a smoother? Closes issue #12093.

2008-03-26 19:55 +0000 [r111129]  Joshua Colp <jcolp@digium.com>

	* contrib/scripts/autosupport: Update autosupport script. (closes
	  issue #12310) Reported by: angler Patches: autosupport.diff
	  uploaded by angler (license 106)

2008-03-26 19:51 +0000 [r111126]  Kevin P. Fleming <kpfleming@digium.com>

	* /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
	  2008) | 2 lines update UPGRADE notes to document usage of the
	  script ........

2008-03-26 19:37 +0000 [r111049-111121]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: This code change is made just for
	  clarification. It does exactly the same thing as before. It just
	  doesn't look as wrong.

	* apps/app_voicemail.c: Add a lock to the vm_state structure and
	  use the lock around mail_open calls to prevent concurrent access
	  of the same mailstream. This, along with trunk's ability to
	  configure TCP timeouts for IMAP storage will help to prevent
	  crashes and hangs when using voicemail with IMAP storage. (closes
	  issue #10487) Reported by: ewilhelmsen

2008-03-26 19:06 +0000 [r111024]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
	  Merged revisions 111019 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
	  2008) | 2 lines add a script to make getting the iLBC source code
	  simple for end users ........

2008-03-26 19:04 +0000 [r111014-111020]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If we are requested to authenticate a
	  reinvite make sure that it contains T38 SDP if need be. (closes
	  issue #11995) Reported by: fall

	* channels/chan_iax2.c: Make sure that full video frames are sent
	  whenever the 15 bit timestamp rolls over. (closes issue #11923)
	  Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
	  mihai (license 94)

2008-03-26 17:43 +0000 [r110880-110962]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: add note that the user will need to enable
	  codec_ilbc to get it to build

	* codecs/ilbc/StateConstructW.h (removed),
	  codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
	  (removed), codecs/ilbc/getCBvec.c (removed),
	  codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
	  (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
	  (removed), codecs/ilbc/getCBvec.h (removed),
	  codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
	  (removed), codecs/ilbc/FrameClassify.c (removed),
	  codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
	  codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
	  (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
	  (removed), codecs/ilbc/anaFilter.c (removed),
	  codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
	  (removed), codecs/ilbc/doCPLC.h (removed),
	  codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
	  codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
	  (removed), codecs/ilbc/createCB.h (removed), CHANGES,
	  codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
	  (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
	  codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
	  codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
	  (removed), codecs/ilbc/iCBSearch.h (removed),
	  codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
	  codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
	  (removed), codecs/ilbc/hpOutput.h (removed),
	  codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
	  codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
	  (removed), codecs/ilbc/iCBConstruct.c (removed),
	  codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
	  (removed), codecs/ilbc/syntFilter.h (removed),
	  codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
	  (removed): Merged revisions 110869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
	  2008) | 2 lines due to licensing restrictions, we cannot
	  distribute the source code for iLBC encoding and decoding... so
	  remove it, and add instructions on how the user can obtain it
	  themselves ........

2008-03-25 22:51 +0000 [r110779]  Jason Parker <jparker@digium.com>

	* cdr/cdr_custom.c: Make file access in cdr_custom similar to
	  cdr_csv. Fixes issue #12268. Patch borrowed from r82344

2008-03-25 20:03 +0000 [r110727]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: This one line change makes an if inside a
	  for loop (in realtime_peer) check all the ast_variables the loop
	  was intending to test rather than just the first one.

2008-03-25 15:40 +0000 [r110635]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: When reverting a commit, I accidentally left
	  in this bit which was an experiment to see what would happen. It
	  passed the compile test, and I didn't notice I had left this
	  change in too. So this is a revert of a revert...sort of.

2008-03-25 14:37 +0000 [r110628]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/options.h, main/asterisk.c, Makefile,
	  main/app.c: Add an option (transmit_silence) which transmits
	  silence during both Record() and DTMF generation. The reason this
	  is an option is that in order to transmit silence we have to
	  setup a translation path. This may not be needed/wanted in all
	  cases. (closes issue #10058) Reported by: tracinet

2008-03-24 19:17 +0000 [r110618]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This is a revert for revision 108288. The
	  reason is that that revision was not for an actual bug fix per
	  se, and so it really should not have been in 1.4 in the first
	  place. Plus, people who compile with DO_CRASH are more likely to
	  encounter a crash due to this change. While I think the usage of
	  DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
	  beyond the scope of 1.4 and should be done instead in a developer
	  branch based on trunk so that all scheduler functions are fixed
	  at once. I also am reverting the change to trunk and 1.6 since
	  they also suffer from the DO_CRASH potential. (closes issue
	  #12272) Reported by: qq12345

2008-03-24 17:34 +0000 [r110614]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.

2008-03-21 14:32 +0000 [r110474]  Jason Parker <jparker@digium.com>

	* codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
	  mips platforms either. (closes issue #12270) Reported by:
	  zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
	  (license 33)

2008-03-20 23:13 +0000 [r110163-110395]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
	  to 50 ms in the autoservice thread. This really should not make a
	  difference except in very rare cases. That case would be that all
	  of the channels in autoservice are not generating any frames. In
	  that case, this change reduces the potential amount of time that
	  a thread waits in ast_autoservice_stop() for the autoservice
	  thread to wrap back around to the beginning of its loop. (closes
	  issue #12266, reported by dimas)

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  110335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
	  | 6 lines Fix some very broken code that was introduced in 1.2.26
	  as a part of the security fix. The dnsmgr is not appropriate
	  here. The dnsmgr takes a pointer to an address structure that a
	  background thread continuously updates. However, in these cases,
	  a stack variable was passed. That means that the dnsmgr thread
	  would be continuously writing to bogus memory. ........

	* apps/app_meetme.c: Fix a bug where when calls on the trunk side
	  hang up while on hold, the state is not properly reflected.
	  (closes issue #11990, reported by anakaoka, patched by me)

2008-03-19 20:33 +0000 [r110083]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Add a missing unlock in the case that memory
	  allocation fails in app_chanspy. Thanks to Russell for confirming
	  that this was an issue.

2008-03-19 19:11 +0000 [r110019-110035]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Add sanity checking for position resuming.
	  We *have* to make sure that the position does not exceed the
	  total number of files present, and we have to make sure that the
	  position's filename is the same as previous. These values can
	  change if a music class is reloaded and give unpredictable
	  behavior. (closes issue #11663) Reported by: junky

	* main/rtp.c: Make sure that the mark bit does not incorrectly
	  cause video frame timestamps to be calculated as if they are
	  audio frames. (closes issue #11429) Reported by: sperreault
	  Patches: 11429-frametype.diff uploaded by qwell (license 4)

2008-03-19 17:12 +0000 [r109973]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
	  (added): People report bugs about Asterisk crashing with DO_CRASH
	  enabled was getting a little silly... Now we only show certain
	  cflags when you run configure with --enable-dev-mode
	  (corresponding menuselect change to follow)

2008-03-19 15:41 +0000 [r109908]  Steve Murphy <murf@digium.com>

	* main/config.c: (closes issue #11442) Reported by: tzafrir
	  Patches: 11442.patch uploaded by murf (license 17) Tested by:
	  murf I didn't give tzafrir very much time to test this, but if he
	  does still have remaining issues, he is welcome to re-open this
	  bug, and we'll do what is called for. I reproduced the problem,
	  and tested the fix, so I hope I am not jumping by just going
	  ahead and committing the fix. The problem was with what file_save
	  does with templates; firstly, it tended to print out multiple
	  options: [my_category](!)(templateref) instead of
	  [my_category](!,templateref) which is fixed by this patch.
	  Nextly, the code to suppress output of duplicate declarations
	  that would occur because the reader copies inherited declarations
	  down the hierarchy, was not working. Thus: [master-template](!)
	  mastervar = bar [template](!,master-template) tvar = value
	  [cat](template) catvar = val would be rewritten as: ;! ;!
	  Automatically generated configuration file ;! Filename:
	  experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
	  Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
	  [master-template](!) mastervar = bar
	  [template](!,master-template) mastervar = bar tvar = value
	  [cat](template) mastervar = bar tvar = value catvar = val This
	  has been fixed. Since the config reader 'explodes' inherited vars
	  into the category, users may, in certain circumstances, see
	  output different from what they originally entered, but it should
	  be both correct and equivalent.

2008-03-19 04:06 +0000 [r109763-109838]  Russell Bryant <russell@digium.com>

	* main/utils.c: Tweak spacing in a recent change because I'm very
	  picky.

	* apps/app_chanspy.c: Fix one place where the chanspy datastore
	  isn't removed from a channel. (issue #12243, reported by atis,
	  patch by me)

2008-03-18 20:52 +0000 [r109713]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This patch makes it so that all queue member
	  status changes are handled through device state code. This
	  removes several problems people were seeing where their queue
	  members would get into an "unknown" state. Huge props go to atis
	  on this one since he was the one who found the code section that
	  was causing the problem and proposed the solution. I just wrote
	  what he suggested :) (closes issue #12127) Reported by: atis
	  Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
	  by: atis, jvandal

2008-03-18 19:23 +0000 [r109648]  Jason Parker <jparker@digium.com>

	* codecs/log2comp.h: Allow codecs that use log2comp (g726) to
	  compile correctly on x86 with gcc4 optimizations. (closes issue
	  #12253) Reported by: fossil Patches: log2comp.patch uploaded by
	  fossil (license 140)

2008-03-18 17:58 +0000 [r109575]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Make sure an agent doesn't try to send
	  dtmf to a NULL channel closes issue #12242 Reported by Yourname

2008-03-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.19-rc3 released.

2008-03-18 16:25 +0000 [r109482]  Terry Wilson <twilson@digium.com>

	* include/asterisk/astobj.h: Fix character string being treated ad
	  format string

2008-03-18 15:10 +0000 [r109393]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
	  3 lines Do not return with a successful authentication if the
	  From header ends up empty. (AST-2008-003) ........

2008-03-18 14:58 +0000 [r109386]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
	  number of payloads accepted, and also make sure a given payload
	  does not exceed our maximum value. (AST-2008-002)

2008-03-18 06:37 +0000 [r109309]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-ntest23 (added),
	  pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
	  pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
	  pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
	  pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
	  pbx/ael/ael-test/ael-ntest23/qq.ael (added),
	  pbx/ael/ael-test/ael-ntest23/t1 (added),
	  pbx/ael/ael-test/ael-ntest23/t2 (added),
	  pbx/ael/ael-test/ael-ntest23/t3 (added),
	  pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
	  issue #11903) Reported by: atis Many thanks to atis for spotting
	  this problem and reporting it. The fix was to straighten out how
	  items are placed on and removed from the file stack. Regressions
	  as well as the provided test case helped to straighten out all
	  code paths. valgrind was used to make sure all memory allocated
	  was freed. Sorry for not solving this earlier. I got distracted.
	  Added the ntest23 regression test, which is mainly a copy of
	  ntest22, but with a few juicy errors thrown in, to replicate the
	  kind of error that atis spotted.

2008-03-17 22:05 +0000 [r109226]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: Fix a logic flaw in the code that stores lock info
	  which is displayed via the "core show locks" command. The idea
	  behind this section of code was to remove the previous lock from
	  the list if it was a trylock that had failed. Unfortunately,
	  instead of checking the status of the previous lock, we were
	  referencing the index immediately following the previous lock in
	  the lock_info->locks array. The result of this problem, under the
	  right circumstances, was that the lock which we currently in the
	  process of attempting to acquire could "overwrite" the previous
	  lock which was acquired. While this does not in any way affect
	  typical operation, it *could* lead to misleading "core show
	  locks" output.

2008-03-17 17:55 +0000 [r109171]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Update the directory of placed calls on
	  skinny phones when dialing a channel that does not provide
	  progress (analog ZAP lines) The phone does handle the double
	  update on calls to channels that do provide progress and wont
	  insert duplicate items (closes issue #12239) Reported by: DEA
	  Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)

2008-03-17 16:24 +0000 [r109107]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: 200 OKs in response to a reinvite need to be
	  sent reliably. If the remote side does not receive one the dialog
	  will be torn down. (closes issue #12208) Reported by: atrash

2008-03-17 15:15 +0000 [r109057]  Jason Parker <jparker@digium.com>

	* main/file.c: Backport revision 106439 from trunk. I didn't
	  realize this was broken in 1.4 as well. Closes issue #12222.

2008-03-17 14:18 +0000 [r109012]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Make sure that we release the lock on the
	  spyee channel if the spyee or spy has hung up (closes issue
	  #12232) Reported by: atis

2008-03-16 21:47 +0000 [r108961]  Michiel van Baak <michiel@vanbaak.info>

	* main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
	  (closes issue #12228) Reported by: andrew Patches: SRC.patch
	  uploaded by andrew (license 240)

2008-03-14 20:09 +0000 [r108792-108796]  Russell Bryant <russell@digium.com>

	* channels/chan_oss.c: Fix a channel name issue. chan_oss registers
	  the "Console" channel type, but it created channels with an "OSS"
	  prefix. (closes issue #12194, reported by davidw, patched by me)

	* contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
	  start networking before asterisk, as well. (closes issue #12200,
	  reported by and change suggested by reinerotto)

2008-03-14 16:44 +0000 [r108737]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a race condition in the SIP packet
	  scheduler which could cause a crash. chan_sip uses the scheduler
	  API in order to schedule retransmission of reliable packets (such
	  as INVITES). If a retransmission of a packet is occurring, then
	  the packet is removed from the scheduler and retrans_pkt is
	  called. Meanwhile, if a response is received from the packet as
	  previously transmitted, then when we ACK the response, we will
	  remove the packet from the scheduler and free the packet. The
	  problem is that both the ACK function and retrans_pkt attempt to
	  acquire the same lock at the beginning of the function call. This
	  means that if the ACK function acquires the lock first, then it
	  will free the packet which retrans_pkt is about to read from and
	  write to. The result is a crash. The solution: 1. If the ACK
	  function fails to remove the packet from the scheduler and the
	  retransmit id of the packet is not -1 (meaning that we have not
	  reached the maximum number of retransmissions) then release the
	  lock and yield so that retrans_pkt may acquire the lock and
	  operate. 2. Make absolutely certain that the ACK function does
	  not recursively lock the lock in question. If it does, then
	  releasing the lock will do no good, since retrans_pkt will still
	  be unable to acquire the lock. (closes issue #12098) Reported by:
	  wegbert (closes issue #12089) Reported by: PTorres Patches:
	  12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
	  by: jvandal

2008-03-14 14:29 +0000 [r108682]  Jason Parker <jparker@digium.com>

	* res/res_musiconhold.c: Fix a potential segfault if chan (or
	  chan->music_state) is NULL. Closes issue #12210, credit to
	  edantie for pointing this out.

2008-03-13 21:38 +0000 [r108469-108583]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
	  Fix another issue that was causing crashes in chanspy. This
	  introduces a new datastore callback, called chan_fixup(). The
	  concept is exactly like the fixup callback that is used in the
	  channel technology interface. This callback gets called when the
	  owning channel changes due to a masquerade. Before this was
	  introduced, if a masquerade happened on a channel being spyed on,
	  the channel pointer in the datastore became invalid. (closes
	  issue #12187) (reported by, and lots of testing from atis) (props
	  to file for the help with ideas)

	* channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
	  phones to blink when an extension that has been subscribed to
	  goes on hold. Otherwise, they just stay on like it does when an
	  extension is in use. (closes issue #11263) Reported by: russell
	  Patches: notify_hold.rev1.txt uploaded by russell (license 2)
	  Tested by: russell

	* apps/app_followme.c: Fix a couple uses of sprintf. The second one
	  could actually cause an overflow of a stack buffer. It's not a
	  security issue though, it only depends on your configuration.

2008-03-12 21:53 +0000 [r108227-108288]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
	  for autocongestion in chan_sip. The scheduler callback will
	  always return 0. This means that this id is never rescheduled, so
	  it makes no sense to loop trying to delete the id from the
	  scheduler queue. If we fail to remove the item from the queue
	  once, it will fail every single time. (Yes I realize that in this
	  case, the macro would exit early because the id is set to -1 in
	  the callback, but it still makes no sense to use that macro in
	  favor of calling ast_sched_del once and being done with it) This
	  is the first of potentially several such fixes.

	* include/asterisk/sched.h: Added a large comment before the
	  AST_SCHED_DEL macro to explain its purpose as well as when it is
	  appropriate and when it is not appropriate to use it. I also
	  removed the part of the debug message that mentions that this is
	  probably a bug because there are some perfectly legitimate places
	  where ast_sched_del may fail to delete an entry (e.g. when the
	  scheduler callback manually reschedules with a new id instead of
	  returning non-zero to tell the scheduler to reschedule with the
	  same idea). I also raised the debug level of the debug message in
	  AST_SCHED_DEL since it seems like it could come up quite
	  frequently since the macro is probably being used in several
	  places where it shouldn't be. Also removed the redundant line,
	  file, and function information since that is provided by ast_log.

2008-03-12 19:57 +0000 [r108135]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c, main/channel.c: (closes issue #12187,
	  reported by atis, fixed by me after some brainstorming on the
	  issue with mmichelson) - Update copyright info on app_chanspy. -
	  Fix a race condition that caused app_chanspy to crash. The issue
	  was that the chanspy datastore magic that was used to ensure that
	  spyee channels did not disappear out from under the code did not
	  completely solve the problem. It was actually possible for
	  chanspy to acquire a channel reference out of its datastore to a
	  channel that was in the middle of being destroyed. That was
	  because datastore destruction in ast_channel_free() was done near
	  the end. So, this left the code in app_chanspy accessing a
	  channel that was partially, or completely invalid because it was
	  in the process of being free'd by another thread. The following
	  sort of shows the code path where the race occurred:
	  =============================================================================
	  Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
	  --------------------------------------||-------------------------------------
	  ast_channel_free() || - remove channel from channel list || -
	  lock/unlock the channel to ensure || that no references retrieved
	  from || the channel list exist. ||
	  --------------------------------------||-------------------------------------
	  || channel_spy() - destroy some channel data || - Lock chanspy
	  datastore || - Retrieve reference to channel || - lock channel ||
	  - Unlock chanspy datastore
	  --------------------------------------||-------------------------------------
	  - destroy channel datastores || - call chanspy datastore d'tor ||
	  which NULL's out the ds' || - Operate on the channel ...
	  reference to the channel || || - free the channel || || || -
	  unlock the channel
	  --------------------------------------||-------------------------------------
	  =============================================================================

2008-03-12 19:16 +0000 [r108086]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: if we receive an INVITE with a
	  Content-Length that is not a valid number, or is zero, then don't
	  process the rest of the message body looking for an SDP closes
	  issue #11475 Reported by: andrebarbosa

2008-03-12 18:26 +0000 [r108083]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, include/asterisk/audiohook.h,
	  main/audiohook.c: Add a trigger mode that triggers on both read
	  and write. The actual function that returns the combined audio
	  frame though will wait until both sides have fed in audio, or
	  until one side stops (such as the case when you call Wait).
	  (closes issue #11945) Reported by: xheliox

2008-03-12 16:59 +0000 [r108031]  Russell Bryant <russell@digium.com>

	* main/channel.c: Destroy the channel lock after the channel
	  datastores. (inspired by issue #12187)

2008-03-12 01:52 +0000 [r107877]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
	  Document all of the possible realtime fields

2008-03-11 23:37 +0000 [r107714-107826]  Jason Parker <jparker@digium.com>

	* doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
	  ODBC voicemail. (closes issue #12186) Reported by: jsmith
	  Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
	  15)

	* channels/chan_gtalk.c: Copy voicemail dependency logic for
	  res_adsi to chan_gtalk (for jabber). (closes issue #12014)
	  Reported by: junky

2008-03-11 20:48 +0000 [r107713]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile.rules, channels/Makefile: get chan_vpb to build properly
	  in dev mode

2008-03-11 20:47 +0000 [r107712]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Add a newline on a log

2008-03-11 19:20 +0000 [r107582-107646]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Make sure the visible indication is on the
	  right channel so when the masquerade happens the proper
	  indication is enacted. (closes issue #11707) Reported by: iam

	* apps/app_meetme.c: Add an additional check for setting conference
	  parameter when using the marked user options. It was possible for
	  it to return to a no listen/no talk state if a masquerade
	  happened. (closes issue #12136) Reported by: aragon

	* apps/app_exec.c: Fix a minor spelling error. (closes issue
	  #12183) Reported by: darrylc

2008-03-11  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.19-rc2 released.

2008-03-11 15:18 +0000 [r107352-107472]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_rpt.c: backport a fix from trunk

	* channels/misdn/isdn_lib.c, codecs/Makefile,
	  channels/chan_misdn.c: fix various other problems found by gcc
	  4.3

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  apps/app_sms.c: stop checking for mktime() in the configure
	  script... we don't use it, and the test is buggy under gcc 4.3

	* configure, main/Makefile, configure.ac, makeopts.in: check for
	  compiler support for -fno-strict-overflow before using it (tested
	  with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
	  Reported by: Netview

	* configure, configure.ac: fix small bug in IMAP toolkit testing

	* main/udptl.c, utils/Makefile, main/Makefile,
	  main/editline/readline.c, pbx/Makefile: fix up various compiler
	  warnings found with gcc-4.3: - the output of flex includes a
	  static function called 'input' that is not used, so for the
	  moment we'll stop having the compiler tell us about unused
	  variables in the flex source files (a better fix would be to
	  improve our flex post-processing to remove the unused function) -
	  main/stdtime/localtime.c makes assumptions about signed integer
	  overflow, and gcc-4.3's improved optimizer tries to take
	  advantage of handling potential overflow conditions at compile
	  time; for now, suppress these optimizations until we can fiure
	  out if the code needs improvement - main/udptl.c has some
	  references to uninitialized variables; in one case there was no
	  bug, but in the other it was certainly possibly for unexpected
	  behavior to occur - main/editline/readline.c had an unused
	  variable

2008-03-11 00:59 +0000 [r107290]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: If we fail to alloc a channel, we should
	  re-lock the pvt structure before returning.

2008-03-10 21:32 +0000 [r107230]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Use non-global storage for eswitch

2008-03-10 20:27 +0000 [r107173]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Make sure to reenable echo can after a
	  "failed" (canceled, etc) three-way call. (closes issue #11335)
	  Reported by: rebuild

2008-03-10 20:17 +0000 [r107099-107161]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Fix another bug specifically related to asynchronous
	  call origination. Once the PBX is started on the channel using
	  ast_pbx_start(), then the ownership of the channel has been
	  passed on to another thread. We can no longer access it in this
	  code. If the channel gets hung up very quickly, it is possible
	  that we could access a channel that has been free'd. (inspired by
	  BE-386)

	* main/pbx.c: Fix some bugs related to originating calls. If the
	  code failed to start a PBX on the channel (such as if you set a
	  call limit based on the system's load average), then there were
	  cases where a channel that has already been free'd using
	  ast_hangup() got accessed. This caused weird memory corruption
	  and crashes to occur. (fixes issue BE-386) (much debugging credit
	  goes to twilson, final patch written by me)

	* main/channel.c: Resolve a compiler warning.

	* main/channel.c: Fix a race condition where the generator can go
	  away (closes issue #12175, reported by edantie, patched by me)

2008-03-10 14:33 +0000 [r107016]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
	  unanswered CDRs are dropped to the CDR core, not everything uses
	  app_dial. (closes issue #11516) Reported by: ys Patches:
	  branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
	  anest, jcapp, dartvader

2008-03-08 15:59 +0000 [r106945]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: don't generate D-Channel "up" and "down"
	  messages unless the channel state is actually changing; also,
	  generate the "up" message when an implicit "up" occurs due to
	  reception of a normal event when we thought the channel was
	  "down"

2008-03-07 22:51 +0000 [r106895]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Only start the SLA thread if SLA has actually
	  been configured.

2008-03-07 22:14 +0000 [r106842]  Jason Parker <jparker@digium.com>

	* main/editline/Makefile.in: Fix hardcoded grep in editline, were
	  GNU grep is required. (closes issue #12124) Reported by: dmartin

2008-03-07 19:32 +0000 [r106788]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Ignore source update control frame. (closes issue
	  #12168) Reported by: plack

2008-03-07 17:16 +0000 [r106704]  Russell Bryant <russell@digium.com>

	* include/asterisk/sched.h: Change a warning message to a debug
	  message. This is happening quite frequently, and it is not worth
	  spamming users with these messages unless we are pretty confident
	  that it should never happen. As it stands today, it _will_ and
	  _does_ happen and until that gets cleaned up a reasonable amount
	  on the development side, let's not spam the logs of everyone
	  else. (closes issue #12154)

2008-03-07 16:22 +0000 [r106552-106635]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Warn the user when a temporary greeting
	  exists (Closes issue #11409)

	* main/rtp.c: Properly initialize rtp->schedid (Closes issue
	  #12154)

	* apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
	  apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
	  funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
	  main/manager.c: Safely use the strncat() function. (closes issue
	  #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
	  uploaded by Corydon76 (license 14)

2008-03-06 22:10 +0000 [r106437]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Quell an annoying message that is likely to print
	  every single time that ast_pbx_outgoing_app is called. The reason
	  is that __ast_request_and_dial allocates the cdr for the channel,
	  so it should be expected that the channel will have a cdr on it.
	  Thanks to joetester on IRC for pointing this out

2008-03-06 04:40 +0000 [r106328]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile: Upgrade to the next release of sounds

2008-03-05 22:37 +0000 [r106237]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a potential deadlock and a few
	  different potential crashes. (closes issue #12145, reported by
	  thiagarcia, patched by me)

2008-03-05 22:32 +0000 [r106235]  Joshua Colp <jcolp@digium.com>

	* channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
	  apps/app_dial.c, main/channel.c, channels/chan_phone.c,
	  main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
	  channels/chan_alsa.c, apps/app_followme.c,
	  include/asterisk/frame.h: Add a control frame to indicate the
	  source of media has changed. Depending on the underlying
	  technology it may need to change some things. (closes issue
	  #12148) Reported by: jcomellas

2008-03-05 21:12 +0000 [r106178]  Michiel van Baak <michiel@vanbaak.info>

	* doc/realtime.txt: document var_metric so no bugreports will come
	  in when it's actually a configuration issue. (issue #12151)
	  Reported and patched by: caio1982 1.4 patch by me

2008-03-05 15:32 +0000 [r106038]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: when a PRI call must be moved to a different
	  B channel at the request of the other endpoint, ensure that any
	  DSP active on the original channel is moved to the new one
	  (closes issue #11917) Reported by: mavetju Tested by: mavetju

2008-03-05 15:17 +0000 [r106015]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, include/asterisk/sched.h: Correctly
	  initialize retransid in SIP, and ensure that the warning when
	  failing to delete a schedule entry can actually hit the log.
	  (closes issue #12140) Reported by: slavon Patches: sch2.patch
	  uploaded by slavon (license 288) (Patch slightly modified by me)

2008-03-05 01:52 +0000 [r105932]  Russell Bryant <russell@digium.com>

	* main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
	  that I just noticed in the RTP code. The calculation for setting
	  the len field in an ast_frame of audio was wrong when G.722 is in
	  use. The len field represents the number of ms of audio that the
	  frame contains. It would have set the value to be twice what it
	  should be.

2008-03-04 18:10 +0000 [r105674-105676]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: In addition to setting the marker bit let's change
	  our ssrc so they know for sure it is a different source.

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
	  new source of audio comes in (such as music on hold) make sure
	  the marker bit gets set. (closes issue #10355) Reported by:
	  wdecarne Patches: 10355.diff uploaded by file (license 11)
	  (closes issue #11491) Reported by: kanderson

2008-03-04  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.19-rc1 released.

2008-03-04 04:31 +0000 [r105591]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Backport a minor bug fix from trunk that I found
	  while doing random code cleanup. Properly break out of the loop
	  when a context isn't found when verify that includes are valid.

2008-03-03 18:06 +0000 [r105572]  Jason Parker <jparker@digium.com>

	* res/snmp/agent.c: Fix type for astNumChannels. (closes issue
	  #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
	  (license 192)

2008-03-03 17:16 +0000 [r105563-105570]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c: In the case of an ast_channel allocation
	  failure, take the local_pvt out of the pvt list before destroying
	  it.

	* channels/chan_local.c: Fix a potential memory leak of the
	  local_pvt struct when ast_channel allocation fails. Also, in
	  passing, centralize the code necessary to destroy a local_pvt.

	* main/autoservice.c: Update the copyright information for
	  autoservice. Most of the code in this file now is stuff that I
	  have written recently ...

	* main/asterisk.c, main/channel.c, include/asterisk.h,
	  main/autoservice.c: Merge in some changes from
	  team/russell/autoservice-nochans-1.4 These changes fix up some
	  dubious code that I came across while auditing what happens in
	  the autoservice thread when there are no channels currently in
	  autoservice. 1) Change it so that autoservice thread doesn't keep
	  looping around calling ast_waitfor_n() on 0 channels twice a
	  second. Instead, use a thread condition so that the thread
	  properly goes to sleep and does not wake up until a channel is
	  put into autoservice. This actually fixes an interesting bug, as
	  well. If the autoservice thread is already running (almost always
	  is the case), then when the thread goes from having 0 channels to
	  have 1 channel to autoservice, that channel would have to wait
	  for up to 1/2 of a second to have the first frame read from it.
	  2) Fix up the code in ast_waitfor_nandfds() for when it gets
	  called with no channels and no fds to poll() on, such as was the
	  case with the previous code for the autoservice thread. In this
	  case, the code would call alloca(0), and pass the result as the
	  first argument to poll(). In this case, the 2nd argument to
	  poll() specified that there were no fds, so this invalid pointer
	  shouldn't actually get dereferenced, but, this code makes it
	  explicit and ensures the pointers are NULL unless we have valid
	  data to put there. (related to issue #12116)

2008-03-03 15:28 +0000 [r105557-105560]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: It is possible for no audio to pass between the
	  current digit and next digit so expand logic that clears
	  emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
	  edgreenberg Patches: v1-11911.patch uploaded by dimas (license
	  88) Tested by: tbsky

	* channels/chan_sip.c: Add a comment to describe some logic.
	  (closes issue #12120) Reported by: flefoll Patches:
	  chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
	  244)

2008-02-29 23:34 +0000 [r105409]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Fix a major bug in autoservice. There was a
	  race condition in the handling of the list of channels in
	  autoservice. The problem was that it was possible for a channel
	  to get removed from autoservice and destroyed, while the
	  autoservice thread was still messing with the channel. This led
	  to memory corruption, and caused crashes. This explains multiple
	  backtraces I have seen that have references to autoservice, but
	  do to the nature of the issue (memory corruption), could cause
	  crashes in a number of areas. (fixes the crash in BE-386) (closes
	  issue #11694) (closes issue #11940) The following issues could be
	  related. If you are the reporter of one of these, please update
	  to include this fix and try again. (potentially fixes issue
	  #11189) (potentially fixes issue #12107) (potentially fixes issue
	  #11573) (potentially fixes issue #12008) (potentially fixes issue
	  #11189) (potentially fixes issue #11993) (potentially fixes issue
	  #11791)

2008-02-29 14:47 +0000 [r105326]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Fix a potential memory leak

2008-02-29 14:34 +0000 [r105296]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: If the message file does not exist, just
	  return harmlessly, instead of crashing. (Closes issue #12108)

2008-02-29 13:48 +0000 [r105261]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Bump up the size of the uniqueid variable.
	  (closes issue #12107) Reported by: asgaroth

2008-02-29 13:05 +0000 [r105209]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Automatically create new buddy upon reception
	  of a presence stanza of type subscribed. (closes issue #12066)
	  Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
	  phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
	  (license 73) Tested by: ffadaie, phsultan

2008-02-28 22:23 +0000 [r105116]  Russell Bryant <russell@digium.com>

	* main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
	  tracking code that was discovered by mmichelson. The issue is
	  that if the lock history array was full, then the functions to
	  mark a lock as acquired or not would adjust the stats for
	  whatever lock is at the end of the array, which may not be
	  itself. So, do a sanity check to make sure that we're updating
	  lock info for the proper lock. (This explains the bizarre stats
	  on lock #63 in BE-396, thanks Mark!)

2008-02-28 21:56 +0000 [r105113]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk: Update init script for LSB
	  compat (closes issue #9843) Reported by: ibc Patches:
	  rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
	  paravoid

2008-02-28 20:11 +0000 [r105059]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: When using autofill, members who are in use
	  should be counted towards the number of available members to call
	  if ringinuse is set to yes. Thanks to jmls who brought this issue
	  up on IRC

2008-02-28 19:20 +0000 [r104920-105005]  Jason Parker <jparker@digium.com>

	* main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
	  applications, if we get NULL. This protects against possible
	  segfaults in applications that may try to use data before
	  checking length (ast_strdupa'ing it, for example) (closes issue
	  #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
	  uploaded by qwell (license 4)

	* channels/chan_skinny.c: According to a video at www.cisco.com,
	  the 7921G supports 6 line appearances.

2008-02-28 00:05 +0000 [r104868]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, build_tools/strip_nonapi: Compatibility fix for
	  PPC64 (closes issue #12081) Reported by: jcollie Patches:
	  asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
	  Tested by: jcollie, Corydon76

2008-02-27 21:49 +0000 [r104841]  Mark Michelson <mmichelson@digium.com>

	* main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
	  lockable list. This is because in some cases, the ast_dial may
	  exist in multiple threads due to asynchronous execution of its
	  application, and I found some cases where race conditions could
	  exist. 2. Check in ast_dial_join to be sure that the channel
	  still exists before attempting to lock it, since it could have
	  gotten hung up but the is_running_app flag on the
	  ast_dial_channel may not have been cleared yet. (closes issue
	  #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
	  putnopvut (license 60) Tested by: jvandal

2008-02-27 20:56 +0000 [r104787]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Don't loop around infinitely trying to spy on
	  our own channel, and don't forget to free/detach the datastore
	  upon hangup of the spy.

2008-02-27 20:36 +0000 [r104783]  Mark Michelson <mmichelson@digium.com>

	* main/file.c: Bump a couple of more buffers up by 2 so that
	  annoying warnings aren't generated like crazy on every
	  fileexists_core call.

2008-02-27 18:15 +0000 [r104704]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Ensure the session ID can't be 0.

2008-02-27 17:41 +0000 [r104665]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Bump up the buffer by 2.

2008-02-27 17:33 +0000 [r104625]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
	  stuck in an infinite loop without being able to detect that the
	  calling channel hung up. (closes issue #12076, reported by junky,
	  patched by me)

2008-02-27 17:26 +0000 [r104598]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Inherit language from the transfering channel
	  on a blind transfer. (closes issue #11682) Reported by: caio1982
	  Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
	  (license 22) Tested by: caio1982, victoryure

2008-02-27 17:07 +0000 [r104596]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: Use the lock (which already existed, it just
	  wasn't used) on the updaters list to protect the contents instead
	  of the overall module list lock. (closes issue #12080) Reported
	  by: ChaseVenters

2008-02-27 16:53 +0000 [r104593]  Kevin P. Fleming <kpfleming@digium.com>

	* main/file.c: fallback to standard English prompts properly when
	  using new prompt directory layout (closes issue #11831) Reported
	  by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
	  20) (modified by me to improve code and conform rest of function
	  to coding guidelines)

2008-02-27 16:45 +0000 [r104591]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: When we receive a known alarm, make sure
	  that the unknown alarm flag is not still set to make sure that
	  when we come back out of alarm, it gets reported in the log and
	  manager interface (after discussion with tzafrir on the -dev
	  list)

2008-02-27 15:52 +0000 [r104536]  Joshua Colp <jcolp@digium.com>

	* res/res_smdi.c: Only stop the MWI monitor thread if it was
	  actually started. (closes issue #12086) Reported by: francesco_r

2008-02-27 01:15 +0000 [r104332-104334]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: Avoid some recursion in the cleanup code for
	  the chanspy datastore (closes issue #12076, reported by junky,
	  patched by me)

	* channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
	  an alarm. However, Asterisk 1.4 does not know what to do with
	  these alarms. Only Asterisk 1.6 cares about it. So, if we get an
	  unknown alarm in chan_zap, don't generate confusing log messages
	  about it.

2008-02-26 18:26 +0000 [r104132-104141]  Jason Parker <jparker@digium.com>

	* Makefile: Add badshell to .PHONY target (thanks Kevin)

	* Makefile: Since all shells aren't as awesome as bash, we have to
	  fail if somebody tries to use a literal "~" in DESTDIR.

	* sounds/Makefile: Revert previous abspath change. ...abspath is
	  new in GNU make 3.81. I feel so...defeated. Must find new fix!

	* sounds/Makefile: Fix a very bizarre issue we were seeing with our
	  buildbot when using a DESTDIR that wasn't an absolute path (such
	  as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
	  that some of the targets were being expanded to the full path, so
	  $@ ended up being /root/asterisk-1.4/[...]/ rather than
	  ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
	  in GNU make. (*cough*
	  http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)

2008-02-26 00:25 +0000 [r104119]  Russell Bryant <russell@digium.com>

	* include/asterisk/smdi.h, apps/app_voicemail.c,
	  channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
	  Merge changes from team/russell/smdi-1.4 This commit brings in a
	  significant set of changes to the SMDI support in Asterisk. There
	  were a number of bugs in the current implementation, most notably
	  being that it was very likely on busy systems to pop off the
	  wrong message from the SMDI message queue. So, this set of
	  changes fixes the issues discovered as well as introducing some
	  new ways to use the SMDI support which are required to avoid the
	  bugs with grabbing the wrong message off of the queue. This code
	  introduces a new interface to SMDI, with two dialplan functions.
	  First, you get an SMDI message in the dialplan using
	  SMDI_MSG_RETRIEVE() and then you access details in the message
	  using the SMDI_MSG() function. A side benefit of this is that it
	  now supports more than just chan_zap. For example, with this
	  implementation, you can have some FXO lines being terminated on a
	  SIP gateway, but the SMDI link in Asterisk. Another issue with
	  the current implementation is that it is quite common that the
	  station ID that comes in on the SMDI link is not necessarily the
	  same as the Asterisk voicemail box. There are now additional
	  directives in the smdi.conf configuration file which let you map
	  SMDI station IDs to Asterisk voicemail boxes. Yet another issue
	  with the current SMDI support was related to MWI reporting over
	  the SMDI link. The current code could only report a MWI change
	  when the change was made by someone calling into voicemail. If
	  the change was made by some other entity (such as with IMAP
	  storage, or with a web interface of some kind), then the MWI
	  change would never be sent. The SMDI module can now poll for MWI
	  changes if configured to do so. This work was inspired by and
	  primarily done for the University of Pennsylvania. (also related
	  to issue #9260)

2008-02-26 00:03 +0000 [r104111]  Jason Parker <jparker@digium.com>

	* channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
	  (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
	  uploaded by putnopvut (license 60)

2008-02-25 23:42 +0000 [r104102-104106]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: This patch fixes some pretty significant
	  problems with how app_chanspy handles pointers to channels that
	  are being spied upon. It was very likely that a crash would occur
	  if the channel being spied upon hung up. This was because the
	  current ast_channel handling _requires_ that the object is locked
	  or else it could disappear at any time (except in the owning
	  channel thread). So, this patch uses some channel datastore magic
	  on the spied upon channel to be able to detect if and when the
	  channel goes away. (closes issue #11877) (patch written by me,
	  but thanks to kpfleming for the idea, and to file for review)

	* main/utils.c: Improve the lock tracking code a bit so that a
	  bunch of old locks that threads failed to lock don't sit around
	  in the history. When a lock is first locked, this checks to see
	  if the last lock in the list was one that was failed to be
	  locked. If it is, then that was a lock that we're no longer
	  sitting in a trylock loop trying to lock, so just remove it.
	  (inspired by issue #11712)

2008-02-25 21:37 +0000 [r104095]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make it so a users.conf user creates both a
	  SIP peer and a SIP user. The user will be used for inbound
	  authentication for the device, and peer will be used for placing
	  calls to the device. (closes issue #9044) Reported by: queuetue
	  Patches: sip-gui-friend.diff uploaded by qwell (license 4)

2008-02-25 21:31 +0000 [r104094]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: If the destination folder is full, don't
	  delete a message when exiting. (closes issue #12065) Reported by:
	  selsky Patch by: (myself)

2008-02-25 20:49 +0000 [r104092]  Jason Parker <jparker@digium.com>

	* main/config.c: Allow the use of #include and #exec in situations
	  where the max include depth was only 1. Specifically, this fixes
	  using #include and #exec in extconfig.conf. This was basically
	  caused because the config file itself raises the include level to
	  1. I opted not to raise the include limit, because recursion here
	  could cause very bizarre behavior. Pointed out, and tested by
	  jmls (closes issue #12064)

2008-02-25 18:38 +0000 [r104086]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Ensure that the channel doesn't disappear
	  in agent_logoff(). If it does, it could cause a crash. (fixes the
	  crash reported in BE-396)

2008-02-25 16:16 +0000 [r104082-104084]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If a resubscription comes in for a dialog we
	  no longer know about tell the remote side that the dialog does
	  not exist so they subscribe again using a new dialog. (closes
	  issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
	  uploaded by file (license 11)

	* channels/chan_sip.c: Due to recent changes tag will no longer be
	  NULL if not present so we have to use ast_strlen_zero to see if
	  it's actually blank. (closes issue #12061) Reported by: flefoll
	  Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
	  flefoll (license 244)

2008-02-22 22:45 +0000 [r104037]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Backwards debug message. (closes issue
	  #12052) Reported by: flefoll Patches:
	  chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
	  244)

2008-02-21 21:05 +0000 [r104026-104027]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: And as a followup to revision 104026,
	  completely remove event-related calls from a section of code
	  where we know there was no event to handle or get.

	* channels/chan_zap.c: Remove an incorrect debug message. It
	  reported that it had received a specific event and tried to
	  report which event was received. What actually was happening was
	  that it was reporting the number of bytes returned from a call to
	  read(). Thanks to Jared Smith for bringing the issue up on IRC

2008-02-21 14:33 +0000 [r104015]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: reduce the likelihood that HTTP Manager session
	  ids will consist of primarily '1' bits

2008-02-20 22:32 +0000 [r103956]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Clear up confusion when viewing the
	  QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
	  user's perspective, the queue does exist, we shouldn't tell them
	  we couldn't find the queue. Instead since it is a dead queue,
	  report a 0 waiting count This issue was brought up on IRC by jmls

2008-02-20 22:06 +0000 [r103953]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Don't wait for additional digits when
	  overlap dialing is enabled if the setup message contains the
	  sending_complete information element. (closes issue #11785)
	  Reported by: klaus3000 Patches:
	  sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
	  klaus3000 (license 65)

2008-02-20 21:40 +0000 [r103904]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Fix a crash if the channel becomes NULL
	  while attempting to lock it. (closes issue #12039) Reported by:
	  danpwi

2008-02-20 17:53 +0000 [r103845]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Compat fix for Solaris (closes issue
	  #12022) Reported by: asgaroth Patches:
	  20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: asgaroth

2008-02-19 20:28 +0000 [r103823]  Joshua Colp <jcolp@digium.com>

	* channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
	  (closes issue #11241) Reported by: tusar Patches:
	  h323id_as_callerid_name.patch uploaded by tusar (license 344)

2008-02-19 20:02 +0000 [r103821]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c: Account for the fact that the "other"
	  channel can disappear while the local pvt is not locked. (fixes a
	  problem introduced in rev 100581) (closes issue #12012) Reported
	  by: stevedavies Patch by me

2008-02-19 17:31 +0000 [r103807-103812]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Don't look for launchd when cross
	  compiling. (closes issue #12029) Reported by: ovi

	* channels/chan_sip.c: Fix building of chan_sip.

2008-02-19 10:27 +0000 [r103806]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure we send error replies correctly by
	  checking the via header.

2008-02-18 23:56 +0000 [r103801]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Ensure that emulated DTMFs do not get interrupted
	  by another begin frame. (closes issue #11740) Reported by: gserra
	  Patches: v1-11740.patch uploaded by dimas (license 88) (closes
	  issue #11955) Reported by: tsearle (closes issue #10530) Reported
	  by: xmarksthespot

2008-02-18 22:28 +0000 [r103790-103795]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Fix previous commit so that we actually
	  disable echocanbridged if echocancel is off.

	* channels/chan_zap.c: Correct a message when echocancelwhenbridged
	  is on, but echocancel is not. Issue #12019

2008-02-18 20:52 +0000 [r103786]  Mark Michelson <mmichelson@digium.com>

	* main/app.c: There was an invalid assumption when calculating the
	  duration of a file that the filestream in question was created
	  properly. Unfortunately this led to a segfault in the situation
	  where an unknown format was specified in voicemail.conf and a
	  voicemail was recorded. Now, we first check to be sure that the
	  stream was written correctly or else assume a zero duration.
	  (closes issue #12021) Reported by: jakep Tested by: putnopvut

2008-02-18 17:31 +0000 [r103780]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c, channels/chan_sip.c: When a SIP channel is being
	  auto-destroyed, it's possible for it to still be in bridge code.
	  When that happens, we crash. Delay the RTP destruction until the
	  bridge is ended. (closes issue #11960) Reported by: norman
	  Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
	  (license 14) Tested by: norman

2008-02-18 16:37 +0000 [r103770]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_zap.c: Fix a linked list corruption that under the
	  right circumstances could lead to a looped list, meaning it will
	  traverse forever. (closes issue #11818) Reported by: michael-fig
	  Patches: 11818.patch uploaded by putnopvut (license 60) Tested
	  by: michael-fig

2008-02-18 16:11 +0000 [r103763-103768]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c: Backport fix from issue #9325. (closes issue
	  #11980) Reported by: rbrunka

	* channels/chan_sip.c: Don't care if the extension given doesn't
	  exist for subscription based MWI.

2008-02-15 23:31 +0000 [r103726-103741]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
	  condition (closes issue #11780) Reported by: guillecabeza
	  Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
	  380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
	  380)

	* main/loader.c: In the case that you try to directly reload a
	  module has returned AST_MODULE_LOAD_DECLINE, log a message
	  indicating that the module is not fully initialized and must be
	  initialized using "module load".

	* main/loader.c: Don't attempt to execute the reload callback for a
	  module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
	  that was reported against chan_console in trunk. (closes issue
	  #11953, reported by junky, fixed by me)

2008-02-15 17:26 +0000 [r103688-103722]  Mark Michelson <mmichelson@digium.com>

	* doc/imapstorage.txt, configure, configure.ac: Final round of
	  changes for configure script logic for IMAP Now if a directory is
	  specified, then we will search that directory for a source
	  installation of the IMAP toolkit. If none is found, then we will
	  use that directory as the basis for detecting a package
	  installation of the IMAP c-client. If that check fails, then
	  configure will fail.

	* configure, configure.ac: Fix a bit of wrong logic in the
	  configure script that caused problems when trying to configure
	  without IMAP. Patch suggestion from phsultan, but I modified it
	  slightly. (closes issue #12003) Reported by: pj Tested by:
	  putnopvut

	* doc/imapstorage.txt, configure, configure.ac: I apparently
	  misunderstood one of the requirements of this configure change.
	  Now, if a source directory is specified with the --with-imap
	  option, and a valid source installation is not detected there,
	  then configure will fail and will not check for a package
	  installation.

	* doc/imapstorage.txt: Make a small clarification in the
	  documentation

	* doc/imapstorage.txt: Update documentation regarding configuration
	  of IMAP

	* apps/app_voicemail.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Change to the
	  configure logic regarding IMAP. Prior to this commit, if you
	  wished to configure Asterisk with IMAP support, you would use the
	  --with-imap configure switch in one of the following two ways:
	  --with-imap=/some/directory would look in the directory specified
	  for a UW IMAP source installation --with-imap would assume that
	  you had imap-2004g installed in .. relative to the Asterisk
	  source With this set of changes the two above options still work
	  the same, but there are two new behaviors, too.
	  --with-imap=system will assume that you have -libc-client.so
	  where you store your shared objects and will attempt to find
	  c-client headers in your include path either in the imap or
	  c-client directory. If either of the two original methods of
	  specifying the imap option should fail, then the check for
	  --with-imap =system will be performed in addition. It is only
	  after this "system" check that failure can happen.

	* apps/app_voicemail.c: Fix build for non-IMAP builds

	* apps/app_voicemail.c: Fix the new message count if delete=yes
	  when using IMAP storage. (closes issue #11406) Reported by:
	  jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
	  50) Tested by: jaroth

2008-02-14 19:51 +0000 [r103683-103684]  Jason Parker <jparker@digium.com>

	* funcs/func_cdr.c: swap location for this..

	* funcs/func_cdr.c: Document the 'l' option to the CDR() function.
	  (Thanks voipgate for pointing out the option, and Leif for
	  providing text for it.) Closes issue #11695.

2008-02-13 06:25 +0000 [r103556-103607]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_agent.c: We aren't talking to ourselves; we're
	  talking to someone else. (closes issue #11771) Reported by:
	  msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
	  (license 22) Tested by: caio1982, msetim

	* apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
	  not loaded (which is a symbol dependency) (closes issue #11760)
	  Reported by: non-poster Patches: 20080114__bug11760.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: Corydon76,
	  non-poster, jamesgolovich

2008-02-12 22:24 +0000 [r103503-103504]  Jason Parker <jparker@digium.com>

	* main/asterisk.c: revert accidental change from last commit. oops

	* contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
	  that was impossible.

2008-02-12 15:09 +0000 [r103324-103385]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Even if no CallerID name or number has been
	  provided by the remote party still use the configured sip.conf
	  ones. (closes issue #11977) Reported by: pj

	* apps/app_meetme.c: If entering a conference with the 'w' option
	  ensure that we can't listen or speak until the marked user
	  appears. (closes issue #11835) Reported by: alanmcmillan

2008-02-11 17:05 +0000 [r103315]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/zapata.conf.sample: improve 2BCT documentation a bit
	  (thanks Jared)

2008-02-09 06:23 +0000 [r103197]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Commit fix for being unable to send
	  voicemail from VoiceMailMain Reported by: William F Acker (via
	  the -users mailing list) Patch by: Corydon76 (license 14)

2008-02-08 18:48 +0000 [r103070-103120]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Prevent a potential three-thread deadlock. Also
	  added a comment block to explicitly state the locking order
	  necessary inside app_queue. (closes issue #11862) Reported by:
	  flujan Patches: 11862.patch uploaded by putnopvut (license 60)
	  Tested by: flujan

	* channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
	  fails for Zaptel timing device. (closes issue #11891) Reported
	  by: tzafrir

2008-02-08 15:08 +0000 [r102968]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make sure the presence of dbsecret is
	  factored into user scoring. (closes issue #11952) Reported by:
	  bbhoss

2008-02-07 19:53 +0000 [r102858]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Specify which digit string was matched in
	  debug message. (closes issue #11949) Reported by: dimas Patches:
	  v1-feature-debug.patch uploaded by dimas (license 88)

2008-02-07 16:41 +0000 [r102807]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/zapata.conf.sample: document usage of 'transfer'
	  configuration option for ISDN PRI switch-side transfers

2008-02-06 17:59 +0000 [r102653-102725]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only consider a T.38-only INVITE compatible
	  if we have both a joint capability between us and them and if
	  they provided T.38.

	* main/global_datastores.c: Add missing header file and
	  ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
	  snuffy

2008-02-06 15:19 +0000 [r102651]  Russell Bryant <russell@digium.com>

	* configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
	  that it gets inherited by outbound channels. (due to a discussion
	  between me and a user via email)

2008-02-06 11:48 +0000 [r102627]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx/Makefile, res/Makefile: ensure that all remaining
	  multi-object modules are built using their proper CFLAGS and
	  include directory paths

2008-02-06 00:26 +0000 [r102576]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Move around some defines to unbreak ODBC
	  storage. (closes issue #11932) Reported by: snuffy

2008-02-05 20:02 +0000 [r102453]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
	  issue #11919) Reported by: eferro Patches:
	  mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
	  Tested by: eferro

2008-02-05 19:52 +0000 [r102450]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If a REGISTER attempt comes in that is a
	  retransmission of a previous REGISTER do not create a new nonce
	  value. (issue #BE-381)

2008-02-05 17:15 +0000 [r102425]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/Makefile: ensure that components of chan_misdn.so are
	  built using any special build options that the configure script
	  generated (reported by Philipp Kempgen on asterisk-dev)

2008-02-05 15:09 +0000 [r102378]  Joshua Colp <jcolp@digium.com>

	* res/res_clioriginate.c: Perform dialing asynchronously when using
	  the originate CLI command so the CLI does not appear to block.
	  (closes issue #11927) Reported by: bbhoss

2008-02-04 21:06 +0000 [r102214-102323]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, utils/muted.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
	  fix: OS X now deprecates the use of the daemon(3) API. (closes
	  issue #11908) Reported by: oej Patches:
	  20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76

	* funcs/func_strings.c: Missing braces. (closes issue #11912)
	  Reported by: dimas Patches: sprintf.patch uploaded by dimas
	  (license 88)

2008-02-03 16:38 +0000 [r102090-102142]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
	  (according to RFC 3261) (closes issue #9492) Reported by:
	  kryptolus Patches: bug9492.txt uploaded by oej (license 306)
	  Tested by: oej

	* channels/chan_sip.c: Handle ACK and CANCEL in an invite
	  transaction - even if we get INFO transactions during the actual
	  call setup. (closes issue #10567) Reported by: jacksch Tested by:
	  oej Patch by: oej inspired by suggestions from neutrino88 in the
	  bug tracker

2008-02-01 23:06 +0000 [r101989]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
	  out that even though G.722 is 16 kHz, it is supposed to specified
	  as 8 kHz in the RTP, and RTP timestamps are supposed to be
	  calculated based on 8 kHz. (Apparently this is due to a bug in a
	  spec, but people follow it anyway, because it's the spec ...)

2008-02-01 21:54 +0000 [r101894-101942]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
	  voicemail, and fixed several other bugs while I'm in the area.
	  (closes issue #11615) Reported by: jamessan Patches:
	  20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76, jamessan

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: Change detection of getifaddrs to use
	  AST_C_COMPILE_CHECK, backported from trunk (as suggested by
	  kpfleming)

2008-02-01 17:41 +0000 [r101822]  Jason Parker <jparker@digium.com>

	* apps/app_authenticate.c: Remove a needless (and incorrect) call
	  to feof() after fgets(). This would have exited the loop early if
	  you had an authentication file with no newline at the end.

2008-02-01 17:27 +0000 [r101818-101820]  Russell Bryant <russell@digium.com>

	* apps/app_authenticate.c: off by one error

	* apps/app_authenticate.c: Don't overwrite the last character of a
	  line if it's not a newline. This would happen if the last line in
	  the file doesn't have a newline. (pointed out by Qwell)

2008-02-01 15:55 +0000 [r101772]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/acl.c: Compatibility fix for OpenWRT (reported by Brian
	  Capouch via the mailing list)

2008-02-01 00:32 +0000 [r101693]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
	  strings for the case that no peer or hostname was provided, which
	  is the one part of the dial string that is absolutely required.
	  If it's not there, bail out. (closes issue #11897) Reported by
	  sokhapkin Patch by me

2008-02-01 00:06 +0000 [r101649]  Mark Michelson <mmichelson@digium.com>

	* apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
	  periods of no channel activity" (closes issue #9256) Reported by:
	  cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
	  uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
	  rjain

2008-01-31  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.18 released.

2008-01-31 23:10 +0000 [r101601]  Russell Bryant <russell@digium.com>

	* main/translate.c, main/file.c: Fix a couple of places where
	  ast_frfree() was not called on a frame that came from a
	  translator. This showed itself by g729 decoders not getting
	  released. Since the flag inside the translator frame never got
	  unset by freeing the frame to indicate it was no longer in use,
	  the translators never got destroyed, and thus the g729 licenses
	  were not released. (closes issue #11892) Reported by: xrg
	  Patches: 11892.diff uploaded by russell (license 2) Tested by:
	  xrg, russell

2008-01-31 21:00 +0000 [r101531]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
	  beginning of an absolute pathname. 2. If ast_monitor_change_fname
	  is called and the new filename is the same as the old, then exit
	  early and don't set the filename_changed field in the monitor
	  structure. Setting it in this case was causing ast_monitor_stop
	  to erroneously delete them. (closes issue #11741) Reported by:
	  garlew Tested by: putnopvut

2008-01-31 19:52 +0000 [r101482]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
	  for struct in_addr funkiness. Issue #11885, patch by snuffy.

2008-01-31 19:30 +0000 [r101480]  Steve Murphy <murf@digium.com>

	* main/pbx.c: closes issue #11845; that's the one where there's a
	  1004 byte cdr leak with every AMI Redirect to a zap channel

2008-01-31 19:17 +0000 [r101413-101433]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Add more missing locking of the agents
	  list ...

	* channels/chan_agent.c: Move the locking from find_agent() into
	  the agent dialplan function handler to ensure that the agent
	  doesn't disappear while we're looking at it.

	* channels/chan_agent.c: Add missing locking to the find_agent()
	  function.

2008-01-30 15:41 +0000 [r101222]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Fix an issue where if a frame of higher
	  sample size preceeded a frame of lower sample size and
	  ast_slinfactory_read was called with a sample size of the
	  combined values or higher a crash would happen. (closes issue
	  #11878) Reported by: stuarth

2008-01-30 15:34 +0000 [r101219]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample: Change default config to use
	  descending channel order of groups, rather than ascending. Fixes
	  a potential source of confusion in glare-type situations. Issue
	  11875, reported by JimVanM.

2008-01-30 15:23 +0000 [r101216]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a logic error with regards to autofill.
	  Prior to this change, it was possible for a caller to go out of
	  turn if autofill were enabled and callers ahead in the queue were
	  attempting to call a member. This change fixes this.

2008-01-30 11:20 +0000 [r101152]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Stop musiconhold on attended transfer.
	  (closes issue #11872) Reported by: gareth Patches:
	  svn-101018.patch uploaded by gareth (license 208)

2008-01-29 23:50 +0000 [r101080]  Dwayne M. Hubbard <dhubbard@digium.com>

	* build_tools/make_version: updated build_tools to handle the
	  autotag directory structure changes; changes related to BE-353.
	  Patch by The Russell and reviewed by The Me.

2008-01-29 23:02 +0000 [r100973-101035]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Remove a memory leak from updating realtime
	  queues

	* apps/app_queue.c: Fixing an erroneous return value returned when
	  attempting to pause or unpause a queue member fails. Fixes
	  BE-366, thanks to John Bigelow for writing the patch.

2008-01-29 17:57 +0000 [r100934]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Don't forget to record the channel so we
	  know whether it is bridged or not later. (closes issue #11811)
	  Reported by: slavon

2008-01-29 17:43 +0000 [r100932]  Russell Bryant <russell@digium.com>

	* main/Makefile: Fix the last couple of issues related to building
	  from a path that contains spaces. (closes issue #11834)

2008-01-29 17:41 +0000 [r100930]  Jason Parker <jparker@digium.com>

	* channels/misdn_config.c: Initialize an array to 0s if config
	  option not specified. (closes issue #11860) Patches:
	  misdn_get_config.v1.diff uploaded by IgorG (license 20)

2008-01-29 17:21 +0000 [r100882-100922]  Russell Bryant <russell@digium.com>

	* Makefile: Use GNU make magic instead of shell magic to escape
	  spaces in the working directory. (related to issue #11834)

	* Makefile: Fix building Asterisk when the working path has spaces
	  in it. (closes issue #11834) Reported by: spendergrass Patched
	  by: me

2008-01-29 16:10 +0000 [r100835]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Allow zap groups above 30 to work properly.
	  (closes issue #11590) Reported by: tbsky

2008-01-29 10:36 +0000 [r100793]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed potential segfault in misdn show
	  channels CLI command

2008-01-29 08:26 +0000 [r100740]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: (closes issue #11736) Reported by: MVF
	  Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
	  oej, MVF, revolution (russellb: This was the showstopper for the
	  release.)

2008-01-28 21:02 +0000 [r100675]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
	  (went to 't' instead of 'T')

2008-01-28 20:55 +0000 [r100673]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
	  chan_vpb. It is alive and well.

2008-01-28 20:42 +0000 [r100672]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
	  greeting files into the database like we do with the others.
	  Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
	  by dimas (license 88)

2008-01-28 18:34 +0000 [r100626-100629]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: For some reason, the use of this strdupa()
	  is leading to memory corruption on freebsd sparc64. This trivial
	  workaround fixes it. (closes issue #10300, closes issue #11857,
	  reported by mattias04 and Home-of-the-Brave)

	* res/res_features.c: Fix a crash in ast_masq_park_call() (issue
	  #11342) Reported by: DEA Patches: res_features-park.txt uploaded
	  by DEA (license 3)

2008-01-28 18:23 +0000 [r100624]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Correct a comment which made little/no
	  sense.

2008-01-28 17:15 +0000 [r100581]  Russell Bryant <russell@digium.com>

	* main/channel.c, channels/chan_local.c,
	  include/asterisk/channel.h: Make some deadlock related fixes.
	  These bugs were discovered and reported internally at Digium by
	  Steve Pitts. - Fix up chan_local to ensure that the channel lock
	  is held before the local pvt lock. - Don't hold the channel lock
	  when executing the timing function, as it can cause a deadlock
	  when using chan_local. This actually changes the code back to be
	  how it was before the change for issue #10765. But, I added some
	  other locking that I think will prevent the problem reported
	  there, as well.

2008-01-27 21:59 +0000 [r100465]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
	  channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
	  channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
	  pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
	  the scheduler, ignoring the return value could possibly cause
	  memory to be accessed after it is freed, which causes all sorts
	  of random memory corruption. Instead, if a deletion fails, wait a
	  bit and try again (noting that another thread could change our
	  taskid value). (closes issue #11386) Reported by: flujan Patches:
	  20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76, flujan, stuarth`

2008-01-25 22:32 +0000 [r100418]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
	  now preferred that users of Voicetronix products use chan_zap in
	  combination with their zaptel drivers.

2008-01-25 21:24 +0000 [r100378]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: This would have never been true, since we're
	  passing (sizeof(req.data) - 1) as the len to recvfrom().

2008-01-24 21:57 +0000 [r100264]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/app.h: make these macros not assume that the
	  only other field in the structure is 'argc'... this is true when
	  someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
	  to define your own structure as long as it has the right fields

2008-01-24 17:22 +0000 [r100164]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Update main Asterisk copyright info to 2008

2008-01-24 16:41 +0000 [r100138]  Jason Parker <jparker@digium.com>

	* main/acl.c: Fix compilation on Solaris. (closes issue #11832)
	  Patches: bug-11832.diff uploaded by snuffy (license 35)

2008-01-23 21:07 +0000 [r99977-99978]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Second attempt. Don't change invitestate
	  when receiving 18x messages in CANCEL state. (issue #11736)
	  Reported by: MVF Patch by oej.

	* channels/chan_sip.c: Make sure we don't cancel destruction on
	  calls in CANCEL state, even if we get 183 while waiting for
	  answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
	  bug11736.txt uploaded by oej (license 306) Tested by: MVF

2008-01-23 20:25 +0000 [r99975]  Mark Michelson <mmichelson@digium.com>

	* apps/app_externalivr.c: Fixing a typo.

2008-01-23 17:46 +0000 [r99923]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
	  beginning of a list of channels to potentially spy on. However,
	  if there were no matching channels, it would beep at you over and
	  over, which is pretty annoying. Now, it will only beep once in
	  the case that there are no channels to spy on, but it will still
	  beep again once it reaches the beginning of the channel list
	  again. (closes issue #11738, patched by me)

2008-01-23 16:18 +0000 [r99878]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: These flag tests were illogical. They were
	  testing sip_peer flags on a sip_pvt. Thanks to Russell for
	  helping to get this odd problem figured out.

2008-01-23 04:31 +0000 [r99718-99777]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: When we reset the password via an external
	  command, we should also reset the password stored in the
	  in-memory list, too (otherwise it doesn't really take effect).
	  (closes issue #11809) Reported by: davetroy Patches:
	  fix_externpass.diff uploaded by davetroy (license 384)

	* res/res_odbc.c: Oops, should have checked for a NULL obj, here,
	  too

	* main/acl.c: Just confirmed that all current platforms need this
	  header file

2008-01-22 20:56 +0000 [r99652]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Thanks to Russell's education I realize that
	  BUFSIZ has changed since I learned the C language over 20 years
	  ago... Resetting chan_sip to the size of BUFSIZ that I expected
	  in my old head to avoid to heavy memory allocations on some
	  systems.

2008-01-22 20:34 +0000 [r99643]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Fix the defines for OS X (and Solaris, too)

2008-01-22 17:41 +0000 [r99592-99594]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
	  apps/app_followme.c: Add more dependencies on chan_local and add
	  a note to the description of chan_local so that people don't
	  disable it in menuselect just to clean up.

	* apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
	  still runs without chan_local, but will be missing forwarding
	  functionality.

2008-01-22 16:54 +0000 [r99540]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Ensure that we can get an address even when we don't
	  have a default route. (closes issue #9225) Reported by: junky
	  Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
	  (license 14) Tested by: oej, loloski, sergee

2008-01-22 15:08 +0000 [r99501]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Cleaning up some documentation that led to
	  confusion in a bug report

2008-01-21 23:55 +0000 [r99426]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Fixing an issue wherein monitoring local
	  channels was not possible. During a channel masquerade, the
	  monitors on the two channels involved are swapped. In 99% of the
	  cases this results in the desired effect. However, if monitoring
	  a local channel, this caused the monitor which was on the local
	  channel to get moved onto a channel which is immediately hung up
	  after the masquerade has completed. By swapping the monitors
	  prior to the masquerade, we avoid the problem by tricking the
	  masquerade into placing the monitor back onto the channel where
	  we want it. During the investigation of the issue, the channel's
	  monitor was the only thing that was swapped in such a manner
	  which did not make sense to have done. All other variable
	  swapping made sense.

2008-01-21 18:11 +0000 [r99341]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample,
	  include/asterisk/res_odbc.h: Permit the user to specify number of
	  seconds that a connection may remain idle, which fixes a crash on
	  reconnect with the MyODBC driver. (closes issue #11798) Reported
	  by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: mvanbaak

2008-01-21 16:01 +0000 [r99301]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Bump the buffer size for Via headers up to
	  512. There are some exceptionally large Via headers out there.
	  (closes issue #11783) Reported by: ofirroval

2008-01-19 10:05 +0000 [r99187]  Russell Bryant <russell@digium.com>

	* main/slinfactory.c: Fix a couple of memory leaks with frame
	  handling. Specifically, ast_frame_free() needed to be called on
	  the frame that came from the translator to signed linear.

2008-01-18 22:57 +0000 [r99127]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/channel.h: Remove the __ in front of the unused
	  variable. This causes some compilers to freak out.

2008-01-18 21:37 +0000 [r99079-99081]  Russell Bryant <russell@digium.com>

	* include/asterisk/translate.h, main/frame.c: Revert adding the
	  packed attribute, as it really doesn't make sense why that would
	  do any good. Fix the real bug, which is to do the check to see if
	  the frame came from a translator at the beginning of
	  ast_frame_free(), instead of at the end. This ensures that it
	  always gets checked, even if none of the parts of the frame are
	  malloc'd, and also ensures that we aren't looking at free'd
	  memory in the case that it is a malloc'd frame. (closes issue
	  #11792, reported by explidous, patched by me)

	* include/asterisk/translate.h: Since we're relying on the offset
	  between the frame and the beginning of the translator pvt struct,
	  set the packed attribute to make sure we get to the right place.
	  (potential fix for issue #11792)

2008-01-18 17:13 +0000 [r99032]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: This should at least temporarily fix a
	  problem where the 't' Dial option is incorrectly passed to the
	  transferee when built-in attended transfers are used. There is
	  still a problem with 'T', but better to fix some problems than no
	  problems while we work on it. (closes issue #7904) Reported by:
	  k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
	  (license 138) Tested by: sergee, otherwiseguy

2008-01-17 23:42 +0000 [r99007-99014]  Pari Nannapaneni <paripurnachand@digium.com>

	* configs/cdr.conf.sample: doh! revert a revert of a revert
	  (changed by mistake in 99010)

	* main/manager.c, configs/cdr.conf.sample: missed that one while
	  reverting

	* main/manager.c: reverting 99001 - We need the Max-Age for
	  extending the life of cookie mansession_id

2008-01-17 22:37 +0000 [r99004]  Russell Bryant <russell@digium.com>

	* main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
	  Have IAX2 optimize the codec translation path just like chan_sip
	  does it. If the caller's codec is in our codec list, move it to
	  the top to avoid transcoding. (closes issue #10500) Reported by:
	  stevedavies Patches: iax-prefer-current-codec.patch uploaded by
	  stevedavies (license 184) iax-prefer-current-codec.1.4.patch
	  uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
	  sheldonh

2008-01-17 21:31 +0000 [r99001]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: we should only send the Set-Cookie header to the
	  browser on the first response after creating a manager session,
	  not on every response (doing so causes the browser to clear any
	  local cookies it may have associated with the session)

2008-01-17 16:19 +0000 [r98991]  Jason Parker <jparker@digium.com>

	* configs/zapata.conf.sample: Add a clarification about the
	  immediate= option of zapata.conf Issue 11784, patch by klaus3000.

2008-01-16 22:36 +0000 [r98982]  Russell Bryant <russell@digium.com>

	* .cleancount, include/asterisk/channel.h: Add an unused pointer to
	  the ast_channel struct. This makes the ast_channel structure
	  retain the same size as it had in previous 1.4 releases. Also,
	  all of the offsets for members in the structure are still the
	  same (except for the two pointers that got replaced for the new
	  spy/whisper architecture.)

2008-01-16 20:34 +0000 [r98966-98973]  Joshua Colp <jcolp@digium.com>

	* .cleancount: Bump up cleancount due to previous commit that
	  changed the channel structure.

	* apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
	  main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
	  (added), main/Makefile, include/asterisk/chanspy.h (removed),
	  include/asterisk/channel.h, main/audiohook.c (added): Replace
	  current spy architecture with backport of audiohooks. This should
	  take care of current known spy issues.

	* channels/chan_iax2.c: Add missing NULLs at end of two
	  ast_load_realtimes. (closes issue #11769) Reported by: tequ
	  Patches: chaniax.patch uploaded by dimas (license 88)

2008-01-16 17:20 +0000 [r98964]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Fix a deadlock in chan_local in
	  local_hangup. There was contention because the local_pvt was held
	  and it was attempting to lock a channel, which is the incorrect
	  locking order. (closes issue #11730) Reported by: UDI-Doug
	  Patches: 11730.patch uploaded by putnopvut (license 60) Tested
	  by: UDI-Doug

2008-01-16 15:08 +0000 [r98951-98960]  Joshua Colp <jcolp@digium.com>

	* main/dial.c: Introduce a lock into the dialing API that protects
	  it when destroying the structure. (closes issue #11687) Reported
	  by: callguy Patches: 11687.diff uploaded by file (license 11)

	* main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
	  issue #11777) Reported by: tootai

	* channels/chan_sip.c: Don't drop the old record route information
	  when dealing with packets related to a reinvite. (closes issue
	  #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
	  uploaded by kebl0155 (license 356)

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
	  versions use a separate library for some things so we need to use
	  it if present in codec_speex. (closes issue #11693) Reported by:
	  yzg

2008-01-15 23:50 +0000 [r98943-98946]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Change a buffer in check_auth() to be a
	  thread local dynamically allocated buffer, instead of a massive
	  buffer on the stack. This fixes a crash reported by Qwell due to
	  running out of stack space when building with LOW_MEMORY defined.
	  On a very related note, the usage of BUFSIZ in various places in
	  chan_sip is arbitrary and careless. BUFSIZ is a system specific
	  define. On my machine, it is 8192, but by definition (according
	  to google) could be as small as 256. So, this buffer in
	  check_auth was 16 kB. We don't even support SIP messages larger
	  than 4 kB! Further usage of this define should be avoided, unless
	  it is used in the proper context.

	* main/rtp.c, include/asterisk/translate.h, main/frame.c,
	  main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
	  codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
	  some memory access errors pointed out by the valgrind2.txt output
	  on issue #11698. The issue here is that it is possible for an
	  instance of a translator to get destroyed while the frame
	  allocated as a part of the translator is still being processed.
	  Specifically, this is possible anywhere between a call to
	  ast_read() and ast_frame_free(), which is _a lot_ of places in
	  the code. The reason this happens is that the channel might get
	  masqueraded during this time. During a masquerade, existing
	  translation paths get destroyed. So, this patch fixes the issue
	  in an API and ABI compatible way. (This one is for you,
	  paravoid!) It changes an int in ast_frame to be used as flag
	  bits. The 1 bit is still used to indicate that the frame contains
	  timing information. Also, a second flag has been added to
	  indicate that the frame came from a translator. When a frame with
	  this flag gets released and has this flag, a function is called
	  in translate.c to let it know that this frame is doing being
	  processed. At this point, the flag gets cleared. Also, if the
	  translator was requested to be destroyed while its internal frame
	  still had this flag set, its destruction has been deffered until
	  it finds out that the frame is no longer being processed.
	  Admittedly, this feels like a hack. But, it does fix the issue,
	  and I was not able to think of a better solution ...

2008-01-15 20:08 +0000 [r98894-98934]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Based on the boundary found move over the
	  correct amount. (closes issue #11750) Reported by: tasker

	* channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
	  in the multipart mixed content type. (closes issue #11750)
	  Reported by: tasker

2008-01-14 20:59 +0000 [r98849]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Adding in appropriate unlocks for the locks
	  I added. Thanks to joetester on IRC for pointing this out.

2008-01-14 17:38 +0000 [r98774]  Russell Bryant <russell@digium.com>

	* main/translate.c: Revert a change that introduces an unacceptable
	  performance hit and is causing memory leaks ... (from rev 97973)

2008-01-14 16:35 +0000 [r98733-98737]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing another compilation error. I'm a bit off
	  today :(

	* apps/app_queue.c: Oops. Last commit had compilation error.

	* apps/app_queue.c: Adding explicit defaults for missing options to
	  init_queue. This is necessary because if a user either removes or
	  comments one of these options and reloads their queues, the
	  option will not reset to its default, instead maintaining the
	  value from prior to the reload. Thanks to John Bigelow for
	  pointing this error out to me.

2008-01-12 00:05 +0000 [r98467]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Add a connection timeout attribute, as that was
	  what was intended with the login timeout, but ODBC divides it up
	  into 2 different timeouts. (Closes issue #11745)

2008-01-11 22:46 +0000 [r98390]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
	  (closes issue #11646) Reported by: caio1982 Patches:
	  dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
	  dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
	  by: caio1982, mvanbaak

2008-01-11 21:28 +0000 [r98372]  Pari Nannapaneni <paripurnachand@digium.com>

	* main/http.c: Comment explaining how to force browser to always
	  read some html files from server.

2008-01-11 19:51 +0000 [r98317-98325]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If the incoming RTP stream changes codec force the
	  bridge to break if the other side does not support it. (closes
	  issue #11729) Reported by: tsearle Patches:
	  new_codec_patch_udiff.patch uploaded by tsearle (license 373)

	* res/res_agi.c: If the channel is hungup during RECORD FILE send a
	  result code of -1 to be uniform with everything else. (closes
	  issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
	  by davevg (license 209)

2008-01-11 19:10 +0000 [r98315]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Properly report the hangup cause as no answer
	  when someone does not answer (closes issue #10574, reported by
	  boch, patched by moy)

2008-01-11 18:25 +0000 [r98266]  Tilghman Lesher <tlesher@digium.com>

	* codecs/gsm/Makefile: Add another exception (which doesn't work)
	  for -march optimization flag. Reported by: thomasmebes Patch by:
	  tilghman (Closes issue #11563)

2008-01-11 18:25 +0000 [r98265]  Russell Bryant <russell@digium.com>

	* doc/security.txt, main/asterisk.c, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
	  makeopts.in: Backport the ability to set the ToS bits on Linux
	  when not running as root. Normally, we would not backport
	  features into 1.4, but, I was convinced by the justification
	  supplied by the supplier of this patch. He pointed out that this
	  patch removes a requirement for running as root, thus reducing
	  the potential impacts of security issues. (closes issue #11742)
	  Reported by: paravoid Patches: libcap.diff uploaded by paravoid
	  (license 200)

2008-01-11 17:22 +0000 [r98219]  Joshua Colp <jcolp@digium.com>

	* apps/app_followme.c: Ensure the return value of ast_bridge_call
	  is passed back up as the application return value. This is needed
	  for transfers to function so the PBX core knows to continue
	  execution. (closes issue #10327) Reported by: kkiely

2008-01-11 15:52 +0000 [r98164]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Back out changes from revision 97077, since
	  it wasn't perfect

2008-01-11 03:39 +0000 [r97976-98082]  Russell Bryant <russell@digium.com>

	* main/frame.c: Fix samples vs. length calculations for g722

	* main/translate.c: Simplify this code with a suggestion from Luigi
	  on the asterisk-dev list. Instead of using is16kHz(), implement a
	  format_rate() function.

	* main/translate.c: Fix various timing calculations that made
	  assumptions that the audio being processed was at a sample rate
	  of 8 kHz.

2008-01-10 23:08 +0000 [r97973]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, main/translate.c: 1) When we get a
	  translated frame out, clone it, because if the translator pvt is
	  freed before we use the frame, bad things happen. 2) Getting a
	  failure from ast_sched_delete means that the schedule ID is
	  currently running. Don't just ignore it. (Closes issue #11698)

2008-01-10 21:57 +0000 [r97925]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Let us leave a voicemail for ourself if we
	  have logged into VoiceMailMain and chosen to leave a message.
	  (closes issue #11735, reported and patched by jamessan)

2008-01-10 21:37 +0000 [r97849-97889]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
	  same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
	  overrode the normally generate yyfree func with our own version
	  that checks the pointer for non-null before passing to free().
	  Also takes care of a little problem with 2.5.33 and the use of
	  the __STDC_VERSION__ macro.

	* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
	  fix for 2 things: a problem Terry was having in OSX with null
	  pointers, which was my fault, as I probably forgot to run the sed
	  script last time I made mods. So, I moved the fix into the flex
	  input itself. Then, I found when I used flex 2.5.33, that it was
	  using __STDC_VERSION__, and that's not real good; so I added back
	  in a DIFFERENT sed script to fix that little mess. Tested
	  everything, a couple different ways. Hope I did no harm, at the
	  least.

2008-01-10 20:12 +0000 [r97847]  Jason Parker <jparker@digium.com>

	* include/asterisk/frame.h: Fix a comment that is no longer true.

2008-01-10 16:19 +0000 [r97734-97753]  Russell Bryant <russell@digium.com>

	* pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
	  pbx/kdeconsole_main.cc (removed): Remove other remnants of
	  pbx_kdeconsole

	* pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
	  configure, include/asterisk/autoconfig.h.in, configure.ac,
	  makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
	  worked in ages, and nobody has complained. (closes issue #11706,
	  reported by caio1982)

2008-01-10 15:07 +0000 [r97697]  Joshua Colp <jcolp@digium.com>

	* funcs/func_groupcount.c: Don't try to copy the category from the
	  group if no category exists. (closes issue #11724) Reported by:
	  IgorG Patches: group_count.v1.patch uploaded by IgorG (license
	  20)

2008-01-09 23:01 +0000 [r97640-97645]  Russell Bryant <russell@digium.com>

	* pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
	  messages

	* pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
	  actually load on my system still (related to issue #11706)

2008-01-09 20:28 +0000 [r97618-97622]  Jason Parker <jparker@digium.com>

	* main/cli.c: Correctly display a message if a command could not be
	  found. Also fix a comment which may have led to this happening.
	  Issue 11718, reported by kshumard.

	* main/cli.c: Fix some locking and return value funkiness. We
	  really shouldn't be unlocking this lock inside of a function,
	  unless we locked it there too.

2008-01-09 18:48 +0000 [r97575]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
	  smaller functions this time

2008-01-09 18:02 +0000 [r97529]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Fix saying the parking space number to the
	  caller doing the parking ...

2008-01-09 17:21 +0000 [r97491]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: report the same message whether Zaptel does
	  not have transcoder support loaded or no transcoders were found

2008-01-09 16:44 +0000 [r97489]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
	  function. As underlined in issue #10437 by Josh, we need to
	  prevent a possible memory leak. We only set the name part of the
	  caller id, the number part is not relevant when dealing with
	  JIDs. Closes issue #11549.

2008-01-09 16:11 +0000 [r97450]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Don't do conferencing totally in Zaptel if
	  Monitor is running on the channel. (closes issue #11709) Reported
	  by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
	  (license 371)

2008-01-09 15:43 +0000 [r97410-97448]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: pass the right variable to get an error
	  string... oops

	* channels/chan_zap.c: add error number output to ioctl failure
	  messages to help with debugging

2008-01-09 00:44 +0000 [r97350]  Tilghman Lesher <tlesher@digium.com>

	* main/cli.c, main/editline/readline.c: Allow filename completion
	  on zero-length modules, remove a memory leak, remove a file
	  descriptor leak, and make filename completion thread-safe.
	  Patched and tested by tilghman. (Closes issue #11681)

2008-01-09 00:17 +0000 [r97206-97308]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: use the \retval doxygen command properly

	* apps/app_queue.c: Part 1 of N of adding doxygen comments to
	  app_queue. I picked some of the most common functions used (which
	  also happen to be some the biggest/ugliest functions too) to
	  document first. I'm pretty new to doxygen so criticism is
	  welcome.

	* apps/app_queue.c: Some coding guidelines-related cleanup

2008-01-08 20:48 +0000 [r97195]  Joshua Colp <jcolp@digium.com>

	* channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
	  (closes issue #11443) Reported by: eferro Patches:
	  dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
	  337)

2008-01-08 20:47 +0000 [r97194]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c, main/utils.c: Increase constants to where
	  we're less likely to hit them while debugging. (Closes issue
	  #11694)

2008-01-08 20:42 +0000 [r97192]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Making some changes designed to not allow
	  for a corrupted mailstream for a vm_state. 1. Add locking to the
	  vm_state retrieval functions so that no linked list corruption
	  occurs. 2. Make sure to always grab the persistent vm_state when
	  mailstream access is necessary. 3. Correct an incorrect return
	  value in the init_mailstream function. (closes issue #11304,
	  reported by dwhite)

2008-01-08 19:53 +0000 [r97093-97152]  Joshua Colp <jcolp@digium.com>

	* funcs/func_groupcount.c: If no group has been provided to the
	  GROUP_COUNT dialplan function then use the first one specific to
	  the channel. (closes issue #11077) Reported by: m4him

	* apps/app_queue.c: Make app_queue calls work with directed pickup.
	  (closes issue #11700) Reported by: jbauer

2008-01-08 18:02 +0000 [r97077]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
	  found in issue #11386, but not completely closing that issue.

2008-01-07 20:47 +0000 [r96884-96932]  Russell Bryant <russell@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 96931 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
	  2 lines Change misery.digium.com to pbx.digium.com ........

	* res/res_smdi.c: Don't crash if something happens when setting up
	  an SMDI interface and it gets destroyed before the SMDI port
	  handling thread gets created.

2008-01-07 14:34 +0000 [r96797-96815]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Indentation fix, makes the code easier to read

	* res/res_jabber.c: Compute the base64 value over the
	  [authzid]\0authcid\0password string, thus excluding the trailing
	  NULL byte. This change has already been committed to trunk, see
	  #11644.

2008-01-05 02:09 +0000 [r96644]  Russell Bryant <russell@digium.com>

	* main/devicestate.c: Don't pass an empty string as the device
	  name.

2008-01-04 23:03 +0000 [r96575]  Tilghman Lesher <tlesher@digium.com>

	* main/devicestate.c: Fix the problem of notification of a device
	  state change to a device with a '-' in the name. Could probably
	  do with a better fix in trunk, but this bug has been open way too
	  long without a better solution. Reported by: stevedavies Patch
	  by: tilghman (Closes issue #9668)

2008-01-04 22:55 +0000 [r96573]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Properly continue in the dialplan if using
	  PARKINGEXTEN and the slot is full. Issue 11237, patch by me.

2008-01-04 19:27 +0000 [r96525]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: If you change the bindaddr in sip.conf to a
	  non-bound address and reload, sip goes kablooie. Reported and
	  patched by: one47 (Closes issue #11535)

2008-01-04 16:19 +0000 [r96394-96449]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Make use of the temporary channel pointer
	  while the pvt is unlocked. (closes issue #11675) Reported by:
	  flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
	  uploaded by flefoll (license 244)

	* channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
	  been destroyed before we get to this point (closes issue #11672,
	  reported by snuffy, patched by me)

2008-01-03 21:37 +0000 [r96318]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c: Missed initialization caused crash.
	  Reported and fixed by: tiziano (Closes issue #11671)

2008-01-03 12:12 +0000 [r96198-96199]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: make sure frame is completely clean,
	  before we send it to asterisk as DTMF. If we don't make it clean,
	  it happens that one way audio occurs..

	* channels/chan_misdn.c: when overlapdial was used and no number
	  was dialed, the call was dropped, now we just jump into the s
	  extension, which makes a lot more sense.

2008-01-02 23:46 +0000 [r96102]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: We need to reset the membername to NULL on each
	  iteration of this loop, otherwise the result is that multiple
	  members can have the same name, since the variable was not reset
	  on each iteration of the loop.

2008-01-02 22:14 +0000 [r96020-96024]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: Convert locks of the contexts list in
	  pbx_config to the appropriate rdlock or wrlock

	* pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
	  list.

	* apps/app_macro.c: app_macro only needs a rdlock on the contexts
	  list.

2008-01-02  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.17 released.

2008-01-02 20:24 +0000 [r95946]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Allocate a SIP refer structure when
	  performing a transfer using BYE with Also so that the transfer
	  information is properly stored. (AST-2008-028) (closes issue
	  #11637) Reported by: greyvoip

2008-01-02 17:51 +0000 [r95890]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: A change to improve the accuracy of queue
	  logging in the case where a member does not answer during the
	  specified timeout period. Prior to this change, there was a small
	  chance that the member name recorded in this case would be blank.
	  Also prior to this change, if using the ringall strategy, if no
	  one answered the call during the specified timeout, the member
	  name listed in the queue log would randomly be one of the members
	  that was rung. (closes issue #11498, reported and tested by
	  hloubser, patched by me)

2007-12-31 23:43 +0000 [r95577]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Avoiding a potentially bad locking situation.
	  ast_merge_contexts_and_delete writelocks the conlock, then calls
	  ast_hint_extension, which attempts to readlock the same lock.
	  Recursion with read-write locks is dangerous, so the inner lock
	  needs to be removed. I did this by copying the "guts" of
	  ast_hint_extension into ast_merge_contexts_and_delete (sans the
	  extra lock). (this change is inspired by the locking problems
	  seen in issue #11080, but I have no idea if this is the
	  problematic area experienced by the reporters of that issue)

2007-12-31 20:27 +0000 [r95470]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Allow the default "0" to be returned if the
	  STAT fails (Closes issue #11659)

2007-12-28 18:24 +0000 [r95191]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove duplicate increment of the header
	  count in the add_header() function. (closes issue #11648)
	  Reported by: makoto Patch provided by sergee, committed patch by
	  me, inspired by comments from putnopvut

2007-12-28 00:16 +0000 [r95095]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: I found a bug while browsing the queue code and
	  managed to reproduce it in a small setup. If a queue uses the
	  ringall strategy, it was possible through unfortunate coincidence
	  for a single member at a given penalty level to make app_queue
	  think that all members at that penalty level were unavailable and
	  cause the members at the next penalty level to be rung. With this
	  patch, we will only move to the next penalty level if ALL the
	  members at a given penalty level are unreachable.

2007-12-27 21:40 +0000 [r95024]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't report a syntax error when an empty string
	  is passed to ast_get_group. Just return 0. (closes issue #11540)
	  Reported by: tzafrir Patches: group_empty.diff uploaded by
	  tzafrir (license 46) -- slightly changed by me

2007-12-27 20:09 +0000 [r94977]  Mark Michelson <mmichelson@digium.com>

	* main/io.c: Fixing a typo in a comment.

2007-12-27 17:32 +0000 [r94905-94924]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Include types.h in chan_h323 as without it
	  it can not be compiled on some operating systems like FreeBSD to
	  name one. (closes issue #11585) Reported by: sobomax Patches:
	  chan_h323.c.diff uploaded by sobomax (license 359)

	* channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
	  set or not on the dialog. It is possible for it to be a pointer
	  to NULL. (closes issue #11557) Reported by: FuriousGeorge

2007-12-27 15:16 +0000 [r94828-94831]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Now that the contexts lock is a read/write lock, it
	  should not be locked here in ast_hint_state_changed(). This makes
	  it get locked recursively which now causes a deadlock. (closes
	  issue #11080, thanks to callguy for the access to a deadlocked
	  machine)

	* include/asterisk/translate.h, main/translate.c: Use the constant
	  that I really meant to use here ...

	* main/translate.c: Change ast_translator_best_choice() to only pay
	  attention to audio formats. This fixes a problem where Asterisk
	  claims that a translation path can not be found for channels
	  involving video. (closes issue #11638) Reported by: cwhuang
	  Tested by: cwhuang Patch suggested by cwhuang, with some
	  additional changes by me.

2007-12-27 01:01 +0000 [r94824]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: make this comment explain the situation in an
	  even more explicit fashion

2007-12-26 20:43 +0000 [r94808]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Workaround for what is probably a glibc bug (but
	  we'll see this crop up again and again, if we don't add the
	  workaround). Reported by: rolek Patch by: tilghman (Closes issue
	  #11601, closes issue #11426)

2007-12-26 19:04 +0000 [r94789-94801]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
	  was already set before starting autoservice, remember it and
	  ensure that the channel has the same setting when autoservice
	  gets stopped. (pointed out by d1mas, patched up by me)

	* main/autoservice.c: When a channel is in autoservice, mark a flag
	  on the channel that says that we only care about the END of a
	  digit. That way, no magic digit emulation stuff will happen when
	  all we're doing is queueing up END frames.

	* res/res_features.c: Don't try to send a parked call back to
	  itself. (closes issue #11622, reported by djrodman, patched by
	  me)

	* main/autoservice.c: Don't store DTMF BEGIN frames while a channel
	  is in autoservice. It's just going to make ast_read() do a lot of
	  extra work when the channel comes back out of autoservice.
	  (closes issue #11628, patched by me)

	* Makefile: List include/asterisk/version.h as a .PHONY target
	  because we want the commands listed for this target to be
	  executed regardless of whether the file exists or not. This fixes
	  having the version not up to date when running from svn. (closes
	  issue #11619, reported by plack, fixed by me)

2007-12-25 02:27 +0000 [r94769]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: file says... build on the builders.

2007-12-24 19:36 +0000 [r94763-94767]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Race: we need to wait to queue a NewChannel event
	  until after the channel is inserted into the channel list. The
	  reason is because some manager users immediately queue requests
	  from the channel when they see that event and are confused when
	  Asterisk reports no such channel. (Closes issue #11632)

	* channels/chan_sip.c: More deadlock avoidance code (this time
	  between sip_monitor and sip_hangup) Reported by: apsaras Patch
	  by: tilghman (Closes issue #11413)

	* channels/chan_sip.c: Another bit of bad logic in realtime_peer
	  Reported by: dimas Patch by: dimas (Closes issue #11631)

2007-12-23 01:21 +0000 [r94660]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Argh... I suppose third time's the charm.

2007-12-21 20:21 +0000 [r94468-94543]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Bunch of coding guidelines cleanup

	* apps/app_voicemail.c: Better quota support for using IMAP storage
	  voicemail (closes issue #11415, reported by jaroth) (closes issue
	  #11152, reported by selsky) Patch provided by jaroth

	* apps/app_voicemail.c: The mail_copy c-client function does not
	  expect a full imap mailbox string, just the name of the mailbox.
	  (closes issue #11419, reported and patched by jaroth, with
	  additional patchwork from me)

	* main/dial.c: Since we are freeing list elements within a list
	  traversal, we need to use the safe traversal and remove the item
	  from the list before freeing it. (closes issue 11612, reported by
	  dtyoo)

2007-12-21 16:37 +0000 [r94466]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
	  a read/write lock to resolve a deadlock. This has a nice side
	  benefit of improving performance. :) (closes issue #11609)
	  (closes issue #11080)

2007-12-21 16:11 +0000 [r94420-94464]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing a debug message I accidentally just
	  committed

	* main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
	  dates and times. Also some coding guidelines cleanup. (closes
	  issue #11599, reported and patched by caio1982, coding guidelines
	  cleanup by me)

2007-12-21 15:07 +0000 [r94418]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Fix for restart-as-user problem reported via the
	  -dev list

2007-12-20  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16.2 released.

2007-12-20 20:22 +0000 [r94215-94256]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
	  5 lines Fix another potential seg fault ... (closes issue #11606)
	  Reported by: dimas ........

	* channels/chan_zap.c: Fix a deadlock in d-channel handling in
	  chan_zap. This deadlock was introduced by the fix to ensure that
	  channels are properly locked when handling channel variables.
	  There were sections of this code where the channel pvt was locked
	  before the channel lock, when in fact it _must_ be the other way
	  around. (closes issue #11582) Reported by: bugi

2007-12-19 23:02 +0000 [r94122]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: Sox versions 13.0.0 and newer do not have
	  "soxmix" and instead use sox -m. res_monitor needs to use this if
	  the user does not have soxmix. (closes issue #11589, reported by
	  amessina, patch inspired by amessina but with a flourish from me)

2007-12-19 22:48 +0000 [r94077]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Check
	  for the existence of the soxmix application on the target
	  platform and have the result available in autoconfig.h. (part of
	  issue #11589)

2007-12-19  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16.1 released.

2007-12-19 17:29 +0000 [r93955]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
	  NULL.

2007-12-19 17:04 +0000 [r93949]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
	  defined (Closes issue #11602)

2007-12-18 22:42 +0000 [r93764]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: FreeBSD also does not have byte swap
	  functions. Issue 11586, patch by sobomax.

2007-12-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16 released.

2007-12-18 18:45 +0000 [r93668-93676]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  93667 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
	  | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........

2007-12-18 17:02 +0000 [r93625]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Rework deadlock avoidance used in ast_write,
	  since it meant that agent channels which were being monitored had
	  one audio file recorded and one empty audio file saved. (closes
	  issue #11529, reported by atis patched by me)

2007-12-17 22:56 +0000 [r93381-93420]  Jason Parker <jparker@digium.com>

	* main/translate.c: What was I thinking when I wrote this
	  masterpiece? -1 + 1 = 0.. who woulda thunk it?.

2007-12-17 22:28 +0000 [r93377]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Do not try to access information about a lock when
	  printing out a trylock attempt. It is possible for the lock that
	  it references to no longer be valid. This would have caused
	  segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
	  Reported by: callguy (closes issue #11100) Reported by: callguy

2007-12-17 21:12 +0000 [r93336]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/time.h: Today is tomorrow's yesterday, and
	  yesterday's tomorrow is today, and tomorrow's tomorrow is the day
	  after tomorrow, so who cares if you recycle anyway? If this
	  confuses you, that's nothing compared to what this fixes. ;-)

2007-12-17 19:53 +0000 [r93291]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: We need to create the directory for a
	  voicemail user even if they are using IMAP storage since
	  greetings are stored in the filesystem. (closes issue #11388,
	  reported by spditner, patch by me inspired by a patch by
	  spditner)

2007-12-17 18:05 +0000 [r93250]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: If a call is received with a called number
	  IE containing nothing go to the 's' extension. (closes issue
	  #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
	  uploaded by Corydon76 (license 14)

2007-12-17 07:21 +0000 [r93183]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
	  res/Makefile, channels/Makefile, formats/Makefile: fix some
	  copy-and-paste leftovers

2007-12-17 07:15 +0000 [r93182]  Olle Johansson <oej@edvina.net>

	* channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
	  apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
	  dependencies on res_monitor and res_features. I wonder if
	  Asterisk can run at all without res_features. My guess is that
	  there's propably a lot of more modules and the core that depends
	  on it. Reported by: caio1982 (closes issue #11574)

2007-12-17 06:44 +0000 [r93180]  Kevin P. Fleming <kpfleming@digium.com>

	* formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
	  configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
	  formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
	  include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
	  configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
	  (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
	  http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
	  rizzo brought up some issues related to the way that the metadata
	  required for menuselect and the rest of the build system is
	  extracted from the source files. Since I had a few hours to kill
	  on an airplane today, I decided to improve this situation... so
	  now the system caches the extracted metadata and uses it to build
	  the menuselect 'tree' as much as it can. The result of this is
	  that when a single source file is changed, only the metadata for
	  that file needs to be extracted again, and the rest is used from
	  the cache files. I also reduced the number of forked processes
	  required to do the metadata extraction; it was actually possible
	  to do most of what we needed in the Makefiles themselves without
	  using any shell scripts at all! On my laptop, these changes
	  resulted in an 80% decrease in the time required for the
	  'menuselect.makeopts' automatic check to occur after editing a
	  single source file. While doing this work I also cleaned up a few
	  minor things in the Makefiles, adding a check for 'awk' to the
	  configure script and changed all remaining places we use 'grep'
	  or 'awk' to use the ones found by the configure script, and
	  changed the 'prep_tarball' script to build the menuselect
	  metadata so that tarballs of Asterisk will include it and won't
	  require the user to wait while it is extracted after unpacking.

2007-12-14 17:36 +0000 [r93000]  Russell Bryant <russell@digium.com>

	* main/config.c: There are a lot of existing systems that #include
	  non-existent files. So, to make the transition to treating this
	  as an error a bit less painless, just issue a huge error message
	  for now. Then, later, we can reinstate the code that treats it as
	  a failure. (Thanks to philippel for the feedback)

2007-12-14 15:16 +0000 [r92937]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Up the length of the format on the SIP
	  channel since it can now be rather long. (closes issue #11552)
	  Reported by: francesco_r

2007-12-14 15:05 +0000 [r92934]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
	  state setting and overlap_task thread starting.

2007-12-14 15:01 +0000 [r92933]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Change help documentation to match actual behavior
	  (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
	  (Closes issue #11548)

2007-12-14 01:24 +0000 [r92875]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
	  don't spam the CLI with messages about possible deadlocks.
	  Instead just print the intended single message every five
	  seconds. (closes issue 11537, reported and patched by dimas)

2007-12-13 21:28 +0000 [r92815]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Properly initialize polarity statuses, so
	  that they are detected properly. Reported by: julianjm Patch by:
	  julianjm (Closes issue #10238)

2007-12-13 20:13 +0000 [r92809]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Make application help text a little more clear about
	  the use of extensions in a filename.

2007-12-13 20:03 +0000 [r92803-92807]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Prevent another potential fd leak

	* apps/app_voicemail.c: Prevent a possible fd leak.

2007-12-13 00:11 +0000 [r92696]  Jason Parker <jparker@digium.com>

	* main/config.c, channels/chan_sip.c, channels/chan_h323.c,
	  channels/chan_iax2.c: If a typo is found in a config file, we
	  previous continued on with what was already loaded. We do not
	  want to do this (see bug below for details). This makes it so
	  that if a [ is found without a ], the entire config will fail,
	  and nothing in it will be loaded. Isue #10690.

2007-12-12 22:00 +0000 [r92656]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: emit a warning message when we drop a G.729B
	  CNG frame destined for the transcoder

2007-12-12 21:15 +0000 [r92617]  Jason Parker <jparker@digium.com>

	* apps/app_meetme.c: Don't increment user count until after name
	  has been recorded (if enabled). Issue 11048, tested by pep.

2007-12-12 19:40 +0000 [r92556]  Russell Bryant <russell@digium.com>

	* res/res_features.c: resolve compiler warning

2007-12-12 17:46 +0000 [r92510]  Mark Michelson <mmichelson@digium.com>

	* res/res_features.c: Correctly detect where a dynamic feature was
	  activated. Before this patch, the channel which initiated the
	  bridge was always assumed to have been the one which activated
	  the dynamic feature. This patch corrects this. (closes issue
	  #11529, reported and patched by nic_bellamy)

2007-12-12 16:52 +0000 [r92463]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Test
	  directly for the API that fixed AST-2007-026, to ensure that
	  older versions of PostgreSQL are no longer acceptable. (Closes
	  issue #11526)

2007-12-12 16:08 +0000 [r92443]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing an unused variable.

2007-12-11 19:51 +0000 [r92363]  Joshua Colp <jcolp@digium.com>

	* main/global_datastores.c: Fix potential memory leak with the
	  dialed interfaces list if another memory allocation fails.
	  (closes issue #11507) Reported by: eliel Patches:
	  global_datastores.c.patch uploaded by eliel (license 64)

2007-12-11 17:42 +0000 [r92323]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing autofill to be more accurate.
	  Specifically, if calls ahead of the current caller were ringing
	  members (but not yet bridged) there could be available members
	  and waiting callers who would not get matched up. The member
	  availability checker was correctly determining the number of
	  available members in this scenario, but the queue itself did not
	  parallelly reflect this status on the pending calls. This commit
	  corrects the issue. (closes issue #11459, reported by
	  equissoftware, patched by me)

2007-12-10 16:36 +0000 [r92204]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Add G729A as another possible payload name for G729.
	  Some devices use this instead of G729, which is perfectly normal
	  since the payload number itself is defined and can't be used by
	  anything else so the name doesn't matter that much. (closes issue
	  #11483) Reported by: revolution Patches: rtp.diff uploaded by
	  revolution (license 346)

2007-12-10 16:29 +0000 [r92202]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: If there are no members in a queue, then the
	  loop where the datastore for detecting duplicate dialed numbers
	  will be skipped, meaning the datastore isn't created. This means
	  that when we try to free it, there's a crash. This stops that
	  crash from occurring. (closes issue #11499, reported by slavon,
	  patched by eliel)

2007-12-10 16:13 +0000 [r92200]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is possible for nativeformats to contain
	  more then one codec, so print out multiple ones. (closes issue
	  #11366) Reported by: ovi

2007-12-10 14:04 +0000 [r92158]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Avoid reinvite race situations with two
	  Asterisks trying to reinvite each other in 1.4 and trunk. This
	  patch implements support for the 491 error code that Asterisk 1.4
	  generates on situations where we get an incoming INVITE and
	  already has one in progress. Thanks to mavetju for reporting and
	  to Raj Jain for an excellent explanation of the problem. Patch by
	  myself. Tested with 8 Asterisk servers connected to each other in
	  a training network. Closes issue #10481

2007-12-07 23:29 +0000 [r91890]  Jason Parker <jparker@digium.com>

	* main/dsp.c: We need to make sure we free the input frame if we
	  return a different frame in ast_dsp_process. Issue 11273, pointed
	  out by dimas, with a patch by eliel.

2007-12-07 22:30 +0000 [r91870]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: even though Asterisk explicitly requests that
	  endpoints using G.729 do *not* use Annex B (silence detection and
	  comfort noise generation) some do anyway; the transcoder card
	  interface does not currently work properly with CNG frames, so
	  trim off the CNG before sending the data

2007-12-07 21:24 +0000 [r91777-91830]  Russell Bryant <russell@digium.com>

	* main/utils.c: Make the lock protecting each thread's list of
	  locks it currently holds recursive. I think that this will fix
	  the situation where some people have said that "core show locks"
	  locks up the CLI. (related to issue #11080)

	* include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
	  code. The ast_mutex_init() function had the mutex attribute
	  object marked as static. This means that multiple threads
	  initializing locks at the same time could step on each other and
	  end up with improperly initialized locks. (found when tracking
	  down locking issues related to issue #11080)

	* include/asterisk/lock.h: I love fixing lock related errors in the
	  lock debugging code. That's about as ironic as it gets in
	  Asterisk programming land. Anyway, I spotted this bug while
	  trying to track down why systems are locking up and acting weird
	  in issue #11080. The mutex attribute object was marked as static
	  in this function when it should not have been.

	* apps/app_dial.c: * Add channel locking around datastore
	  operations that expect the channel to be locked. * Document why
	  we don't record Local channels in the dialed interfaces list. *
	  Remove the dialed variable as it isn't needed. * Restructure some
	  code for clarity and coding guidelines stuff

	* apps/app_queue.c: * Add channel locking around datastore
	  operations that expect the channel to be locked. * Document why
	  we don't record Local channels in the dialed interfaces list. *
	  Handle memory allocation failure. * Remove the dialed variable,
	  as it wasn't actually needed. * Tweak some formatting to conform
	  to coding guidelines.

	* main/autoservice.c: * Add a bit more of a verbose comment as to
	  why a hangup frame needs to be queued up if autoservice gets a
	  NULL return from ast_read(). * Make the process of queueing the
	  hangup frame more efficient by putting the frame where it is
	  going to end up and avoiding some locking and extra memory
	  allocations and freeing.

2007-12-07 15:39 +0000 [r91737]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c: Hangups that happen during autoservice were
	  not processed appropriately. This is because a hangup actually
	  causes a NULL frame to be received, not a hangup frame. Queueing
	  a hangup if we receive a NULL frame during autoservice corrects
	  this problem (closes issue #11467, reported by jmls, patched by
	  me)

2007-12-07 02:51 +0000 [r91675-91693]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: Don't unlock the dialed_interfaces list until
	  we're done messing with the iterator.

	* apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
	  from Queue() and Dial() again. There was a slight flaw in the
	  code to prevent call forwards from looping that caused this
	  problem. (related to issue #11486)

	* apps/app_queue.c: Fix in an issue in the call forwarding handling
	  code that was causing crashes on every call into a queue. I'm not
	  entirely sure about the logic in this part of the code, so I want
	  to look at it some more tomorrow. However, this makes it safe and
	  keeps it from crashing. (closes issue #11486, reported by adamg,
	  patched by me)

2007-12-07 00:52 +0000 [r91637]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c: At the end of a call, when we're reporting, RTCP may
	  already be partially torn down, so check for NULL dereference
	  Reported by: blitzrage Patch by: tilghman (Closes issue #11450)

2007-12-06 20:25 +0000 [r91541]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: IMAP storage did not honor the maxmsg
	  setting in voicemail.conf, and it also had the possibility of
	  crashing if a user had more than 256 messages in their voicemail.
	  This patch kills two birds with one stone by adding maxmsg
	  support and also setting a hard limit on the number of messages
	  at 255 so that the crashes cannot happen. (closes issue #11101,
	  reported by Skavin, patched by me)

2007-12-06 19:11 +0000 [r91501]  Russell Bryant <russell@digium.com>

	* main/loader.c, include/asterisk/module.h: Add a new module flag
	  to indicate that a build sum is present. Modules built against
	  older Asterisk 1.4 headers will now load properly with just a
	  warning indicating that they are old and may cause problems.
	  (patch by paravoid)

2007-12-06 16:49 +0000 [r91439-91450]  Joshua Colp <jcolp@digium.com>

	* main/udptl.c: Fix various in the udptl implementation. It could
	  return empty modem frames, have an incorrect sequence number on
	  packets, and display the wrong sequence number in the debug
	  messages. (closes issue #11228) Reported by: Cache Patches:
	  udptl-4.patch uploaded by dimas (license 88)

	* channels/chan_sip.c: Add support for accepting and sending T.38
	  in the initial INVITE. (closes issue #9402) Reported by: thdei

2007-12-06 12:54 +0000 [r91366]  Olle Johansson <oej@edvina.net>

	* main/loader.c, include/asterisk/logger.h, main/logger.c: Make
	  sure logger is reloaded at general reload in the cli. (Discovered
	  during Asterisk training in Portugal)

2007-12-05 22:57 +0000 [r91273-91292]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Reverting extra stuff I didn't mean to
	  commit

	* apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
	  did not properly handle the case where only a label was provided.
	  This was due to the fact that the answering channel did not have
	  an extension set, so ast_parseable_goto would fail. This fix
	  eliminates the call to ast_parseable_goto on the answering
	  channel since it is a wasteful call. The answering channel and
	  the calling channel are both directed to the same extension and
	  context, just different priorities, so we can just copy the
	  values from the calling channel to the answering channel and
	  increment the answering channel's priority. (closes issue #11382,
	  reported by jon, patch by me with correction by jon)

2007-12-05 21:38 +0000 [r91237]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile: Upgrade to the latest version of extra sounds

2007-12-05 17:31 +0000 [r90967-91192]  Russell Bryant <russell@digium.com>

	* main/threadstorage.c: Make the lock in the threadstorage
	  debugging code untracked to avoid a deadlock on thread
	  destruction. (closes issue #11207) Reported by: ys Patches:
	  threadstorage.c.diff uploaded by ys (license 281) Also fixes an
	  open bug report: (closes issue #11446)

	* main/utils.c: When DEBUG_THREADS is enabled, we only have the
	  details about who is holding a lock that we are waiting on for a
	  mutex, not rwlocks. This should fix the problem where people have
	  reported "core show locks" crashing sometimes.

	* include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
	  reported as happening on Mac systems. It turns out that the
	  problem was the Mac version of the ast_atomic_fetchadd_int()
	  function. The Mac atomic add function returns the _new_ value,
	  while this function is supposed to return the old value. So, the
	  crashes happened on unreferencing objects. If the reference count
	  was decreased to 1, ao2_ref() thought that it had been decreased
	  to zero, and called the destructor. However, there was still an
	  outstanding reference around. (closes issue #11176) (closes issue
	  #11289)

	* include/asterisk/file.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/compiler.h: Modify file.h to maintain API
	  compatibility with earlier versions. If a recent compiler is
	  being used, then a warning will show up for any modules still
	  using the old name "private" instead of "_private". (patch
	  suggested by paravoid)

	* main/pbx.c: Make some changes to some additions I made recently
	  for doing channel autoservice when looking up extensions. This
	  code was added to handle the case where a dialplan switch was in
	  use that could block for a long time. However, the way that I
	  added it, it did this for all extension lookups. However, lookups
	  in the in-memory tree of extensions should _not_ take long enough
	  to matter. So, move the autoservice stuff to be only around
	  executing a switch.

2007-12-04 17:28 +0000 [r90876]  Jason Parker <jparker@digium.com>

	* main/channel.c: If we fail to create a channel after allocating a
	  timing fd, we need to make sure to close it. Issue 11454, patch
	  by eliel.

2007-12-04 05:29 +0000 [r90798]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Fix build issue on the build cluster.

2007-12-03 23:50 +0000 [r90736-90753]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/compat.h: Solaris requires the inclusion of
	  sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
	  snuffy,tilghman (Closes issue #11430)

	* res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
	  NULL deref could result Reported by: xrg Patch by: tilghman
	  (Closes issue #11387)

2007-12-03 23:12 +0000 [r90735]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, main/channel.c, main/global_datastores.c
	  (added), channels/chan_local.c, main/Makefile,
	  include/asterisk/channel.h, include/asterisk/global_datastores.h
	  (added), apps/app_queue.c: A big one... This is the merge of the
	  forward-loop branch. The main change here is that call-forwards
	  can no longer loop. This is accomplished by creating a datastore
	  on the calling channel which has a linked list of all devices
	  dialed. If a forward happens, then the local channel which is
	  created inherits the datastore. If, through this progression of
	  forwards and datastore inheritance, a device is attempted to be
	  dialed a second time, it will simply be skipped and a warning
	  message will be printed to the CLI. After the dialing has been
	  completed, the datastore is detached from the channel and
	  destroyed. This change also introduces some side effects to the
	  code which I shall enumerate here: 1. Datastore inheritance has
	  been backported from trunk into 1.4 2. A large chunk of code has
	  been removed from app_dial. This chunk is the section of code
	  which handles the call forward case after the channel has been
	  requested but before it has been called. This was removed because
	  call-forwarding still works fine without it, it makes the code
	  less error-prone should it need changing, and it made this set of
	  changes much less painful to just have the forwarding handled in
	  one place in each module. 3. Two new files, global_datastores.h
	  and .c have been added. These are necessary since the datastore
	  which is attached to the channel may be created and attached in
	  either app_dial or app_queue, so they need a common place to find
	  the datastore info. This approach was taken in case similar
	  datastores are needed in the future, there will be a common place
	  to add them.

2007-12-03 22:06 +0000 [r90696]  Jason Parker <jparker@digium.com>

	* apps/app_meetme.c: Make sure we always close the conference fd if
	  we have an open one. Issue 11383, reported by markmhy, patch by
	  eliel.

2007-12-03 20:59 +0000 [r90639]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_mgcp.c: Changing some bad logic when calculating
	  the interdigit timeout. (closes issue #11402, reported and
	  patched by eferro)

2007-12-03 20:51 +0000 [r90607]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Fix crash in ParkAndAnnounce application.
	  Issue #11436, reported by lytledd, patch by eliel.

2007-12-03 20:05 +0000 [r90548-90588]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Do not create a smoother for G723.1 frames, they need
	  to be left alone to their native 20/24 byte size.

	* .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
	  the indication currently playing on a channel when a masquerade
	  operation happens. (issue #BE-88)

2007-12-03 18:20 +0000 [r90546]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Only log debug messages if debug is
	  enabled. Closes issue #11416, patch by casper.

2007-12-02 18:18 +0000 [r90470]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: The other day when I went through making
	  changes as a result of the ao2_link() change, I added some code
	  to set pointers to NULL after they were unreferenced. This
	  pointed out that in this place, the object was unreferenced
	  before the code was done using it. So, move the unref down a
	  little bit. (crash reported by jmls on IRC)

2007-12-02 09:34 +0000 [r90432]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c: Clarify the return value on autoservice.
	  Specifically, if you started autoservice and autoservice was
	  already on, it would erroneously return an error. Reported by:
	  adiemus Patch by: dimas (Closes issue #11433)

2007-11-30 19:26 +0000 [r90310-90348]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
	  apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
	  ao2_link(). Previously, in inherited a reference. Now, it
	  automatically increases the reference count to reflect the
	  reference that is now held by the container. This was done to be
	  more consistent with ao2_unlink(), which automatically releases
	  the reference held by the container. It also makes it so it is no
	  longer possible for a pointer to be invalid after ao2_link()
	  returns.

	* include/asterisk/astobj2.h: Add some notes on the behavior of
	  ao2_unlink() after a discussion with Tilghman

2007-11-30 14:43 +0000 [r90269]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix locking issues under one legged replaces
	  scenarios. (closes issue #11420) Reported by: irroot Patches:
	  chan_sip_oneleg.patch uploaded by irroot (license 52)

2007-11-30 00:16 +0000 [r90231]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_mgcp.c: Clear the DTMF buffer if the call times
	  out. (closes issue #11418, reported and patched by eferro)

2007-11-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.15 released.

2007-11-29 19:48 +0000 [r90166]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
	  we use thread-safe escaping (Fixes AST-2007-026)

2007-11-29 19:38 +0000 [r90163]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This patch handles the case where a queue
	  member with a negative penalty is added via the manager. If a
	  negative value is submitted for a member penalty, we set it to 0.
	  (closes issue #11411, reported and patched by Laureano)

2007-11-29 19:24 +0000 [r90154-90160]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c: Properly escape input buffers (Fixes
	  AST-2007-025)

	* formats/format_g726.c, include/asterisk/file.h,
	  formats/format_wav.c, formats/format_pcm.c,
	  formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
	  as a field name in a header file messes with C++ projects
	  Reported by: chewbacca Patch by: casper (Closes issue #11401)

	* sounds/Makefile: Upgrade the core sounds release version

2007-11-29 00:36 +0000 [r90142-90147]  Russell Bryant <russell@digium.com>

	* funcs/func_callerid.c: fix some formatting i accidentally changed

	* funcs/func_callerid.c, main/channel.c,
	  include/asterisk/channel.h: This set of changes is to make some
	  callerID handling thread-safe. The ast_set_callerid() function
	  needed to lock the channel. Also, the handlers for the CALLERID()
	  dialplan function needed to lock the channel when reading or
	  writing callerid values directly on the channel structure.

	* include/asterisk/file.h, main/file.c: Merge a change from
	  team/russell/chan_refcount ... This makes ast_stopstream()
	  thread-safe.

2007-11-28 22:59 +0000 [r90101]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
	  Reported by: eliel Patches: load_realtime.patch uploaded by eliel
	  (license 64)

2007-11-28 22:30 +0000 [r90098]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/users.conf.sample, main/manager.c: it is impossible to
	  set permissions for manager accounts created by users.conf
	  (reported internally, patched by me)

2007-11-28 22:08 +0000 [r89999-90059]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Removing some seemingly pointless code. This sets a
	  channel variable for every priority executed in the dialplan if
	  you have debug set to anything non-zero. This seems pointless due
	  to the fact that these channel variables are not referenced
	  anywhere else in the code and their names are esoteric enough
	  that they would not be practical to reference in the dialplan.
	  Plus the fact that this behavior isn't documented anywhere means
	  that the change is not likely to cause any disruption. If
	  anything, this may actually cause a slight performance increase
	  if running with debug on. The motivating influence for this code
	  change is the eventwhencalled option for queues. If set to vars,
	  all channel variables will be output to the manager. These
	  unnecessary channel variables make the output a lot more
	  difficult to deal with.

	* apps/app_voicemail.c: Recording greetings when using IMAP storage
	  was causing zero-length files to be stored. Since greetings are
	  not retrieved from IMAP anyway, it is pointless to attempt
	  storing them there. (closes issue #11359, reported by spditner,
	  patched by me)

2007-11-28 00:20 +0000 [r89839-89893]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: - update documentation for
	  some of the goto functions to note that they handle locking the
	  channel as needed - update ast_explicit_goto() to lock the
	  channel as needed

	* main/autoservice.c: Don't do frame processing if ast_read()
	  returned NULL.

	* apps/app_queue.c: Instead of depending on the return value of
	  ast_true(), explicitly set the eventwhencalled variable to 1.

	* main/pbx.c: Don't start/stop autoservice in
	  pbx_extension_helper() unless a channel exists

2007-11-27 23:10 +0000 [r89837]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Two changes with regards to the
	  'eventwhencalled' option of queues.conf 1) Due to some signed vs.
	  unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
	  did exactly the same thing. Thus the sign change of the ast_true
	  call. 2) The vars2manager function overwrote a \n for every
	  channel variable it parsed, resulting in bizarre output for the
	  channel variables. This patch remedies this. (related to issue
	  #11385, however I'm not sure if this will actually be enough to
	  close it)

2007-11-27 21:45 +0000 [r89790]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, main/pbx.c: Merge changes from
	  team/russell/autoservice_1.4 This set of changes fixes an issue
	  that was reported to me on IRC yesterday. The user, d1mas, was
	  using chan_zap for incoming calls and was having DTMF recognition
	  issues in some situations. Specifically, he noticed that the
	  problem occurred when using DISA or WaitExten. He also noticed
	  that when using Read, the problem did not occur. His system also
	  used DUNDi for dialplan lookups. So, he theorized that if the
	  DUNDi lookups blocked for some period of time, that audio from
	  the zap channel could get lost. If the audio got lost, then it
	  wouldn't be run through the DTMF detector, and digits could get
	  lost. He was correct, and the following set of changes fixes the
	  problem. However, the changes go a little bit further than what
	  was necessary to fix this exact problem. 1) I updated
	  pbx_extension_helper() to autoservice the associated channel to
	  handle cases where extension lookups may take a long time. This
	  would normally be a dialplan switch that does some lookup over
	  the network, such as the DUNDi or IAX2 switches. This ensures
	  that even while a DUNDi lookup is blocking, the channel will be
	  continuously serviced. 2) I made a change to the autoservice
	  code. This is actually something that has bothered me for a long
	  time. When a channel is in autoservice, _all_ frames get thrown
	  away. However, some frames really shouldn't be thrown away. The
	  most notable examples are signalling (CONTROL) frames, and DTMF.
	  So, this patch queues up important frames while a channel is in
	  autoservice. When autoservice is stopped on the channel, the
	  queued up frames get stuck back on the channel so that they can
	  get processed instead of thrown away. 3) I made another change to
	  the autoservice code to handle the case where autoservice is
	  started on channels recursively. Previously, you could call
	  ast_autoservice_start() multiple times on a channel, and it would
	  stop the first time ast_autoservice_stop() gets called. Now, it
	  will ensure that autoservice doesn't actually stop until the
	  final call to ast_autoservice_stop().

2007-11-27 20:22 +0000 [r89727]  Mark Michelson <mmichelson@digium.com>

	* res/res_config_pgsql.c: Changing some calls from free() to
	  ast_free() since they were allocated with ast_calloc(). (closes
	  issue #11390, reported and patched by Laureano)

2007-11-27 20:16 +0000 [r89701-89709]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: on second thought... revert all the other changes
	  i've made in app options parsing leaving only one: if an empty
	  argument is supplied for an option, set that argument pointer to
	  point to an empty string rather than NULL, so that the
	  application can do normal checks on it without worrying about it
	  being NULL

	* main/app.c: generate a warning when an application option that
	  requires an argument is ignored due to lack of an argument

2007-11-27 16:12 +0000 [r89634]  Russell Bryant <russell@digium.com>

	* configs/voicemail.conf.sample: Add a note to the sample voicemail
	  config noting that when using IMAP storage, only the first format
	  specified will be attached to the message.

2007-11-27 15:38 +0000 [r89631]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Default result of STAT should be "0" not "".
	  Reported via the -users mailing list, fixed by me.

2007-11-27 15:23 +0000 [r89624-89630]  Olle Johansson <oej@edvina.net>

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
	  get a codec offer using a well-known payload type, but using it
	  for another codec that we don't know, Asterisk did not remove
	  that codec from the list. With this patch, we remove the codec
	  from audio and video rtp objects and deny it ever existed. Thanks
	  to lasse for testing. (closes issue #11376) Reported by: lasse
	  Patches: bug11376.txt uploaded by oej (license 306) Tested by:
	  lasse

	* configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
	  #11304) Reported by: pj

2007-11-27 06:24 +0000 [r89622]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
	  include/asterisk/cdr.h: closes issue #11379; OK, this is an
	  attempt to make both sides happy. To the cdr.conf file, I added
	  the option 'unanswered', which defaults to 'no'. In this mode,
	  you will see a cdr for a call, whether it was answered or not.
	  The disposition will be NO ANSWER or ANSWERED, as appropriate.
	  The src is as you'd expect, the destination channel will be one
	  of the channels from the Dial() call, usually the last in the
	  list if more than one chan was specified. With unanswered set to
	  'yes', you will still see this cdr entry in both cases. But in
	  the case where the dial timed out, you will also see a cdr for
	  each line attempted, marked NO ANSWER, with no destination
	  channel name. The new option defaults to 'no', so you don't see
	  the pesky extra cdr's by default, and you will not see the
	  irritating 'not posted' messages.

2007-11-26 23:10 +0000 [r89616-89618]  Mark Michelson <mmichelson@digium.com>

	* apps/app_playback.c: After issuing a "say load new", if a caller
	  hangs up during the middle of playback of a number, app_playback
	  will continue to try to play the remaining files. With this
	  change, no more files will be played back upon hangup. (closes
	  issue #11345, reported and patched by IgorG)

	* apps/app_playback.c: After issuing a "say load new" tons of
	  warning messages are printed out to the CLI every time do_say in
	  app_playback is called. Removing these warnings

2007-11-26 21:10 +0000 [r89599-89610]  Joshua Colp <jcolp@digium.com>

	* main/dial.c: Fix issues with async dialing with an application
	  executing. The application has to be terminated and control
	  returned to the thread before hanging things up. (issue #BE-252)

	* res/res_features.c: Add module counting removal for error
	  conditions. (closes issue #11333) Reported by: Laureano Patches:
	  res_features_v2.c.patch uploaded by Laureano (license 265)

2007-11-26 17:41 +0000 [r89594]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Add channel locking to a function that needed to be
	  doing it. This is just a little something I noticed while working
	  on a completely unrelated issue.

2007-11-26 17:36 +0000 [r89587-89592]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: Use ast_free to free memory, or else we shall
	  implode if MALLOC_DEBUG is enabled. (closes issue #11347)
	  Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
	  (license 281)

	* apps/app_mixmonitor.c: Close the audio file before sending it to
	  the post processing application. (closes issue #11357) Reported
	  by: reformed Patches: mixmonitor.patch uploaded by reformed
	  (license 330)

2007-11-26 17:20 +0000 [r89586]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: when parsing application options that take arguments,
	  don't indicate that the option was supplied unless a
	  non-zero-length argument was found for it

2007-11-26 15:48 +0000 [r89580]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Revert vmu->email back to an empty string
	  if it was empty when imap_store_file was called. This prevents
	  sending a duplicate e-mail. (closes issue #11204, reported by
	  spditner, patched by me)

2007-11-26 15:34 +0000 [r89571-89577]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: If channel allocation fails because the alert
	  pipe could not be created also free the scheduler context.
	  (closes issue #11355) Reported by: eliel Patches:
	  main.channel.c.patch uploaded by eliel (license 64)

	* apps/app_meetme.c: When unloading app_meetme destroy any auto
	  created contexts created by SLA. (closes issue #11367) Reported
	  by: eliel

2007-11-25 17:17 +0000 [r89559]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample,
	  include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
	  attempted to use the ESCAPE clause to set the escape delimiter to
	  a backslash. Unfortunately, this does not universally work on all
	  databases, since on databases which natively use the backslash as
	  a delimiter, the backslash itself needs to be delimited, but on
	  other databases that have no delimiter, backslashing the
	  backslash causes an error. So the only solution that I can come
	  up with is to create an option in res_odbc that explicitly
	  specifies whether or not backslash is a native delimiter. If it
	  is, we use it natively; if not, we use the ESCAPE clause to make
	  it one. Reported by: elguero Patch by: tilghman (Closes issue
	  #11364)

2007-11-24 16:59 +0000 [r89534-89545]  Tilghman Lesher <tlesher@digium.com>

	* res/res_adsi.c: Free some frames that would otherwise leak on
	  error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
	  issue #11351)

	* apps/app_voicemail.c, main/app.c: Currently, zero-length
	  voicemail messages cause a hangup in VoicemailMain. This change
	  fixes the problem, with a multi-faceted approach. First, we do
	  our best to avoid these messages from being created in the first
	  place, and second, if that fails, we detect when the voicemail
	  message is zero-length and avoid exiting at that point. Reported
	  by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)

	* main/manager.c: Up until this point, the XML output of the
	  manager has been technically invalid, due to the repetition of
	  certain parameters in a single event. This caused various issues
	  for XML parsers, some of which refused to parse at all, given the
	  invalidity of the rendered XML. So this commit fixes the XML
	  output, ensuring that each entity parameter has a unique name,
	  thus ensuring valid XML. Reported by: msetim Patch by: tilghman
	  (Closes issue #10220)

	* res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
	  not just 2nd-Nth parameters. Reported by: apsaras Patch by:
	  tilghman (Closes issue #11353)

2007-11-22 17:29 +0000 [r89527]  Russell Bryant <russell@digium.com>

	* configs/agents.conf.sample: mvanbaak pointed out a spelling error
	  in this sample configuration file. While I was at it, I went
	  ahead and tweaked it a little bit more.

2007-11-21 19:27 +0000 [r89493-89495]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a small error I made in my previous commit

	* apps/app_queue.c: Changing an inaccurate debug message to be less
	  inaccurate. Under the circumstances, this message would always
	  report that there were 0 members available, even though that may
	  not be true.

2007-11-21 18:59 +0000 [r89491]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: If a channel gets masqueraded in the middle
	  of a park, don't play the announcement to the masqueraded
	  channel, and dial back to the original channel on timeout.

2007-11-20 19:16 +0000 [r89461-89462]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/module.h: re-doxygen some comments

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: bring back compile-option checking
	  when loading modules, only this time use a string-based storage
	  and comparison mechanism because it is easier to support on other
	  platforms

2007-11-20 17:50 +0000 [r89457]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: According to comments in main/pbx.c, it is essential
	  that if we are going to lock the conlock as well as the hints
	  lock, it must be locked in that respective order. In order to
	  prevent a potential deadlock, we need to lock the conlock prior
	  to locking the hints lock in ast_hint_state_changed (see the call
	  stack example on issue #11323 for how this can happen). (closes
	  issue #11323, reported by eelcob, suggestion for patch by eelcob,
	  patch by me)

2007-11-20 15:22 +0000 [r89450]  Steve Murphy <murf@digium.com>

	* doc/queues-with-callback-members.txt: closes issue #11324; break
	  statements missing in switch cases.

2007-11-20 13:40 +0000 [r89445]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: added RR patch from iroot #10908, thanks.

2007-11-19 15:53 +0000 [r89416-89419]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Print out the correct filename
	  (features.conf) in the log message when parkpos options are
	  incorrect. (closes issue #11295) Reported by: Laureano Patches:
	  res_features.c.patch uploaded by Laureano (license 265)

	* doc/localchannel.txt: Clarify documentation a bit, include that a
	  frame has to pass through the core in order for the Local channel
	  optimization to happen. (closes issue #11246) Reported by: jon

2007-11-16  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.14 released.

2007-11-16 22:26 +0000 [r89339]  Russell Bryant <russell@digium.com>

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: Temporarily revert revision 89325,
	  which added md5 magic for keeping track of what build options
	  were used. We agreed that we should remove this before making a
	  1.4 release, and then we can put it back in. Then, we can take a
	  month or so to play around with it to get it how we want it.

2007-11-16 16:47 +0000 [r89325]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: To help combat problems where
	  people build external modules (asterisk-addons or others) and
	  then change the build options of the Asterisk build in a way that
	  makes the incompatible without warning, this commit introduces an
	  MD5 signature of the important build-time options and includes
	  that signature into modules when they are built. When the loader
	  loads one of these modules and notices the problem, it will emit
	  a warning to console and refuse to initialize the module, as
	  doing so could cause the system to be unstable or even crash. If
	  you upgrade to this version of Asterisk, you must rebuild *all*
	  of your modules that came from other sources before trying to run
	  this version. If you are using Digium's G.729 binary codec
	  module, you will need v33 or newer.

2007-11-16 15:28 +0000 [r89323]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make realtime queues accessible from the
	  QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
	  patched by atis, with small modifications from me)

2007-11-15 18:37 +0000 [r89298-89302]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Start Asterisk in Debian at a more reasonable time
	  (since zaptel is at level 20)

	* channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
	  by valgrind

	* channels/chan_iax2.c: Yet another memory corruption issue.
	  Reported by: atis Patch by: tilghman Fixes issue #10923

2007-11-15 17:19 +0000 [r89296]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Update the SLAStation application to account
	  for the case where the SLA thread has a call out to the station,
	  but the user has pressed a line button to answer the call instead
	  of picking up the handset. If they do, the phone sends out a new
	  INVITE. So, the SLAStation app must check to see if it is picking
	  up a ringing trunk, and ensure that the other stations stop
	  ringing. (reported internally, patched by me, tested by mogorman)

2007-11-15 14:57 +0000 [r89286-89288]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Undoing previous commit since I realize it was
	  wrong

	* main/manager.c: Adding a missing mutex unlock. (closes issue
	  11256, reported and patched by ys)

2007-11-15 11:26 +0000 [r89280-89281]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't send re-invites during pending INVITE
	  transactions. Patch by one47 - thanks! Closes issue #9305

	* channels/chan_sip.c: Improve support for multipart messages. Code
	  by gasparz, changes by me (mostly formatting). Thanks, gasparz!
	  Closes issue #10947

2007-11-14 23:23 +0000 [r89275]  Tilghman Lesher <tlesher@digium.com>

	* main/app.c: When a recording ends with '#', we are improperly
	  trimming an extra 200ms from the recording. Reported by: sim
	  Patch by: tilghman Closes issue #11247

2007-11-14 01:15 +0000 [r89260]  Joshua Colp <jcolp@digium.com>

	* main/srv.c: Return the proper value when the srv_callback
	  function executes properly. (closes issue #11240) Reported by:
	  jtodd

2007-11-13 21:07 +0000 [r89248-89254]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
	  systems which require a third arg to open() when using O_CREAT.
	  Issue 11238, reported by puzzled.

	* res/res_features.c: Revert change from revision 67064. It is
	  documented behavior that if a parking extension already exists
	  while using PARKINGEXTEN, dialplan execution will continue. If
	  blind transferring to a Park with PARKINGEXTEN, you must keep
	  this in mind, and handle the failure yourself. Issue 11237,
	  reported by jon.

2007-11-13 17:34 +0000 [r89246]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: If we set a value for qualify, we should
	  actually pay attention to it, instead of overriding the value

2007-11-13 16:02 +0000 [r89241]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Reverting commit made in revision 89205
	  since it is unnecessary. Thanks to Kevin for pointing this out

2007-11-13 13:51 +0000 [r89239]  Tilghman Lesher <tlesher@digium.com>

	* main/utils.c: Debugging is running into the 16-lock limit.
	  Increase to avoid. (This define is only effective when debugging
	  is turned on, so there's no effect for most installations.)

2007-11-13 00:56 +0000 [r89205]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
	  only 1 argument is given, then the args.options and
	  args.post_process strings are uninitialized and could contain
	  garbage. This change handles this situation properly by only
	  using arguments that we have parsed.

2007-11-12 20:46 +0000 [r89194]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev

2007-11-12 20:16 +0000 [r89184-89191]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c: If two config writes collide, file corruption
	  could result. Use a mkstemp() file, instead. Reported by:
	  paravoid Patch by: tilghman Closes issue #10781

	* main/channel.c, channels/chan_sip.c: Fix two cases of memory
	  corruption caused by background threads. Reported by: atis Patch
	  by: tilghman Fixes issue #10923

2007-11-12 11:26 +0000 [r89169-89173]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
	  no number was dialed and overlapdial is set, we wait for the ISDN
	  timeout instead of starting our own timer. added a comment for
	  the misdn.conf.sample for the overlapdial config option.

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
	  channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
	  restart all interfaces Restart_Indicator, to automatically send a
	  RESTART after the L2 of a PTP Port comes up. Also fixed some
	  places where we have send a RELEASE without need for it.

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
	  state/event issue with overlapdial=yes when no extension matched.
	  removed the general sending of a RELEASE_COMPLETE when we receive
	  a RELEASE, this is done by mISDNuser/mISDN. This makes it
	  possible to use asterisk-1.4 with mISDN trunk, but requires users
	  of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
	  (when using the NT mode at all)

	* channels/misdn/isdn_lib.c: fixed the support for CW and therefore
	  for the reject_cause option.

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  aded ntkeepcalls option, to avoid droÃpping calls when the L2
	  goes down on a PTP link. There are some pbx which do turn off the
	  L1 for a very short while and restart it immediately. normally
	  T310 should be started and after 10 seconds or so the calls
	  should be dropped, this is a simple fix wihtout this timer.

2007-11-08 23:52 +0000 [r89125]  Jason Parker <jparker@digium.com>

	* main/say.c: Properly say the seconds here.. Issue 11203, fix
	  described by vma.

2007-11-08 21:00 +0000 [r89119]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Rework of the commit I made yesterday to use
	  the already built-in ast_uri_decode function as opposed to my
	  home-rolled one. Also added comments. Thanks to oej for pointing
	  me in the right direction

2007-11-08 18:45 +0000 [r89115]  Jason Parker <jparker@digium.com>

	* configs/res_odbc.conf.sample: Avoid warnings on load when using
	  sample configuration files. Issue 11195, patch by eliel.

2007-11-08 16:47 +0000 [r89111]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: I made this same adjustment in trunk to fix
	  a bug, and it makes sense to do it in 1.4 as well. If an
	  imapfolder is specified in voicemail.conf, don't ever explicitly
	  connect to INBOX since it may not exist.

2007-11-08 05:26 +0000 [r89105]  Kevin P. Fleming <kpfleming@digium.com>

	* main/srv.c: fix a glaring bug in the new SRV record handling that
	  would cause incorrect weight sorting

2007-11-08 04:55 +0000 [r89103]  Tilghman Lesher <tlesher@digium.com>

	* doc/valgrind.txt: Typo

2007-11-08 02:26 +0000 [r89095-89101]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Do not add a sip: to the beginning of the To
	  URI unless needed. (closes issue #10756) Reported by: goestelecom

	* channels/chan_sip.c: Improve the devicestate logic for multiple
	  devices. If any are available then the extension is considered
	  available. (closes issue #10164) Reported by: nic_bellamy
	  Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
	  (license 299)

	* channels/chan_sip.c: Add support for allowing one outgoing
	  transaction. This means if a response comes back out of order
	  chan_sip will still handle it. I dream of a chan_sip with real
	  transaction support. (closes issue #10946) Reported by: flefoll
	  (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
	  Reported by: atca_pres

	* channels/chan_sip.c: If callerid is configured in sip.conf use
	  that for checking the presence of an extension in the dialplan.
	  (closes issue #11185) Reported by: spditner

2007-11-07 23:39 +0000 [r89093]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: The member refcount must be incremented, to
	  avoid using it after deallocation. A huge thanks go to lvl- for
	  patiently providing the necessary valgrind output that was
	  necessary to finding this problem of memory corruption. Reported
	  by: lvl- Patch by: tilghman Closes issue #11174

2007-11-07 22:40 +0000 [r89090]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This patch makes it possible for SIP phones
	  to dial extensions defined with '#' characters in extensions.conf
	  AND maintain their escaped characters when forming URI's (closes
	  issue #10681, reported by cahen, patched by me, code review by
	  file)

2007-11-07 21:40 +0000 [r89088]  Steve Murphy <murf@digium.com>

	* cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
	  10578, I just ran 1.4 thru valgrind; some of the config leakage
	  I've already fixed, but it doesn't hurt to double check. I found
	  and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
	  tho.

2007-11-07 15:56 +0000 [r89085]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Fixing a segfault in the manager "core show
	  channels concise" command. (closes issue #11183, reported by arnd
	  and patched by ys)

2007-11-07 04:07 +0000 [r89079]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.ael.sample: Suppress AEL warnings on load.
	  Reported by: eliel Patch by: eliel Closes issue #11178

2007-11-06 20:18 +0000 [r89053]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix init_classes() so that classes that
	  actually do have files loaded aren't treated as empty, and
	  immediately destroyed ...

2007-11-06 19:09 +0000 [r89046]  Jason Parker <jparker@digium.com>

	* codecs/codec_zap.c: Correctly set the total number of channels
	  from a zaptel transcoder board. SPD-49, patch by Matthew
	  Nicholson.

2007-11-06 19:09 +0000 [r89045]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: We went to the trouble of creating a
	  method of tracking failed trylocks, then never turned it on
	  (oops).

2007-11-06 18:53 +0000 [r89042]  Olle Johansson <oej@edvina.net>

	* main/tdd.c: Bug fixes to tdd support in zaptel.

2007-11-06 18:20 +0000 [r89037]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: If someone were to delete the files used
	  by an existing MOH class, and then issue a reload, further use of
	  that class could result in a crash due to dividing by zero. This
	  set of changes fixes up some places to prevent this from
	  happening. (closes issue #10948) Reported by: jcomellas Patches:
	  res_musiconhold_division_by_zero.patch uploaded by jcomellas
	  (license 282) Additional changes added by me.

2007-11-06 17:52 +0000 [r89036]  Steve Murphy <murf@digium.com>

	* main/config.c: closes issue #8786 - where the [catname](!) and
	  [catname](othercat1,othercat2,...) notation gets dropped across a
	  ConfigUpdate (or any other thing that would cause a config file
	  to be written). While I was at it, I also cleaned up some of the
	  destroy routines to free up comments, which was not being done.
	  Made sure the new struct I introduced is also cleaned up properly
	  at destruction time. My code handles multiple template
	  inclusions. Many thanks to ssokol for his patch, which, while not
	  literally used in the final merge, served as a foundation for the
	  fix.

2007-11-06 17:08 +0000 [r88994-89032]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make it so that if a peer is determined to
	  be unreachable using qualify their devicestate will report back
	  unavailable. (closes issue #11006) Reported by: pj

	* channels/chan_zap.c: Fix improbable but possible memory leaks in
	  chan_zap. (closes issue #11166) Reported by: eliel Patches:
	  chan_zap.c.patch uploaded by eliel (license 64)

2007-11-06 13:50 +0000 [r88931]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Remove some checks to see if locks are
	  initialized from the non-DEBUG_THREADS versions of the lock
	  routines. These are incorrect for a number of reasons: - It
	  breaks the build on mac. - If there is a problem with locks not
	  getting initialized, then the proper fix is to find that place
	  and fix the code so that it does get initialized. - If additional
	  debug code is needed to help find the problem areas, then this
	  type of things should _only_ be put in the DEBUG_THREADS
	  wrappers.

2007-11-06 02:52 +0000 [r88862]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/srv.h: update comment to match the state of the
	  code

2007-11-05 23:29 +0000 [r88826]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Reworked deadlock avoidance in __ast_read.
	  Restored audio to callback agents. (closes issue #11071, reported
	  by callguy, patched by me, tested by callguy and Ted Brown)

2007-11-05 22:07 +0000 [r88709-88805]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
	  to channel variables, I went looking around at the ways that
	  channel variables are handled. In general, they were not handled
	  in a thread-safe way. The channel _must_ be locked when reading
	  or writing from/to the channel variable list. What I have done to
	  improve this situation is to make pbx_builtin_setvar_helper() and
	  friends lock the channel when doing their thing. Asterisk API
	  calls almost all lock the channel for you as necessary, but this
	  family of functions did not. (closes issue #10923, reported by
	  atis) (closes issue #11159, reported by 850t)

	* channels/chan_sip.c: When traversing the list of channel
	  variables here in transmit_invite(), the asterisk channel must be
	  locked, as this data may change at any time. (I have seen
	  numerous reports of crashes related to the handling of channel
	  variables. There are a couple of issues on the bug tracker
	  related to it, but it has also been noted on IRC and mailing
	  lists. So, I am finding and fixing some places where channel
	  variables are handled improperly.)

	* channels/chan_sip.c: Fix up some indentation.

	* main/srv.c, include/asterisk/srv.h: Merge changes from
	  asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
	  record support in Asterisk was broken. There was no guarantee on
	  what record Asterisk would choose to actually use. This set of
	  changes improves the situation by ensuring that Asterisk will
	  choose the highest priority record.

	* main/channel.c: Merge the last bit of changes from
	  asterisk/team/russell/readq-1.4 The issue here is that the
	  channel frame readq handling got broken when the code was
	  converted to use the linked list macros. It caused corruption of
	  the list head and tail pointers. So, I fixed up the usage of the
	  linked list macros and in passing, simplified the code. I also
	  documented what the code is doing, as it was a bit difficult to
	  figure out at first. This bug showed itself with crashes showing
	  messed up head/tail pointers for the readq. However, there are a
	  couple of crashes that aren't quite as obvious, but I think may
	  be related. So, if your bug gets closed by this commit, but you
	  still have a problem, please reopen or create a new bug report.
	  (closes issue #10936) (closes issue #10595) (closes issue #10368)
	  (closes issue #11084) (closes issue #10040) (closes issue #10840)

2007-11-05 18:47 +0000 [r88671]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If a SIP channel is put on hold multiple
	  times do not keep incrementing the onHold value. (closes issue
	  #11085) Reported by: francesco_r Tested by: blitzrage (closes
	  issue #10474) Reported by: acennami

2007-11-05 17:46 +0000 [r88624]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix up datastore handling in ast_do_masquerade().
	  The code is intended to move any channel datastores from the old
	  channel to the new one. However, it did not use the linked list
	  macros properly to accomplish the task. The existing code would
	  only work if there was only a single datastore on the old
	  channel.

2007-11-05 17:19 +0000 [r88585]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Make sure we destroy the config structure on
	  configuration failure. Issue 11163, patch by eliel.

2007-11-05 16:20 +0000 [r88539]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Don't check used pooled connections for
	  connection status, as it will cause issues for prepared queries.
	  Reported by: Nick Gorham (via -dev list) Patch by: tilghman

2007-11-04 22:38 +0000 [r88471]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/stringfields.h, main/channel.c,
	  apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
	  Rename ast_string_field_free_pool to
	  ast_string_field_free_memory, and ast_string_field_free_all to
	  ast_string_field_reset_all to avoid misuse (due to too similar
	  names and an error in documentation). Fix two related memory
	  leaks in app_meetme. No need to merge to trunk, different fix
	  already applied there. Not applicable to 1.2

2007-11-02 20:49 +0000 [r88328-88366]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make subscribecontext behave as advertised.
	  It will now look for the presence of a hint in the given context
	  (be it subscribecontext or context). (closes issue #10702)
	  Reported by: slavon

	* channels/chan_sip.c: If an INFO request within a dialog is
	  received with a content length of 0 simply send back a 200 OK. It
	  is valid to do this and the remote side is probably using it to
	  make sure the signalling is still alive. (closes issue #5747)
	  Reported by: chandi Patches: infofix-81430-1.patch uploaded by
	  IgorG (license 20)

2007-11-02 16:51 +0000 [r88283]  Jason Parker <jparker@digium.com>

	* main/say.c: We need to make sure to specify a language to
	  ast_fileexists, otherwise it may fail for anything besides en
	  Issue 11147, fix discovered by both citats and myself
	  (independently), with input from Corydon76

2007-11-02 13:03 +0000 [r88116-88210]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
	  Patch by: ys Closes issue #11143

	* doc/valgrind.txt (added): Add some notes on using valgrind

2007-11-01 16:21 +0000 [r88078]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Make sure we set the poll fds to NULL after
	  free()ing it. Part of issue 11017, patch by tzafrir.

2007-11-01 13:27 +0000 [r87970-88026]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Fix up commit for my Zap channel with spies in
	  Meetme fix. (thanks Tony Mountifield!)

	* apps/app_meetme.c: If a Zap channel contains a spy or a spy is
	  added take it out of the conference in kernel space and make it
	  go through Asterisk so the spy gets audio from both sides.
	  (closes issue #10060) Reported by: mparker

2007-10-31 21:23 +0000 [r87906-87908]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Make sure we free some allocated memory before
	  returning. Issue 11131, patch by eliel.

	* channels/chan_gtalk.c: Don't try to allocate memory that we're
	  just going to re-allocate later anyways. Issue 11130, patch by
	  eliel.

2007-10-31 18:03 +0000 [r87852]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Create samples for ALL of the available options in
	  asterisk.conf

2007-10-31 17:49 +0000 [r87775-87849]  Steve Murphy <murf@digium.com>

	* pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
	  save' cli command saves a file where the semicolon is not
	  escaped. Fixed this; User also wanted comments to be preserved
	  across dialplan save, but this is impossible at this point in
	  time, because comments are not stored in the dialplan. They are
	  'compiled' out of extensions.conf. The only way to preserve those
	  comments is to use the config file reader/writer that the GUI
	  uses to allow online user edits. extensions.conf is first and
	  foremost, a config file, and is read in by the normal config-file
	  reading routines. Then, it is processed into a dialplan
	  (context/exten structs).

	* pbx/pbx_ael.c: Included some verbage in the check_includes func,
	  to inform the user that included contexts that have no match in
	  the AEL, might be OK, as AEL cannot check in the extensions.conf
	  or the in-memory contexts, as they may not be there at the time
	  of the check.

2007-10-30 23:02 +0000 [r87739]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
	  Reported by: ys Fixed by: ys Closes issue #11116

2007-10-30 21:19 +0000 [r87686]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Merge the changes from
	  team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
	  race condition related to the handling of POKEing peers.
	  Essentially, a reference to a peer is held by the scheduler when
	  there are pending callbacks, but the reference count didn't
	  reflect it. So, it was possible for a peer to hit a reference
	  count of zero and have its destructor begin to be called at the
	  same time that the scheduler thread ran a POKE related callback.
	  If that happened, a crash would likely occur. (closes issue
	  #11082, closes issue #11094)

2007-10-30 20:29 +0000 [r87650]  Jason Parker <jparker@digium.com>

	* channels/Makefile: Only try to clean out h323/ if the
	  h323/Makefile exists.

2007-10-30 16:13 +0000 [r87571]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Add two more checks before printing out a
	  warning message about bridging. If either channel has hungup of
	  course the bridge will have failed. (closes issue #10009)
	  Reported by: dimas

2007-10-30 15:45 +0000 [r87567]  Jason Parker <jparker@digium.com>

	* main/editline/np/vis.c: Fix build of editline on Solaris. Issue
	  11113, patch by snuffy.

2007-10-30 15:10 +0000 [r87534]  Joshua Colp <jcolp@digium.com>

	* apps/app_followme.c: Return 1.4 to a state where it builds.
	  Changing the arguments to a function and not changing where they
	  are used is bad, mmmk?

2007-10-30 14:31 +0000 [r87514]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Fix issue where the recorded name wasn't
	  getting removed correctly. (closes issue #11115) Reported by:
	  davevg Patches: followme-v3.diff

2007-10-29 22:13 +0000 [r87460-87465]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/gsm: missed one directory

	* codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
	  codecs/lpc10, main/db1-ast, main/editline, main,
	  codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
	  codecs/lpc10/Makefile, utils, codecs, agi,
	  main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
	  clean up (and ignore) assembler and preprocessor intermediate
	  files if any are created during the build

	* Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
	  already there (used when debugging preprocessor issues) because
	  the compiler will whine about each compile command

2007-10-29 21:06 +0000 [r87427]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing a completely unnecessary quota
	  check from IMAP code.

2007-10-29 20:22 +0000 [r87373-87396]  Russell Bryant <russell@digium.com>

	* main/utils.c, include/asterisk/lock.h: Add some more details to
	  the output of "core show locks". When a thread is waiting for a
	  lock, this will now show the details about who currently has it
	  locked. (inspired by issue #11100)

	* main/astmm.c: Remove a lock that doesn't make any sense. The
	  regions lock needs to be held when traversing the list of
	  allocated chunks so that they can be printed out to the CLI.
	  (Thanks to eliel on #asterisk-dev for pointing this out!)

2007-10-29 17:20 +0000 [r87342]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue where if both sides of the dialog
	  cancelled the dialog at the same time chan_sip could kepe
	  retransmitting a response for no reason. (closes issue #9566)
	  Reported by: atca_pres Patches: bug9566.patch uploaded by oej

2007-10-29 17:13 +0000 [r87340]  Jason Parker <jparker@digium.com>

	* funcs/func_realtime.c, funcs/func_cut.c: Allow some function
	  modules to compile under dev mode. Issue 11104, patch by andrew.

2007-10-29 14:23 +0000 [r87294]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Fix issue with ast_unescape_semicolon going into an
	  endless loop. (closes issue #10550) Reported by: ramonpeek
	  Patches: unescape-85177-1.patch uploaded by IgorG (license 20)

2007-10-28 13:46 +0000 [r87262]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
	  funcs/func_cut.c: Add autoservice to several more functions which
	  might delay in their responses. Also, make sure that func_odbc
	  functions have a channel on which to set variables. Reported by
	  russell Fixed by tilghman Closes issue #11099

2007-10-26 16:34 +0000 [r87168]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
	  pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
	  include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
	  utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
	  pbx/ael/ael.flex: closes issue #11086 where a user complains that
	  references to following contexts report a problem; The problem
	  was REALLy that he was referring to empty contexts, which were
	  being ignored. Reporter stated that empty contexts should be OK.
	  I checked it out against extensions.conf, and sure enough, empty
	  contexts ARE ok. So, I removed the restriction from AEL. This,
	  though, highlighted a problem with multiple contexts of the same
	  name. This should be OK, also. So, I added the extend keyword to
	  AEL, and it can preceed the 'context' keyword (mixed with
	  'abstract', if nec.). This will turn off the warnings in AEL if
	  the same context name is used 2 or more times. Also, I now call
	  ast_context_find_or_create for contexts now, instead of just
	  ast_context_create; I did this because pbx_config does this. The
	  'extend' keyword thus becomes a statement of intent. AEL can now
	  duplicate the behavior of pbx_config,

2007-10-26 13:54 +0000 [r87120]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_curl.c: The addition of autoservice to func_curl
	  additionally made func_curl dependent on the existence of a
	  channel, with no real reason. This should make func_curl once
	  again work without a channel. Reported by jmls. Fixed by
	  tilghman. Closes issue #11090

2007-10-25 23:03 +0000 [r87069]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, include/asterisk/linkedlists.h: appending one
	  list to another should leave the first list empty, and not
	  require the user to do that

2007-10-25 22:53 +0000 [r87067]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_cut.c: Backport alternate encoding of newline
	  delimiters from trunk to 1.4, as approved by Russell Reported by
	  blitzrage Closes issue #10903

2007-10-24 20:56 +0000 [r86982]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Correctly respect hidecalleridname
	  configuration option. Simplify code slightly in the process.
	  Issue 11079, reported by ddv2005

2007-10-24 04:14 +0000 [r86880-86936]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
	  to specify app:spec in hint arguments

	* funcs/func_logic.c: closes issue #11052 -- where nothing after
	  the ? will allow un-initialized variable values to corrupt and
	  crash asterisk on 64-bit platforms

	* main/Makefile: this update to Makefile corrects how ast_expr2f.c
	  should be generated

	* main/ast_expr2f.c: This should get rid of a really, really
	  irritating warning generated by some 64-bit platforms from libc,
	  where free(0) is frowned upon

2007-10-22 21:36 +0000 [r86836]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: If lock tracking is not enabled, then we
	  can not attempt to log any mutex failures. If so, we could end up
	  in infinite recursion. The only lock that is affected by this is
	  a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
	  issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
	  (license 281)

2007-10-22 17:38 +0000 [r86787]  Tilghman Lesher <tlesher@digium.com>

	* main/astmm.c: Minor FreeBSD build fix

2007-10-22 16:35 +0000 [r86754-86756]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: After reading online I have confirmed that
	  Record-Route headers should be copied to 1xx responses as well.
	  (closes issue #10113) Reported by: makoto

	* apps/app_controlplayback.c: Make sure res is a positive value
	  before performing the check to determine whether the user stopped
	  it or not. (closes issue #11023) Reported by: cfc

2007-10-22 15:52 +0000 [r86726-86750]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't leak a frame in the case that an END frame
	  is received and the time since the BEGIN is less than that of the
	  defined minimum DTMF duration. (closes issue #11051) Reported by:
	  casper Patches: channel.c.86664.diff uploaded by casper (license
	  55)

	* include/asterisk/lock.h: Update the static mutex initializer to
	  include the initialization of the internal mutex used to protect
	  the lock debugging data. (closes issue #11044, patch suggested by
	  Ivan)

2007-10-22 14:48 +0000 [r86694]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Account for the fact that sometimes headers
	  may be terminated with \r\n instead of just \n (closes issue
	  #11043, reported by yehavi)

2007-10-22 14:27 +0000 [r86630-86663]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Move log message to before the frame it
	  references is freed. (closes issue #11050) Reported by: slavon
	  Patches: channel.c.86662.diff uploaded by casper (license 55)

	* pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
	  issue #11041) Reported by: jsmith Patches:
	  asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
	  (license 176)

	* main/loader.c: Fixes for building under OpenSolaris. (closes
	  issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
	  uploaded by snuffy (license 35)

2007-10-22 09:21 +0000 [r86598]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
	  DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
	  does not match after an overlap call. Also added out_cause=1

2007-10-19 16:38 +0000 [r86469-86502]  Joshua Colp <jcolp@digium.com>

	* main/app.c: When returning a DTMF digit from
	  ast_control_streamfile cast it as a char so that 0 does not
	  overlap with the success return code. (closes issue #11023)
	  Reported by: cfc

	* channels/chan_sip.c: Fix two issues with domains and transfers.
	  If a port was given in the hostname it was treated as part of the
	  hostname. If domains were configured but external domains were
	  not enabled all transfers would be considered remote. (closes
	  issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
	  uploaded by ramonpeek (license 266)

	* channels/chan_sip.c: Set port number in received as information
	  for registrations as well. (closes issue #11028) Reported by:
	  brad-x

2007-10-19 01:45 +0000 [r86438]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Fixed OSP module did not report
	  source/devinfo IP in correct format.

2007-10-18 22:01 +0000 [r86405-86406]  Jason Parker <jparker@digium.com>

	* Makefile: Correct documentation. I removed the wrong line..

	* Makefile: Add documentation for options in asterisk.conf Issue
	  11029, patch by eserra

2007-10-18 21:16 +0000 [r86330-86372]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
	  commit.

	* configs/iax.conf.sample, channels/chan_iax2.c: Add support for
	  setting the maximum trunk size for IAX2 trunking

	* main/channel.c, include/asterisk/channel.h: The channel needs to
	  stay locked while running timer callbacks, as they access and
	  modify channel data that may change elsewhere. I went through
	  every timer callback in the source tree to make sure that none of
	  them did any additional locking that could introduce deadlocks,
	  and all is well. (closes issue #10765) Reported by: Ivan Patches:
	  ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
	  229)

2007-10-18 17:38 +0000 [r86328]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: If a non-existent file is specified to be
	  played either as a periodic announcement or as a hold/position
	  announcement, the caller would be kicked out of the queue. No
	  longer does this happen.

2007-10-18 15:45 +0000 [r86237-86296]  Russell Bryant <russell@digium.com>

	* codecs/codec_zap.c: Execute the RELEASE operation on transcoder
	  channels in the destroy callback. (patch from jsloan)

	* main/utils.c: Revert a change that I made for issue #10979 which,
	  as has been pointed out to me in issue #11018, doesn't really
	  make sense. There is no reason to have the base64 decode function
	  force a '\0' terminated buffer, when the result is almost always
	  binary, anyway. In fact, this caused some breakage, as some code
	  in res_crypto passed in a buffer exactly the right size to get
	  its binary result, which got stomped on by this patch. (closes
	  issue #11018, reported by dimas)

2007-10-17 21:39 +0000 [r86202]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changing the strategy field of the call_queue
	  struct to be signed instead of unsigned, since the code attempts
	  to set the strategy to -1 if you specify a bogus strategy. While
	  this isn't a huge issue in 1.4, it could be a problem for someone
	  who, say, tries to use the roundrobin strategy in trunk (despite
	  all the deprecation warnings in 1.4).

2007-10-17 17:57 +0000 [r86149]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: If Asterisk is in the middle of shutting
	  down, respond to OPTIONS with 503 Unavailable. (closes issue
	  #10994) Reported by: eserra Patches: sip-options-503.patch
	  uploaded by eserra (license 45)

2007-10-17 16:58 +0000 [r86117]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Whoops, forgot to remove the original
	  sip_scheddestroy. (closes issue #11010) Reported by: vadim

2007-10-17 15:23 +0000 [r86066]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: When runuser/rungroup is specified, a remote
	  console could only be attained by root (Closes issue #9999)

2007-10-17 15:06 +0000 [r86063]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't schedule dialog destruction if a
	  MESSAGE is received using an existing dialog. (closes issue
	  #11010) Reported by: vadim

2007-10-16 23:35 +0000 [r86028-86032]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Since monitor-join is deprecated now,
	  remove the example from the sample queues.conf file

	* UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
	  the monitor-join queue option

	* apps/app_queue.c: Adding deprecated warning to monitor-join
	  option, since the plan is to no longer support this in favor of
	  monitor-type = mixmonitor (related to issue #10885)

2007-10-16 22:36 +0000 [r85994-85997]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: really picky formatting tweak ...

	* include/asterisk/lock.h: Some locking errors exposed the fact
	  that the lock debugging code itself was not thread safe. How
	  ironic! Anyway, these changes ensure that the code that is
	  accessing the lock debugging data is thread-safe. Many thanks to
	  Ivan for finding and fixing the core issue here, and also thanks
	  to those that tested the patch and provided test results. (closes
	  issue #10571) (closes issue #10886) (closes issue #10875) (might
	  close some others, as well ...) Patches: (from issue #10571)
	  ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
	  229) - a few small changes by me

2007-10-16 21:14 +0000 [r85958]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Trying to remove a non-dynamic queue member via
	  dynamic means can lead to some interesting (read nasty)
	  situations. This patch clears up the issue by making only dynamic
	  queue members removable via dynamic methods.

2007-10-16 19:41 +0000 [r85921]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Also set up gmtoff (this is used in the
	  %z gnu extension to strftime) Reported and fixed by jcmoore
	  Closes issue #11002

2007-10-16 19:10 +0000 [r85896]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Remove a pointless lock.

2007-10-16 15:21 +0000 [r85852]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing a double free which happens in the
	  statechange thread. (closes issue #10987, reported by andrew)

2007-10-16 14:52 +0000 [r85818-85850]  Joshua Colp <jcolp@digium.com>

	* apps/app_hasnewvoicemail.c: Check to make sure a value has been
	  given to the VMCOUNT dialplan function. (closes issue #10996)
	  Reported by: marsosa

	* main/threadstorage.c: Fix memory allocation issue in
	  threadstorage. (closes issue #10995) Reported by: snuffy Patches:
	  new-patch.diff uploaded by snuffy (license 35)

2007-10-16 10:46 +0000 [r85800]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Fix the output for this channel help CLI
	  command

2007-10-15 21:10 +0000 [r85717-85720]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Ensure that no pending state changes are leaked
	  when the device state change thread gets stopped on module
	  unload.

	* apps/app_queue.c: Previously, app_queue created a thread to
	  handle every single device state change. I changed this a while
	  ago in trunk for performance reasons. However, bug 8407 points
	  out that it is actually a race condition, causing device state
	  changes to get processed in random order. So, I backported my
	  changes from trunk to 1.4. (closes issue #8407, patch provided by
	  tim_ringenbach, committed patch by me)

2007-10-15 20:29 +0000 [r85687]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Don't execute a gosub if the arguments is
	  zero-len (not just NULL) Reported by davevg Fixed by me Closes
	  issue #10985

2007-10-15 20:21 +0000 [r85686]  Russell Bryant <russell@digium.com>

	* main/say.c: Add a small fix for the tw version of saying dates.
	  (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
	  uploaded by sharkey (license 172)

2007-10-15 20:15 +0000 [r85684]  Jason Parker <jparker@digium.com>

	* Makefile: Properly use DESTDIR in 'config' target. Do not try to
	  run chkconfig or similar if using DESTDIR. Issue 10938, patch by
	  cabal95.

2007-10-15 19:22 +0000 [r85604-85649]  Russell Bryant <russell@digium.com>

	* main/utils.c: Be pedantic about handling memory allocation
	  failure.

	* main/utils.c: The loop in the handler for the "core show locks"
	  could potentially block for some amount of time. Be a little bit
	  more careful and prepare all of the output in an intermediary
	  buffer while holding a global resource. Then, after releasing it,
	  send the output to ast_cli().

	* channels/chan_sip.c: Make the default for the srvlookup option to
	  be yes. It doesn't really make sense for it to default to off.
	  The default configuration file has it on, and proper RFC
	  behavior, as indicated by a comment in the code, is for it to be
	  on. So, let's have it on by default to make lives easier. (closes
	  issue #10954, suggested by jtodd)

2007-10-15 16:39 +0000 [r85571]  Joshua Colp <jcolp@digium.com>

	* configs/features.conf.sample: Document that DTMF based features
	  only work when two channels are bridged together. (closes issue
	  #10773) Reported by: pbayley

2007-10-15 16:34 +0000 [r85561]  Russell Bryant <russell@digium.com>

	* include/asterisk/strings.h: Make a few changes so that characters
	  in the upper half of the ISO-8859-1 character set don't get
	  stripped when reading configuration. (closes issue #10982,
	  dandre)

2007-10-15 16:22 +0000 [r85559]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Bring both DTMF begin and end frames up through to
	  the core for DTMF feature handling. (closes issue #10826)
	  Reported by: dimas

2007-10-15 15:40 +0000 [r85556]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Ensure the buffer passed to
	  ast_canmatch_extension() is properly initialized so that it is
	  null terminated. (issue #10977) Reported by: dimas Patches:
	  pbxdundi.patch uploaded by dimas (license 88) - small mods by me

2007-10-15 14:55 +0000 [r85552]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If Monitor or a spy was added to a P2P or native
	  bridged channel bring the channel back to the generic bridging
	  core so the monitor or spy operations work. (closes issue #10943)
	  Reported by: julianjm

2007-10-15 13:16 +0000 [r85540-85548]  Russell Bryant <russell@digium.com>

	* main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
	  (closes issue #10980) Reported by: casper Patches:
	  db.c.84633.diff uploaded by casper (license 55)

	* main/asterisk.c: Make sure remote consoles unmute themselves
	  again after reconnecting. (closes issue #10847) Reported by: atis
	  Patches: console_unmute_on_reconnect.patch uploaded by atis
	  (license 242)

	* main/utils.c: Make sure that the base64 decoder returns a
	  terminated string. (closes issue #10979) Reported by: ys Patches:
	  util.c.diff uploaded by ys (license 281) - small mods by me

	* pbx/pbx_config.c: Don't create the context for users in
	  users.conf until we know at least one user exists. (closes issue
	  #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
	  dimas (license 88)

2007-10-13 15:26 +0000 [r85536]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.ael.sample: Remove deprecated syntax from
	  sample ael file Reported and patched by: dimas Closes issue
	  #10967

2007-10-13 05:48 +0000 [r85532-85533]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
	  issue with console verbosity when running asterisk -rx to execute
	  a command and retrieve its output. The issue was that there was
	  no way for the main Asterisk process to know that the remote
	  console was connecting in the -rx mode. The way that James has
	  fixed this is to have all remote consoles muted by default. Then,
	  regular remote consoles automatically execute a CLI command to
	  unmute themselves when they first start up. (closes issue #10847)
	  Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
	  by jamesgolovich (license 176)

	* main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
	  handle the case where read() may return the text for more than
	  one CLI command at once for a remote console. (closes issue
	  #10888) Reported by: jamesgolovich Patches:
	  asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
	  176)

2007-10-12 18:30 +0000 [r85523]  Tilghman Lesher <tlesher@digium.com>

	* doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address

2007-10-12 15:45 +0000 [r85515-85517]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
	  SDMI. (closes issue #10959)

	* pbx/pbx_realtime.c: Fix the potential use of an uninitialized
	  buffer in a log message. (closes issue #10958) Reported by: dimas
	  Patches: realtime.patch uploaded by dimas (license 88)

2007-10-11 15:26 +0000 [r85397]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When creating a new packet don't try to stop
	  retransmission of it. It was just allocated/created so it's
	  impossible for it to have already been scheduled. (closes issue
	  #10945) Reported by: flefoll Patches:
	  chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
	  (license 244)

2007-10-11 04:35 +0000 [r85356]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: A dollar sign by itself, not indicating a start of a
	  variable or expression prematurely ends substitution (closes
	  issue #10939)

2007-10-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.13 released.

2007-10-10 15:56 +0000 [r85316]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h: I introduced a new member to the
	  ast_filestream struct in 1.4.12, but put it in the middle of the
	  struct, instead of at the end. One of the Debian folks, paravoid,
	  pointed out that this breaks binary compatability with modules
	  compiled against older headers. So, I'm moving the new member to
	  the end of the struct to resolve the situation.

2007-10-10 15:51 +0000 [r85315]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: The thread ID should be unsigned.

2007-10-10 14:42 +0000 [r85277-85280]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If devicestate is passed a port number strip
	  it out. (closes issue #10930) Reported by: ibc

	* channels/chan_sip.c: Add support for handling a 182 Queued
	  response. (closes issue #10924) Reported by: ramonpeek Patches:
	  queued-182.diff uploaded by ramonpeek (license 266)

2007-10-10 14:26 +0000 [r85276]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: A bunch of changes from sprintf to
	  snprintf. See security advisory AST-2002-022

2007-10-10 14:14 +0000 [r85242]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Close voicemail message description file if
	  duration did not meet the minimum, or else we will eventually run
	  out of file descriptors. (closes issue #10918) Reported by:
	  brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
	  279)

2007-10-10 06:24 +0000 [r85195]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/frame.h: use a macro instead of an inline
	  function, so that backtraces will report the caller of
	  ast_frame_free() properly

2007-10-09 21:55 +0000 [r85158]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, main/utils.c, include/asterisk/lock.h: This
	  commit fixes the following issues: - Deadlock in ast_write (issue
	  #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
	  initialization error in lock.h (issue #10571)

2007-10-09 14:30 +0000 [r84990-85093]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't perform a reinvite if a transfer is in
	  progress. (issue #10915) Reported by: ramonpeek

	* main/rtp.c: Only update codec information if the channel has a
	  technology private structure. (issue #10915) Reported by:
	  ramonpeek

	* main/rtp.c: Update codec information as well as address when
	  doing hold reinvites. (issue #10868) Reported by: mavince

	* main/channel.c: Don't keep trying to native bridge if either of
	  the channels are involved in a masquerade operation to be done.
	  (closes issue #10696) Reported by: tbelder

2007-10-08 03:28 +0000 [r84957]  Russell Bryant <russell@digium.com>

	* Makefile.rules: Enable file dependency tracking for _all_ builds,
	  and not just for builds with dev-mode enabled. I have seen enough
	  problems caused by this that I don't think it's worth keeping. I
	  want to continue to encourage anybody that is interested to
	  continue to run Asterisk from svn. Furthermore, I do not want
	  their systems to break when we change a structure definition in a
	  header file. :)

2007-10-07 16:15 +0000 [r84890-84902]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Presence packets from a client who's connected
	  with our Jabber ID are valid, therefore, those clients must be
	  considered as buddies. The resource string helps us make the
	  distinction between clients. Closes issue #10707, reported by
	  yusufmotiwala.

	* res/res_jabber.c: Prevent Asterisk from crashing when receiving a
	  presence packet without resource from a buddy that is known to
	  have a resource list. Revert a change I previously made, where
	  Asterisk could point to a freed memory location.

2007-10-05 19:42 +0000 [r84851]  Tilghman Lesher <tlesher@digium.com>

	* main/db.c: Log exactly why we can't open the database, if we fail
	  (closes issue #10887)

2007-10-05 18:55 +0000 [r84818]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Update the remembered RTP peer information when
	  putting an endpoint on hold or taking it off hold so that the RTP
	  stack does not initiate a needless reinvite. (closes issue
	  #10868) Reported by: mavince

2007-10-05 16:44 +0000 [r84783]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Do deadlock avoidance in a couple more
	  places. You can't lock two channels at the same time without
	  doing extra work to make sure it succeeds. (closes issue #10895,
	  patch by me)

2007-10-05  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.12.1 released. (This is mainly to include the 
	  app_queue fix for a memory leak on reload, but includes a couple
	  of other bug fixes, as well.)

2007-10-05 01:39 +0000 [r84742]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix a copy/paste error in the description of
	  UpdateConfig that was pointed out by JerJer on #asterisk-dev

2007-10-04 21:57 +0000 [r84692]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Don't allocate space for queue members unless
	  it's needed. You end up deleting dynamic members on a reload. Not
	  good. closes issue (#10879, reported by dazza76, patched by me)

2007-10-04 21:36 +0000 [r84690]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: callers of sig2str already add the word
	  'signalling' in the appropriate place, so don't duplicate it

2007-10-04 14:51 +0000 [r84637]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Create a duplicate of the channel's member name
	  as the tab completion stuff will free it. (closes issue #10884)
	  Reported by: adamg

2007-10-03 22:59 +0000 [r84581]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c: When an RFC 2833 event is sent that we don't
	  recognize, ignore it, don't queue a NULL digit (closes issue
	  #10877)

2007-10-03 18:20 +0000 [r84511-84544]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
	  in a switch expr doesn't execute correctly, because the commas in
	  the function args are not converted to vertbars before the func
	  is called. I modified just the switch code to convert the commas
	  to vertbars if there, but if more of these sort of probs are
	  found, I may have to resort to something a little more
	  fundamental. We'll see, I guess.

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
	  pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
	  null input to a switch statement results in a hangup; since
	  switch is implemented with extensions, and the default case is
	  implemented with a '.', and the '.' matches 1 or more remaining
	  characters, the case where 0 characters exist isn't matched, and
	  the extension isn't matched, and the goto fails, and a hangup
	  occurs. Now, when a default case is generated, it also generates
	  a single fixed extension that will match a null input. That
	  extension just does a goto to the default extension for that
	  switch. I played with an alternate solution, where I just tack an
	  extra char onto all the patterns and the goto, but not the
	  default case's pattern. Then even a null input will still have at
	  least one char in it. But it made me nervous, having that extra
	  char in , even if that's a pretty secret and low-level issue.

2007-10-02  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.12 released.

2007-10-02 20:06 +0000 [r84474]  Russell Bryant <russell@digium.com>

	* Makefile, build_tools/prep_tarball: * Don't build the
	  menuselect-tree for the tarball, as it requires running the
	  configure script first * Change the Makefile to note that
	  menuselect-tree depends on the configure script.

2007-10-02 19:01 +0000 [r84410-84437]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Fix some odd formatting I missed..

	* res/res_features.c: Finish up on transferee channel before return
	  on failure. Issue 10821, patch by Ivan

2007-10-02 14:12 +0000 [r84370]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Use snprintf instead of sprintf in one
	  place. There is no vulnerability here due to various buffer sizes
	  around the code, but I still didn't like seeing a non
	  length-limited copy of data coming off of the wire into a stack
	  buffer, as this would be a problem in the future if buffer sizes
	  elsewhere got changed or size limitations removed ...

2007-10-02 09:48 +0000 [r84345]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: terminate USERUSER String with 0

2007-10-01 21:52 +0000 [r84291]  Jason Parker <jparker@digium.com>

	* Makefile, Makefile.rules, channels/Makefile: Add dist-clean
	  support for subdirs. Change h323 to only remove the Makefile on a
	  dist-clean, rather than a clean. This fixes a bug I found with
	  trying to run make after a make clean

2007-10-01 21:25 +0000 [r84274]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/channel.c, main/manager.c, channels/chan_agent.c: moved
	  get_base_channel() code from action_redirect to
	  ast_channel_masquerade() for issue 7706 and BE-160

2007-10-01 21:18 +0000 [r84273]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...

2007-10-01 21:07 +0000 [r84271]  Russell Bryant <russell@digium.com>

	* main/utils.c, include/asterisk/lock.h: Fulfull a feature request
	  from Qwell on the "core show locks" output. It will now note the
	  lock type for each lock that a thread holds. (mutex, rdlock, or
	  wrlock)

2007-10-01 20:27 +0000 [r84239]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
	  #10777 -- by returning a null for the parse tree when there's
	  really nothing there, and making sure we don't try to do checking
	  on a null tree.

2007-10-01 19:56 +0000 [r84166-84236]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Add another sanity check in the AGI read loop. We
	  really don't care about EAGAIN unless we didn't read an entire
	  line. If there is a newline at the end if the read buffer, break,
	  because we got the whole thing. (reported and patched by bmd)

	* include/asterisk/lock.h: Show rwlocks in the "core show locks"
	  output. Before, it only showed mutexes.

	* channels/Makefile: Remove another file in "make clean". (closes
	  issue #10814, paravoid)

	* apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.

2007-10-01 14:10 +0000 [r84158-84163]  Joshua Colp <jcolp@digium.com>

	* configs/usbradio.conf.sample (removed): Remove chan_usbradio
	  config file from tree, it is not present in here. (closes issue
	  #10839) Reported by: casper

	* res/res_musiconhold.c: Fix randomness. save_pos was being set to
	  0 initially instead of -1, causing it to jump to position 0 when
	  moh started. (closes issue #10859) Reported by: jamesgolovich
	  Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
	  (license 176)

	* apps/app_dial.c: Only attempt early bridging if the options given
	  to Dial() permit it. (closes issue #10861) Reported by: peekyb

2007-09-30 20:02 +0000 [r84146]  Russell Bryant <russell@digium.com>

	* include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
	  modules. The load and reload parameters were in the wrong place.
	  (closes issue #10846, alebm)

2007-09-29 23:00 +0000 [r84133-84135]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
	  (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
	  pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
	  pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
	  pbx/ael/ael-test/ael-ntest22/qq.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1 (added),
	  pbx/ael/ael-test/ael-ntest22/t2 (added),
	  pbx/ael/ael-test/ael-ntest22/t3 (added),
	  pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
	  pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
	  that matches what I did in 84134 for AEL regressions.

	* pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
	  10786; All config files support #include with globbing (you know,
	  *,[chars],?,{list,list},etc), so I've updated the AEL system to
	  support this also.

2007-09-28 14:13 +0000 [r84049-84078]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Correct pronunciations of numbers for .nl (Closes
	  issue #10837)

	* main/channel.c: Avoid a deadlock with ALL of the locks in the
	  masquerade function, not just the pairs of channels. (Closes
	  issue #10406)

2007-09-27 23:12 +0000 [r84018]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/manager.c, channels/chan_agent.c,
	  include/asterisk/channel.h: if an Agent is redirected, the base
	  channel should actually be redirected. This was causing multiple
	  issues, especially issue 7706 and BE-160

2007-09-27 00:01 +0000 [r83976]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: remove a todo item that has been completed

2007-09-26 23:53 +0000 [r83974]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_alsa.c: avoid the weird usage of assert() in the
	  ALSA header files that gcc 4.2 wants to complain about

2007-09-26 21:35 +0000 [r83910-83943]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: I changed my mind ... I think this should be
	  a LOG_NOTICE.

	* channels/chan_sip.c: Add a log message that was requested by the
	  masses in the developer tutorial session at Astricon. chan_sip
	  did not output any message when a call was rejected because the
	  extension was not found. This adds a verbose message (at verbose
	  level 3) to note when this happens.

	* channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
	  (please run the configure script with --enable-dev-mode so this
	  doesn't happen again ...)

2007-09-26 18:35 +0000 [r83879]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Remove unused 4k of memory on the program
	  stack (closes issue #10827)

2007-09-25 14:13 +0000 [r83637-83773]  Tilghman Lesher <tlesher@digium.com>

	* main/app.c: jmls pointed out that unsetting the group and setting
	  the group to the blank string aren't quite the same.

	* build_tools/make_defaults_h: In the source, keys are relative to
	  the datadir, not varlib (which is the same in most cases, but
	  it's good to be accurate). Closes issue #10811

	* doc/realtime.txt: Oops. Removed the unworkable workaround. This
	  note should never have been in the release.

	* main/app.c: Making change to group splitting, as discussed on the
	  -dev list. The main effect of this will be to permit
	  Set(GROUP([cat])=), i.e. unsetting a group.

2007-09-24 07:54 +0000 [r83620]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed round_robin group dial method, this
	  never worked well on BRI Ports (2 channels)

2007-09-22 19:39 +0000 [r83558-83589]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
	  are made here for the first arg in the for(arg1; arg2; arg3) {}
	  statement, as were done for the 3rd arg. It can now be an
	  assignment that will embedded in a Set() app, or a macro call, or
	  an app call.

	* pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
	  statement is now wrapped in Set() only if there's an '=' in that
	  string. Otherwise, if it begins with '&', then a Macro call is
	  generated; otherwise it is made into an app call. A bit more
	  accomodating, keeps the new guys happy, and the guys with ael-1
	  code should be happy, too

2007-09-21 14:37 +0000 [r83432]  Russell Bryant <russell@digium.com>

	* main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
	  channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
	  main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
	  pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
	  channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
	  main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
	  main/db1-ast/hash/hash.c, include/asterisk/channel.h,
	  channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
	  with cosnt pointers. This set of changes gets all of Asterisk
	  (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
	  #10774, patch from qwell)

2007-09-21 13:34 +0000 [r83400]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix video under certain circumstances. It
	  would have been possible for the formats on the channel to not
	  contain the video format. (closes issue #10782) Reported by:
	  cwhuang

2007-09-20 21:16 +0000 [r83316-83348]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: When daemonizing, don't change working directory
	  to "/". It makes it not be able to do a core dump when not
	  running as uid=root. (closes issue #10766, xrg)

	* contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
	  ask for /bin/bash, as it uses bashisms. (closes issue #10772,
	  reported by culrich)

2007-09-20 17:09 +0000 [r83246]  Jason Parker <jparker@digium.com>

	* apps/app_disa.c: If # is pressed after dialing an extension in
	  DISA, stop trying to collect more digits. (issue #10754) Reported
	  by: atis Patches: app_disa.c.branch.patch uploaded by atis
	  (license 242) app_disa.c.trunk.patch uploaded by atis (license
	  242)

2007-09-20 16:25 +0000 [r83230-83232]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make sure the minimum T1 timer value is
	  obeyed in all cases. (closes issue #10768) Reported by: flefoll
	  Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
	  (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
	  flefoll (license 244)

	* channels/chan_sip.c: Fix a minor spelling error. (closes issue
	  #10769) Reported by: flefoll Patches:
	  chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
	  244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
	  (license 244)

2007-09-19 19:50 +0000 [r83121-83179]  Russell Bryant <russell@digium.com>

	* apps/app_system.c: The System() and TrySystem() applications can
	  take a substantial amount of time to execute while not servicing
	  the channel. So, put the channel in autoservice while the command
	  is being executed. (closes issue #10726, reported by mnicholson)

	* funcs/func_curl.c: Using curl can take a substantial amount of
	  time, so the channel should be autoserviced while waiting for it
	  to complete. (closes issue #10725, reported by mnicholson)

	* channels/chan_iax2.c: When handling a reload of chan_iax2, don't
	  use an ao2_callback() to POKE all peers. Instead, use an
	  iterator. By using an iterator, the peers container is not locked
	  while the POKE is being done. It can cause a deadlock if the
	  peers container is locked because poking a peer will try to lock
	  pvt structs, while there is a lot of other code that will hold a
	  pvt lock when trying to go lock the peers container. (reported to
	  me directly by Loic Didelot. Thank you for the debug info!)

	* main/manager.c: Fix up another potential race condition. Do the
	  loop decrementing use count on events with the eventq protected
	  from being changed. (reported on IRC by Ivan)

2007-09-19 13:47 +0000 [r83070-83074]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Protect the CDR record from modification by
	  pbx_exec so that the application data contains the Queue data.
	  (closes issue #10761) Reported by: snar Patches:
	  app-queue-mixmonitor.patch uploaded by snar (license 245)

	* channels/chan_sip.c: (closes issue #10760) Reported by: dimas
	  Patches: chan_sip.patch uploaded by dimas (license 88) Read in
	  subscribecontext option in general to be the default.

2007-09-19 09:32 +0000 [r83023-83024]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: removed comment which violates the coding
	  guidelines.

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
	  allow configuring the asterisk dtmf detector instead of the
	  mISDN_dsp ones. also added the patch from irroot #10190, so that
	  dtmf tones detected by the asterisk detector are passed outofband
	  to asterisk, to make any use of dtmf tones at all.

2007-09-19 00:19 +0000 [r82992]  Russell Bryant <russell@digium.com>

	* apps/app_flash.c: Change the description of app_flash to note how
	  it can be a useful tool instead of just saying that it is
	  generally a worthless feature. (Thanks to Jim Van Meggelen for
	  pointing it out and providing the proposed text)

2007-09-18 23:41 +0000 [r82961]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Initialize a variable to NULL to make the world
	  happy.

2007-09-18 22:42 +0000 [r82929]  Russell Bryant <russell@digium.com>

	* include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
	  interrupting the fgets() call when using FastAGI. This version of
	  the patch maintains the original behavior of the code when not
	  using FastAGI. (closes issue #10553) Reported by: juggie Patches:
	  res_agi_fgets-4.patch uploaded by juggie (license 24)
	  res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
	  mods by me Tested by: juggie, festr

2007-09-18 21:49 +0000 [r82887-82913]  Doug Bailey <dbailey@digium.com>

	* main/manager.c: Corrected patch applied in revision r82887.

	* main/manager.c: Fixed a bug where http manager sessions prevented
	  the eventq from being cleaned out because http manager sessions
	  do not have a valid file descriptor.

2007-09-18 20:56 +0000 [r82867]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix a memory leak that can occur on systems under
	  higher load. The issue is that when events are appended to the
	  master event queue, they use the number of active sessions as a
	  use count so it will know when all active sessions at the time
	  the event happened have consumed it. However, the handling of the
	  number of sessions was not properly synchronized, so the use
	  count was not always correct, causing an event to disappear
	  early, or get stuck in the event queue for forever. (closes issue
	  #9238, reported by bweschke, patch from Ivan, modified by me)

2007-09-18 20:09 +0000 [r82865]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Moving the logic for handling an empty
	  membername to the create_member function so that there is a
	  common place where this occurs instead of being spread out to
	  several different places.

2007-09-18 18:59 +0000 [r82834]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: there is no need for conditional logic to
	  select ->interface or ->membername, snince ->membername will
	  always be populated

2007-09-18 16:31 +0000 [r82802]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: When copying the contents from the wildcard
	  peer, do a deep copy instead of shallow copy so that it doesn't
	  crash when beging destroyed. (closes issue #10546, patch by me)

2007-09-18 15:28 +0000 [r82751]  Jason Parker <jparker@digium.com>

	* configs/sip.conf.sample: Correct the allowexternaldomains option
	  in SIP sample config. Issue 10753

2007-09-17 20:16 +0000 [r82594-82676]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
	  ast_localtime() instead of a couple places in app_voicemail to
	  prevent the problem everywhere instead of just a couple of
	  places. (related to issue #10746)

	* apps/app_voicemail.c: Initialize some memory to fix crashes when
	  leaving voicemail. This problem was fixed by running Asterisk
	  under valgrind. (closes issue #10746, reported by arcivanov,
	  patched by me) *** IMPORTANT NOTE: We need to check to see if
	  this same bug exists elsewhere.

	* res/res_features.c: Handle the case where there are multiple
	  dynamic features with the same digit mapping, but won't always
	  match the activated on/by access controls. In that case, the code
	  needs to keep trying features for a match. (reported by Atis on
	  the asterisk-dev list, patched by me)

2007-09-17 16:40 +0000 [r82590-82592]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: revert a change that wasn't supposed to be
	  committed... doh!

	* apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
	  where a logical member name (if specified) was not used, but
	  instead the direct interface was listed

2007-09-17 02:00 +0000 [r82514]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
	  of passing a NULL pointer into snprintf pass "". It makes Solaris
	  much happier.

2007-09-14 21:19 +0000 [r82444]  Steve Murphy <murf@digium.com>

	* main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
	  had to leave out the bit about ending the previous cdr in the
	  fork; it would destroy current implementations.

2007-09-14 21:17 +0000 [r82435]  Russell Bryant <russell@digium.com>

	* configs/zapata.conf.sample: Add a note to help clarify the value
	  set with the echocancel option. (inspired by Malcolm's blog post
	  on blogs.digium.com about HPEC)

2007-09-14 18:35 +0000 [r82396-82398]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Crap, I broke the build. Fixed.

	* apps/app_queue.c: Adding member name field to manager events
	  where they were missing before (closes issue #10721, reported by
	  snar)

2007-09-14 17:48 +0000 [r82394]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: If a channel does not have an owner, do not
	  try to set a channel variable. This will end up making the
	  channel variable global, which is not right. Closes issue #10720,
	  patch by flefoll.

2007-09-14 15:50 +0000 [r82382-82385]  Russell Bryant <russell@digium.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  checking for libusb here, so nobody has to deal with conflicts in
	  the chan_usbradio-1.4 branch every time the configure script gets
	  changed

	* channels/chan_usbradio.c (removed), channels/xpmr (removed),
	  channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
	  It can't live here because we have a strict policy to not include
	  new features in release branches. However, I'm going to merge it
	  into trunk, and I also have a special 1.4 based branch that
	  includes this module. svn co
	  http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4

2007-09-14 14:42 +0000 [r82376]  Mark Michelson <mmichelson@digium.com>

	* doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
	  (closes issue #10717, reported and patched by leedm777)

2007-09-14 01:24 +0000 [r82368]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c: Fixed problem with changes made to cdr
	  functionality

2007-09-14 00:52 +0000 [r82367]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_usbradio.c: this new driver may not live in this
	  branch for long (since it is a new feature), but it definitely
	  should not be built by default

2007-09-14 00:34 +0000 [r82366]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
	  channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
	  channels/xpmr (added), channels/xpmr/LICENSE (added),
	  channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
	  (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
	  channel driver for USB Radio device and support thereof.

2007-09-13 23:11 +0000 [r82358]  Jason Parker <jparker@digium.com>

	* pbx/pbx_spool.c: Fix a small typo. retrytime > waittime

2007-09-13 20:16 +0000 [r82346]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Preemptively fixing a possible segfault. It is
	  possible that queuename is NULL (meaning pause ALL queues), so
	  use q->name instead.

2007-09-13 20:11 +0000 [r82344]  Jason Parker <jparker@digium.com>

	* cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
	  mutliple threads tried to close the same file. Do we actually
	  need the locking here? What happens if you open the same file
	  twice, and two threads try to write to it at the same time? Is
	  fputs() going to write out the entire line at once? I suspect
	  that it could be possible for the second fopen to run during the
	  first fputs, so the position could be in the middle of the
	  previously written line... Issue 10347, initial patch by
	  explidous (but I removed all of the paranoia stuff..)

2007-09-13 18:57 +0000 [r82337-82339]  Russell Bryant <russell@digium.com>

	* main/astobj2.c: resolve a warning when not building under dev
	  mode

	* main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
	  in tracking astobj2 statistics if dev-mode is enabled. Also, when
	  dev mode is enabled, register the CLI command that can be used to
	  run the astobj2 test and print out statistics.

2007-09-13 18:12 +0000 [r82335]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: Merged revisions 82334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
	  | 2 lines clarify the OpenSSL and OpenH323 license exceptions
	  ........

2007-09-13 16:25 +0000 [r82326]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added logic to handle the unlikely case that
	  someone has two queues with the same name. Asterisk will log a
	  warning message letting the user know that one was already
	  defined with that name and is it skipping all further instances.
	  This also will work for realtime queues but in order for that to
	  happen, the user would have to trigger a perfectly timed reload
	  as a realtime queue is being looked up, which is highly unlikely
	  (but taken care of nonetheless).

2007-09-13 11:47 +0000 [r82309]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Closes issue #9401, reported and patched
	  by irrot, with slight modifications by me. Handle DTMF sent by
	  Asterisk properly.

2007-09-12 21:56 +0000 [r82296]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Fix a check of the wrong pointer, as pointed out
	  by an XXX comment left in the code. The problem was harmless,
	  however.

2007-09-12 21:28 +0000 [r82291]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
	  files

2007-09-12 20:24 +0000 [r82286]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_meetme.c: remove a race condition for the creation of
	  recordthread's, and fix a small memory leak. This closes issue#
	  10636

2007-09-12 20:12 +0000 [r82285]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/private.h, main/stdtime/tzfile.h,
	  include/asterisk/localtime.h, main/stdtime/localtime.c: Working
	  on issue #10531 exposed a rather nasty 64-bit issue on
	  ast_mktime, so we updated the localtime.c file from source. Next
	  we'll have to write ast_strptime to match.

2007-09-12 15:16 +0000 [r82278-82280]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Clean up the output of "asterisk -h". This
	  tweaks the wording and wraps lines at 80 characters. (closes
	  issue #10699, seanbright)

	* res/res_agi.c: revert patch from issue #10553, as someone not
	  using fastagi reported that this broke their system.

2007-09-12 14:30 +0000 [r82274-82276]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Accidentally committed changes to
	  app_voicemail which do NOT need to be in the 1.4 branch yet.
	  reverting...

	* apps/app_voicemail.c, apps/app_queue.c: We should only initialize
	  a realtime queue when it is allocated, not every time we access
	  it. This prevents the members ao2_container from being
	  reallocated every time the queue is accessed. I also removed a
	  debug message I had accidentally left in on a previous commit.

2007-09-11 22:37 +0000 [r82267]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
	  these calls was reading bogus memory ...

2007-09-11 21:41 +0000 [r82265]  Joshua Colp <jcolp@digium.com>

	* codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
	  #10679) Reported by: andrew Build under dev mode when K6OPTS is
	  enabled.

2007-09-11 20:49 +0000 [r82263]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix another missing unref of member objects.
	  This one was pointed out by Marta. When building the outgoing
	  list in try_calling(), a member reference is stored in each
	  outgoing entry. However, when this list got destroyed, the
	  reference was not released.

2007-09-11 20:36 +0000 [r82261]  Steve Murphy <murf@digium.com>

	* main/cdr.c: this change should fix issue # 10659 -- what I worry
	  about is how many other bug reports it may generate. Hopefully,
	  we can please the/a majority. Hopefully. We shall see. Calls not
	  marked ANSWERED and with only one channel name will not be
	  posted. This should eliminate the double CDR's.

2007-09-11 16:05 +0000 [r82252]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: All instances of ao2_iterators which were just
	  named 'i' have been renamed to 'mem_iter' so that when refcounted
	  queues are merged into trunk, there will be little confusion
	  regarding iterator names, especially when a queue and member
	  iterator are used in the same function.

2007-09-11 16:03 +0000 [r82250]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: The sample dundi.conf claims support for a
	  wildcard peer entry - [*], but the code did not support it. This
	  patch makes it work. (closes issue #10546, patch by dds, with
	  some changes by me)

2007-09-11 16:01 +0000 [r82249]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
	  hold/retrieve issue.

2007-09-11 15:26 +0000 [r82245]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
	  res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
	  juggie When using fastagi, fgets() can return before a full line
	  is read. Add explicit handling for the case where it gets
	  interrupted.

2007-09-11 14:56 +0000 [r82243]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
	  Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
	  (license 176) Don't leak memory when unloading DUNDi.

2007-09-11 14:34 +0000 [r82198-82240]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Add a couple more missing unrefs of queue
	  member objects

	* apps/app_queue.c: Add a missing unref of a queue member in an
	  error handling block

	* apps/app_queue.c: Document why membercount can not simply be
	  replaced by ao2_container_count()

	* main/astobj2.c: backport astobj2 race condition fix. This
	  function is the exact same as trunk so it applies here as well.

2007-09-10 18:02 +0000 [r82155]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Convert struct member to use refcounts (closes
	  issue #10199)

2007-09-10 15:02 +0000 [r82091]  Mark Michelson <mmichelson@digium.com>

	* configs/misdn.conf.sample: Removing non-existent options from
	  misdn configuration sample. (closes issue #10678, reported and
	  patched by IgorG)

2007-09-09 02:35 +0000 [r82028]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix inline compiles on really old
	  compilers (who uses gcc 2.7 anymore, really?)

2007-09-08 18:41 +0000 [r81952-81997]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
	  did not free the entry it removed.

	* .cleancount: (closes issue #10672) Bump the cleancount so that a
	  "make clean" will be forced. This is needed because my fix in
	  revision 81599 made a change to a data structure in file.h, and
	  since file dependency tracking is only on with dev-mode enabled,
	  file format modules that don't get rebuilt may crash, as is the
	  case with this issue. This makes me wonder - how much faster does
	  the code build without the file dependency tracking enabled? If
	  it doesn't make much of a difference, then it may be worth just
	  keeping it on all of the time, or perhaps just not in release
	  tarballs, so that this type of issue is avoided.

2007-09-07 19:48 +0000 [r81923]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
	  as the mixmonitor filename. This moves the setting of the
	  MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
	  by sim.

2007-09-07 15:25 +0000 [r81886]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Moving the explanation for joinempty
	  to a more appropriate place

2007-09-06 22:28 +0000 [r81832]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: (closes issue #9724, closes issue #10374)
	  Reported by: kenw Patches: 9724.txt uploaded by russell (license
	  2) Tested by: kenw, russell Resolve a deadlock that occurs when
	  doing a SIP transfer to parking. I come across this type of
	  deadlock fairly often it seems. It is very important to mind the
	  boundary between the channel driver and the core in respect to
	  the channel lock and the channel-pvt lock. Channel drivers lock
	  to lock the pvt and then the channel once it calls into the core,
	  while the core will do it in the opposite order. The way this is
	  avoided is by having channel drivers either release their pvt
	  lock while calling into the core, or such as in this case,
	  unlocking the pvt just long enough to acquire the channel lock.

2007-09-06 22:05 +0000 [r81778-81826]  Jason Parker <jparker@digium.com>

	* Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
	  for ASTLDFLAGS. This adds LDOPTS

	* include/asterisk/astobj2.h: This should fix a build issue that
	  people building against uClibc were seeing with the addition of
	  astobj2

2007-09-06 19:40 +0000 [r81776]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: (closes issue #10122) Reported by:
	  stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
	  qwell (license 4) Tested by: stevefeinstein After looking over
	  the code I agree with Qwell. Setting the file descriptor to
	  conference each time just causes a fight back and forth.

2007-09-06 16:56 +0000 [r81743]  Philippe Sultan <philippe.sultan@gmail.com>

	* include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
	  length fixes. Removed an unused variable in aji_client structure
	  (context)

2007-09-06 16:25 +0000 [r81682-81713]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixes an issue where valid DTMF had to be
	  pressed twice to exit a queue if a member's phone was ringing.
	  (closes issue #10655, reported by strider2k, patched by me)

	* res/res_features.c: Fixes a memory leak (closes issue #10658,
	  reported and patched by Ivan)

2007-09-06 14:20 +0000 [r81650]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
	  and Google's XMPP server complaint, if set, the 'from' attribute
	  must be set to the user's full JID.

2007-09-05 20:53 +0000 [r81599]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, main/say.c, res/res_features.c,
	  main/file.c, include/asterisk/channel.h: Fix an issue that can
	  occur when you do an attended transfer to parking. If you
	  complete the transfer before the announcement of the parking spot
	  finishes, then the channel being parked will hear the remainder
	  of the announcement. These changes make it so that will not
	  happen anymore. Basically, res_features sets a flag on the
	  channel is playing the announcement to so that the file streaming
	  core knows that it needs to watch out for a channel masquerade,
	  and if it occurs, to abort the announcement. (closes BE-182)

2007-09-05 17:18 +0000 [r81569]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Solaris x86 compatibility fix

2007-09-05 15:19 +0000 [r81525]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing the build...

2007-09-05 15:14 +0000 [r81523]  Jason Parker <jparker@digium.com>

	* channels/chan_phone.c: Do not try to unregister a NULL channel
	  tech. Also changed load_module function to use defines rather
	  than numbers for return values. Issue 10651, patch by
	  rbraun_proformatique, with additions by me.

2007-09-05 15:03 +0000 [r81520]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
	  only count members who are logged in and available. (related to
	  issue #10652, reported by wuwu)

2007-09-05 13:11 +0000 [r81492]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: (closes issue #10650) Reported by: tacvbo Only
	  print out that the spy was removed while holding the spy lock.

2007-09-04 20:54 +0000 [r81453-81455]  Jason Parker <jparker@digium.com>

	* apps/app_followme.c: Rather than attempt to play a file, we can
	  just check whether it exists. Issue 10634, patch by me, testing
	  by pabelanger, sanity checked by bweschke

	* configs/followme.conf.sample: Change default followme config file
	  to point to the correct files. Issue 10644, patch by pabelanger

2007-09-04 18:37 +0000 [r81448]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
	  Remove the typedefs on ao2_container and ao2_iterator. This is
	  simply because we don't typedef objects anywhere else in
	  Asterisk, so we might as well make this follow the same
	  convention.

2007-09-04 16:40 +0000 [r81442]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: there is no point in sending 401
	  Unauthorized to a UAS that sent us a properly-formatted
	  Authentication header with the expected username and nonce but an
	  incorrect response (which indicates the shared secret does not
	  match)... instead, let's send 403 Forbidden so that the UAS
	  doesn't retry with the same authentication credentials repeatedly

2007-09-04 14:23 +0000 [r81435-81439]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: (closes issue #10632) Reported by:
	  jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
	  by jamesgolovich (license 176) Fix memory leak when unloading
	  chan_iax2. The firmware files were not being freed.

	* main/channel.c: (closes issue #10476) Reported by: mdu113 Only
	  look for the end of a digit when waiting for a digit. This in
	  turn disables emulation in the core.

	* main/dns.c: (closes issue #10610) Reported by: john Patches:
	  dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
	  Don't return a match if no SRV record actually exists.

2007-09-03 18:57 +0000 [r81433]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove a couple of calls to
	  ast_string_field_free_pools() on peers in error handling blocks
	  in the code for building peers. The peer object destructor does
	  this and doing it twice will cause a crash. (closes issue #10625,
	  reported by and patched by pnlarsson)

2007-09-01 15:57 +0000 [r81426-81428]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changed a comment to be more accurate. (really
	  this is just a test to make sure I can commit properly from home)

	* main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
	  into ao2_match_by_addr and making it available everywhere since
	  it could be a handy callback to have

2007-08-31 21:27 +0000 [r81418]  Russell Bryant <russell@digium.com>

	* include/asterisk/astobj2.h: Remove references to a debugging
	  parameter that does not exist

2007-08-31 19:48 +0000 [r81416]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixed broken behavior of a reload on realtime
	  queues. Prior to this patch, if a reload was issued and a
	  realtime queue had callers waiting in it, then the queue would be
	  removed from the queue list, but it would not actually be freed
	  (in fact, a debug message warning about a memory leak would come
	  up). With this patch, reloads do not touch realtime queues at
	  all.

2007-08-31 19:16 +0000 [r81415]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_logic.c: The IF() function was not allowing true
	  values that had embedded colons (closes issue #10613)

2007-08-31 18:44 +0000 [r81412]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c: Re-order dial options to be in line with the
	  existing alpha order. Issue 10621, initial patch by junky

2007-08-31 17:38 +0000 [r81410]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
	  available. Closes issue 10548, reported by keepitcool.

2007-08-31 15:53 +0000 [r81406]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c: Make it the engine's responsible to check for
	  the presence of results.

2007-08-31 15:51 +0000 [r81405]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: add missing "transcoder show" (and deprecated
	  "show transcoder") CLI commands that were in 1.2 but never added
	  to 1.4

2007-08-31 14:38 +0000 [r81401-81403]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10618) Reported by: dimas
	  Don't pass through the stopped sounds frame.... just drop it.

	* res/res_features.c: (closes issue #10009) Reported by: dimas
	  Don't output a bridge failed warning message if it failed because
	  one of the channels was part of the masquerade process. That is
	  perfectly normal.

2007-08-30 22:05 +0000 [r81397]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing an extraneous (and possibly
	  misleading) log message. Firstly, if the announce file isn't
	  found, the streaming functions will report it. Secondly, not all
	  non-zero returns from play_file mean that the announce file
	  wasn't found. Positive return values simply mean that a digit was
	  pressed (most likely to skip through the announcement). (closes
	  issue #10612, reported and patched by dimas)

2007-08-30 21:23 +0000 [r81395]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10514) Reported by: casper
	  Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
	  Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
	  for it to ever be that value.

2007-08-30 21:11 +0000 [r81392]  Steve Murphy <murf@digium.com>

	* main/cdr.c: via issue 10599, where 'CDR already initialized'
	  messages are being generated. Since all channels will have an
	  init'd CDR attached at creation time, this message is now
	  particularly useless. Removed.

2007-08-30 15:38 +0000 [r81383]  Russell Bryant <russell@digium.com>

	* channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
	  define. (closes issue #10559, paravoid)

2007-08-30 15:35 +0000 [r81381]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changed some manager event messages to reflect
	  whether a queue member is a realtime member or not

2007-08-30 15:33 +0000 [r81379]  Russell Bryant <russell@digium.com>

	* configs/modem.conf.sample (removed), configs/enum.conf.sample,
	  configs/extensions.ael.sample: Fix a typo, update a reload
	  command, and remove an unused configuration file. (closes issue
	  #10606, casper)

2007-08-30 14:53 +0000 [r81375]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
	  pbx.diff uploaded by jmls (license 141) Backport changes from
	  81372. Add REASON dialplan variable for when an originated call
	  fails and the failed extension is executed.

2007-08-30 14:43 +0000 [r81373]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: Fixed some warnings.

2007-08-30 14:23 +0000 [r81369]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (issue #10599) Reported by: dimas Handle the
	  -1 control subclass during feature dialing (it indicates to stop
	  sounds).

2007-08-30 08:31 +0000 [r81367]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
	  issue where a misdn_read would lock the channel, but read would
	  not return because it blocks. later chan_misdn would try to queue
	  a frame like a AST_CONTROL_ANSWER which could result in a
	  deadlock situation. misdn_read will now not block forever
	  anymore, and we don't queue the ANSWER frame at all when we
	  already was called with misdn_answer -> answer would be called
	  twice. Also we don't explicitly send a RELEASE_COMPLETE on
	  receiption of a RELEASE anymore, because mISDN does that for us,
	  this resulted in a problem on some switches, which would block
	  our port after some calls for a short while.

2007-08-29 16:35 +0000 [r81346-81349]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This patch, in essence, will correctly pause a
	  realtime queue member and reflect those changes in the realtime
	  engine. (issue #10424, reported by irroot, patch by me) This
	  patch creates a new function called update_realtime_member_field,
	  which is a generic function which will allow any one field of a
	  realtime queue member to be updated. This patch only uses this
	  function to update the paused status of a queue member, but it
	  lays the foundation for persisting the state of a realtime member
	  the same way that static members' state is maintained when using
	  the persistentmembers setting

	* apps/app_queue.c: Changed some tabs to spaces

2007-08-29 15:57 +0000 [r81342]  Russell Bryant <russell@digium.com>

	* main/Makefile: If chan_h323 is not being built, don't use g++ to
	  do the final link of Asterisk. (in response to a question on the
	  asterisk-dev list)

2007-08-29 15:52 +0000 [r81340]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This fix creates a more accurate way of
	  detecting whether realtime members were deleted. (closes issue
	  10541, reported by Alric, patched by me) The REALLY nice things
	  about this patch is that queue members now have a "realtime"
	  field which will be true if the member is a realtime member. This
	  means we can check this value prior to certain processing if it
	  should ONLY be done for realtime members.

2007-08-29 14:13 +0000 [r81331]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
	  rtp timeouts work even if two RTP streams are directly bridged in
	  the RTP stack.

2007-08-28 21:38 +0000 [r81226-81291]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Change the message about receiving a
	  mini-frame before the first full voice frame to a DEBUG message.

	* pbx/pbx_dundi.c: revert unintentional changes in rev 81226

	* configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
	  tones. (closes issue #7953, hanabana)

2007-08-28 14:12 +0000 [r81120-81189]  Mark Michelson <mmichelson@digium.com>

	* contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
	  res_config_mysql (closes issue #10573, reported by chrisvaughan,
	  patch suggested by chrisvaughan as well)

	* apps/app_queue.c: Resolve a potential deadlock. In this case, a
	  single queue is locked, then the queue list. In changethread(),
	  the queue list is locked, and then each individual queue is
	  locked. Under the right circumstances, this could deadlock. As
	  such, I have unlocked the individual queue before locking the
	  queue list, and then locked the queue back after the queue list
	  is unlocked.

	* channels/chan_agent.c: DTMF begin frames should be ignored so
	  that when an agent acks a call with the '#' key, he doesn't cause
	  a queue's announce file to be interrupted. Also went ahead and
	  did the same for the '*' key and for ending a call. (closes issue
	  #10528, reported by deskhack, patched by me)

2007-08-27 17:27 +0000 [r81042-81074]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
	  of the memory it allocates on unload and should be fixed (when
	  I'm not in the middle of something else ...).

	* pbx/pbx_dundi.c: explicity define a variable as a boolean

	* res/res_musiconhold.c: (closes issue #10419) Reported by:
	  mustardman Patches: asterisk-mohposition.diff.txt uploaded by
	  jamesgolovich (license 176) This patch fixes a few problems with
	  music on hold. * Fix issues with starting at the beginning of a
	  file when it shouldn't. * Fix the inuse counter to be decremented
	  even if the class had not been set to be deleted when not in use
	  anymore * Don't arbitrarily limit the number of MOH files to 255

2007-08-27 15:01 +0000 [r81012]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
	  Patches: chan_sip-ChannelReload-20080825.patch uploaded by
	  jesselang (license 202) Remove an extra \r\n to make the
	  ChannelReload event conform with every other event.

2007-08-27 14:55 +0000 [r81010]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Found a case where the queue's membercount is
	  off. It does not take into account dynamic members on a reload.

2007-08-27 13:20 +0000 [r80974]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
	  jitter value output in the CLI to be as expected.

2007-08-26 18:11 +0000 [r80932]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove an extra signal_condition() for the
	  scheduler thread. (closes issue #10564, patch from casper)

2007-08-25 17:37 +0000 [r80895]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some issues with the handling of the
	  scheduler in chan_iax2. Most of the places that scheduled items
	  to be executed by the scheduler thread did not signal the
	  scheduler thread to wake up so that it could recalculate the time
	  until the next action. These changes will make the scheduler
	  thread more responsive and ensure that actions get executed as
	  close to when intended as possible instead of it being possible
	  for very long delays.

2007-08-24 22:59 +0000 [r80878]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_zapateller.c: An empty string is an empty callerid ...
	  so zap it. This closes issue #10502, which was pointed out by
	  dswartz. Thank you, and may the swartz be with you

2007-08-24 21:22 +0000 [r80820-80849]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
	  available when Asterisk first starts, then don't give up on
	  poking peers. Allow the poke to get rescheduled so that it will
	  work once the dnsmgr is able to resolve the host. (closes issue
	  #10521, patch by jamesgolovich)

	* main/dsp.c: Improve the debouncing logic in the DTMF detector to
	  fix some reliability issues. Previously, this code used a shift
	  register of hits and non-hits. However, if the start of the digit
	  isn't clean, it is possible for the leading edge detector to miss
	  the digit. These changes replace the flawed shift register logic
	  and also does the debouncing on the trailing edge as well.
	  (closes issue #10535, many thanks to softins for the patch)

2007-08-24 19:52 +0000 [r80818]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: A minor correction to the available logic of
	  autofill. If a queue member is paused, they're not really
	  "available" so don't count them as such. Somewhat related to
	  issue #10155

2007-08-24 18:52 +0000 [r80789]  Steve Murphy <murf@digium.com>

	* main/cdr.c: From a complaint by jmls, I realize that the message
	  in cdr_disposition is unnecessary. To get failure disposition,
	  just return -1; no use having more than one case do that.

2007-08-24 15:51 +0000 [r80750]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.

2007-08-24 15:41 +0000 [r80747]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
	  the code, instead of only in documentation (closes issue #10549)

2007-08-24 15:28 +0000 [r80722]  Russell Bryant <russell@digium.com>

	* utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
	  I think this would have caused a "*" to get in the
	  menuselect-tree file.

2007-08-24 14:48 +0000 [r80689-80717]  Steve Murphy <murf@digium.com>

	* utils/ael_main.c: This change addresses JerJer's complaint that
	  aelparse builds and installs even if pbx_ael is unchecked in the
	  menuselect stuff.

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
	  backport of 80649, a fix to an unreported problem in the ael
	  parser, that results in a crash on a 64bit machine

2007-08-24 11:42 +0000 [r80661]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
	  answered too early, which results in CDRs wrongly stating that a
	  call was ANSWERED when the calling party cancelled a call before
	  before being established. We must not answer the call upon
	  reception of a 'transport-accept' iq packet, but this packet
	  still needs to be acknowledged, otherwise the remote peer would
	  close the call (like in #8970).

2007-08-23 21:34 +0000 [r80601-80617]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
	  warnings

	* channels/chan_misdn.c: make chan_misdn compile without warnings

2007-08-23 20:16 +0000 [r80539-80573]  Russell Bryant <russell@digium.com>

	* include/asterisk/features.h, res/res_features.c: When executing a
	  dynamic feature, don't look it up a second time by digit pattern
	  after we already looked it up by name. This causes broken
	  behavior if there is more than one feature defined with the same
	  digit pattern. (closes issue #10539, reported by bungalow, patch
	  by me)

	* funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
	  none of these values take ms so I don't know what I was thinking

	* funcs/func_timeout.c: Fix func_timeout to take values in floating
	  point so 1.5 actually means 1.5 seconds instead of being rounded.
	  (closes issue #10540, reported by spendergrass, patch by me)

2007-08-23 17:14 +0000 [r80505-80507]  Jason Parker <jparker@digium.com>

	* /: *sigh*

	* /: use autotagged externals

2007-08-23 17:08 +0000 [r80501]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: report the actual channel number that was
	  unregistered, instead of assuming that the interface list
	  consists of channels 1 through <x> with no gaps in the sequence

2007-08-23 17:02 +0000 [r80360-80499]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some code where it was possible for a
	  reference to a peer to not get released when it should. Thank you
	  to Marta Carbone for pointing this out!

	* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
	  This is a hack to maintain old behavior of chan_iax2. This
	  ensures that if the peers and users are being stored in a linked
	  list, that they go in the list in the same order that the older
	  code used. This is necessary to maintain the behavior of which
	  peers and users get matched when traversing the container.

	* res/res_agi.c: Revert res_agi fix that didn't quite work until we
	  get it right ...

	* include/asterisk/astobj2.h: Add some more documentation on
	  iterating ao2 containers. The documentation implies that is
	  possible to miss an object or see an object twice while
	  iterating. After looking through the code and talking with
	  mmichelson, I have documented the exact conditions under which
	  this can happen (which are rare and harmless in most cases).

	* main/astobj2.c: When converting this code to use the list macros,
	  I changed it so objects are added to the head of a bucket instead
	  of the tail. However, while looking over code with mmichelson, we
	  noticed that the algorithm used in ao2_iterator_next requires
	  that items are added to the tail. This wouldn't have caused any
	  huge problem, but it wasn't correct. It meant that if an object
	  was added to a container while you were iterating it, and it was
	  added to the same bucket that the current element is in, then the
	  new object would be returned by ao2_iterator_next, and any other
	  objects in the bucket would be bypassed in the traversal.

	* channels/chan_sip.c: Don't crash when using realtime in chan_sip
	  without an insecure setting in the database. (closes issue
	  #10348, reported by link55, fixed by me)

	* main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
	  (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
	  channels/chan_iax2.c: Merge changes from
	  team/russell/iax_refcount. This set of changes fixes problems
	  with the handling of iax2_user and iax2_peer objects. It was very
	  possible for a thread to still hold a reference to one of these
	  objects while a reload operation tries to delete them. The fix
	  here is to ensure that all references to these objects are
	  tracked so that they can't go away while still in use. To
	  accomplish this, I used the astobj2 reference counted object
	  model. This code has been in one of Luigi Rizzo's branches for a
	  long time and was primarily developed by one of his students,
	  Marta Carbone. I wanted to go ahead and bring this in to 1.4
	  because there are other problems similar to the ones fixed by
	  these changes, so we might as well go ahead and use the new
	  astobj if we're going to go through all of the work necessary to
	  fix the problems. As a nice side benefit of these changes, peer
	  and user handling got more efficient. Using astobj2 lets us not
	  hold the container lock for peers or users nearly as long while
	  iterating. Also, by changing a define at the top of chan_iax2.c,
	  the objects will be distributed in a hash table, drastically
	  increasing lookup speed in these containers, which will have a
	  very big impact on systems that have a large number of users or
	  peers. The use of the hash table will be made the default in
	  trunk. It is not the default in 1.4 because it changes the
	  behavior slightly. Previously, since peers and users were stored
	  in memory in the same order they were specified in the
	  configuration file, you could influence peer and user matching
	  order based on the order they are specified in the configuration.
	  The hash table does not guarantee any order in the container, so
	  this behavior will be going away. It just means that you have to
	  be a little more careful ensuring that peers and users are
	  matched explicitly and not forcing chan_iax2 to have to guess
	  which user is the right one based on secret, host, and access
	  list settings, instead of simply using the username. If you have
	  any questions, feel free to ask on the asterisk-dev list.

	* res/res_agi.c: Juggie in #asterisk-dev was reporting problems
	  where fgets would return without reading the whole line when
	  using fastagi. When this happens, errno was set to EINTR or
	  EAGAIN. This patch accounts for the possibility and lets fgets
	  continue in that case.

2007-08-22 18:53 +0000 [r80302-80330]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
	  build_tools/strip_nonapi, build_tools/prep_moduledeps,
	  build_tools/make_buildopts_h: Fix a few build issues in Solaris
	  (and likely others). Use GREP and ID variables from autoconf.
	  Reported to me in #asterisk-dev I forgot who reported this -
	  sorry. :(

	* Makefile: Change a syntax that the GNU make in Solaris dislikes.

	* build_tools/make_version: Fix a bashism (we explicitly request
	  /bin/sh). Remove some oddly placed quotes I found in passing.

2007-08-22 16:21 +0000 [r80257]  Russell Bryant <russell@digium.com>

	* Makefile: Honor the contents of the COPTS variable as custom
	  target CFLAGS. Apparently this is what openwrt does. (reported by
	  Brian Capouch on the asterisk-dev list, patch by me)

2007-08-22 16:14 +0000 [r80255]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10526) Reported by: sinistermidget
	  Revert commit from issue #10355 and return timestamp skew to 640.

2007-08-21  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.11 released.

2007-08-21 18:42 +0000 [r80183]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Don't record SIP dialog history if it's not
	  turned on. Also, put an upper limit on how many history entires
	  will be stored for each SIP dialog. It is currently set to 50,
	  but can be increased if deemed necessary. (closes issue #10421,
	  closes issue #10418, patches suggested by jmoldenhauer, patches
	  updated by me) (Security implications documented in AST-2007-020)

2007-08-21 16:39 +0000 [r80166-80167]  Steve Murphy <murf@digium.com>

	* include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
	  the diffs from ulaw.h and alaw.h for now; accidentally added them
	  in 80166

	* main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
	  This patch solves problem 1 in 8126; it should not slow down the
	  alaw codec, but should prevent signal degradation via multiple
	  trips thru the codec. Fossil estimates the twice thru this codec
	  will prevent fax from working. 4-6 times thru would result
	  hearable, noticeable, voice degradation.

2007-08-21 15:22 +0000 [r80132]  Russell Bryant <russell@digium.com>

	* channels/chan_mgcp.c: Don't try to dereference the owner channel
	  when it may not exist (issue #10507, maxper)

2007-08-21 15:03 +0000 [r80130]  Jason Parker <jparker@digium.com>

	* configs/cdr.conf.sample: (issue #10510) Reported by: casper
	  Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
	  errors in sample cdr config file.

2007-08-20 21:57 +0000 [r80088]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix the build of app_queue

2007-08-20 21:39 +0000 [r80049-80086]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: After a discussion on #asterisk-dev, it was
	  decided that this should be in 1.4 as well. (issue #10424,
	  reported and patched by irroot)

	* apps/app_queue.c: Found a pointless ternary if. member->dynamic
	  was set to 1 and has no opportunity to change between then and
	  this line, so "dynamic" will ALWAYS be output.

2007-08-20 16:08 +0000 [r80047]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample: (issue #10499) Reported by:
	  casper Patches: extensions.conf.sample.diff uploaded by casper
	  (license 55) Update CLI examples in extensions.conf.sample to
	  reflect command changes.

2007-08-20 15:34 +0000 [r80044]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Ukrainian language voicemail support.
	  (closes issue #10458, reported and patched by Oleh)

2007-08-20 02:42 +0000 [r79998]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
	  snuffy via IRC)

2007-08-18 14:30 +0000 [r79947]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Don't allocate vmu for messagecount when we
	  could just use the stack instead (closes issue #10490) Also,
	  remove a useless (and leaky) SQLAllocHandle (closes issue #10480)

2007-08-17 21:01 +0000 [r79912]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Avoid a crash in the handling of DTMF based
	  Caller ID. It is valid for ast_read to return NULL in the case
	  that the channel has been hung up. (crash reported by
	  anonymouz666 on IRC in #asterisk-dev)

2007-08-17 19:14 +0000 [r79906]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Patch allows for more seamless transition
	  from file storage voicemail to ODBC storage voicemail. If a
	  retrieval of a greeting from the database fails, but the file is
	  found on the file system, then we go ahead an insert the greeting
	  into the database. The result of this is that people who switch
	  from file storage to ODBC storage do not need to rerecord their
	  voicemail greetings.

2007-08-17 19:12 +0000 [r79902-79904]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
	  Don't send a semicolon over the wire in sip notify messages.
	  Caused by fix for issue 9938. I basically took the code that
	  existed before 9938 was fixed, and copied it into a new function
	  - ast_unescape_semicolon There should be very few places this
	  will be needed (pbx_config does NOT need this (see issue 9938 for
	  details)) Issue 10430, patch by me, with help/ideas from murf
	  (thanks murf).

	* channels/chan_local.c: Re-add the setting of callerid name and
	  number. Issue 10485, reported by and fix explained by paradise.

2007-08-17 13:37 +0000 [r79857]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix some crashes in chan_sip. This patch
	  changes various places that add items to the scheduler to ensure
	  that they don't overwrite the ID of a previously scheduled item.
	  If there is one, it should be removed. (closes issue #10391,
	  closes issue #10256, probably others, patch by me)

2007-08-17 08:22 +0000 [r79833]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: sometimes we don't need to signal dtmf
	  tones to asterisk, we just want them to go through as inband.
	  Otherwise they might be generated by the other channel partner
	  and then there is a double tone.

2007-08-16 22:32 +0000 [r79756-79792]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix a little race condition that could
	  cause a crash if two channels had MOH stopped at the same time
	  that were using a class that had been marked for deletion when
	  its use count hits zero.

	* res/res_musiconhold.c: This patch fixes a bug where reloading the
	  module with "module reload" did not delete classes from memory
	  that were no longer in the config. This patch fixes that problem
	  as well as another one. Previously, if you reloaded MOH using the
	  "moh reload" CLI command, which behaved differently than "module
	  reload ...", MOH had to be stopped on every channel and started
	  again immediately. However, there was no way to tell what class
	  was being used, so they would all fall back to the default class.
	  (closes issue #10139) Reported by: blitzrage Patches:
	  asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
	  (license 176) Tested by: jamesgolovich

	* channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
	  introduced by making frame handling and scheduling
	  multi-threaded. Unfortunately, we have to do some expensive
	  deadlock avoidance when queueing frames on to the ast_channel
	  owner of the IAX2 pvt struct. This was already handled for
	  regular frames, but ast_queue_hangup and ast_queue_control were
	  still used directly. Making these changes introduced even more
	  places where the IAX2 pvt struct can disappear in the context of
	  a function holding its lock due to calling a function that has to
	  unlock/lock it to avoid deadlocks. I went through and fixed all
	  of these places to account for this possibility. (issue #10362,
	  patch by me)

2007-08-16 21:16 +0000 [r79690-79748]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fixes a problem where agents would get
	  stuck busy due to their wrapuptime being longer than the queue's
	  wrapuptime and ringinuse=no for the queue. (closes issue #10215,
	  reported by Doug, repaired by me) Special thanks to fkasumovic
	  for pointing out the source of the problem and to bweschke for
	  helping to come up with a solution!

	* apps/app_voicemail.c: base_encode is not trying to open a log
	  file, so we should not call it a log file in the warning.
	  (related to issue #10452, reported by bcnit)

2007-08-16 09:37 +0000 [r79665]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: A fix for two critical problems detected while
	  working with Daniel McKeehan in issue #10184. Upon priority
	  change, the resource list is not NULL terminated when moving an
	  item to the end of the list. This makes Asterisk endlessy loop
	  whenever it needs to read the list. Jids with different resource
	  and priority values, like in Gmail's and GoogleTalk's jabber
	  clients put that problem in evidence. Upon reception of a 'from'
	  attribute with an empty resource string, Asterisk crashes when
	  trying to access the found->cap pointer if the resource list for
	  the given buddy is not empty. This situation is perfectly valid
	  and must be handled. The Gizmoproject's jabber client put that
	  problem in evidence. Also added a few comments in the code as
	  well as a handle for the capabilities from Gmail's jabber client,
	  which are stored in a caps:c tag rather than the usual c tag.
	  Closes issue #10184.

2007-08-16 08:21 +0000 [r79642]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
	  we want to send IA5 Chars.

2007-08-15 14:40 +0000 [r79553]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10440) Reported by: irroot (closes
	  issue #10454) Reported by: flo_turc Increase maximum timestamp
	  skew to 120. 20 was apparently far too low.

2007-08-15 14:26 +0000 [r79527]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed an error in the Russian language
	  voicemail intro. (issue #10458, reported and patched by Oleh)

2007-08-15 14:18 +0000 [r79523]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10456) Reported by: irroot
	  Patches: sip_timeout.patch uploaded by irroot (license 52) Change
	  hardcoded timer value to defined value. I'm doing this in 1.4 as
	  well so if it needs to be changed in the future this place would
	  not have been forgotten.

2007-08-14 18:49 +0000 [r79436-79470]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another spot where an iax2_peer would
	  be leaked if realtime was in use.

	* channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
	  related to the use of realtime. I found these while working on
	  iax2_peer object reference tracking.

2007-08-14 15:27 +0000 [r79397]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10415) Reported by: atis
	  Revert fix for #10327 as it causes more issues then it solves.

2007-08-13 22:40 +0000 [r79363]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: memset really, really needs to be used here.

2007-08-13 21:57 +0000 [r79334]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Instead of accepting a single DTMF
	  character accept a full string.

2007-08-13 20:37 +0000 [r79272-79301]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Don't call find_peer in
	  registry_authrequest with the pvt lock held to avoid a deadlock.

	* channels/chan_iax2.c: Release the pvt lock before calling
	  find_peer in register_verify to avoid a deadlock. Also, remove
	  some unnecessary locking in auth_fail that was only done
	  recursively.

	* channels/chan_iax2.c: Don't call find_peer within update_registry
	  with a pvt lock held. This can cause a deadlock as the code will
	  eventually call find_callno.

	* channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
	  have tracked them down to a single core issue. You can not call
	  find_callno() while holding a pvt lock as this function has to
	  lock another (every) other pvt lock. Doing so can lead to a
	  classic deadlock. So, I am tracking down all of the code paths
	  where this can happen and fixing them. The fix I committed
	  earlier today was along the same theme. This patch fixes some
	  code down the path of authenticate_reply.

2007-08-13 17:49 +0000 [r79255]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
	  pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
	  fixes bug 10411. I added a new regression test, some regression
	  test cleanups

2007-08-13 15:28 +0000 [r79214]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a potential deadlock in socket_process.
	  check_provisioning can eventually call find_callno. You can't
	  hold a pvt lock while calling find_callno because it goes through
	  and locks every single one looking for a match.

2007-08-13 14:51 +0000 [r79174-79207]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Add an API call to allow the engine to
	  know that DTMF was received.

	* channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
	  channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
	  Don't set the callerid name and number a second time on a newly
	  created channel. ast_channel_alloc itself already sets it and
	  setting it twice would cause a memory leak.

2007-08-11 05:23 +0000 [r79142]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Ensure the connection gets marked as used at
	  allocation time (closes issue #10429, report and fix by
	  mnicholson)

2007-08-10 20:53 +0000 [r79044-79099]  Steve Murphy <murf@digium.com>

	* main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
	  a user complaint on #asterisk, I have forced pbx_spool to explain
	  what reason codes mean, when they are logged

	* main/cdr.c: Re bug behavior mentioned in #asterisk, made this
	  tweak to code, to prevent hundreds of log messages from being
	  generated

	* main/cdr.c: This will help debug; from a question asked on
	  #asterisk

2007-08-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.10.1 released.
	
2007-08-10 15:20 +0000 [r78995]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: The last set of changes that I made to
	  "core show locks" made it not able to track mutexes unless they
	  were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
	  with ast_mutex_init() were not tracked. It should work now.

2007-08-10 14:15 +0000 [r78951-78955]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Don't bother having the core pass through or emulate
	  begin DTMF frames when in an ast_waitstream. It only cares about
	  the end of DTMF.

	* configs/queues.conf.sample: (closes issue #10422) Reported by:
	  bhowell Add note to sample configuration about module load order
	  and how it can cause perfectly good queue members to be marked as
	  invalid.

2007-08-10 13:24 +0000 [r78936]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
	  information element. We send them now also in the disconnect
	  message.

2007-08-09 23:47 +0000 [r78907]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Improved a bit of logic regarding
	  comma-separated mailboxes in has_voicemail. Also added some
	  braces to some compound if statements since unbraced if
	  statements scare me in general.

2007-08-09 23:10 +0000 [r78891]  Steve Murphy <murf@digium.com>

	* Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
	  output

2007-08-09 22:03 +0000 [r78826-78860]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing some extra debug code I left in my
	  last commit

	* apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
	  1. instead of using inboxcount as the core message counting
	  function, we use messagecount instead. This makes it possible to
	  count messages in folders besides just INBOX and Old. 2.
	  inboxcount and hasvoicemail now use messagecount as their means
	  of determining return values. 3. Added a copy_message function
	  for IMAP storage. Unfortunately I don't have the means to test
	  it, but it seems like a pretty straightforward function. 4.
	  Removed a #ifndef IMAP_STORAGE and matching #endif from
	  leave_voicemail for a couple of reasons. One, we want to support
	  copying mail to multiple IMAP boxes, and two, IMAP was broken
	  because a STORE macro had been moved into this section of code.

	* channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
	  put some new code in the wrong place and so I've reverted the
	  canreinvite section to how it was and put my new code where it
	  should be.

2007-08-09 17:58 +0000 [r78717-78778]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: add a comment to indicate that inboxcount
	  for ODBC_STORAGE needs to be fixed to support multiple mailboxes

	* apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
	  ODBC_STORAGE. Also, leave a comment for this to be fixed for
	  IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
	  working on this code right now for another reason. This is broken
	  even worse in trunk, but for a different reason. The fact that
	  the mailbox option supported multiple mailboxes is completely not
	  obvious from the code in the channel drivers. Anyway, I will fix
	  that in another commit ...

	* apps/app_meetme.c: Fix a problem with the combination of the 'F'
	  option to pass DTMF through a conference and options that use
	  DTMF to activate various features. The problem was that the BEGIN
	  frame would be passed through, but the END frame would get
	  intercepted to activate a feature. Then, the other conference
	  members would hear DTMF for forever, which they didn't seem to
	  like very much. (closes issue #10400, reported by stevefeinstein,
	  fixed by me)

2007-08-08 19:29 +0000 [r78646]  Jason Parker <jparker@digium.com>

	* doc/jabber.txt: Fix mogs email address.

2007-08-08 18:16 +0000 [r78575-78620]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed some compiler warnings so that
	  compiling with dev-mode and IMAP storage would not have any
	  errors. This section of code may get changed again shortly since
	  my change uncovers a rather silly bit of logic.

	* apps/app_queue.c: Changing a bit of logic so that someone will
	  NEVER exit the queue on timeout unless they have enabled the 'n'
	  option. This commit relates to issue #10320. Thanks to
	  jfitzgibbon for detailing the idea behind this code change.

2007-08-08 13:51 +0000 [r78569]  Joshua Colp <jcolp@digium.com>

	* configs/sip.conf.sample: (closes issue #10335) Reported by:
	  adamgundy Update sip.conf to include another scenario where
	  directrtpsetup will fail.

2007-08-07  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.10 released.
	
2007-08-07 20:57 +0000 [r78488]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c: Fix the build of this module on 64-bit
	  platforms

2007-08-07 19:43 +0000 [r78450]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: The logic behind inboxcount's return value
	  was reversed in has_voicemail and message_count. (closes issue
	  #10401, reported by st1710, patched by me)

2007-08-07 19:34 +0000 [r78437]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Don't free the environment handle when the
	  connection fails, because other connections might be depending
	  upon it

2007-08-07 19:11 +0000 [r78416]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Allow chan_sip to build in devmode

2007-08-07 19:09 +0000 [r78415]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, res/res_config_odbc.c,
	  apps/app_directory.c: Reconnection doesn't happen automatically
	  when a DB goes down (fixes issue #9389)

2007-08-07 18:25 +0000 [r78375]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Properly check the capabilities count to
	  avoid a segfault. (ASA-2007-019)

2007-08-07 17:45 +0000 [r78371]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
	  4 lines Revert patch committed for issue #9660. It broke E&M
	  trunks. (closes issue #10360) (closes issue #10364) ........

2007-08-06 21:41 +0000 [r78275]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Add additional DTMF log messages to help when
	  debugging issues.

2007-08-06 20:44 +0000 [r78184-78242]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix an issue where dynamic threads can get
	  free'd, but still exist in the dynamic thread list. (closes issue
	  #10392, patch from Mihai, with credit to his colleague, Pete)

	* include/asterisk/linkedlists.h: Fix the return value of
	  AST_LIST_REMOVE(). This shouldn't be causing any problems,
	  though, because the only code that uses the return value only
	  checks to see if it is NULL. (closes issue #10390, pointed out by
	  mihai)

2007-08-06 16:32 +0000 [r78182]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is possible for a transfer to occur
	  before the remote device has our tag in which case they send none
	  in the transfer. In this case we need to not fail the transfer
	  dialog lookup.

2007-08-06 16:30 +0000 [r78180]  Jason Parker <jparker@digium.com>

	* main/config.c: Fix an issue with using UpdateConfig (manager
	  action) where escaped semicolons in a config would be converted
	  to just semicolons (\; to ;) Issue 9938

2007-08-06 15:27 +0000 [r78166-78172]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
	  we pass through RTP timestamp information we need to make the
	  allowed timestamp skew considerably less. There are situations
	  where a source may change and due to the timestamp difference the
	  receiver will experience an audio gap since we did not indicate
	  by setting the marker bit that the source changed.

	* configure, configure.ac: (closes issue #10383) Reported by: rizzo
	  Include stdlib.h so NULL gets defined for gethostbyname_r checks.

2007-08-06 13:33 +0000 [r78164]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fixed a mistake I made in realtime_peer
	  which caused it to return NULL every time. Thanks to Jon Fealy
	  for emailing me the correction.

2007-08-05 14:18 +0000 [r78146]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
	  bug #10382)

2007-08-05 04:15 +0000 [r78143]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Fix compilation failure when
	  MALLOC_DEBUG is enabled, but DEBUG_THREADS is not

2007-08-05 03:29 +0000 [r78139]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: If peer is not found, the error message is
	  misleading (should be peer not found, not ACL failure)

2007-08-03 20:25 +0000 [r78103]  Mark Michelson <mmichelson@digium.com>

	* main/config.c, channels/chan_sip.c, include/asterisk/config.h:
	  Changed the behavior of sip's realtime_peer function to match the
	  corresponding way of matching for non-realtime peers. Now matches
	  are made on both the IP address and port number, or if the
	  insecure setting is set to "port" then just match on the IP
	  address. In order to accomplish this, I also added a new API
	  call, ast_category_root, which returns the first variable of an
	  ast_category struct

2007-08-03 20:14 +0000 [r78028-78101]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: (closes issue #10194) Reported by:
	  blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
	  blitzrage Fix a problem when you call Voicemail() with multiple
	  mailboxes specified and ODBC_STORAGE is in use. The audio part of
	  the message was only given to the first mailbox specified.

	* main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
	  improvements to lock debugging. These changes take effect with
	  DEBUG_THREADS enabled and provide the following: * This will keep
	  track of which locks are held by which thread as well as which
	  lock a thread is waiting for in a thread-local data structure. A
	  reference to this structure is available on the stack in the
	  dummy_start() function, which is the common entry point for all
	  threads. This information can be easily retrieved using gdb if
	  you switch to the dummy_start() stack frame of any thread and
	  print the contents of the lock_info variable. * All of the
	  thread-local structures for keeping track of this lock
	  information are also stored in a list so that the information can
	  be dumped to the CLI using the "core show locks" CLI command.
	  This introduces a little bit of a performance hit as it requires
	  additional underlying locking operations inside of every
	  lock/unlock on an ast_mutex. However, the benefits of having this
	  information available at the CLI is huge, especially considering
	  this is only done in DEBUG_THREADS mode. It means that in most
	  cases where we debug deadlocks, we no longer have to request
	  access to the machine to analyze the contents of ast_mutex_t
	  structures. We can now just ask them to get the output of "core
	  show locks", which gives us all of the information we needed in
	  most cases. I also had to make some additional changes to astmm.c
	  to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
	  enabled. I disabled tracking of one of the locks in astmm.c
	  because it gets used inside the replacement memory allocation
	  routines, and the lock tracking code allocates memory. This
	  caused infinite recursion.

	* channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
	  frames when the mohinterpret option is set to "passthrough". This
	  was pointed out by Kevin in the middle of a training session.

	* channels/chan_iax2.c: Don't reuse the timespec that was set to 0
	  in the previous timedwait as it will just return immediately.
	  Also, fix some logic so the thread's lock isn't unlocked twice in
	  the weird case of dynamic threads getting acquired right after a
	  timeout. (pointed out by SteveK)

2007-08-02 21:53 +0000 [r77993-77996]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
	  actually allow 6 chars to be sent. Also make note of the "A"
	  option of date format. Issue 9779, modifications by DEA, wedhorn,
	  and myself.

	* channels/chan_skinny.c: If a device disconnects, the session will
	  go away. If this happens during call setup, we need to give up.
	  Issue 10325.

2007-08-02 19:25 +0000 [r77949]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix the case where a dynamic thread times
	  out waiting for something to do during the first time it runs.
	  This shouldn't ever happen, but we should account for it anyway.
	  (pointed out by pete, who works with mihai)

2007-08-02 18:42 +0000 [r77947]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure we clear the prompt status
	  message on a hangup. Also rearrange messages to better fit with
	  what a wireshark trace shows it should be. Issue 10299, initial
	  patch and solution by sbisker, modified by me to fit with
	  wireshark trace.

2007-08-02 18:21 +0000 [r77945]  Steve Murphy <murf@digium.com>

	* main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
	  line This patch hopefully solves 10141; The user is running with
	  it, and it doesn't appear to harm asterisk's operation, and may
	  prevent a crash. I'll store it in 1.2, as we have shut down
	  support on 1.2, but since I developed the patch before support
	  finished, and it might affect 1.4 and trunk, I'm going ahead with
	  it. ........

2007-08-02 18:04 +0000 [r77939-77943]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another race condition in the handling
	  of dynamic threads. If the dynamic thread timed out waiting for
	  something to do, but was acquired to perform an action
	  immediately afterwords, then wait on the condition again to give
	  the other thread a chance to finish setting up the data for what
	  action this thread should perform. Otherwise, if it immediately
	  continues, it will perform the wrong action. (reported on IRC by
	  mihai, patch by me) (related to issue #10289)

	* channels/chan_iax2.c: Add another sanity check to
	  vnak_retransmit(). This check ensures that frames that have
	  already been marked for deletion don't get retransmitted. (closes
	  issue #10361, patch from mihai)

2007-08-02 15:15 +0000 [r77890-77894]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure that we show the correct
	  extension if dialed from a macro "From: 5555" rather than "From:
	  s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
	  tested by sbisker

	* channels/chan_skinny.c: Put in some additional debug information
	  for softkey/stimulus messages. Issue 10291, patch by DEA.

2007-08-01 22:16 +0000 [r77887]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some race conditions which have been
	  causing weird problems in chan_iax2. The most notable problem is
	  that people have been seeing storms of VNAK frames being sent due
	  to really old frames mysteriously being in the retransmission
	  queue and never getting removed. It was possible that a dynamic
	  thread got created, but did not acquire its lock before the
	  thread that created it signals it to perform an action. When this
	  happens, the thread will sleep until it hits a timeout, and then
	  get destroyed. So, the action never gets performed and in some
	  cases, means a frame doesn't get transmitted and never gets freed
	  since the scheduler never gets a chance to reschedule
	  transmission. Another less severe race condition is in the
	  handling of a timeout for a dynamic thread. It was possible for
	  it to be acquired to perform at action at the same time that it
	  hit a timeout. When this occurs, whatever action it was acquired
	  for would never get performed. (patch contributed by Mihai and
	  SteveK) (closes issue #10289) (closes issue #10248) (closes issue
	  #10232) (possibly related to issue #10359)

2007-08-01 22:14 +0000 [r77886]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
	  not compile cleanly (missing def)

2007-08-01 21:08 +0000 [r77883]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an issue that caused one-way audio on
	  some newer devices (specifically the 7921), due to sending
	  packets in the wrong order during hangup. Also make sure we clear
	  tones/messages on the correct line/instance. Issue 10291, patch
	  by DEA, tested by sbisker and myself.

2007-08-01 18:08 +0000 [r77863-77871]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: (closes issue #10351) Reported by: ftarz Some
	  platforms don't like it when you pass NULL to vsnprintf so pass
	  "" instead.

	* include/asterisk/threadstorage.h, channels/chan_mgcp.c,
	  apps/app_voicemail.c, main/acl.c, utils/smsq.c,
	  channels/chan_iax2.c: Add some fixes for building on Solaris.

	* main/utils.c: Whoops, I meant R_5 not R5.

	* configure, configure.ac: And for my last trick... make sure that
	  if gethostbyname_r is exported by a library that it is used.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: Extend autoconf logic to determine which version of
	  gethostbyname_r is on the system.

2007-08-01 14:08 +0000 [r77852-77854]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixes an issue I introduced to queues wherein a
	  queue with joinempty=yes would kick people out of the queue
	  because of erroneously thinking the 'n' option was in use.
	  (closes issue #10320, reported by jfitzgibbon, patched by me,
	  tested by blitzrage and me) Thank you blitzrage for all the
	  testing you've done lately with queues! It's much appreciated!

	* apps/app_queue.c: If a queue uses dynamic realtime members, then
	  the member list should be updated after each attempt to call the
	  queue. This fixes an issue where if a caller calls into a queue
	  where no one is logged in, they would wait forever even if a
	  member logged in at some point. (closes issue #10346, reported by
	  and tested by blitzrage, patched by me)

2007-07-31 21:09 +0000 [r77845-77846]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c: Much newer version, 0.70 with much additions

	* main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
	  receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
	  logic in TONE_VERIFY/RELAX mode in chan_zap.

2007-07-31 20:59 +0000 [r77844]  Steve Murphy <murf@digium.com>

	* /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
	  line This probably isn't super-general, but it's a first stab at
	  using kill -11 to generate a core file instead of gcore. ........

2007-07-31 16:17 +0000 [r77831]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, include/asterisk/speech.h: Add a flag to the
	  speech API that allows an engine to set whether it received
	  results or not.

2007-07-31 15:53 +0000 [r77827]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
	  DEBUG_THREADS or it does nothing

2007-07-31 15:21 +0000 [r77824]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This patch makes Asterisk send 100 Trying
	  provisional responses upon receipt of re-invites. This makes it
	  so that if there are two or more Asterisk servers between
	  endpoints, the Asterisk servers will not keep retransmitting the
	  re-invites. (closes issue #10274, reported by cstadlmann, patched
	  by me with approval from file)

2007-07-30 20:17 +0000 [r77795]  Jason Parker <jparker@digium.com>

	* main/say.c: Applications like SayAlpha() should not hang up the
	  channel if you request an "unknown" character such as a comma.
	  Instead, skip the character and move on. Issue 10083, initial
	  patch by jsmith, modified by me.

2007-07-30 20:16 +0000 [r77785-77794]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix an issue that could potentially cause
	  corruption of the global iax frame queue. In the network_thread()
	  loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
	  macro. However, to remove an element of the list within this
	  loop, it used AST_LIST_REMOVE, instead of
	  AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
	  internal variables of the SAFE macro invalid. Mihai says that he
	  already made this change in his local copy and it didn't help his
	  VNAK storm issues, but I still think it's wrong. :)

	* res/res_agi.c: (closes issue #10279) Reported by: seanbright
	  Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
	  seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
	  uploaded by seanbright (license 71) Allow the "agi_network: yes"
	  line to be printed out in the AGI debug output. Also, allow
	  partial writes to be handled when writing out this line just like
	  it is for all of the others.

	* main/channel.c: file and I both committed changes for issue
	  #10301. Remove a duplicated assignment to restore the original
	  value of the previous channel.

2007-07-30 18:43 +0000 [r77783]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, res/res_agi.c: Merged revisions 77782 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
	  | 2 lines Revert change in revision 71656, even though it fixed a
	  bug, because many people were depending upon the (broken)
	  behavior. ........

2007-07-30 17:29 +0000 [r77780]  Russell Bryant <russell@digium.com>

	* main/channel.c: (closes issue #10301) Reported by: fnordian
	  Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
	  (license 110) Additional changes by me Fix some problems in
	  channel_find_locked() which can cause an infinite loop. The
	  reference to the previous channel is set to NULL in some cases.
	  These changes ensure that the reference to the previous channel
	  gets restored before needing it again. I'm not convinced that the
	  code that is setting it to NULL is really the right thing to do.
	  However, I am making these changes to fix the obvious problem and
	  just leaving an XXX comment that it needs a better explanation
	  that what is there now.

2007-07-30 17:11 +0000 [r77768-77778]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10327) Reported by: kkiely
	  Instead of directly mucking with the extension/context/priority
	  of the channel we are transferring when it has a PBX simply call
	  ast_async_goto on it. This will ensure that the channel gets
	  handled properly and sent to the right place.

	* main/channel.c: (closes issue #10301) Reported by: fnordian
	  Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
	  (license 110) Restore previous behavior where if we failed to
	  lock the channel we wanted we would return to exactly the same
	  point as if we had just reentered the function.

	* /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
	  lines (closes issue #10334) Reported by: ramonpeek Pass through
	  the return value from macro_exec through the MacroIf application.
	  ........

2007-07-27 18:15 +0000 [r77571]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Missing newline

2007-07-27 17:04 +0000 [r77536-77540]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
	  Patches: cdr_pgsql.patch uploaded by prashant (license 114)
	  Finish the Postgresql connection after the log messages are
	  printed so we don't access invalid memory.

	* channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
	  Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
	  julianjm (license 99) Clear ONHOLD flag when decrementing the
	  onHold peer count. If we did not do this the count may keep
	  decreasing.

2007-07-27 14:30 +0000 [r77490]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: "re-invite" was misspelled

2007-07-26 23:19 +0000 [r77460]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: (closes issue #10302) Reported by: litnialex If a
	  DTMF end frame comes from a channel without a begin and it is
	  going to a technology that only accepts end frames (aka INFO)
	  then use the minimum DTMF duration if one is not in the frame
	  already.

2007-07-26 22:16 +0000 [r77424-77429]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/mp3.txt: change protocol for downloads as well

	* doc/mp3.txt, sounds/Makefile: use new canonical name for download
	  server

2007-07-26 21:23 +0000 [r77410]  Russell Bryant <russell@digium.com>

	* Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
	  in trunk, but not in 1.4. When Asterisk is compiled under dev
	  mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
	  define it in the same way that trunk does. Also, revert the
	  change that added this define in the Makefile The advantage to
	  doing it this way is that buildopts.h gets installed when you
	  install Asterisk. Then, when building any out of tree modules, or
	  building asterisk-addons, these modules know which options the
	  rest of Asterisk was built with.

2007-07-26 20:35 +0000 [r77380]  Mark Michelson <mmichelson@digium.com>

	* Makefile, main/logger.c: Fixes to get ast_backtrace working
	  properly. The AST_DEVMODE macro was never defined so the majority
	  of ast_backtrace never attempted compilation. The makefile now
	  defines AST_DEVMODE if configure was run with --enable-dev-mode.
	  Also, changes were made to acccomodate 64 bit systems in
	  ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
	  their roles in allowing me to get this committed

2007-07-26 19:32 +0000 [r77348-77350]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/logger.c: Missed one

	* main/logger.c: Oops, that builtin define should be all-lowercase.

2007-07-26 18:30 +0000 [r77318]  Mark Michelson <mmichelson@digium.com>

	* cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
	  meaning free gets called on the same variable twice. This patch
	  sets the connection to NULL after calls to PQfinish so that the
	  problem does not occur. Also in this patch, prashant_jois
	  informed me that it is safe to pass a null pointer to PQfinish,
	  so I have removed the check for conn's existence from
	  my_unload_module. (closes issue 10295, reported by junky, patched
	  by me with input from prashant_jois)

2007-07-25 22:39 +0000 [r77191]  Steve Murphy <murf@digium.com>

	* apps/app_meetme.c: This fix solves problem with intense squelch
	  noise when someone joins conf in bug 9430; We repro'd the problem
	  with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
	  I'm applying it. It looks like playing the recorded username will
	  louse up the next thing played into the channel. Josh rearranged
	  the code so as to start things over before playing data directly
	  into the conference.

2007-07-25 22:16 +0000 [r77176]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
	  Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
	  digit to terminate a DTMF string with. If none is specified then
	  no terminator will be used.

2007-07-25 21:52 +0000 [r77154]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
	  not a signed int, so use %u instead of %d in the format string

2007-07-25 20:23 +0000 [r77116-77136]  Jason Parker <jparker@digium.com>

	* /: so are my fingers...

	* /: autotagexternals script is still obviously misbehaving...

	* /: use autotagged externals

2007-07-25 17:14 +0000 [r77071]  Joshua Colp <jcolp@digium.com>

	* configure, acinclude.m4: Fix autoconf logic for finding OpenH323
	  when it is not in the first place searched (/usr/share/openh323).

2007-07-25 09:34 +0000 [r77022]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: set the sequence number in a frame for all frame
	  types

2007-07-25 00:18 +0000 [r76983]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
	  line this fixes bug 10293, where the error message because
	  defaultzone or loadzone was not defined was confusing ........

2007-07-24 22:12 +0000 [r76891-76937]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
	  Jul 2007) | 2 lines Oops, res contains the error code, not errno.
	  I was wondering why a mutex was reporting "No such file or
	  directory"... ........

	* main/app.c: Found another place where we should be using the
	  umask (thanks jcmoore)

2007-07-24  Jason Parker <jparker@digium.com>

	* Asterisk 1.4.9 released.

2007-07-24 16:42 +0000 [r76803-76805]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Don't create the Asterisk channel until we
	  are starting the PBX on it. (ASA-2007-018)

2007-07-24 16:26 +0000 [r76801]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added a membercount variable to call_queue
	  struct which keeps track of the number of logged in members in a
	  particular queue. This makes it so that the 'n' option for
	  Queue() can act properly depending on which strategy is used. If
	  the strategy is roundrobin, rrmemory, or ringall, we want to ring
	  each phone once before moving on in the dialplan. However, if any
	  other strategy is used, we will only ring one phone since it
	  cannot be guaranteed that a different phone will ring on
	  subsequent attempts to ring a phone. As a side effect of this,
	  the QUEUE_MEMBER_COUNT dialplan function now just reads the
	  membercount variable instead of traversing through the member
	  list to figure out how many members there are. Special thanks to
	  blitzrage for helping to test this out. (closes issue #10127,
	  reported by bcnit, patched by me, tested by blitzrage)

2007-07-23 22:38 +0000 [r76708]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: It was our stated intention for 1.4 that
	  files created in app_voicemail should depend upon the umask.
	  Unfortunately, mkstemp() creates files with mode 0600, regardless
	  of the umask. This corrects that deficiency.

2007-07-23 18:59 +0000 [r76656]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix some incorrect softkey labels in
	  messages. Don't try to play dialtone in some unimplemented
	  features.

2007-07-23 18:29 +0000 [r76654]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
	  2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
	  force channel format changes when a generator is present. The
	  generator may have changed the formats itself and changing them
	  back would cause issues. ........

2007-07-23 17:57 +0000 [r76620]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Don't try to queue up hold/unhold frames
	  on a non-existent channel. Issue 10276.

2007-07-23 17:48 +0000 [r76519-76618]  Joshua Colp <jcolp@digium.com>

	* apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
	  by putting the value of the digit char in an int.

	* /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
	  lines (closes issue #10236) Reported by: homesick Patches:
	  rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
	  Remote Party ID on guest calls. ........

	* channels/chan_skinny.c: (closes issue #10268) Reported by:
	  mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
	  (license 7) Add another OS that has to use the Macros for byte
	  ordering.

2007-07-23 12:25 +0000 [r76485]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use a signed integer for storing the number
	  of bytes in the packet read from the network. Using an unsigned
	  value here made it impossible to handle an error returned from
	  recvfrom(). Furthermore, in the case that recvfrom() did return
	  an error, this would cause a crash due to a heap overflow.
	  (closes issue #10265, reported by and fix suggested by
	  timrobbins)

2007-07-21 02:02 +0000 [r76227]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
	  4 lines Backport a fix for a memory leak that was fixed in trunk
	  in reivision 76221 by rizzo. The memory used for the localaddr
	  list was not freed during a configuration reload. ........

2007-07-20 21:36 +0000 [r76211]  Steve Murphy <murf@digium.com>

	* sounds/Makefile: This patch from 10249 is worth applying! It
	  prevents downloading sound files if they are already downloaded.
	  Darn Practical, if you ask me

2007-07-20 21:03 +0000 [r76174-76178]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Allow getting a call from an existing
	  "sub" channel. Cancel ringing if endpoint hangs up before
	  answering. Fixes were backported from trunk (there was apparently
	  a bit of confusion during merge of a previous patch). (closes
	  issue #10241)

	* main/manager.c: Eliminate a compiler warning with gcc 4.2 by
	  constifying a char *

	* channels/chan_skinny.c: It's possible for sub->owner to be NULL
	  here if you cancel the call immediately after/during sending a
	  digit.

2007-07-20 18:42 +0000 [r76139]  Mark Michelson <mmichelson@digium.com>

	* apps/app_directory.c: When using users.conf for the entries in
	  the directory, if multiple users had the same last name, only the
	  first user listed would be available in the directory. (closes
	  issue #10200, reported by mrskippy, patched by me)

2007-07-20 18:22 +0000 [r76132]  Russell Bryant <russell@digium.com>

	* main/channel.c: Use the define that specifies the default length
	  of an artificially created DTMF digit in the ast_senddigit()
	  function. The define is set to 100ms by default, which is the
	  same thing that this function was using. But, using the define
	  lets changes take effect in this case, as well as the others
	  where it was already used.

2007-07-20 17:20 +0000 [r76054-76087]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
	  lines (closes issue #10247) Reported by: fkasumovic Patches:
	  chan_sip.patch uploaded by fkasumovic (license #101) Drop any
	  peer realm authentication entries when reloading so multiple
	  entries do not get added to the peer. ........

	* res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
	  Patches: res_conver.patch uploaded by fkasumovic (license #101)
	  Use the last occurance of . to find the extension, not the first
	  occurance.

	* apps/app_queue.c: Move makeannouncement variable declaration to
	  proper place.

2007-07-19 20:36 +0000 [r75980]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Remove some duplicate code.

2007-07-19 18:59 +0000 [r75969-75978]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: The diff on this looks pretty big but all I did
	  was remove a pointless if statement (always evaluates true).

	* apps/app_queue.c: Changes in handling return values of several
	  functions in app_queue. This all started as a fix for issue
	  #10008 but now includes all of the following changes: 1.
	  Simplifying the code to handle positive return values from ast
	  API calls. 2. Removing the background_file function. 3. The fix
	  for issue #10008 (closes issue #10008, reported and patched by
	  dimas)

2007-07-19 15:53 +0000 [r75928]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
	  6 lines When processing full frames, take sequence number
	  wraparound into account when deciding whether or not we need to
	  request retransmissions by sending a VNAK. This code could cause
	  VNAKs to be sent erroneously in some cases, and to not be sent in
	  other cases when it should have been. (closes issue #10237,
	  reported and patched by mihai) ........

2007-07-18 22:59 +0000 [r75807]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Need to make sure we set milliseconds and
	  timestamp - pointed out by the recent ast_ time stuff from
	  Tilghman

2007-07-18 21:09 +0000 [r75759]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
	  5 lines When traversing the queue of frames for possible
	  retransmission after receiving a VNAK, handle sequence number
	  wraparound so that all frames that should be retransmitted
	  actually do get retransmitted. (issue #10227, reported and
	  patched by mihai) ........

2007-07-18 20:40 +0000 [r75749]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
	  | 2 lines Store prior to copy (closes issue #10193) ........

2007-07-18 20:17 +0000 [r75732]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Umm, why are we transmitting dialtone on
	  cfwdall?

2007-07-18 20:00 +0000 [r75712]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
	  pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
	  Asterisk won't build under devmode using GCC 4.2.

2007-07-18 19:54 +0000 [r75707-75711]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
	  Issue 9245, patch by slimey.

	* channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
	  9887, patches by DEA

2007-07-18 17:56 +0000 [r75658]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
	  | 1 line removed the word 'pissed' from ast_log(...) function
	  call for BE-90 ........

2007-07-18 15:44 +0000 [r75583-75623]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Few more places that needs to check for
	  onhold state.

	* channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
	  It is possible for hold status to exist without call limits set,
	  so we need to ensure update_call_counter is executed regardless.

	* channels/chan_h323.c: Don't bother reloading chan_h323 if it did
	  not load successfully in the first place. This would otherwise
	  cause a crash.

	* pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
	  the threadid of each running thread before shutting them down as
	  the thread themselves may change the value.

2007-07-18 12:29 +0000 [r75529]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_meetme.c: Using a freed frame causes crashes (closes
	  issue #9317)

2007-07-17  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.8 released.

2007-07-17 20:57 +0000 [r75441-75450]  Russell Bryant <russell@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
	  Jul 2007) | 3 lines Properly check for the length in the skinny
	  packet to prevent an invalid memcpy. (ASA-2007-016) ........

	* main/rtp.c: cast arguments to ast_log so that it builds without
	  warnings for me

	* channels/iax2-parser.c, channels/iax2-parser.h, /,
	  channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
	  5 lines Ensure that when encoding the contents of an ast_frame
	  into an iax_frame, that the size of the destination buffer is
	  known in the iax_frame so that code won't write past the end of
	  the allocated buffer when sending outgoing frames. (ASA-2007-014)
	  ........

	* /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
	  4 lines After parsing information elements in IAX frames, set the
	  data length to zero, so that code later on does not think it has
	  data to copy. (ASA-2007-015) ........

2007-07-17 20:40 +0000 [r75439]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Ensure that the pointer to STUN data does not go to
	  unaccessible memory. (ASA-2007-017)

2007-07-17 20:33 +0000 [r75437]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: (issue #10210) Reported by: juggie Patches:
	  10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
	  juggie, blitzrage Log a warning if someone uses DeadAGI on a live
	  channel.

2007-07-17 20:03 +0000 [r75405]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
	  can return -1 on failure, so I need to be sure that we only enter
	  the if statement if it is successful. Related to my fix to issue
	  #10186

2007-07-17 20:01 +0000 [r75401-75403]  Russell Bryant <russell@digium.com>

	* main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
	  10209-trunk-2.patch uploaded by juggie Tested by: juggie,
	  blitzrage In ast_pbx_run(), mark a channel as hung up after an
	  application returned -1, or when it runs out of extensions to
	  execute. This is so that code can detect that this channel has
	  been hung up for things like making sure DeadAGI is used on
	  actual dead channels, and is beneficial for other things, like
	  making sure someone doesn't try to start spying on a channel that
	  is about to go away.

	* res/res_agi.c: Remove a duplicated newline character in AGI debug
	  output. (closes issue #10207, patch by seanbright)

2007-07-16 20:53 +0000 [r75258-75306]  Kevin P. Fleming <kpfleming@digium.com>

	* main/dns.c, /: Merged revisions 75304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
	  | 3 lines provide proper copyright/license attribution for this
	  structure that was copied from a BSD-licensed header file long,
	  long ago... ........

	* /: another fix that is not needed here (finishing up 75251)

2007-07-16 18:16 +0000 [r75253]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Restoring functionality from 1.2 wherein
	  Retrydial will not exit if there is no announce file specified.
	  This change makes it so that if there is no announce file
	  specified, the application will continue until finished (or
	  caller hangs up). If a bogus announce file is specified, then a
	  warning message will be printed saying that the file could not be
	  found, but execution will still continue. (closes issue #10186,
	  reported by jon, patched by me)

2007-07-16 18:12 +0000 [r75252]  Kevin P. Fleming <kpfleming@digium.com>

	* /: block change that is not relevant here

2007-07-13 20:36 +0000 [r75108]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
	  Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
	  load_moh_classes() could return without destroying the loaded
	  configuration. ........

2007-07-13 20:15 +0000 [r75078]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
	  2007) | 5 lines Fixed an issue where chanspy flags were
	  uninitialized if no options were passed. What triggered this
	  investigation was an IRC chat where some people's quiet flags
	  were set while others' weren't even though none of them had
	  specified the q option. ........

2007-07-13 20:10 +0000 [r75053-75067]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
	  Jul 2007) | 6 lines Ensure that adding a user to the list of
	  users of a specific music on hold class is not done at the same
	  time as any of the other operations on this list to prevent list
	  corruption. Using the global moh_data lock for this is not ideal,
	  but it is what is used to protect these lists everywhere else in
	  the module, and I am only changing what is necessary to fix the
	  bug. ........

	* channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
	  12 lines (closes issue #9660) Reported by: mmacvicar Patches
	  submitted by: bbryant, russell Tested by: mmacvicar, marco,
	  arcivanov, jmhunter, explidous When using a TDM400P (and probably
	  other analog cards) there was a chance that you could hang up and
	  pick the phone back up where it has been long enough to be not
	  considered a flash hook, but too soon such that the device
	  reports that it is busy and the person on the phone will only
	  hear silence. This patch makes chan_zap more tolerant of this and
	  gives the device a couple of seconds to succeed so the person on
	  the phone happily gets their dialtone. ........

2007-07-12 23:00 +0000 [r74998]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Change to my previous fix regarding agent
	  logoff soft. Now uses deferlogoff instead of loginstart since
	  loginstart is used after logoff. Thanks to makoto for pointing
	  this out and suggesting the fix. (closes issue #10178, reported
	  and patched by makoto, with modification by me)

2007-07-12 20:42 +0000 [r74955]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This patch resolves 10143; thanks to irroot
	  for the patch; looked acceptable. Let the community decide if it
	  messes things up

2007-07-12 19:17 +0000 [r74888-74922]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Whoops... didn't want this to be returned to 0
	  each iteration.

	* main/channel.c: When waiting for a digit ensure that a begin
	  frame was received with it, not just an end frame. (issue #10084
	  reported by rushowr)

2007-07-12 16:53 +0000 [r74839-74866]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: It helps if I actually add this stuff for
	  the 7921 too - otherwise it won't actually do much of anything.

	* channels/chan_skinny.c: Add device ID for 7921 wireless skinny
	  phone

	* channels/chan_skinny.c: Fix dialing in skinny that was broken in
	  some cases. Issue 10136, fix provided by DEA.

2007-07-12 15:53 +0000 [r74815]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
	  2007) | 2 lines Only print out a warning for situations where it
	  is actually helpful. (issue #10187 reported by denke) ........

2007-07-11 22:57 +0000 [r74767]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
	  5 lines The function make_trunk() can fail and return -1 instead
	  of a valid new call number. Fix the uses of this function to
	  handle this instead of treating it as the new call number. This
	  would cause a deadlock and memory corruption. (possible cause of
	  issue #9614 and others, patch by me) ........

2007-07-11 21:14 +0000 [r74722]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
	  Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
	  did not work...at all. Now it does. (closes issue #10178,
	  reported and patched by makoto, with slight modification for 1.4
	  and trunk by me) ........

2007-07-11 18:34 +0000 [r74657]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
	  Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
	  the condition that uses LIKE. This fixes realtime extensions with
	  ODBC. (closes issue #10175, reported by stuarth, patch by me)
	  ........

2007-07-11 18:18 +0000 [r74628-74642]  Steve Murphy <murf@digium.com>

	* Makefile: This fixes 10172, where the entire man8 dir gets
	  removed during an uninstall of asterisk

	* utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
	  further reversion of previously applied floating point stuff for
	  expr2

2007-07-11 17:16 +0000 [r74515-74590]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Instead of
	  figuring out kernel versions that have compiler.h and not...
	  let's just use autoconf to check for it's presence. (issue #10174
	  reported by francesco_r)

	* channels/chan_phone.c: Only check if we need to do a SIGMA based
	  tone generation if we have a card. (issue #10179 reported by
	  mikowhy)

2007-07-10 23:32 +0000 [r74476]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Forwarding a message with IMAP storage was
	  storing the message in the sender's box instead of the forwarded
	  mailbox. (closes issue #10138, reported and patched by jaroth)

2007-07-10 19:58 +0000 [r74374-74428]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
	  lines Fix an issue where it was possible to have a service level
	  of over 100% Between the time recalc_holdtime and update_queue
	  was called, it was possible that the call could have been hungup.
	  Move both additions to the same place, so this won't happen.
	  Issue 10158, initial patch by makoto, modified by me. ........

	* main/dns.c: Don't use #if to check if something is defined - use
	  #ifdef instead. Pointed out by kpfleming

	* /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
	  2007) | 4 lines Fix an issue with wrapuptime not working when
	  using AgentLogin. Issue 10169, patch by makoto, with a minor mod
	  by me to not re-break issue 9618 ........

	* main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 74373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
	  lines Use res_ndestroy on systems that have it. Otherwise, use
	  res_nclose. This prevents a memleak on NetBSD - and possibly
	  others. Issue 10133, patch by me, reported and tested by scw
	  ........

2007-07-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.7.1 released.

2007-07-10 16:00 +0000 [r74323]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: fix an uninitialized variable

2007-07-10 15:38 +0000 [r74317]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
	  lines Fix a small typo in description in of Voicemail()
	  application. Issue 10170, patch by casper. ........

2007-07-10 15:31 +0000 [r74314]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
	  Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
	  #10075, this part reported by jmls on IRC, patch by me) ........

2007-07-10 14:50 +0000 [r74262-74265]  Joshua Colp <jcolp@digium.com>

	* /, main/app.c: Merged revisions 74264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
	  lines Ensure the group information category exists before trying
	  to do a string comparison with it. (issue #10171 reported by
	  mlegas) ........

	* channels/chan_sip.c: Only spit out an inringing warning message
	  when it is applicable. Since call limits are already toast in
	  realtime let's not scare the user if they are using it. (issue
	  #10166 reported by bcnit)

2007-07-09  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.7 released.

2007-07-09 21:31 +0000 [r74162-74211]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Update the configure script to check for
	  a required function that is not present in the 1.2 version of
	  libpri. This will prevent the configure script from thinking that
	  it has compatible libpri support for Asterisk 1.4, when it
	  actually does not because the installed version is from 1.2.

	* res/res_musiconhold.c: (closes issue #10123) Reported by:
	  blitzrage Patches submitted by: juggie, qwell, me Tested by:
	  blitzrage When trying to find a music on hold class to use, try
	  all of the options, instead of only the first one that is set.
	  Also, change the MusicOnHold applications to not hang up on the
	  channel when a class can not be found.

2007-07-09 20:19 +0000 [r74159]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
	  lines Several chan_zap options were not working on reload because
	  they were arbitrarily disallowed when reloading some/most PRI
	  options (such as signalling) was disallowed. Options such as
	  polarityonanswerdelay and answeronpolarityswitch can safely be
	  changed on a reload. This corrects that behavior. Issue 9186,
	  patch by tzafrir. ........

2007-07-09 18:38 +0000 [r74120-74122]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Forgot to get rid of an extraneous debug
	  message.

	* apps/app_queue.c: The n option for Queue should make the queue
	  exit immediately after failure to reach any members and should
	  not be dependent on the timeout value passed to Queue (closes
	  issue #10127, reported by bcnit, repaired by me)

2007-07-09 15:32 +0000 [r74082]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c: Only destroy the scheduler context if it
	  was allocated. (issue #10124 reported by gzero)

2007-07-09 14:57 +0000 [r74047]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
	  Pass the mailbox instead of the context to inbox_count when the
	  context is "default." (closes issue #10135, reported by yannj,
	  repaired by me)

2007-07-09 14:49 +0000 [r74043-74045]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
	  synchronization tweaks. (issue #10124 reported by gzero)

	* configure, acinclude.m4: Use AC_CHECK_HEADER to check for
	  ptlib/openh323 to allow for cross compiling. (issue #9675
	  reported by zandbelt)

2007-07-09 04:03 +0000 [r73985]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
	  'make progdocs'. (Closes issue #10104)

2007-07-09 03:13 +0000 [r73930-73980]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c: Give Agent channel names priority when doing CDR
	  merging. (issue #10011 reported by krtorio)

	* pbx/pbx_config.c: Add a few sanity checks when writing out the
	  dialplan. (issue #10157 reported by dome)

2007-07-08 09:47 +0000 [r73849]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: While tracking down a bug, I need some more
	  history. Dumphistory is very useful, indeed.

2007-07-06 23:02 +0000 [r73769]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
	  4 lines If a sip_pvt struct has already registered an extension
	  state callback, remove the old one before adding a new one. If
	  this isn't done, Asterisk will crash. (issue #10120) ........

2007-07-06 16:36 +0000 [r73727]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing a rare case which causes voicemail
	  to crash when compiled with IMAP storage. inboxcount has the
	  possibility of finding an "interactive" vm_state when no
	  persistent "non-interactive" vm_state exists for that mailbox. If
	  this should happen when someone attempts to leave a message, it
	  results in a crash. This patch, along with my commit in revision
	  72670 fix issue 10053, reported by jaroth. closes issue #10053

2007-07-06 16:12 +0000 [r73679-73696]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
	  Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
	  Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
	  with MSSQL 2005 by explicitly stating that '\' is being used as
	  an escape character. ........

	* /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
	  7 lines (closes issue #10125) Reported by: makoto Patches
	  submitted by: makoto This fixes a crash in chan_sip that happens
	  when the bindaddr setting is not valid on Asterisk startup, gets
	  fixed, and then a reload gets issued. ........

2007-07-06 15:27 +0000 [r73675]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
	  Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
	  (issue 9618, reported by jiddings, patched by moi) closes issue
	  #9618 ........

2007-07-06 03:34 +0000 [r73551-73629]  Russell Bryant <russell@digium.com>

	* BUGS: fix a little spelling error

	* channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
	  the monitor thread if it was never started. (closes issue #10124,
	  reported by gzero, fixed by me)

	* channels/chan_iax2.c: copy from the correct buffer when deferring
	  a full frame (related to issue #9937)

	* channels/chan_iax2.c: * Store the call number that a thread is
	  processing without the full frame bit set to ease debugging *
	  When deferring a full frame for processing, stick it into the
	  queue for the thread that is processing frames for that call, not
	  the one that read the current frame and is about to go back into
	  the idle list (related to issue #9937)

2007-07-05 22:20 +0000 [r73548]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
	  | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
	  like we don't support it for G.729 ........

2007-07-05 20:50 +0000 [r73512]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Pass HOLD and UNHOLD frames to the other
	  channel when they are returned from a native bridge function.
	  This fixes a problem where when two zap channels are natively
	  bridged and one does a flash hook, the other channel did not
	  receive music on hold. (Reported to me directly by Doug Bailey at
	  Digium)

2007-07-05 19:18 +0000 [r73467]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
	  lines Copy language information to the dialog structure when
	  calling a peer for situations where a PBX may be started on the
	  dialed channel. (issue #10121 reported by clegall_proformatique)
	  ........

2007-07-05 15:59 +0000 [r73400]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Correcting a minor CLI bug I found. When
	  issuing the queue show command, if you type queue show and then
	  press tab, you can continue pressing tab and it will keep
	  auto-completing queue names even though only 1 queue can be used
	  as an argument.

2007-07-05 15:28 +0000 [r73398]  Russell Bryant <russell@digium.com>

	* channels/chan_vpb.cc, channels/Makefile: Make this module build
	  for me in dev-mode

2007-07-05 14:21 +0000 [r73316-73355]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
	  lines Tweak spy locking. (issue #9951 reported by welles)
	  ........

	* channels/chan_local.c, /: Merged revisions 73318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
	  2007) | 2 lines Actually check to make sure a PBX was started on
	  one of the Local channels instead of blindly assuming it was.
	  (issue #10112 reported by makoto) ........

	* /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
	  lines Reset ServicelevelPerf variable back to 0 if we are unable
	  to calculate it each time... otherwise we will get previous
	  values. (issue #10117 reported by noriyuki) ........

2007-07-04 14:53 +0000 [r73208-73253]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
	  Jul 2007) | 1 line bchannel configurations like echocancel and
	  volume control, need to be setuped on inbound calls too. ........

	* channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
	  Jul 2007) | 1 line bad bug in overlapdial case, we called
	  start_pbx multiple times, because the state wasn't changed..
	  ........

2007-07-03 20:17 +0000 [r73143]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
	  main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
	  expr floating patch from 1.4; too much of a behavior change. If
	  you want this fix, try trunk instead. bug 9508.

2007-07-03 15:42 +0000 [r73104-73106]  Jason Parker <jparker@digium.com>

	* /: What the heck. This should not have happened.

	* /: use autotagged externals

2007-07-03 12:38 +0000 [r73053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
	  | 2 lines RetryDial should accept a 0 argument, but it does not,
	  because atoi does not distinguish between 0 and error (closes
	  issue #10106) ........

2007-07-03 08:17 +0000 [r73005]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
	  Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
	  be called from mISDN Source channels.. #9449 ........

2007-07-02 20:16 +0000 [r72933]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
	  main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
	  main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
	  for floating point numbers added to ast_expr2 $\[...\] exprs.
	  Fixes bug 9508, where the expr code fails with fp numbers. The
	  MATH function returns fp numbers by default, so this fix is
	  considered necessary.

2007-07-02 18:18 +0000 [r72926]  Russell Bryant <russell@digium.com>

	* main/manager.c: Remove a bogus comment and add proper locking to
	  the handler function for the CLI command to show information on
	  manager actions.

2007-07-02 14:32 +0000 [r72888]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Added additional DTMF debug messages for when
	  emulation occurs.

2007-07-02 08:41 +0000 [r72850-72852]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 72585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
	  1 line check if the bchannel stack id is already used, if so
	  don't use it a second time. Also added a release_chan lock, so
	  that the same chan_list object cannot be freed twice. chan_misdn
	  does not crash anymore on heavy load with these changes. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 72099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
	  1 line simplified generation for dummy bchannels, also we mark
	  them as dummies, so they are not used later as real-bchannels,
	  optimized the RESTART mechanisms, we block a channel now on
	  cause:44, and send out a RESTART automatically, then on reception
	  of RESTART_ACKNOWLEDGE we unblock the channel again. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
	  revisions 72087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
	  1 line simplified channel finding and locking a lot. removed
	  unnecessary #ifdefed areas. ........

2007-07-01 23:52 +0000 [r72806]  Russell Bryant <russell@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
	  5 lines When appending lines to call files to keep track of
	  retries, write a leading newline just in case the original call
	  file did not have a newline at the end. This fix is in response
	  to a problem I saw reported on the asterisk-users mailing list.
	  ........

2007-06-30 16:50 +0000 [r72705-72766]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Tweak the configure script so that error
	  output isn't spewed to the console when searching for GTK2 libs,
	  and they aren't found.

	* formats/format_pcm.c: give format_pcm a more concise destription

2007-06-29 19:07 +0000 [r72665]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
	  absence of the function. This was already done in trunk.

2007-06-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.6 released.

2007-06-29 14:26 +0000 [r72597-72599]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c: Minor change for older GCC versions.

	* Makefile, configure, configure.ac, makeopts.in: Backport fix for
	  GCC versions without support for declaration-after-statement.

2007-06-29 04:47 +0000 [r72554-72556]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/manager.c: Issue 10055 - Change memory allocation to use the
	  heap for a command, since the output has the potential to
	  overflow the stack (as it did here)

	* res/res_jabber.c: Fix 1.4 breakage

2007-06-28 19:44 +0000 [r72493]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in: regenerate the
	  configure script for rizzo

2007-06-28 19:29 +0000 [r72453-72489]  Luigi Rizzo <rizzo@icir.org>

	* configure.ac: add a check for gethostbyname_r so we can simplify
	  the handling e.g. in utils.c Also add comments on a couple of
	  features which are not working on FreeBSD. All the above has been
	  already done in trunk so the merge must be blocked. Can someone
	  please regenerate ./configure ?

	* Makefile, channels/chan_zap.c, main/say.c: Add
	  -Wdeclaration-after-statement to AST_DEVMODE flags to catch
	  variable declarations in the middle of a block. Fix the few
	  instances of the above spotted out by the compiler. All of this
	  has been already done or is not applicable in trunk, so the merge
	  of this change will be blocked.

	* apps/app_meetme.c: cast a time_t so that it does not conflict
	  with the print format. This change was already done on trunk so
	  this change needs to be blocked from merging.

2007-06-27 23:29 +0000 [r72383]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
	  3 lines Reinstating patch. This actually fixes the problem,
	  however I was running a development branch without it and
	  mistakenly thought it wasn't fixed. Fixes issue #10010, and
	  #9654: 100% CPU usage caused by an asterisk console losing it's
	  controlling terminal. ........

2007-06-27 23:25 +0000 [r72381]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
	  2007) | 2 lines Update documentation to clarify variable usage
	  with MixMonitor. (issue #9494 reported by netoguy) ........

2007-06-27 23:03 +0000 [r72335]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72333 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
	  2 lines Reverted changes for earlier revisions 72259 to 72261.
	  Issue #9654, #10010 ........

2007-06-27 22:58 +0000 [r72328-72331]  Joshua Colp <jcolp@digium.com>

	* channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
	  our list. Since these are dynamic payloads the other side
	  shouldn't care. (issue #9426 reported by irroot)

	* /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
	  lines Fix issue where queue log events might be missing. (issue
	  #7765 reported by mtryfoss) ........

2007-06-27 21:08 +0000 [r72272]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
	  5 lines Fix a minor issue with parsing the priority number. You
	  could have as much whitespace as you want around a numeric
	  priority, but you couldn't have any whitespace around a special
	  priority like "n" or "hint". (issue #10039, reported by mitheloc,
	  fixed by me) ........

2007-06-27 20:46 +0000 [r72260]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72259 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
	  4 lines Fixes 100% load when controlling terminal disappears.
	  Issue #9654, #10010 ........

2007-06-27 20:25 +0000 [r72257]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 72256 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
	  lines I may possibly get shot for doing this... but... defer CDR
	  processing until after the channel has been dealt with. This
	  should eliminate all of the issues with channels going funky
	  (SIP/PRI) when you are posting CDRs to a database that is either
	  slow or unavailable and do not want to enable batching. ........

2007-06-27 19:13 +0000 [r72205]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: use the proper type for storing group number
	  bits so that if someone specifies 'group=42' it will actually
	  work instead of being silently ignored

2007-06-27 18:40 +0000 [r72182-72185]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix another problem in voicemail with
	  missing symbols. Issue 10074, patch by kryptolus, extended to
	  include #if 0'd blocks (just in case)

2007-06-27 17:31 +0000 [r72148]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Make the ast_read_noaudio API call behave better
	  under circumstances where DTMF emulation was happening and a
	  generator was setup. (issue #10065 reported by stevefeinstein)

2007-06-27 17:10 +0000 [r72125]  Jason Parker <jparker@digium.com>

	* channels/chan_gtalk.c: Don't modify a variable that we don't want
	  modified. Make a copy of it instead. Issue 10029, patch by
	  phsultan with slight modifications by me (to remove needless
	  casts).

2007-06-27 16:34 +0000 [r72112]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Only output debug information related to RTCP
	  timestamps when RTCP debug is turned on (issue #10066, patch by
	  me)

2007-06-27 07:58 +0000 [r72042]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
	  1 line for inbound TE calls, we setup the bchannel when we get
	  the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
	  removed some #if 0 areas which weren't used anymore. ........
	  r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
	  1 line isdn_lib.c didn't compile ........

2007-06-27 00:58 +0000 [r72006]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.

2007-06-26 23:02 +0000 [r71953]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing a pointless line. This variable
	  was already set earlier and between then and this line, there is
	  no way that the values on the right side of the assignment could
	  have changed.

2007-06-26 20:36 +0000 [r71915]  Jason Parker <jparker@digium.com>

	* main/rtp.c: Don't dereference a pointer that may be NULL here.
	  Issue 10017.

2007-06-26 19:00 +0000 [r71877]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: A few changes, the ultimate goal of which
	  is to keep better track of the number of messages that a mailbox
	  currently has. A description of the changes: 1. Changed the
	  "updated" field of the vm_state struct to act more as a binary
	  semaphore than a counting semaphore, since its current
	  implementation made the inboxcount function not work properly.
	  This change falls in line with a change made by UPenn with their
	  IMAP setup and helps to sync our changes with theirs. 2.
	  Eliminated some redundant calls to get_vm_state_by_mailbox inside
	  leave_voicemail 3. Use the play_folder variable to keep track of
	  the number of old and new messages in a mailbox as the messages
	  are deleted 4. Added an increment to the number of new messages
	  that was not there previously in the leave_voicemail function

2007-06-26 15:47 +0000 [r71796]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing bug where the authuser was
	  mistakenly pulled from the mailbox string instead of the IMAP
	  user. (closes issue 10054, reported and patched by jaroth)

2007-06-26 12:27 +0000 [r71657-71751]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
	  | 2 lines Issue 10062 - Trying to move a message without
	  selecting one first results in memory corruption ........

	* /, res/res_agi.c: Merged revisions 71656 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
	  | 2 lines Issue 10035 - handle_exec returns a result inconsistent
	  with all of the other AGI commands ........

2007-06-25 14:13 +0000 [r71522-71576]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Build a peer as well when hash323 is
	  enabled in users.conf (issue #9599 reported by asagage)

	* channels/chan_agent.c: Minor tweak for queueing up the unhold
	  frame... this will teach me to do bugs while half asleep. (issue
	  #10046 reported by dimas)

2007-06-25 12:40 +0000 [r71519]  Russell Bryant <russell@digium.com>

	* doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
	  #10048, Matti)

2007-06-25 01:10 +0000 [r71412-71430]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
	  lines Ignore other URIs after the first in a 300 Multiple Choice
	  response. (issue #10041 reported by homesick) ........

	* main/cdr.c: Fix it so 1.4 actually compiles on my box.

	* channels/chan_agent.c: Check to make sure the channel pointer is
	  present before queueing up an unhold frame on it. (issue #10046
	  reported by dimas)

2007-06-24 20:16 +0000 [r71362-71371]  Russell Bryant <russell@digium.com>

	* build_tools/prep_tarball: Include the menuselect-tree file in
	  tarballs to make builds from tarballs a little bit faster

	* main/asterisk.c, /: Merged revisions 71358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
	  2 lines Revert the patch from issue 9654 due to an unexpected
	  side effect ........

2007-06-24 17:50 +0000 [r71289-71291]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
	  peer->cdr is what we really wanted to use

	* main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
	  Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
	  be able to set variables to the empty string. ........

2007-06-23 03:29 +0000 [r71230]  Steve Murphy <murf@digium.com>

	* main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
	  where clid and src are lost via bridging.

2007-06-22 22:44 +0000 [r71214]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
	  Jun 2007) | 1 line fixed a bug that was introduced by copy and
	  paste in the last commit ..bchannels weren't cleaned properly.
	  ........

2007-06-22 15:38 +0000 [r71096-71123]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 70672 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
	  1 line we activate the bchannels in TE mode on incoming calls
	  only when we want to connect the call. ........

	* channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
	  Jun 2007) | 1 line forgot one place .. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
	  Jun 2007) | 1 line on receiption of cause:44 we mark the channel
	  as in use and inform the user about the situation, we need to
	  test the RESTART stuff then. Also shuffled the
	  empty_chan_in_stack function after the bchannel cleaning
	  functions, to avoid race conditions. ........

	* channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
	  Jun 2007) | 1 line when we send out a SETUP, but get no response,
	  we should cleanup everything after reception of a hangup.
	  ........

	* /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
	  1 line restart indicator 0x80 is correct, at least that's what
	  libpri does. ........

	* channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
	  Jun 2007) | 1 line if the bridged partner is mISDN too we should
	  not send dtmf tones, they are transmitted inband always ........

	* channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
	  Jun 2007) | 1 line if we have already some digits, we just stop
	  the tones. ........

2007-06-22 15:00 +0000 [r71068]  Jason Parker <jparker@digium.com>

	* apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
	  revisions 71065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
	  lines Fix a few silly usages of ast_playstream() - it only ever
	  returns 0... Issue 10035 ........

2007-06-22 14:53 +0000 [r71066]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 71064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
	  10 lines Fixed infinite loop when controlling terminal was lost
	  and return value of input function wasn't checked for errors.
	  This would cause 100% cpu to be taken up. (closes issue #9654,
	  issue #10010) Reported by: mnicholson, and eserra Idea for the
	  patch from mnicholson, patched by me ........

2007-06-22 14:10 +0000 [r71063]  Steve Murphy <murf@digium.com>

	* main/cdr.c: My conditions for merging amaflags info was naive;
	  DOCUMENTATION is the default, although null is possible; theft of
	  user-settable fields is not good. Just copy them, leave them
	  alone.

2007-06-22 03:14 +0000 [r71003]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a small typo which ... well ...
	  completely broke chan_iax2. oops! (issue #9937, patch by me)

2007-06-21 22:34 +0000 [r70949]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 70948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
	  line This little fix is in response to bug 10016, but may not
	  cure it. The code is wrong, clearly. In a situation where you set
	  the CDR's amaflags, and then ForkCDR, and then set the new CDR's
	  amaflags to some other value, you will see that all CDRs have had
	  their amaflags changed. This is not good. So I fixed it. ........

2007-06-21 21:40 +0000 [r70899]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
	  lines Don't explode if the gain option is specified without a
	  value. (issue #9274 reported by mfarver) ........

2007-06-21 21:14 +0000 [r70866-70883]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Put the thread reading from the socket back
	  in the idle list if it deferred the processing of a full frame to
	  another thread

	* channels/chan_iax2.c: If a full frame is received while one of
	  the iax2 threads is in the middle of handling a full frame for
	  the same call, queue it up for processing by that same thread
	  later instead of dropping it. (issue #9937, patch by me)

2007-06-21 20:19 +0000 [r70841]  Steve Murphy <murf@digium.com>

	* cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
	  line it was pointed out that the cdr_custom config load could get
	  a lock, and under certain circumstances, would never release it.
	  I also noted that the situation where more than one mapping spec
	  was warned about, but did not ignore further mappings as it had
	  promised. I think I have fixed both situations. ........

2007-06-21 19:49 +0000 [r70808]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: When volgain is used don't leave a
	  temporary file behind. (Closes Issue 8514, Reported and patched
	  by ulogic, code reviewed by Jason Parker)

2007-06-21 15:22 +0000 [r70727]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Do not Packet2Packet bridge if packetization settings
	  do not allow it. (issue #9117 reported by phsultan)

2007-06-21 15:21 +0000 [r70726]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Remove a couple of duplicate unlocks

2007-06-21 13:58 +0000 [r70677]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Fix building with ODBC storage enabled.
	  (issue #10025 reported by denisgalvao)

2007-06-21 13:00 +0000 [r70656]  Steve Murphy <murf@digium.com>

	* main/cdr.c: Via complaints aired in asterisk-users, I submit
	  these changes, which allow cdr updates to see macro
	  context/exten, whether hung up or not

2007-06-20 23:32 +0000 [r70554-70612]  Jason Parker <jparker@digium.com>

	* cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
	  Issue 10020, patch by my, with credit to prashant_jois for
	  pointing out the problem.

	* cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
	  condition fix

	* cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
	  when reloading the module. Issue 10022, patch by me, with credit
	  to prashant_jois for finding the bug.

2007-06-20 22:22 +0000 [r70552]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
	  lines Don't overwrite the configured username setting upon a
	  REGISTER. (issue #8565 reported by jsmith) ........

2007-06-20 20:53 +0000 [r70494]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure we clear the previously dialed
	  number if it did not exist. Issue 9958.

2007-06-20 19:29 +0000 [r70445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
	  | 2 lines Issue 9997 - Timelimit times out the wrong channel
	  ........

2007-06-20 18:46 +0000 [r70397]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
	  5 lines Fix a problem where an established call would not be
	  properly disconnected when a PRI disconnect is received depending
	  on which cause code was received. (issue #9588, original patch by
	  softins, updated patch from jtexter3, and some additional
	  feedback from mhardeman) ........

2007-06-20 17:52 +0000 [r70198-70360]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, main/frame.c: Put the speex packetization values back
	  in but disable it when setting up the smoother.

	* main/frame.c: Don't do packetization/smoother stuff with speex,
	  it doesn't work.

2007-06-20 00:03 +0000 [r70084-70164]  Russell Bryant <russell@digium.com>

	* contrib/scripts/ast_grab_core: don't delete the backtrace in
	  ast_grab_core

	* channels/chan_gtalk.c: Only attempt to queue a hangup on the
	  owner channel if it actually exists. (issue #9795, patch from
	  zandbelt)

2007-06-19 18:23 +0000 [r70062]  Steve Murphy <murf@digium.com>

	* main/channel.c, /: Merged revisions 70053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
	  line This fixes 9246, where channel variables are not available
	  in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
	  consolidate the channel variables during a masquerade, and then
	  copy the merged variables back onto the clone, so the zombie has
	  the same vars that the 'original' has. ........

2007-06-19 17:07 +0000 [r70003]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 69992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
	  lines Handle the CC field in the RTP header. (issue #9384
	  reported by DoodleHu) ........

2007-06-19 16:24 +0000 [r69987]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 69986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
	  lines Update BRIDGEPEER variable if set to the new channel name
	  when a masquerade happens. (issue #9699 reported by dimas)
	  ........

2007-06-19 15:22 +0000 [r69944]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a crash that could occur when handing
	  device state changes. When the state of a device changes, the
	  device state thread tells the extension state handling code that
	  it changed. Then, the extension state code calls the callback in
	  chan_sip so that it can update subscriptions to that extension. A
	  pointer to a sip_pvt structure is passed to this function as the
	  call which needs a NOTIFY sent. However, there was no locking
	  done to ensure that the pvt struct didn't disappear during this
	  process. (issue #9946, reported by tdonahue, patch by me, patch
	  updated to trunk to use the sip_pvt lock wrappers by eliel)

2007-06-19 13:55 +0000 [r69805-69895]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
	  lines Perform an extra hangup check just in case. (issue #9589
	  reported by bcnit) ........

	* /, res/res_features.c: Merged revisions 69846 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
	  lines Add parked call extension AFTER the parking slot has been
	  announced, otherwise two threads will try to handle the same
	  channel and it will go kaboom. (issue #9191 reported by japple)
	  ........

	* main/callerid.c: Fix for building on PowerPC under Linux.

2007-06-18 19:48 +0000 [r69796]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Issue 10005 - Segfault with missing
	  arguments, plus fix a missing define for SIP INFO channels

2007-06-18 19:00 +0000 [r69775-69794]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't count RTP timeout when involved in a
	  T38 fax session. (issue #9222 reported by ivoc)

	* /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
	  lines Set the peer name on the dialog to the one configured in
	  sip.conf and NOT the username to be used for authentication
	  attempts. (issue #9967 reported by achauvin) ........

2007-06-18 17:46 +0000 [r69744]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
	  | 2 lines Issue 9998 - Remove SIG prefix, since it's not
	  supported by ksh ........

2007-06-18 16:51 +0000 [r69708]  Joshua Colp <jcolp@digium.com>

	* main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
	  for the first time so that it does not needlessly spit out
	  changed messages when the host really didn't change.

2007-06-18 16:35 +0000 [r69689-69702]  Russell Bryant <russell@digium.com>

	* res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
	  build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
	  include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
	  To prevent 92138749238754 more reports of "I have unixodbc
	  installed, but still can't build *_odbc.so!", check for ltdl
	  directly, instead of just listing it as another library to
	  include in the unixodbc check in the configure script. This also
	  makes ltdl show up as a dependency in menuselect so people know
	  what to go install. (related to issue #9989, patch by me)

	* build_tools/prep_moduledeps: Change the use of "echo -e" to
	  "printf". On systems where /bin/sh is not bash, most of the lines
	  in menuselect-tree were getting a "-e" at the beginning of every
	  line. I'm surprised nobody noticed this, but I think the XML
	  parser was being very nice and ignoring them.

2007-06-18 16:04 +0000 [r69661-69668]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't defer the BYE till later on a transfer
	  when the transfer itself goes kaboom and has no hope of working.

	* channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
	  something we never locked, and better handle a specific scenario
	  with doing an attended transfer between two non-bridged calls.

2007-06-18 15:46 +0000 [r69660]  Russell Bryant <russell@digium.com>

	* Makefile: Tweak paths for BSD systems (issue #10001, stuarth)

2007-06-18 13:55 +0000 [r69625]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue where it would be possible for the
	  negotiated codecs to get set back to nothing. (issue #9992
	  reported by yehavi)

2007-06-15  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.5 released.

2007-06-15 20:18 +0000 [r69579]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Fix a silly deadlock in res_features that I
	  found while debugging on one of blitzrage's test machines. It was
	  one of the situations where he was seeing hung channels, and may
	  be the cause of some of the reports from other people. (related
	  to issue #9235)

2007-06-15 19:23 +0000 [r69558]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Add support for setting the maximum
	  length of acceptable DTMF in SpeechBackground.

2007-06-15 15:27 +0000 [r69518]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
	  "SUCCESS" for both an answer of the incoming call on the trunk,
	  or if the trunk reached its ring timeout. This patch changes the
	  variable to say "RINGTIMEOUT" in that case. (issue #9973,
	  reported by n00dle, patch by me)

2007-06-14 23:22 +0000 [r69434-69470]  Jason Parker <jparker@digium.com>

	* main/config.c, /: Merged revisions 69469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
	  lines Fix an issue where the line number in an unterminated
	  comment block error message would show the wrong line number.
	  "Reported" to me on #asterisk (somebody posted an error message,
	  and I happened to catch it) ........

	* sounds/Makefile: Update to latest versions of sound files.

2007-06-14 21:50 +0000 [r69392]  Kevin P. Fleming <kpfleming@digium.com>

	* cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
	  cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
	  main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
	  apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
	  main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
	  channels/chan_iax2.c: use ast_localtime() in every place
	  localtime_r() was being used

2007-06-14 21:08 +0000 [r69358]  Russell Bryant <russell@digium.com>

	* main/say.c: Fix some problems with saying dates and times for the
	  "tw" langauge (issue #9964, ljmid)

2007-06-14 15:21 +0000 [r69259]  Jason Parker <jparker@digium.com>

	* funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
	  2007) | 4 lines Change a quite broken while loop to a for loop,
	  so "continue;" works as expected instead of eating 99% CPU...
	  Issue 9966, patch by me. ........

2007-06-13 21:19 +0000 [r69184-69222]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Whoops...

	* channels/chan_iax2.c: Let's make chan_iax2 media only native
	  transfers actually work. (issue #9376 reported by simone
	  cittadini)

	* channels/iax2-parser.c: Add TXMEDIA to list so that it is
	  properly displayed during iax2 packet output.

2007-06-13 19:57 +0000 [r69183]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Move the logic for destroying a call when no
	  response is received to a BYE outside of the block that checks
	  for FLAG_FATAL to be set. This flag is only set when the packet
	  is transmitted with the reliability set to XMIT_CRITICAL when the
	  original packet is transmitted. A BYE is always sent with it set
	  to XMIT_RELIABLE, meaning this code could never be encountered.
	  This resulted in seeing some SIP channels that would never go
	  away with the last packet sent being a BYE. (part of issue #9235,
	  patch from jcmoore)

2007-06-13 19:41 +0000 [r69181]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Contains a patch for fixing an encoding
	  problem when using Outlook to view voicemail emails and
	  attachments. This fix has also been tested on Thunderbird,
	  Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
	  patched by mutterc)

2007-06-13 19:08 +0000 [r69128-69144]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Really ignore NULL frames and check whether
	  the channel hungup or not. (issue #9912 reported by junky)

	* /, main/app.c: Merged revisions 69127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
	  lines Return group counting to previous behavior where you could
	  only have one group per category. (issue #9711 reported by
	  irroot) ........

2007-06-13 16:56 +0000 [r69016-69071]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Clarify a bit of logic. This doesn't change
	  behavior in any way, but it is helpful when following the logic
	  to debug problems like 9235.

	* channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
	  was accessed without the lock held. This issue was reported to me
	  via email by Dmitry Mishchenko. Thanks!

	* cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
	  in #asterisk-bugs. PQclear() was not called on the result
	  structure after doing a PQexec(). Also, fix up some formatting in
	  passing.

2007-06-12 19:36 +0000 [r69012-69014]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Change the full frame dropping log message
	  to debug to avoid future bug reports.

	* channels/chan_iax2.c: Schedule the sending of a PING packet a
	  second later than previously so that it does not collide with the
	  LAGRQ.

2007-06-12 19:13 +0000 [r69010]  Russell Bryant <russell@digium.com>

	* main/channel.c: In ast_channel_make_compatible(), just return if
	  the channels' read and write formats already match up. There are
	  code paths that call this function on a pair of channels multiple
	  times. This made calls fail that were using g729 in some cases.
	  The reason is that codec_g729a will unregister itself from the
	  list of available translators will all licenses are in use. So,
	  the first time the function got called, the right translation
	  path was allocated. However, the second time it got called, the
	  code would not find a translation path to/from g729 and make the
	  call fail, even if the channel actually already had a g729
	  translation path allocated. (SPD-32)

2007-06-12 14:23 +0000 [r68922]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 68921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
	  lines Bring RTP back to Asterisk at the end of a native bridge no
	  matter what. ........

2007-06-11 21:20 +0000 [r68814]  Jason Parker <jparker@digium.com>

	* include/asterisk/time.h: Solaris 10 sometimes (?) needs this
	  include in order to have NULL defined.

2007-06-11 20:45 +0000 [r68781]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
	  used

2007-06-11 16:57 +0000 [r68733]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 68732 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
	  1 line added check for NULL Pointer when calling misdn_new.
	  Asterisk does not allow us to create channels anymore when stop
	  gracefully is used :). also modified the restart_indicator to 0
	  ........

2007-06-11 14:33 +0000 [r68683]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 68682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
	  lines Improve deadlock handling of the channel list. (issue #8376
	  reported by one47) ........

2007-06-11 10:29 +0000 [r68644]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
	  1 line fixed problem that the dummybc chanels had no lock,
	  checking for the lock now. Also fixed the channel restart stuff,
	  we can now specify and restart particular channels too. ........

2007-06-11 04:21 +0000 [r68595]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_config.c: "dialplan save" produced garbage in the config
	  file

2007-06-08 22:23 +0000 [r68527]  Russell Bryant <russell@digium.com>

	* /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
	  4 lines Don't automatically hang up after running Dictate so that
	  callers can exit cleanly using '#' (closes issue #9577, patch
	  from Thomas Andrews) ........

2007-06-08 15:52 +0000 [r68450]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: actually remember the type/subclass of full
	  frames that are in process

2007-06-08 00:17 +0000 [r68370-68401]  Joshua Colp <jcolp@digium.com>

	* /, main/say.c: Merged revisions 68397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
	  lines Don't call ast_waitstream_full when the control file
	  descriptor and audio file descriptor are not set, simply call
	  ast_waitstream! (issue #8530 reported by rickead2000) ........

	* main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
	  lines Do a DNS lookup immediately upon calling the dnsmgr
	  function, don't wait until a refresh happens. (issue #9097
	  reported by plack) ........

2007-06-07 23:14 +0000 [r68354]  Russell Bryant <russell@digium.com>

	* /, main/say.c: Merged revisions 68351 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
	  3 lines Fix a problem where saying a character wouldn't properly
	  break out when the caller pressed '#' (issue #8113, reported by
	  patbaker82, patch from jamesgolovich (hey, long time no see!) and
	  patbaker82) ........

2007-06-07 23:00 +0000 [r68326]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix incorrect French syntax of "old
	  messages". Request for feedback was sent to asterisk-dev mailing
	  list, with little response. Issue 9118, patch by junky.

2007-06-07 22:14 +0000 [r68313]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: some improvements to the IAX2 full frame
	  dropping logic recently added: - use inaddrcmp(), since we have
	  it - output the type of frame and subclass being dropped, and the
	  type/subclass that is already being processed (which caused the
	  drop)

2007-06-07 21:16 +0000 [r68280]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
	  queue members when using realtime configuration for queues. Also,
	  remove an unneeded leading slash for the astdb family. (issue
	  #9911, patch by atis)

2007-06-07 20:25 +0000 [r68211-68249]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an issue with newer phones which
	  require packets be padded out to the correct length. Issue 9887,
	  patch by DEA.

	* apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
	  lines Don't try to save voicemail greetings unless the user
	  presses '1' to accept/save. Issue 9904, patch by me. ........

2007-06-07 19:47 +0000 [r68198]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
	  check to make sure that greetings get stored properly. (Issue
	  8016, reported by edhorton, patched by alamantia with
	  modification by me. Thanks to Jason Parker for the advice on
	  this).

2007-06-07 19:46 +0000 [r68196]  Olle Johansson <oej@edvina.net>

	* channels/chan_features.c: Disable chan_features by default in
	  menuselect

2007-06-07 19:30 +0000 [r68192]  Russell Bryant <russell@digium.com>

	* main/strcompat.c: Include stdarg.h for build issues on Solaris
	  (issue #9381)

2007-06-07 18:39 +0000 [r68071-68157]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Fix logic when doing a name based channel search
	  for a structure when you want to start from a specific point in
	  the channel list. (issue #9324 reported by slavon)

	* apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
	  lines Allow the 'g' option to work if used with the 'S' option.
	  (issue #9888 reported by gasparz) ........

2007-06-07 10:00 +0000 [r67993-68030]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
	  forget.

	* res/res_jabber.c: Ok, we found out that this is not about if you
	  have any *active* clients using TLS, but if you have initialized
	  TLS at all during the lifetime of the module. So if you reload to
	  disable TLS, it won't help.

	* res/res_jabber.c: If you have a jabber client that uses TLS,
	  refuse unload. Bad fix, but will prevent crashes while we are
	  trying to find a workaround. Iksemel development seems to have
	  stalled and we might have to stop using the TCP/TLS connections
	  in that library and use our own, which would scale better from a
	  poll/select perspective I guess. It would also make it easier to
	  migrate to OpenSSL and stop Asterisk from depending on both
	  OpenSSL and GnuTLS.

	* include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
	  sure we can unload res_jabber. Patch by phsultan - thanks! Due to
	  a bug in the iksemel library, this will not work if you are using
	  GTLS in the connection. That's being investigated. If you figure
	  out a way to handle that without us having to patch iksemel, let
	  us know in the bug report. Thanks.

2007-06-07 00:10 +0000 [r67924-67941]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
	  lines Only notify the devicestate system of a peer state change
	  when the peer is built from the config file. (issue #9900
	  reported by arkadia) ........

	* main/file.c: Properly handle cases where a stream can't be
	  written to. (issue #9757 reported by junky)

2007-06-06 22:08 +0000 [r67862-67872]  Russell Bryant <russell@digium.com>

	* res/res_snmp.c: Disable reload functionality in res_snmp. It is
	  not possible to initialize the snmp library more than once
	  without completely unloading the module and loading it again.
	  (issue #9571, reported by hristo, additional helpful debug
	  information from festr, patch from me)

	* channels/chan_sip.c: Fix a crash when doing call pickups with SIP
	  phones. The code unlocked the channel when it should not have.
	  (issue #9652, reported by corruptor, fixed by me)

2007-06-06 19:26 +0000 [r67804]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
	  under a specific set of circumstances: 1. VoiceMailMain was
	  configured in the dialplan with an extension as its argument 2. A
	  message was left for this mailbox 3. Tried to call VoiceMailMain
	  but hung up before entering password. This was fixed by checking
	  that a pointer was non-null prior to trying to dereference it.
	  (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
	  modifications by me).

2007-06-06 16:55 +0000 [r67716]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/linkedlists.h: Merged
	  revisions 67715 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
	  5 lines We have some bug reports showing crashes due to a double
	  free of a channel. Add a sanity check to ast_channel_free() to
	  make sure we don't go on trying to free a channel that wasn't
	  found in the channel list. (issue #8850, and others...) ........

2007-06-06 13:30 +0000 [r67594-67650]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 67649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
	  lines Reinvite the RTP back to the Asterisk machine when the
	  timeout happens. (issue #9888 reported by gasparz) ........

	* main/translate.c: Fix plc_samples warning when registering a
	  translator. (issue #9897 reported by xylome)

	* apps/app_directed_pickup.c: Include macroexten while searching
	  for a channel to pick up in case they are in a macro. (issue
	  #9491 reported by jamesb63)

	* res/res_agi.c: Make the new "agi debug off" CLI command work.
	  (issue #9890 reported by eliel)

	* /, main/devicestate.c: Merged revisions 67593 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
	  lines Revert channel name splitting fix for Zap. The moral of the
	  story is don't use - in your user/peer names. (issue #9668
	  reported by stevedavies) ........

2007-06-05 23:01 +0000 [r67558]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Fix some crashes related to the use of the
	  "meetme" CLI command. The code for this command was not locking
	  the conference list at all. (issue #9351, reported by and patch
	  submitted by Junk-Y, committed patch is different and by me)

2007-06-05 21:30 +0000 [r67526]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
	  9883, wherein macros were not allowing the includes construct.
	  fixed and tested, looks OK. Now includes can serve as an adjunct
	  to catch.

2007-06-05 20:53 +0000 [r67457-67492]  Russell Bryant <russell@digium.com>

	* include/asterisk/linkedlists.h: This bug has been hanging over my
	  head ever since I wrote this SLA code. Every time I tried to go
	  debug it by adding some debug output, the behavior would change.
	  It turns out I wasn't crazy. I had the following piece of code:
	  if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
	  AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
	  conditional statement didn't do much good at all. It always ran
	  at least all of the macro minus the first statement, so I was
	  seeing list entries magically disappear when they weren't
	  supposed to. After many hours of debugging, I have come to this
	  extremely irritating fix. :) (issues #9581, #9497)

	* channels/chan_zap.c: Suppress a bunch of debug output unless
	  option_debug is on

2007-06-05 18:32 +0000 [r67424]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
	  saved to IMAP storage using extensions other than gsm were unable
	  to be played over the phone. (Issue 9786, reporter:
	  xmarksthespot, Patched by xmarksthe spot with revisions by me,
	  reviewed by Russell Bryant).

2007-06-05 18:18 +0000 [r67421]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Correctly update date/time on devices
	  throughout the life of the device, instead of just at
	  registration. Issue 9152, yet another patch by DEA.

2007-06-05 18:17 +0000 [r67420]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Added code to automatically add a default case to
	  switches that don't have one. In some cases, rather than fall
	  thru, it results in a goto with -1 result, which terminates the
	  extension; a sort of dialplan seqfault, sort of. This was
	  required to fix bug reported in 9881

2007-06-05 17:07 +0000 [r67360-67372]  Russell Bryant <russell@digium.com>

	* main/channel.c: Handle a failure in malloc() in
	  ast_safe_string_alloc()

	* main/channel.c: Fix a problem that showed itself by causing Zap
	  channel names to be completely bogus on my machine.
	  ast_safe_string_alloc() was broken. It called vsnprintf() on a
	  va_args list twice without re-initializing it. After the first
	  usage, va_end() and va_start() must be called again.

2007-06-05 16:14 +0000 [r67329-67334]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
	  1 line briding is a bool, fixed copy and paste issue. ........

	* channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
	  Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
	  cb_events function a lot. Commented the different possibilities a
	  bit and made functions of shared code. When the dialed extension
	  does not exist in the extensions.conf we'll jump into the 'i'
	  extension if this does exist, else we disconnect the call with
	  the cause:1 = No Route to Destination. ........

2007-06-05 15:51 +0000 [r67308]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, main/loader.c, include/asterisk/module.h: When
	  shutting down "gracefully", go through and run the unload()
	  callbacks for all of the modules. "stop now" is considered a
	  non-graceful shutdown and will not go through this process.
	  (issue #9804, reported by chrisost, patch by me)

2007-06-05 15:22 +0000 [r67304]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Only muck with the thread structure if an
	  idle one was found/created.

2007-06-05 14:35 +0000 [r67270]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: ensure that a burst of full frames
	  (AST_FRAME_DTMF being the prime example) will not be processed
	  out of order... this is a brute force fix, but seems to be the
	  safest fix for now (thanks to the Digium PQ department for
	  finding this bug)

2007-06-05 10:25 +0000 [r67210]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
	  1 line added possibility to deactivate bridging per port ........

2007-06-04 23:43 +0000 [r67162]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
	  | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
	  ........

2007-06-04 23:31 +0000 [r67158]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
	  structure can disappear and the code did not account for it and
	  crashes. (issues #9642, #9569, #9666, probably others ... based
	  on the work by stevedavies and mihai, with additional changes
	  from me)

2007-06-04 23:26 +0000 [r67121-67156]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix for skinny keepalives. If there is no
	  traffic from the phone for (keep_alive * 1100) ms (arbitrarily
	  adding 10% for network issues, etc), unregister the device. Issue
	  8394, patch by DEA.

	* channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
	  to the recent fix for chan_skinny) Issue 9855, patch by DEA.

2007-06-04 22:28 +0000 [r67119]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Add comments for two functions that get
	  called with the appropriate call locked, but perform operations
	  that could result in the pvt structure getting destroyed before
	  returning again, causing numerous seg faults all over the module.
	  (inspired by issues #9642, #9569, and #9666, and the work done by
	  stevedavies and mihai)

2007-06-04 21:59 +0000 [r67073]  Steve Murphy <murf@digium.com>

	* main/cdr.c: This typo has been here since 1.4 forked. It has been
	  the source of heartburn to many a dialplan/CDR programmer.

2007-06-04 21:47 +0000 [r67071]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)

2007-06-04 19:31 +0000 [r67064-67068]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Better handle SIP devices that say they have
	  SDP content... but really don't. (issue #9398 reported by
	  mthomasslo)

	* apps/app_dial.c: Initialize cidname variable to nothing since it
	  may be used without having been touched. (issue #9661 reported by
	  dimas)

	* res/res_features.c: Returning a value that indicates the parking
	  of a call was a success when it really wasn't (because the
	  parking slot selected was in use) is the wrong thing to do.
	  (issue #9723 reported by mdu113)

2007-06-04 17:11 +0000 [r67061]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk, /,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.mandrake.zaptel,
	  contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
	  | 2 lines Add revision Id tags (by request of tzafrir) ........

2007-06-04 16:02 +0000 [r67026]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Change the configure script to build a
	  test program against libcurl to make sure the results from
	  curl-config can be used to compile successfully. This is intended
	  to help prevent a situation where you are cross compiling, and
	  the configure script finds the curl library installed on the
	  host. (issue #9865, reported and patched by zandbelt)

2007-06-04 15:50 +0000 [r67021]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_jabber.c: Issue 9739 - Malformed jid causes a crash

2007-06-04 15:47 +0000 [r67018-67020]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
	  handling an implicit ACK to a frame that was marked as the final
	  transmission for a call, don't call iax2_destroy() for that call
	  while the global frame queue is still locked. There is a very
	  nice explanation of the deadlock in the report. (issue #9663,
	  thorough report and patch from stevedavies, additional positive
	  test reports from mihai and joff_oconnell)

	* include/asterisk/stringfields.h: Fix some compiler warnings in
	  C++ modules. (issue #9866, reported by osk, patch by Corydon76)

2007-06-01 21:45 +0000 [r66919]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_odbc.c: On some drivers, deallocating the statement
	  handle isn't enough. We also have to clear the cursor (nice,
	  Oracle)

2007-06-01 21:31 +0000 [r66897-66917]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing extraneous debugging lines from
	  revision 66897. Sorry :)

	* apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
	  storage. Attachments with format specified as gsm were duplicated
	  (i.e. two attachments) were left. Thank you very much to
	  xmarksthespot for submitting the patch that fixed this. (Issues
	  9787 and 8873, Reported by xmarksthespot and jerjer, patched by
	  xmarksthespot)

2007-06-01 19:41 +0000 [r66879-66881]  Russell Bryant <russell@digium.com>

	* channels/chan_skinny.c: Changes to the way DTMF is handled in the
	  core broke dialing in chan_skinny. This patch makes chan_skinny
	  usable again. I did not end up testing this, but there are
	  multiple positive test reports listed in the bug report. (issue
	  #9596, reported by pj, testing by pj and mvanbaak, and the fix
	  was written by DEA)

	* apps/app_page.c: List app_meetme as a module that app_page
	  depends on.

2007-05-31 23:03 +0000 [r66821]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* doc/asterisk.8: Issue 9850 - update preferred command line syntax

2007-05-31 18:41 +0000 [r66775]  Russell Bryant <russell@digium.com>

	* res/res_speech.c, include/asterisk/app.h,
	  include/asterisk/speech.h: Change a couple of header files to not
	  use "new", which is a reserved keyword in C++. (issue #9830,
	  reported by osk)

2007-05-31 17:15 +0000 [r66770]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
	  | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
	  Issue 8329 will remain unfixed for pbx_realtime, but only because
	  we lack core API to do it. ........

2007-05-31 16:14 +0000 [r66768]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
	  lines It is now possible for this path of execution to have the
	  frame pointer be NULL, therefore we need to check for it before
	  trying to access it. (issue #9836 reported by barthpbx) ........

2007-05-30 23:26 +0000 [r66671]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed seg-faults when recording greetings
	  in voicemail with IMAP enabled. (Issue No. 9735, reported by
	  xmarksthespot, patched by me)

2007-05-30 17:28 +0000 [r66602-66639]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Silly me for having out of date source! Oh
	  well... I'm still leaving my comment.

	* channels/chan_sip.c: When calling some peer/host that may not
	  exist/reply back... don't keep the dialog in memory for all of
	  eternity.

	* channels/chan_zap.c, channels/chan_features.c: Change how channel
	  names are generated a bit. (issue #9825 reported by eldadran)

2007-05-29 21:56 +0000 [r66538]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
	  | 2 lines If the value of a variable passed to FIELDQTY is blank,
	  then FIELDQTY should return 0, not 1. ........

2007-05-29 19:32 +0000 [r66474-66503]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Properly handle 408 request timeout -
	  according to the RFC, the dialog dies if a request in a dialog
	  gets this response.

	* channels/chan_sip.c: Don't issue hangup on hangup on hangup on
	  hangup (for jcmoore)

2007-05-29 16:44 +0000 [r66437]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Handle cases where a frame may have no data. (issue
	  #9519 reported by dmb)

2007-05-29 16:07 +0000 [r66404-66414]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't reset hangupcause if we already have
	  one

	* channels/chan_sip.c: Tracking down hanging channels, killing them
	  one by one. Issue #9235 and related

2007-05-29 15:43 +0000 [r66398]  Joshua Colp <jcolp@digium.com>

	* doc/datastores.txt: Update datastores documentation. (issue #9801
	  reported by mnicholson)

2007-05-29 09:41 +0000 [r66363]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
	  lines Issue #9802 - Change inuse counter on CANCEL ........

2007-05-28 23:16 +0000 [r66312]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Make the usedistinctiveringdetection option
	  work again. (issue #9823 reported by premeau)

2007-05-27 04:12 +0000 [r66244]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: I don't know what this was trying to do, but
	  it's clearly incorrect. Issues 9808 and 9809.

2007-05-25 14:43 +0000 [r66160]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: have to check for OSP toolkit _after_
	  checking for OpenSSL

2007-05-25 14:41 +0000 [r66159]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, main/say.c: Merged revisions 66127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
	  | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
	  ........

2007-05-25 14:28 +0000 [r66157]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
	  res/res_jabber.c: handle the GNUTLS library properly in the
	  configure script and build system don't build in OSP support
	  unless we have found and are allowed to use SSL support

2007-05-24 22:23 +0000 [r66076]  Russell Bryant <russell@digium.com>

	* main/channel.c: if the string field init fails, clean up the
	  stuff that was allocated already

2007-05-24 22:16 +0000 [r66074]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Fix slinfactory logic when dealing with
	  frames coming in that may already be in the signed linear format.

2007-05-24 22:07 +0000 [r66068-66070]  Russell Bryant <russell@digium.com>

	* main/channel.c: Check the result of ast_string_field_init() in
	  ast_channel_alloc()

	* main/rtp.c: Make 1.4 build on my machine, too..

2007-05-24 20:54 +0000 [r66029-66030]  Jason Parker <jparker@digium.com>

	* configure: Rebuild configure script for previous ar fix.

	* configure.ac: Following moving strip to AC_PATH_TOOL, we need to
	  do something similar for ar.

2007-05-24 20:42 +0000 [r65978-66026]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Checking for the strip application needs to be done with
	  AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
	  compilation environments.

	* Makefile: Clear CFLAGS before running make for menuselect. (issue
	  #9784, reported by ovi, patch by me)

2007-05-24 18:28 +0000 [r65965-65967]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_gtalk.c: oops, use #ifdef instead of #if

	* channels/chan_gtalk.c: don't reference GnuTLS headers and
	  functions unless the configure script found it

	* main/rtp.c: don't use uninitialized variables

2007-05-24 15:27 +0000 [r65902]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Add the ability to blacklist certain commands
	  from being executed using the Command AMI action. (issue #9240
	  reported by junky)

2007-05-24 15:26 +0000 [r65892-65901]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
	  core dump since the GnuTLS interface did not support
	  multithreading correctly.

	* channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
	  Patch by phsultan. Thanks!

2007-05-24 15:16 +0000 [r65877-65883]  Jason Parker <jparker@digium.com>

	* .cleancount: Update cleancount for that last commit - just for
	  good measure.

	* include/asterisk/translate.h, codecs/codec_speex.c,
	  main/translate.c, codecs/codec_ilbc.c: Fix handling of
	  zero-length frames when a codec is capable of native PLC. Issue
	  9183, patch by Mihai.

2007-05-24 15:08 +0000 [r65866]  Dwayne M. Hubbard <dhubbard@digium.com>

	* funcs/func_math.c: merged qwell's func_math patch for issue 9507

2007-05-24 15:08 +0000 [r65863]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: I like it when the RTP stack compiles myself...

2007-05-24 15:05 +0000 [r65857]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
	  when calling from gtalk to SIP over nat.

2007-05-24 15:04 +0000 [r65842-65853]  Russell Bryant <russell@digium.com>

	* apps/app_festival.c: Ensure that frames are fully initialized.
	  This will probably fix getting weird timestamp log messages in
	  logs when using the Festival app. (issue #9781, patch by me)

	* main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
	  code would result in oscillating and incorrect data.
	  Additionally, the RTT would sometimes report negative values due
	  to incorrect calculations. (issue #9601, patch from davetroy)

2007-05-24 14:48 +0000 [r65841]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
	  jingle

2007-05-24 14:42 +0000 [r65839]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
	  lines Allow RFC2833 to be negotiated when an INVITE comes in
	  without SDP and is not matched to a user or peer. (issue #9546
	  reported by mcrawford) ........

2007-05-24 14:38 +0000 [r65836]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
	  Fix "login" as component to jabber server. ...and, by accident,
	  fix a bug in chan_sip for stopping a loop on retransmits of BYE
	  requests.

2007-05-24 09:37 +0000 [r65768]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
	  Mai 2007) | 1 line we should only activate the generator in
	  chan_misdn, when asterisk hask not yet taken the call
	  (WAITING4DIGS state). Alerting audio will be generated fomr
	  asterisk for example. ........

2007-05-23 20:59 +0000 [r65677-65685]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: start the delayed PBX when receive voice or
	  video full frames as well, and comment this delayed-PBX activity

	* /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
	  | 2 lines ensure that variables are set on a newly created
	  channel before we start a PBX on it ........

	* channels/chan_iax2.c: clear the 'delay PBX' flag when we are
	  ready to start the PBX

	* channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
	  channel until we have some sort of response to our ACCEPT (ACK or
	  anything else)

	* /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
	  | 2 lines if we are going to set variables on a newly created
	  channel, it should be done *before* we start the PBX on it
	  ........

2007-05-23 13:07 +0000 [r65589]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
	  3 lines Revert revision 62417 as someone reported problems with
	  it to Mark. This was related to issue #9588. ........

2007-05-22 20:25 +0000 [r65541]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_version: when building a version string for a
	  developer branch, include the base branch in the version string

2007-05-22 18:40 +0000 [r65501]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
	  dependency for app_voicemail and chan_zap (Thanks to mnicholson
	  for pointing it out)

2007-05-22 15:04 +0000 [r65452]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Remove a double const.

2007-05-22 14:02 +0000 [r65408]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Fix a problem with flag recognition.

2007-05-22 13:09 +0000 [r65394]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
	  4 lines Fix a memory leak that I just noticed in the device state
	  handling in app_queue. On most device state changes, it would
	  leak roughly 8 to 64 bytes (the length of the name of the
	  device). ........

2007-05-22 08:12 +0000 [r65342]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
	  Mai 2007) | 1 line we stop the tones only when we're in the
	  pre-call phase, otherwise e.g. when in CONNECTED state we should
	  not stop tones when we receive an Information Message ........

2007-05-20 17:59 +0000 [r65250]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: res_agi needs to export two symbols
	  (ast_agi_register and ast_agi_unregister) for usage by others.
	  (issue #9755 reported by mnicholson)

2007-05-18 22:26 +0000 [r65200-65201]  Steve Murphy <murf@digium.com>

	* main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
	  to main/cdr.c, and neither did I. This is the remainder of the
	  9717 patch, the fix for the run-away FAIL status for a call

	* apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
	  65172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
	  line This update will fix the situation that occurs as described
	  by 9717, where when several targets are specified for a dial, if
	  any one them reports FAIL, the whole call gets FAIL, even though
	  others were ringing OK. I rearranged the priorities, so that a
	  new disposition, NULL, is at the lowest level, and the
	  disposition get init'd to NULL. Then, next up is FAIL, and next
	  up is BUSY, then NOANSWER, then ANSWERED. All the related set
	  routines will only do so if the disposition value to be set to is
	  greater than what's already there. This gives the intended
	  effect. So, if all the targets are busy, you'd get BUSY for the
	  call disposition. If all get BUSY, but one, and that one rings is
	  not answered, you get NOANSWER. If by some freak of nature, the
	  NULL value doesn't get overridden, then the disp2str routine will
	  report NOANSWER as before. ........

2007-05-18 18:16 +0000 [r65041-65123]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
	  lines Not getting an ACK to a 200 OK in the initial invite is
	  critical to the call. ........

	* /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
	  lines Issue 9235 - part of the problem, maybe not all. Please
	  retry with this patch (and no other patch) if you have problems
	  with hanging SIP channels. Thank you. A special Thank You to
	  WeBRainstorm that gave me access to his system. ........

	* channels/chan_sip.c: - Adding support for putting calls OFF hold
	  with a re-invite with blank SDP. This was a bug found while doing
	  tests at SIPit in Antwerp. - In order to not duplicate code, I
	  restructured some of the code for putting calls on/off hold.
	  Thanks DEA for reminding me. This fix has been asleep in the
	  videocaps branch until now.

2007-05-18 12:40 +0000 [r65039]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
	  revisions 65007 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
	  1 line fixed a warning regarding Keypad encoding. encode the IE
	  sending_complete at the right position. ........

2007-05-18 10:37 +0000 [r64974]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
	  call

2007-05-18 08:58 +0000 [r64904]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
	  Mai 2007) | 1 line we *need* to send a PROCEEDING when
	  sending_complete is set, even if need_more_infos is requested.
	  ........

2007-05-18 02:48 +0000 [r64868]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix a small bug I noticed while working on
	  something else. app_queue did not unregister its device state
	  monitoring callback in unload_module(). So, this would make
	  Asterisk crash on the first device state change after you unload
	  the module.

2007-05-17 21:19 +0000 [r64820]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 64819 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
	  | 2 lines How is it that we never caught that this is returning
	  the opposite of our documentation, until now? ........

2007-05-17 16:53 +0000 [r64761]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
	  lines If we have a negative current message, we shouldn't go back
	  even further... Issue 9727. ........

2007-05-17 16:52 +0000 [r64756-64759]  Russell Bryant <russell@digium.com>

	* contrib/scripts/astxs (removed): Remove script that is no longer
	  functional since the build system was redone. (issue #9340,
	  reported by junky)

	* apps/app_dial.c: Increase the size of a buffer to support longer
	  dial strings for channels. (issue #9291, reported and fix
	  suggested by meni)

2007-05-17 16:10 +0000 [r64720-64754]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Even more direct RTP setup fixes! Don't
	  allow a codec that isn't supported to creep into the SDP of
	  either side. (issue #9446 reported by marcelbarbulescu)

	* apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
	  by xmarksthespot)

2007-05-17 06:13 +0000 [r64686]  Russell Bryant <russell@digium.com>

	* README: Update the main README to reflect the new build process
	  for 1.4 and above. (issue #9725, patch by eliel)

2007-05-16 11:01 +0000 [r64516-64609]  Olle Johansson <oej@edvina.net>

	* /: Blocking patch already in this code

	* channels/chan_sip.c: Fix auth on BYE. (Different patch than for
	  1.2)

	* channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE

	* channels/chan_sip.c: Final part of issue #9483 - fixing
	  transfer() of sip calls in the dial plan (twilson)

	* channels/chan_sip.c: Issue #9439 - properly handle username
	  parameters in SIP uri.

	* /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
	  lines Support SIP uri's starting with SIP: and sip: (reported by
	  Tony Mountfield on the mailing list. Thanks!) ........

	* /, channels/chan_sip.c: Merged following patch with a lot of
	  changes for 1.4 ------ Merged revisions 64514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
	  lines Issue #9726 - rlister - Better logging for ACL denials
	  While at it, also added better logging and handling of peers that
	  are not supposed to register. My patch, stole the issue report
	  from Russell. My apologies, Russell :-) ........

2007-05-16 08:44 +0000 [r64515]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
	  Mai 2007) | 1 line in the case immediate=yes, we directly jump
	  into the dialplan, where people can use PlayTones to indicate a
	  Dialtone, so we don't need to to that by ourself. also we should
	  not do a dialtone_indicate for incoming calls on a TE port in
	  overlapdialmode. ........

2007-05-15 19:52 +0000 [r64353-64426]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Properly fix a problem that occurs when you
	  set PARKINGEXTEN to an exten where a call is already parked.
	  (issue #9723, patch by me)

	* res/res_features.c: When someone requests a specific parking
	  space using the PARKINGEXTEN variable, ensure that no other
	  caller is already there. (issue #9723, reported by mdu113, patch
	  by me)

2007-05-14 19:26 +0000 [r64324]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
	  more

2007-05-14 19:13 +0000 [r64306]  Russell Bryant <russell@digium.com>

	* channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
	  just ignoring it. An unknown indication will trigger an error and
	  cause sounds to stop, which in this case, is ringing.

2007-05-14 18:52 +0000 [r64280]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Handle network errors, like host or network
	  unreachable, in a better way. This means that calls to hosts or
	  qualify (OPTION) messages will fail quicker if the TCP/IP stack
	  tells us that there is an issue. Since this is an unconnected UDP
	  socket, we will not get error messages directly in most cases,
	  but maybe on the second and third try. This is already
	  implemented in trunk.

2007-05-14 18:48 +0000 [r64240-64278]  Joshua Colp <jcolp@digium.com>

	* codecs/codec_speex.c: Properly set datalen field when doing PLC
	  in codec_speex. (issue #9722 reported by mihai)

	* /, main/devicestate.c: Merged revisions 64275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
	  lines Only perform stripping of - strings from the channel name
	  for Zap channels. Anywhere else we might remove a legitimate part
	  of a device name. (issue #9668 reported by stevedavies) ........

	* main/channel.c: Fix scenario where if a phone that simply called
	  Echo() put itself on hold it could never get off hold.

2007-05-14 13:58 +0000 [r64193]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
	  worrisome CDR warnings have been removed, that are either not
	  helpful, or not relevant.

2007-05-14 10:39 +0000 [r64157]  Olle Johansson <oej@edvina.net>

	* main/channel.c: Add hangupcause when we lack codecs for
	  transcoding

2007-05-12 22:27 +0000 [r64044-64114]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: This concludes my final adventure with
	  bitmasks and the onhold flag. Would anyone care for some peanuts?

	* channels/chan_sip.c: Tweak hold flags some more. They can be of
	  three states when active: active, inactive, one direction.

	* channels/chan_sip.c: Ensure the onhold flag is set no matter what
	  when being put on hold.

2007-05-11 20:16 +0000 [r63982]  Jason Parker <jparker@digium.com>

	* main/manager.c: Hide manager password from "manager show user
	  foo". I realize that there are other ways to get this, but we
	  really don't need to just show it in plain text so easily. Issue
	  9273, patch by junky

2007-05-11 16:35 +0000 [r63905]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
	  63903 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
	  | 2 lines Issue 9121 - fixups for safe_asterisk script ........

2007-05-11 16:05 +0000 [r63886]  Russell Bryant <russell@digium.com>

	* main/manager.c: When MD5 authentication is not possible because
	  there is no challenge present, either because the Challenge
	  action was never issued, or some other reason, give a proper
	  error message and return an error instead of claiming that the
	  user wasn't found. (reported by jsmith on IRC)

2007-05-11 15:43 +0000 [r63872]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Make the PARKINGEXTEN feature of parking
	  actually work. (issue #9708 reported by mdu113)

2007-05-10 23:15 +0000 [r63830]  Jason Parker <jparker@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
	  lines Fix an issue with trying to kill a thread before it gets
	  created. Issue 9709, patch by nic_bellamy. ........

2007-05-10 22:23 +0000 [r63804]  Russell Bryant <russell@digium.com>

	* main/manager.c: Strip terminal escape sequences from CLI command
	  output that is going to be sent out over the manager interface.
	  (issue #9659, reported by pari, fixed by me)

2007-05-10 20:48 +0000 [r63750]  Doug Bailey <dbailey@digium.com>

	* main/callerid.c: Add test for negative offsets in cid data to
	  prevent infinite loops.

2007-05-10 20:46 +0000 [r63749]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
	  lines Do not allocate SIP pvt's for PEERs we can not reach. This
	  was seen as a lot of dialogs being created then immediately
	  destroyed at reload/restart of the SIP channel. ........

2007-05-09 19:22 +0000 [r63656-63698]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Use the DTMF frame on the channel when returning
	  a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.

	* channels/chan_sip.c: Do not prematurely go on hold if sendonly
	  was not actually set.

2007-05-09 17:25 +0000 [r63654]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
	  lines Make sure we only create a DSP if it's requested on
	  SUB_REAL ........

2007-05-09 16:55 +0000 [r63612]  Russell Bryant <russell@digium.com>

	* main/channel.c: Modify ast_senddigit_begin() to use the same
	  assumptions used elsewhere in the code in that if a channel does
	  not have a send_digit_begin() callback, it only cares about DTMF
	  END events. (pointed out by Michael Neuhauser on the asterisk-dev
	  list)

2007-05-09 16:54 +0000 [r63611]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
	  lines Properly handle hints that point to multiple devices in
	  chan_sip. Why chan_sip is even doing this I have no idea but I
	  would rather not go into a rant. (issue #9536 reported by
	  rlister) ........

2007-05-09 16:43 +0000 [r63608]  Russell Bryant <russell@digium.com>

	* main/channel.c: Only call ast_senddigit_begin() in
	  ast_senddigit() if the channel has a send_digit_begin() callback.
	  Checking the END_DTMF_ONLY flag was the wrong thing to do,
	  because that flag indicates that a *bridged* channel only wants
	  DTMF END events coming from this channel.

2007-05-09 14:50 +0000 [r63566]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
	  | 2 lines Replicate fix from 51158 (app_voicemail) to
	  app_directory (Issue 9224) ........

2007-05-09 13:24 +0000 [r63535]  Russell Bryant <russell@digium.com>

	* Makefile: I have seen multiple people post questions trying to
	  figure out what the message "The configure script must be
	  executed before running 'make'" means. So, add another like that
	  says to specifically run ./configure. If this isn't obvious
	  enough, then they should be using something like AsteriskNOW and
	  not installing from source.

2007-05-09 13:17 +0000 [r63534]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c: Merged revisions
	  62945,63402,63519 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
	  1 line when we're in state WAITING4DIGS, we use the asterisk
	  tone-generator which prods us, so we can't just return -1 in
	  misdn_write in this case. Added a MISDN_KEYPAD channel variable,
	  and fixed the sending of keypad. this enables us to modify the
	  call forward parameters in the switch. ........ r63402 | crichter
	  | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
	  application misdn_check_l2l1 which tries to pull up the L1/L2 on
	  all ports that have the layers down in a group. It waits then for
	  a timeout. This helps for scenarios where multiple PMP BRIs are
	  grouped together, or where a provider has a faulty PTP
	  Implementation, that looses the L2 after a while. ........ r63519
	  | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
	  release_chan frees ch, so we should never touch ch after
	  release_chan, this may cause segfaults. ........

2007-05-09 13:04 +0000 [r63532]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
	  (Reported on asterisk-users)

2007-05-08 22:38 +0000 [r63478]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
	  | 2 lines Issue 9602 - segfault in app_macro ........

2007-05-08 16:53 +0000 [r63403-63448]  Russell Bryant <russell@digium.com>

	* res/res_features.c: I mixed up the use of the find_feature()
	  function, so I renamed it find_dynamic_feature, and changed the
	  code to use the correct lock when using it.

	* res/res_features.c: Use a read/write lock when accessing the
	  built-in features.

	* contrib/scripts/realtime_pgsql.sql (added),
	  contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
	  contrib/scripts to be with the rest of the sql examples. (issue
	  #9676, suretec)

2007-05-08 06:22 +0000 [r63360]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
	  | 2 lines Issue 9527 - upon entering a folder, no message is
	  selected (curmsg == -1), so deleting causes memory corruption
	  (beyond bounds) ........

2007-05-07 22:28 +0000 [r63329]  Russell Bryant <russell@digium.com>

	* configs/res_pgsql.conf.sample (added),
	  configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
	  (added): Add a sample configuration file and example tables for
	  use with res_config_pgsql. (issue #9676, suretec)

2007-05-07 21:45 +0000 [r63283-63286]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
	  revisions 63285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
	  lines Properly handle what happens during a masquerade in
	  relation to group counting. (issue #9657 reported by ramonpeek)
	  ........

	* channels/chan_sip.c: Minor backport of revision 59083 in trunk.
	  Don't queue an unhold frame up if the call was never on hold to
	  begin with.

2007-05-07 20:05 +0000 [r63196-63254]  Olle Johansson <oej@edvina.net>

	* main/config.c: Don't remove configuration from memory just
	  because one section failed.

	* /: Guess svnmerge doesn't handle files that move around. Blocking
	  patch to ./config.c

2007-05-06 12:28 +0000 [r63152]  Olle Johansson <oej@edvina.net>

	* main/file.c: Stop the video stream when you stop playback of all
	  streams for a call

2007-05-04 20:03 +0000 [r63099]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Fix a crash when checking version attribute in
	  an incoming XML caps element. Issue 9667, patch by phsultan.

2007-05-04 16:45 +0000 [r63047]  Pari Nannapaneni <paripurnachand@digium.com>

	* configs/manager.conf.sample: explanation for httptimeout in
	  manager.conf

2007-05-03 16:44 +0000 [r62989]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
	  lines When a peer is seeded or built tell the devicestate core to
	  update it's status. This is easier then having chan_sip load
	  before pbx_config. (issue #9658 reported by dlynes) ........

2007-05-03 16:38 +0000 [r62986]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c: improve loader a bit, by avoiding trying to
	  initialize embedded modules twice and avoiding trying to load
	  modules from disk when they have been loaded already during the
	  'preload' pass (reported by blitzrage on IRC, patch by me)

2007-05-03 15:23 +0000 [r62942]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
	  Bryant, 2007, TM, Patent Pending). This set of changes came from
	  a debugging session I had with Dwayne Hubbard. When he called
	  into his home FXO, ran the Echo application, and pressed a digit,
	  the digit would be echoed back and would never end. This is
	  fixed, along with a couple other little improvements. * When
	  chan_zap is in the middle of playing a digit to a channel, it
	  feeds back null frames, not voice frames. So, I have modified
	  ast_read to check the timing on emulated DTMF when it receives
	  null frames, in addition to where it was doing this on voice
	  frames. * Make a tweak to setting the duration on emulated DTMF
	  digits. If there was no duration specified, it set it to be the
	  minimum, instead of the default. * Instead of timing the emulated
	  digits off of the number of samples in audio frames that pass
	  through, just use time values. Now there is no code in this
	  section that assumes 8kHz audio.

2007-05-03 14:41 +0000 [r62913]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
	  (added), pbx/ael/ael-test/ael-test18/extensions.ael,
	  pbx/ael/ael-test/ael-test19/extensions.ael (added),
	  pbx/ael/ael-test/ael-test19 (added),
	  pbx/ael/ael-test/ref.ael-test20 (added),
	  pbx/ael/ael-test/ael-test20/extensions.ael (added),
	  pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
	  to match what's in trunk

2007-05-03 14:36 +0000 [r62912]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
	  revisions 61357,61770,62885 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
	  1 line some fixes for PMP Hold/Retrieve, it should work now, when
	  briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
	  (Di, 24 Apr 2007) | 1 line added lock for sending messages to
	  avoid double sending. shuffled some empty_chans after the
	  cb_event calls, this avoids that a release_complete from a quite
	  different call releases a fresh created setup by accident.
	  ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
	  Mai 2007) | 1 line fixed the problem that misdn_write did not
	  return -1 when called with 0 samples in a frame this resultet in
	  a deadlock in some circumstances, when the call ended because of
	  a busy extension. added encoding of keypad. ........

2007-05-03 13:54 +0000 [r62883]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test18 (added),
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ael-test18/extensions.ael (added),
	  pbx/ael/ael-test/ael-test18 (added),
	  pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
	  pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
	  These mods fix bug 9623, where an '@' in the eswitch contents
	  causes a syntax error. I also updated the regressions.

2007-05-03 00:23 +0000 [r62797-62842]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
	  May 2007) | 2 lines doh... initializing the pointer variable will
	  work just a bit better ........

	* res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
	  May 2007) | 7 lines increase reliability and efficiency of static
	  Realtime config loading via ODBC: don't request fields we aren't
	  going to use don't request sorting on fields that are pointless
	  to sort on explicitly request the fields we want, because we
	  can't expect the database to always return them in the order they
	  were created (reported by blitzrage in person (!), patch by me)
	  ........

	* res/res_config_pgsql.c: improve static Realtime config loading
	  from PostgreSQL: don't request sorting on fields that are
	  pointless to sort on use ast_build_string() instead of snprintf()
	  don't request the list of fieldnames that resulted from the query
	  when we both knew what they were before we ran the query _AND_ we
	  aren't going to do anything with them anyway (patch by me,
	  inspired by blitzrage's bug report about res_config_odbc)

2007-05-02 22:59 +0000 [r62739-62789]  Russell Bryant <russell@digium.com>

	* main/channel.c: Merge changes from team/russell/inband_dtmf ...
	  Fix some issues related to generating inband DTMF. There are two
	  changes here: 1) The list of DTMF tones in the senddigit_begin()
	  function explicitly specified 100ms of the tone followed by 100ms
	  of silence. This really broke things with the way that Asterisk
	  now wants complete control over when the digit begins and ends.
	  So, regardless of what Asterisk really wanted to do, this was
	  going to play out the tone at the length it wanted to. This
	  caused various problems like DTMF translation to inband to be
	  extremely unreliable. The list of tones has been changed so that
	  the correct DTMF tone is played indefinitely until Asterisk tells
	  it to stop. 2) ast_write() had to be modified to let a DTMF_END
	  frame get processed even when a generator is present. This is how
	  the tone will finally get stopped. (issues #8944, #9250, #9348,
	  maybe others. Thanks to mdu113 from #8944 for the testing and
	  feedback!)

	* main/manager.c: Backport the change that only went in to trunk
	  that fixes the command manager action over http. (reported
	  internally by pari and bkruse)

2007-05-02 20:46 +0000 [r62738]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
	  2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
	  in 'h' extension louses up the dst field ........

2007-05-02 17:43 +0000 [r62692]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
	  | 4 lines Issue 9638 - if a text frame is sent with no
	  terminating NULL through a bridged IAX connection, the remote end
	  will receive garbage characters tacked onto the end. ........

2007-05-02 17:10 +0000 [r62689]  Steve Murphy <murf@digium.com>

	* configs/extensions.conf.sample, main/channel.c, main/pbx.c,
	  channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
	  clid, src fields in channel_alloc call. b)in the channel_alloc
	  func, set the cid_num and name fields from the arglist[blush]. c)
	  don't update the channel app & app data fields if you are in the
	  'h' extension. d)the load_module func in cdr_radius needs to
	  return DECLINE, SUCCESS.

2007-05-02 06:15 +0000 [r62624]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't unlock a channel that we already know
	  does not exist (propably isue 8228)

2007-05-01 21:57 +0000 [r62548]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 62547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
	  4 lines Remove an unnecessary check that makes it so if you hang
	  up after doing an attended transfer before the target extension
	  answers the channel, the transfer is not successful. (issue
	  #9338, patch by svanlund) ........

2007-05-01 21:34 +0000 [r62545]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
	  (found by rayjay, different fixes by me)

2007-05-01 16:26 +0000 [r62497]  Russell Bryant <russell@digium.com>

	* /, configs/indications.conf.sample: Merged revisions 62496 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
	  3 lines Add indications.conf information for the Philippines.
	  (issue #9525, reported and patched by loloski) ........

2007-04-30 15:58 +0000 [r62414-62419]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
	  4 lines This patch fixes an issue where depending on the cause
	  code, when the network sends a PRI disconnect, the call may not
	  be properly hung up. (issue #9588, reported and patched by
	  softins) ........

	* include/asterisk/http.h, main/http.c: When serving dynamic
	  content, include a Cache-Control header to instruct the browsers
	  to not store the resulting content. (issue #9621, reported by
	  Pari, patch by me)

2007-04-30 14:52 +0000 [r62371]  Jason Parker <jparker@digium.com>

	* configs/iax.conf.sample: Remove unused (and potentially
	  confusing) jitterbuffer options from sample config.

2007-04-30 14:36 +0000 [r62369]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c, /: Merged revisions 62368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
	  lines Update copyright notice. It's now the year 2007! ........

2007-04-29 05:50 +0000 [r62299-62331]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a bug that made the "language" setting
	  in zapata.conf not functional. (issue #9626, reported and fixed
	  by sergee)

	* apps/app_meetme.c: Note that the "talker optimization" option
	  will be enabled by default in 1.6

2007-04-27  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.4 released.

2007-04-27 21:10 +0000 [r62218]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Fix a weird problem where when a caller
	  talking to someone sitting behind an agent channel sent a digit,
	  the digit would be played to the agent for forever. This is
	  because chan_agent always returned -1 from its send_digit_begin
	  and _end callbacks. This non-zero return value indicates to the
	  Asterisk core that it would like an inband DTMF generator put on
	  the channel. However, this is the wrong thing to do. It should
	  *always* return 0, instead. When the digit begin and end
	  functions are called on the proxied channel, the underlying
	  channel will indicate whether inband DTMF is needed or not, and
	  the generator will be put on that one, and not the Agent channel.
	  (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
	  by me)

2007-04-27 16:17 +0000 [r62174]  Jason Parker <jparker@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
	  lines This transcoder message needn't be a NOTICE. I've seen it
	  cause confusion more than a few times. ........

2007-04-27 16:14 +0000 [r62171]  Russell Bryant <russell@digium.com>

	* main/pbx.c: If no variables were passed into
	  pbx_substitute_variables_helper_full(), then don't even bother
	  creating a temporary bogus channel, since that is only for
	  allowing certain functions to operate on the variables as if they
	  were on a channel. Most importantly, this fixes a crash. (issue
	  #9613, reported by callguy, fixed by me)

2007-04-27 14:04 +0000 [r62095-62137]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
	  lines Issue #7351 - SIP Cancel fails due to the wrong contact
	  uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
	  - THANKS!!!! THis was a hard one to catch. ........

	* channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
	  annoying DEBUG messages not controlled by option_debug (DEA).
	  Thanks!

2007-04-26 16:33 +0000 [r61959-62038]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
	  lines Revert previous fix for when the IAX2 channel goes funky
	  (that's the technical term). This is causing legit calls to be
	  prematurely hung up. (issue #9600 reported by justdave) ........

	* main/channel.c: Missed an ast_app_group_discard during merge.
	  Thanks blitzrage!

	* res/res_monitor.c: Don't always say that the channel is being
	  paused if it is actually being unpaused in the Manager ack
	  message. (reported by jsmith in #asterisk-bugs)

	* main/config.c, /: Merged revisions 61958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
	  lines Don't count failed include attempts against the
	  configuration include level. (issue #9593 reported by mostyn)
	  ........

2007-04-25 22:29 +0000 [r61914]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
	  | 2 lines handle a very bizarre race condition with channels
	  being redirected before a simple switch can be started on them
	  (issue #9286) ........

2007-04-25 21:59 +0000 [r61863-61870]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
	  2 lines If the callerid= option is specified, but empty, clear
	  any previous data. ........

	* /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
	  2 lines Ensure that callerid settings are reset on a reload.
	  ........

2007-04-25 19:21 +0000 [r61805]  Joshua Colp <jcolp@digium.com>

	* main/cli.c, main/channel.c, include/asterisk/app.h,
	  funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
	  lines Merge rewritten group counting support. No more storing
	  data on the variable list of the channels. That was bad, mmmk?
	  (issue #7497 reported by sabbathbh) ........

2007-04-25 16:22 +0000 [r61799]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
	  3 lines Fix a typo where cid_num got copied instead of cid_ani.
	  (issue #9587, reported and patched by xrg) ........

2007-04-24  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.3 released.

2007-04-24 21:34 +0000 [r61781-61787]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 61786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
	  4 lines Don't crash if a manager connection provides a username
	  that exists in manager.conf but does not have a password, and
	  also requests MD5 authentication. (ASA-2007-012) ........

	* main/channel.c, include/asterisk/channel.h: Improve DTMF handling
	  in ast_read() even more in response to a discussion on the
	  asterisk-dev mailing list. I changed the enforced minimum length
	  of a digit from 100ms to 80ms. Furthermore, I made it now enforce
	  a gap of 45ms in between digits. These values are not
	  configurable in a configuration file right now, but they can be
	  easily changed near the top of main/channel.c.

2007-04-24 18:43 +0000 [r61779]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
	  | 1 line removed #if 0 block from chan_phone, chan_zap, and
	  chan_modem restart_monitor() ........

2007-04-24 16:16 +0000 [r61774]  Russell Bryant <russell@digium.com>

	* main/dial.c: Add a few more state changes in
	  handle_frame_ownerless() so that the SLA code will get notified
	  of these changes even when an owner channel is not provided. This
	  isn't from a specific bug report, it's just something I noticed
	  while poking around.

2007-04-24 16:07 +0000 [r61772]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
	  lines Allow RFC2833 to be sent in the response SDP when an INVITE
	  comes in without SDP. (issue #9546 reported by mcrawford)
	  ........

2007-04-23 18:17 +0000 [r61763-61765]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Some dialplan functions, such as CUT(), expect to
	  operate on variables on a channel. So, this little hack lets them
	  work in places where a channel doesn't exist, such as within
	  DUNDi configuration. (issue #9465, reported and patched by
	  Corydon76, testing by blitzrage)

	* main/channel.c: Ensure that digits passing through Asterisk have
	  a reasonable minimum length. It is currently 100 ms. If someone
	  thinks this should be different, feel free to speak up. (related
	  to issues #8944, #9250, and #9348)

2007-04-20 21:35 +0000 [r61705-61707]  Jason Parker <jparker@digium.com>

	* main/rtp.c: Avoid invalid seqno cycling detection. Per comment
	  from Dave Troy: This adds back in some simple typecasting I had
	  in an earlier version which I realize now may be breaking things.
	  Issue #9554.

	* main/loader.c, /: Merged revisions 61704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
	  lines Fix an issue that I noticed while looking over issue 9571.
	  The reload timestamp was getting set after reloading the built-in
	  stuff, and before the modules. ........

2007-04-20 20:42 +0000 [r61697]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Remove a stray debug message introduced by a recent
	  commit.

2007-04-20 19:51 +0000 [r61694]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
	  lines If the '* to hangup' option is not enabled, we don't need
	  to disable * as a valid exit key. If it was enabled, this
	  statement would've never been checked in the first place. Issue
	  #9552 ........

2007-04-20 18:19 +0000 [r61690]  Russell Bryant <russell@digium.com>

	* main/config.c, apps/app_voicemail.c, main/manager.c,
	  include/asterisk/config.h: Fix the UpdateConfig manager action to
	  properly treat "variables" and "objects" differently (a=b versus
	  a=>b). (issue #9568, reported by pari, patch by me)

2007-04-19 08:37 +0000 [r61686]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
	  lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
	  Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........

2007-04-19 04:36 +0000 [r61681-61683]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/manager.c: Bug 9557 - simple reason why reading a function
	  always returned NULL

	* funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
	  funcs/func_groupcount.c, /, funcs/func_timeout.c,
	  funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
	  | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
	  Channel parameter can cause Asterisk to crash. The reason this
	  needs to be fixed in the functions instead of in AMI is because
	  Channel can legitimately be NULL, such as when retrieving global
	  variables. ........

2007-04-18 22:10 +0000 [r61678]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: allow external build systems to extract the
	  required sound file versions

2007-04-18 20:46 +0000 [r61674-61676]  Olle Johansson <oej@edvina.net>

	* main/rtp.c: Clean upp formatting, add some doxygen stuff while
	  we're in cleaning mode... Thanks Kevin!

	* main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)

2007-04-16 14:47 +0000 [r61664-61666]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: #9483, half of patch by twilson to solve 302
	  redirect issues

	* /: Blocking AstHoloPatch from 1.2

2007-04-13 21:17 +0000 [r61658]  Steve Murphy <murf@digium.com>

	* main/cdr.c: This is a fix to the way CDR merge handles the data
	  that results from ForkCDR.

2007-04-13 19:17 +0000 [r61648-61656]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
	  lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
	  the same as OUTBOUND_GROUP except it will get unset after use so
	  it won't get accidentally inherited. (issue #BE-140) ........

	* apps/app_speech_utils.c: Do not bother looking for a result if
	  none are present.

	* channels/chan_sip.c: For those very verbose SIP implementations
	  that attach tons of info to the Contact header... let's increase
	  our variable sizes. (issue #9535 reported by jeffg)

2007-04-13 17:10 +0000 [r61645]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Eliminate a compiler warning with
	  ODBC_STORAGE enabled so that it will build under dev-mode.

2007-04-13 17:01 +0000 [r61644]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c: A fix for chan_oss that resulted from the
	  CDR changes; it helps to use the right info.

2007-04-13 16:32 +0000 [r61641]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't assume the callid of a dialog will be
	  set, as in some circumstances it may not. (issue #9534 reported
	  by tecnoxarxa)

2007-04-11 16:05 +0000 [r61477]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
	  5 lines If someone sets the "useragent" option in sip.conf to be
	  empty, then don't add the User-Agent header at all. It is an
	  optional header, anyway. Also, the bug report says that some of
	  Japan's SIP providers don't allow it for some weird reason.
	  (issue #9488, reported by makoto, fixed by me) ........

2007-04-11 15:39 +0000 [r61443]  Nadi Sarrar <ns@beronet.com>

	* channels/chan_misdn.c: Don't export AOCD variables on
	  misdn_hangup anymore, this was mainly a fix for trunk..

2007-04-11 15:09 +0000 [r61377-61427]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
	  6 lines Fix a bug with switching between host=dynamic and using
	  specific hosts for peers. The code would only reset the peer's
	  address when it is dynamic if it was a new peer structure. Now,
	  it will also reset the address if it was already in the peer
	  list, but before the reload, it was not dynamic. (issue #9515,
	  reported by caio1982, fixed by me) ........

	* main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
	  bkruse) In passing, constify the elements of the mimetypes table.

	* /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
	  5 lines Remove the attempt at reporting configuration errors in
	  sip.conf. This can cause a bunch of improper messages when using
	  realtime. I give up. As oej tried to convince me when I put this
	  in, there is just no easy way to do it. (inspired by a message on
	  the -dev list) ........

2007-04-11 13:40 +0000 [r61342-61373]  Nadi Sarrar <ns@beronet.com>

	* channels/chan_misdn.c: Export AOCD variables on misdn_hangup.

	* channels/chan_misdn.c: Ignore facility messages in case we don't
	  have a corresponding channel object.

	* channels/chan_misdn.c: AOCD's are now exported to asterisk
	  channel variables.

2007-04-10 16:05 +0000 [r61220]  Russell Bryant <russell@digium.com>

	* main/Makefile, main/http.c, main/minimime (removed): File upload
	  support was added to solve some needs for the Asterisk GUI.
	  However, after much discussion, it has been decided that adding
	  this to 1.4 is not in the best interests of the project. It has
	  been removed here, but will remain in trunk.

2007-04-10 12:43 +0000 [r61183]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
	  2007) | 2 lines msns config parameter defaults to '*' ........

2007-04-10 05:18 +0000 [r61136]  Steve Murphy <murf@digium.com>

	* apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
	  previous fix to overcome a compiler warning; the app NoCDR() has
	  been updated to mark the channel CDR as POST_DISABLED instead of
	  destroying the CDR; this way its flags are propagated thru a
	  bridge and the CDR is actually dropped. The cases where only one
	  channel in a bridge has a CDR was cleaned up.

2007-04-09 19:58 +0000 [r61072]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
	  lines - Don't send ActionID before Response: header. - Don't use
	  a blank in an AMI header ........

2007-04-09 19:55 +0000 [r61062-61070]  Kevin P. Fleming <kpfleming@digium.com>

	* main/minimime/mm_envelope.c, res/res_features.c: fix up some
	  warnings found using --enable-dev-mode

	* main/minimime/Doxyfile (removed),
	  main/minimime/tests/messages/CVS (removed),
	  main/minimime/tests/CVS (removed): remove some more stuff we
	  don't need

2007-04-09 19:41 +0000 [r61042-61044]  Russell Bryant <russell@digium.com>

	* main/minimime/test (removed): Remove another directory that
	  should no longer be there

	* main/minimime/Make.conf (removed), main/minimime/mytest_files
	  (removed), main/minimime/.cvsignore (removed), main/minimime/sys
	  (removed), main/minimime/mm-docs (removed): Remove various files
	  that I thought I already removed.

2007-04-09 19:05 +0000 [r61022]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Use the appropriate interface name with
	  COMPLETECALLER. Issue 9395.

2007-04-09 18:32 +0000 [r60989]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c, main/channel.c, main/cdr.c,
	  channels/chan_phone.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, channels/chan_features.c,
	  channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
	  channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
	  channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
	  include/asterisk/channel.h, channels/chan_gtalk.c,
	  channels/chan_iax2.c: This is a big improvement over the current
	  CDR fixes. It may still need refinement, but this won't have as
	  many folks bothered.

2007-04-09 18:02 +0000 [r60984]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: Add final new line after JabberEvent

2007-04-09 17:22 +0000 [r60936]  Jason Parker <jparker@digium.com>

	* /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
	  lines Allow matching on names shorter than 3 chars. This also
	  fixes the case where somebody wants to match on less then 3
	  chars. Issue 9071 ........

2007-04-09 03:01 +0000 [r60847-60850]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
	  | 2 lines Don't check for error when lowering priority (according
	  to the manpage, it should never happen anyway). It might could
	  happen, though, if another thread messed with the priority, so
	  safeguard against that (reported via -dev list). ........

	* channels/chan_local.c, /: Merged revisions 60846 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
	  Apr 2007) | 2 lines Bug 9505 - If the return value for
	  local_queue_frame is set, then p->lock is no longer valid.
	  ........

2007-04-09 01:03 +0000 [r60762-60798]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
	  lines When calling a device that then forwards us elsewhere... we
	  have to make our channels compatible if it is the only channel
	  being dialed. (issue #9445 reported by marcelbarbulescu) ........

	* apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
	  MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)

2007-04-08 14:14 +0000 [r60661-60713]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
	  | 2 lines Gosub called within a Macro resets the arguments
	  improperly and causes general weirdness. (Issue 8329) ........

	* main/http.c: Fix --enable-dev-mode

	* channels/chan_oss.c: Off by one error, resulting in a crash
	  (Issue 9500)

	* /, main/file.c: Merged revisions 60660 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
	  | 2 lines Bug 9486 - memory leak when opening a filestream
	  ........

2007-04-06 20:58 +0000 [r60603]  Russell Bryant <russell@digium.com>

	* main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
	  main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
	  main/manager.c, main/minimime/mm_mimepart.c,
	  main/minimime/test.sh, configure, include/asterisk/compat.h,
	  main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
	  main/minimime/tests/parse.c, main/minimime/mm_base64.c,
	  main/minimime/mm_mimeutil.c, main/minimime/mm.h,
	  main/minimime/tests, main/minimime/mm_header.c,
	  main/minimime/mm_error.c, main/Makefile,
	  main/minimime/mm_codecs.c, main/minimime/mm_param.c,
	  configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
	  include/asterisk/manager.h, main/minimime/strlcpy.c,
	  configs/http.conf.sample, main/minimime/mm_parse.c,
	  main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
	  main/minimime/mm_util.c, main/minimime/mm_envelope.c,
	  main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
	  main/minimime/tests/messages/test2.txt,
	  main/minimime/tests/messages/test3.txt,
	  main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
	  main/minimime/tests/messages/test4.txt,
	  main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
	  main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
	  main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
	  main/minimime/mimeparser.l, main/minimime/mm_context.c,
	  main/minimime/mimeparser.tab.h, main/minimime (added),
	  main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
	  main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
	  main/minimime/mimeparser.y, Makefile.moddir_rules,
	  main/minimime/sys, main/minimime/tests/Makefile: To be able to
	  achieve the things that we would like to achieve with the
	  Asterisk GUI project, we need a fully functional HTTP interface
	  with access to the Asterisk manager interface. One of the things
	  that was intended to be a part of this system, but was never
	  actually implemented, was the ability for the GUI to be able to
	  upload files to Asterisk. So, this commit adds this in the most
	  minimally invasive way that we could come up with. A lot of work
	  on minimime was done by Steve Murphy. He fixed a lot of bugs in
	  the parser, and updated it to be thread-safe. The ability to
	  check permissions of active manager sessions was added by Dwayne
	  Hubbard. Then, hacking this all together and do doing the
	  modifications necessary to the HTTP interface was done by me.

2007-04-06 20:32 +0000 [r60568-60572]  Dwayne M. Hubbard <dhubbard@digium.com>

	* UPGRADE.txt: clarified a sentence in the format_wav section

	* UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
	  plan to remove GAIN code from trunk

2007-04-06 19:50 +0000 [r60521-60565]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: When a station picks up a trunk that was on
	  hold, make the hints reflect that nobody has the trunk on hold
	  anymore.

	* apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
	  reported by francesco_r, fixed by me) * The original behavior was
	  that if one station put a call on hold, another one picked it up,
	  and then hung up, the code would still consider the call on hold
	  by the first station, so the trunk would not be hung up. However,
	  to better comply with what most people seem to expect it to
	  behave, it will now hang up the trunk. * Fix a problem with
	  "barge=no". This was only intended to prevent people from joining
	  calls that are in progress. However, it also prevented other
	  people from picking up a call that was on hold. This has been
	  fixed. * When there are no active stations on a trunk and it is
	  on hold, the code now indicates the HOLD and UNHOLD conditions to
	  the trunk channel. This allows music on hold to be played to the
	  trunk when it is on hold.

2007-04-06 18:21 +0000 [r60459-60485]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we check the faxdetect option
	  before doing fax processing

	* channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
	  lines There should only be one code path for doing DTMF
	  conditionals on channels. This fixes it. ........

2007-04-06 14:49 +0000 [r60399]  Kevin P. Fleming <kpfleming@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
	  | 2 lines remove undocumented 'cardsmode' parameter and stop
	  searching for transcoders during reload() ........

2007-04-06 01:14 +0000 [r60361]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Add support for returning different
	  types of results (ie: NBest).

2007-04-05 22:58 +0000 [r60325]  Dwayne M. Hubbard <dhubbard@digium.com>

	* formats/format_wav.c: modified default GAIN for issue 5823,
	  thanks jrwalliker

2007-04-05 22:35 +0000 [r60323]  Steve Murphy <murf@digium.com>

	* configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
	  some clarification to the example configs for CDRs, on how to
	  select a backend. Also, made cdr-csv the default if you 'make
	  samples', and no other changes.

2007-04-05 16:10 +0000 [r60268]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
	  lines Just because we can't find the voicemail configuration
	  file, doesn't mean that the module failed to load. The user could
	  be using realtime. Issue #9473 ........

2007-04-05 15:47 +0000 [r60265]  Russell Bryant <russell@digium.com>

	* main/http.c: Add the MIME type for gif by request from Pari

2007-04-05 12:55 +0000 [r60214]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
	  lines Only unlock our pvt and net locks if we are actually going
	  to try to lock the owner again. (issue #9472 reported by zoa)
	  ........

2007-04-04 17:40 +0000 [r60013-60137]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 60134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
	  6 lines It is valid to redirect channels via the manager
	  interface that are not in the UP state. Instead of checking for
	  that to prevent to ensure a dead channel doesn't get redirected,
	  just use the ast_check_hangup() API call. (issue #9457, reported
	  by Callmewind, patch by me) (related to issue #8977) ........

	* channels/chan_sip.c: Add a Content-Length of 0 to the response
	  built by transmit_response_with_unsupported(). (issue #9454,
	  reported by makoto, fixed by me)

	* /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
	  4 lines Fix the return value of handle_common_options() so that
	  it always properly indicates whether it handled the option or
	  not. (issue #9455, reported by Netview, fixed by me) ........

	* apps/app_meetme.c: Fix a problem where if a trunk was hung up
	  while it was on hold, all of the hints would reflect the line
	  still on hold, even though it should reflect that it is back to
	  not in use. (issue #9459, reported by francesco_r, fixed by me)

2007-04-03 19:40 +0000 [r59963]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Don't clash when a person both speaks
	  and uses DTMF.

2007-04-03 19:16 +0000 [r59853-59939]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
	  4 lines Don't attempt to report configuration errors in
	  build_user(). oej pointed out that for a "friend" entry, this
	  won't work, because all user options are valid for peers, but not
	  the other way around. ........

	* /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
	  3 lines Make chan_sip report when it encounters an unknown
	  option. (issue #9440, reported by nightcrawler) ........

	* /, main/app.c: Merged revisions 59886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
	  5 lines When doing a built-in blind or attended transfer, restore
	  the ability to use '#' to terminate the number and immediately do
	  the transfer instead of having to dial the number and just wait
	  for the feature digit timeout. (issue #8366, xueliangliang)
	  ........

	* Makefile: Ensure that menuselect gets executed in dependency
	  check mode every time you run make.

2007-04-03 11:02 +0000 [r59804]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
	  Merged revisions 59788,59803 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
	  lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
	  or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
	  03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........

2007-04-03 07:20 +0000 [r59774]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
	  Merged revisions 59623-59624,59639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
	  1 line we can now make 30 channels on a PRI (before we forgot
	  chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
	  (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
	  r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
	  1 line added option which allows us to accept incoming SETUP
	  Messages without automatically sending Proceeding or Setup
	  Acknowledge, this is useful with some broken switches and if you
	  want to Release incoming calls without previously having
	  acknowledged them. The new option is
	  noautorespond_on_setup=yes|no default is no, so we don't break
	  the existing behaviour ........

2007-04-02 18:58 +0000 [r59724]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
	  lines Increase the maximum size for a string of mailboxes to
	  1024. (issue #9270 reported by rtucker) ........

2007-04-02 17:31 +0000 [r59688]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
	  not the test. As per 9435, thanks to marcelbarbulescu

2007-04-02 15:39 +0000 [r59654]  Russell Bryant <russell@digium.com>

	* main/netsock.c, /: Merged revisions 59608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
	  6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
	  This is needed by the patch that went in for issue 7874.
	  chan_iax2 needs to be able to create socket that is lisetning on
	  INADDR_ANY, but also be able to bind sockets to specific
	  addresses. (Thanks to Stevenson on the asterisk-dev mailing list
	  for explaining why this flag was needed.) ........

2007-03-30 22:50 +0000 [r59573]  Jason Parker <jparker@digium.com>

	* configure, main/Makefile, acinclude.m4: Add linux-uclibc host
	  arch..."thingy". Sorry, I don't know what it's called...

2007-03-30 17:51 +0000 [r59452-59522]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
	  include/asterisk/cdr.h: several changes via kpflemings review

	* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
	  include/asterisk/cdr.h: These mods fix CDR issues from 8221,
	  8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
	  from transfer situations.

	* configs/extensions.conf.sample: A small clarification to keep
	  bugs from being filed, and confusion from rising, if
	  clearglobalvars is set, and globals are set in the AEL file.
	  (9419)

2007-03-29 17:43 +0000 [r59363]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: When building a response to a subscription, the
	  "from" must be the full Jabber ID. This fixes some problems where
	  jabber users are not able to add their Asterisk account to their
	  user list, since they are unable to get Asterisk to approve their
	  subscription. (issue #8210, reported by caspy, and verified by
	  bradtem)

2007-03-29 17:38 +0000 [r59361]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
	  lines Keep a global array of variables indicating whether certain
	  conference rooms are in use. This ensures that two people going
	  into a new dynamic conference when the 'e' option is set don't go
	  into the same conference room. (issue #8835 reported by eliel)
	  ........

2007-03-29 17:17 +0000 [r59304-59358]  Russell Bryant <russell@digium.com>

	* main/rtp.c, /: Merged revisions 59357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
	  5 lines If an error occurs when reading from an RTP socket, and
	  the error code does not indicate that we should try again, then
	  return NULL instead of a "null frame". This will prevent Asterisk
	  from trying over and over again, and eventually causing the
	  system to crash. (issue #8285, john) ........

	* channels/chan_iax2.c: When the IAX2 read callback gets called,
	  return NULL instead of a "null frame". This will cause Asterisk
	  to hangup the call instead of keep trying whatever it was doing.
	  Under normal conditions, this function would *never* be called.
	  However, the author of this patch says an error will occur that
	  will cause it to get called every 100 thousand calls or so. When
	  this does happen, it puts the channel in a loop that eventually
	  brings down the system. So, hangup up the call is certainly a
	  better alternative. (issue #8286, john)

	* Makefile: Export the GTK2 library and include information to sub
	  Makefiles.

2007-03-29 16:07 +0000 [r59300-59302]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
	  | 3 lines Issue 9415 - No point to getting a diagnostic field if
	  we aren't doing anything with the information. (Plus, it tends to
	  crash the Postgres ODBC driver.) ........

2007-03-28 03:38 +0000 [r59281-59289]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Another crash that I thought we had fixed already
	  - Issue 9396

	* apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
	  | 2 lines Oops ........

	* apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
	  | 2 lines Fix a few remaining bad mmap(2) return values ........

2007-03-27 23:20 +0000 [r59262-59278]  Russell Bryant <russell@digium.com>

	* /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
	  3 lines Fix the check of the return value from mmap(). Thanks to
	  Corydon for catching this one. ........

	* apps/app_directory.c: Fix app_directory to actually compile with
	  ODBC_STORAGE, and update the code to the latest res_odbc API.

	* apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
	  The Makefile did not properly ensure that this information got
	  copied from what was selected for app_voicemail. (issue #9224)

	* channels/chan_sip.c: Fix the check that ensures that the CHANNEL
	  function's first argument is "rtpqos". Thanks, Corydon. :)

2007-03-27 18:16 +0000 [r59261]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
	  asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
	  this case the proper thing to do. This change now has it doing
	  the proper thing.

2007-03-27 18:05 +0000 [r59256-59259]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
	  4 lines Fix the use of the "sourceaddress" option when "bindaddr"
	  is set to 0.0.0.0 instead of having each interface explicitly
	  listed. (issue #7874, patch by stevens) ........

	* channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
	  function to just be additional parameter of the CHANNEL function.
	  This way, it will be possible for other RTP based channel drivers
	  to expose this information in the future.

2007-03-27 15:00 +0000 [r59254]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
	  Mär 2007) | 1 line fixed #9355 ........

2007-03-26 21:45 +0000 [r59230]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Oops, this should be case insensitive

2007-03-26 21:41 +0000 [r59228]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
	  asterisk). I turned a duplicate context from a WARNING to an
	  ERROR. Now you get a module load failure, and asterisk just
	  exits. That's better than a crash, right\?

2007-03-26 21:37 +0000 [r59227]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Change this to a single dp function to make
	  oej happy.

2007-03-26 20:06 +0000 [r59225]  Steve Murphy <murf@digium.com>

	* main/config.c: Fix for 9257; by eliminating the globals in
	  main/config.c, we make it thread-safe, which is a minimum
	  requirement.

2007-03-26 19:34 +0000 [r59223]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Add ability to specify no timeout. This
	  means as soon as the prompt is done playing it moves on to the
	  next priority.

2007-03-26 18:33 +0000 [r59215-59217]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Somehow the code for building the email for
	  voicemail got out of sync. This change makes a few tweaks to get
	  1.4 in sync with trunk. (issue #9301)

	* apps/app_meetme.c: Fix some codec negotiation problems when
	  CallerID support is not enabled in SLA. (issue #9308, reported by
	  twilson)

2007-03-26 18:13 +0000 [r59213]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Make SpeechBackground obey the digit
	  timeout value.

2007-03-26 17:53 +0000 [r59207-59209]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Rename the new dialplan functions to match
	  the variable name

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
	  AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
	  some because they get set in sip_hangup. So, there are common
	  situations where the variables will not be available in the
	  dialplan at all. So, this patch provides an alternate method for
	  getting to this information by introducing AUDIORTPQOS and
	  VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
	  with some testing by blitzrage)

2007-03-26 17:38 +0000 [r59206]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
	  pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
	  and STANDALONE_AEL

2007-03-26 15:25 +0000 [r59202]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
	  include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
	  provides a dsp pipeline for i.e. echo cancellation modules, make
	  chan_misdn use it. * add a check for linux/mISDNdsp.h to
	  configure.ac and update the autogenerated files: 'configure',
	  'autoconfig.h.in' (the 'configure' script was not in sync with
	  the latest configure.ac, so the diff is a bit bigger than
	  expected).

2007-03-26 15:16 +0000 [r59200]  Joshua Colp <jcolp@digium.com>

	* pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
	  aelparse binary! DONT_OPTIMIZE should now work once again.

2007-03-24 01:39 +0000 [r59195]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
	  lines Only try to handle a response if it has a response code.
	  (ASA-2007-011) ........

2007-03-23 16:11 +0000 [r59188-59189]  Steve Murphy <murf@digium.com>

	* /: blocking out the fix in 59187... already incorporated here

	* /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
	  line Added a few words in the Macro doc strings about the
	  behavior of macros with hangups (et al.), as per 9337 ........

2007-03-22 23:40 +0000 [r59180-59182]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't allow string input to overrun the
	  buffer to hold it (ASA-2007-010)

	* channels/chan_misdn.c: remove variables that are no longer used
	  (--enable-dev-mode is good, developers should be using it)

2007-03-22 14:40 +0000 [r59145]  Steve Murphy <murf@digium.com>

	* utils/Makefile: The stuff in utils was compiling with -O6 even if
	  DONT_OPTIMIZE is set in menuconfig. Added the include to fix that

2007-03-21 18:08 +0000 [r59081-59089]  Joshua Colp <jcolp@digium.com>

	* main/http.c: Add svg mimetype for pari.

	* res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
	  lines Indicate the filename changed when it is changed. (issue
	  #9311 reported by jsmith) ........

	* channels/chan_sip.c: Until we can do media level parsing for
	  sendrecv/etc just use the first value found. This crept up when a
	  phone was offered audio+video and returned an inactive video
	  stream. chan_sip thought the phone said to put the person on hold
	  but that was totally wrong. (issue #9319 reported by benbrown)

2007-03-20 21:04 +0000 [r59078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/logger.c: Fix defines for inline stack backtraces (only used
	  by developers anyway)

2007-03-20 20:42 +0000 [r59076]  Joshua Colp <jcolp@digium.com>

	* channels/iax2-parser.c: Copy len variable as well, should fix
	  remaining IAX2 DTMF issues.

2007-03-20 17:48 +0000 [r59069-59070]  Steve Murphy <murf@digium.com>

	* apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
	  return it to its previous, untouched, state.

	* apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
	  The fix for the AEL <<security hole>> (bug 9316) is here...

2007-03-20 13:16 +0000 [r59064]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h: Merged revisions
	  58849-58850,59062-59063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
	  1 line added method standard_dec for dialing out on groups, to
	  avoid conflicts, which caused issues with some ISDN providers
	  ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
	  Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
	  crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
	  avoid sending a disconnect when we already received one. ........
	  r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
	  1 line modified a loglevel ........

2007-03-19  Jason Parker  <jparker@digium.com>

	* Asterisk 1.4.2 released.

2007-03-19 22:29 +0000 [r59049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Oops, this should have been a %d all along

2007-03-19 15:52 +0000 [r59042]  Joshua Colp <jcolp@digium.com>

	* funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
	  reported by ajohnson)

2007-03-19 15:42 +0000 [r59040]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
	  via -dev list)

2007-03-18 20:37 +0000 [r59037]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
	  code 0 (reported by qwerty1979)

2007-03-18 16:36 +0000 [r59035]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Don't return a non-zero return code if the
	  profile doesn't exist, to match what the documentation says it
	  already does. (#9307 Reported by kkiely)

2007-03-16 16:12 +0000 [r58992]  Joshua Colp <jcolp@digium.com>

	* apps/app_page.c: Wait for the async thread to exit when hanging
	  up all of the paged phones under all circumstances. (issue #9181
	  reported by PhilSmith)

2007-03-16 01:42 +0000 [r58947-58957]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample: fix a couple SLA documentation
	  references

	* doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
	  (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
	  doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
	  doc/channelvariables.txt (added), doc/ael.txt (added),
	  doc/billing.tex (removed), build_tools/prep_tarball,
	  doc/callingpres.txt (added), doc/enum.txt (added),
	  doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
	  doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
	  doc/security.txt (added), doc/imapstorage.txt (added),
	  doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
	  doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
	  doc/iax.txt (added), doc/ael.tex (removed),
	  doc/channelvariables.tex (removed), doc/enum.tex (removed),
	  doc/security.tex (removed), doc/math.txt (added), Makefile,
	  doc/imapstorage.tex (removed), doc/privacy.tex (removed),
	  doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
	  (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
	  doc/chaniax.txt (added), doc/app-sms.txt (added),
	  doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
	  doc/ices.txt (added), doc/dundi.tex (removed),
	  doc/linkedlists.txt (added), doc/queuelog.txt (added),
	  doc/extconfig.txt (added), doc/radius.txt (added),
	  doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
	  doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
	  (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
	  doc/queuelog.tex (removed), doc/configuration.txt (added),
	  doc/asterisk-conf.txt (added), doc/sla.pdf (added),
	  doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
	  (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
	  doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
	  doc/channels.txt (added), doc/ip-tos.tex (removed),
	  doc/extensions.txt (added), doc/queues-with-callback-members.txt
	  (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
	  doc/misdn.txt (added), doc/manager.txt (added),
	  doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
	  doc/billing.txt (added), doc/localchannel.txt (added),
	  doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
	  (added), doc/00README.1st (added): Making these documentation
	  changes in the 1.4 branch upset various people, so these chanes
	  will only be done in the trunk.

	* build_tools/prep_tarball: Add the --pdf option to the usage of
	  rubber in prep_tarball

	* Makefile, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  configure script checking for GTK2 and some additional Makefile
	  targets to support gmenuselect

2007-03-15 23:52 +0000 [r58946]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
	  common syntax and update the resulting appdocs TeX file

2007-03-15 23:24 +0000 [r58941]  Russell Bryant <russell@digium.com>

	* doc/asterisk.tex: add a link to the rubber homepage

2007-03-15 23:11 +0000 [r58939]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_setcdruserfield.c, main/pbx.c,
	  apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
	  Expand deprecation warnings from simply warning on use to the
	  builtin documentation.

2007-03-15 22:51 +0000 [r58935-58937]  Russell Bryant <russell@digium.com>

	* doc/asterisk.tex, Makefile: Add Asterisk version information to
	  the generated PDF

	* build_tools/prep_tarball: have prep_tarball attempt to build
	  asterisk.pdf

2007-03-15 22:32 +0000 [r58933]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_realtime.c: Function works fine, but the documentation
	  is backwards.

2007-03-15 22:25 +0000 [r58931]  Russell Bryant <russell@digium.com>

	* doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
	  (added), doc/freetds.txt (removed), doc/odbcstorage.txt
	  (removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
	  doc/model.txt (removed), doc/channelvariables.txt (removed),
	  doc/ael.txt (removed), doc/billing.tex (added),
	  doc/callingpres.txt (removed), doc/enum.txt (removed),
	  doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
	  doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
	  doc/security.txt (removed), doc/imapstorage.txt (removed),
	  doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
	  doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
	  doc/iax.txt (removed), doc/ael.tex (added),
	  doc/channelvariables.tex (added), doc/enum.tex (added),
	  doc/security.tex (added), doc/math.txt (removed), Makefile,
	  doc/imapstorage.tex (added), doc/privacy.tex (added),
	  doc/realtime.txt (removed), doc/dundi.txt (removed),
	  doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
	  (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
	  doc/ast_appdocs.tex (added), doc/realtime.tex (added),
	  doc/ices.txt (removed), doc/dundi.tex (added),
	  doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
	  doc/extconfig.txt (removed), doc/radius.txt (removed),
	  doc/cliprompt.tex (added), doc/chaniax.tex (added),
	  doc/hardware.txt (removed), doc/mp3.txt (removed),
	  doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
	  (added), doc/queuelog.tex (added), doc/configuration.txt
	  (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
	  (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
	  doc/h323.txt (removed), doc/mp3.tex (added),
	  doc/configuration.tex (added), doc/asterisk-conf.tex (added),
	  doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
	  doc/ip-tos.tex (added), doc/extensions.txt (removed),
	  doc/queues-with-callback-members.txt (removed), doc/apps.txt
	  (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
	  (removed), doc/manager.txt (removed), doc/jitterbuffer.tex
	  (added), doc/extensions.tex (added), doc/billing.txt (removed),
	  doc/localchannel.txt (removed),
	  doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
	  (removed), doc/00README.1st (removed): Merge changes from
	  svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
	  directory into a single LaTeX formatted document so that we can
	  generate a PDF, HTML, or other formats from this information. *
	  Add a CLI command to dump the application documentation into
	  LaTeX format which will only be include if the configure script
	  is run with --enable-dev-mode. * The PDF turned out to be close
	  to 1 MB, so it is not included. However, you can simply run "make
	  asterisk.pdf" to generate it yourself. We may include it in
	  release tarballs or have automatically generated ones on the web
	  site, but that has yet to be decided.

2007-03-15 18:13 +0000 [r58923]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Don't assume that the pvt structure will
	  still exist after calling schedule_delivery as it may not. (issue
	  #9278 reported by fmachado)

2007-03-14 19:18 +0000 [r58894-58906]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Some people like to put "limitonpeer"
	  instead of "limitonpeers" in their configuration. While we're at
	  it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
	  issue #9172)

	* doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
	  examples section

	* doc/security.txt, /: Merged revisions 58896 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
	  3 lines Add a note to the security file that the Asterisk CLI and
	  log files may contain sensitive information, and that people
	  should keep this in mind. ........

	* configs/sla.conf.sample, apps/app_meetme.c: By default, don't
	  attempt to do any CallerID handling at all with SLA because it is
	  known to not work properly in some situations. However, add an
	  option to enable it for those that would like to use it anyway.
	  The short story behind this is that to properly handle CallerID
	  with SLA, we need the ability to change the CallerID on an
	  existing call, and we are not ready to handle that.

2007-03-14 01:47 +0000 [r58880]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Issue 9162 -
	  pbx_substitute_variables_helper assumes the buffer is initialized
	  to all zeroes. This fixes a case where it wasn't.

2007-03-13 23:19 +0000 [r58870-58872]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Ensure that the blinky lights show that the
	  trunk stopped ringing when the trunk hangs up before a station
	  has answered it. (issue #9234, reported by francesco_r)

	* configs/sla.conf.sample: fix the reference to the SLA
	  documentation

2007-03-13 11:49 +0000 [r58843-58848]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
	  lines Issue #9229 - No port in request URI on register to non
	  default SIP ports (neelakantan) ........

	* channels/chan_sip.c: Don't hangup the call on OK or errors on
	  MESSAGE and INFO inside of a dialog (like video update requests).

	* channels/chan_sip.c: Issue #9251 - Clear From URI from user
	  attributes (tgrman)

2007-03-12 13:08 +0000 [r58825-58826]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 57034,57523,57753,58558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
	  1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
	  bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
	  19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
	  r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
	  1 line fixed another place where the out_cause was hardcoded to
	  16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
	  Mar 2007) | 1 line we can free channel 31 as well, since we can
	  occupy it ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: added UU transceiving and
	  corect handling for rdnis

2007-03-12 01:21 +0000 [r58779-58783]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Allow RFC2833 compensation to compensate for even
	  stupider implementations by queueing up the end frame at the
	  start, not the actual end. (issue #8963 reported by AndrewZ)

	* channels/chan_sip.c, configs/sip.conf.sample: Add
	  matchexterniplocally setting which only substitutes your
	  externip/externhost setting if it matches the localnet setting. I
	  know of at least two people who need opposite settings, so I made
	  it an option! (issue #8821 reported by kokoskarokoska)

2007-03-10 18:11 +0000 [r58638-58705]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a few more places in chan_iax2 where
	  the ast_frame used for receiving a frame was not properly
	  initialized. - Interpolating a frame when the jitterbuffer is in
	  use - decrypting a frame when IAX2 encryption is on - frames in
	  an IAX2 trunk

	* apps/app_meetme.c: Make the compiler happy and initialize a
	  variable.

	* doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
	  Merge some updates to the SLA documentation. I plan to keep
	  working on this to explain all of the expected behavior with call
	  handling, configuration details for specific phones, and other
	  things. However, I got tired of doing it in plain text, so I
	  switched to using LaTeX. I have included the PDF version. I
	  haven't been able to get a nice looking plain text version out of
	  it yet, but I'm not terribly concerned since this is supposed to
	  be more of the manual, while the plain text sample configuration
	  file is the reference.

2007-03-09 21:08 +0000 [r58584-58604]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Fix spelling of unavailable in voicemail
	  documentation. (issue #9248 reported by tensai)

	* /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
	  lines If we are unable to lookup the host in a c line we have to
	  abort, otherwise the previous data is gone and we will
	  (potentially) have no data when all is said and done. ........

2007-03-08 22:15 +0000 [r58510-58512]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Hang up the channel that put the call on hold
	  in the event processing thread to avoid a race condition. Also,
	  if the station originated the call that it is putting on hold,
	  don't hang up the trunk if it was the only station on the call
	  and it is hanging up due to hold and not a normal hangup.

	* channels/chan_zap.c: Add a missing break statement so that
	  handling the above event does not incorrectly destroy the
	  channel. (issue #9242, andrew)

2007-03-08 21:33 +0000 [r58479]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Fix segfault (Issue 9236)

2007-03-08 20:54 +0000 [r58474]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Refactor hold handling a bit so that it does
	  not require keeping the call up when a call is put on hold.

2007-03-08 18:01 +0000 [r58389-58436]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Make early SDP seeding even smarter! We have to check
	  codecs in the make_compatible function too. (issue #9221 reported
	  by marcelbarbulescu)

	* main/dsp.c, /: Merged revisions 58388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
	  lines Only print out debug message if the definition that makes
	  the variables shows up was actually defined. (issue #9233
	  reported by serginuez) ........

2007-03-08 13:23 +0000 [r58351-58354]  Kevin P. Fleming <kpfleming@digium.com>

	* main/http.c: this change was not needed; fclose() handles closing
	  the file descriptor already

	* apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
	  value

	* main/http.c: fix two cases where HTTP session file descriptors
	  would not be closed

2007-03-08 01:01 +0000 [r58243-58320]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, configure, configure.ac: If we receive
	  ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
	  tzafrir) Also, update the configure script to make sure that we
	  don't try to build chan_zap if the installed version of zaptel
	  does not include ZT_EVENT_REMOVED.

	* /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
	  Moore) Merged revisions 58242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
	  7 lines Fix a problem where the Asterisk channel name could be
	  that of the wrong IAX2 user for a call. This is because the first
	  step of choosing this name is to look for an IAX2 peer that
	  happens to have the same IP/port number that this call is coming
	  from and assuming that is it. However, this is not always
	  correct. So, I have made it change this name after authentication
	  happens since at that point, we have an exact match. ........

2007-03-07 17:52 +0000 [r58240]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
	  at least one matching codec before attempting early bridge SDP
	  seeding. (issue #9221 reported by marcelbarbulescu)

2007-03-07 00:27 +0000 [r58165-58168]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 58164 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
	  4 lines If the channels acquired using the manager Redirect
	  action are not up, then don't attempt to do anything with them.
	  It could lead to weird behavior, including crashes. (issue #8977)
	  ........

2007-03-06 23:10 +0000 [r58121]  Steve Murphy <murf@digium.com>

	* /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
	  line Fix for 9220: Eyebeam cannot renew subscriptions for
	  presence info. Reason: re-SUBSCRIBE requests don't include Accept
	  headers, which the rfc says are optional (to put it tersely), (it
	  uses MAY), and luckily, the sip_pvt struct has the format info
	  stored, so we simply leave it if the format is set, and the
	  accept header null. ........

2007-03-06 23:00 +0000 [r58119]  Russell Bryant <russell@digium.com>

	* configs/voicemail.conf.sample: Clarify the documentation of the
	  dialout and sendvoicemail options. (issue #9000, caio1982 and
	  serge-v)

2007-03-06 20:37 +0000 [r58053]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
	  lines Change error message to proper message ........

2007-03-06 18:01 +0000 [r58023]  Russell Bryant <russell@digium.com>

	* channels/chan_skinny.c: Return an error of transmit_response is
	  called without a session. (issue #9002)

2007-03-05 19:19 +0000 [r57870-57914]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Since chan_iax2 does not support reception
	  of DTMF with duration ensure that it is set to 0 on the frame.
	  (issue #8521 reported by gdhgdh)

	* apps/app_meetme.c: Don't create a listen channel and record the
	  conference unless the option is turned on. (issue #9204 reported
	  by francesco_r)

	* apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
	  lines Make create_dirpath use our standard for return values. -1
	  is failure, 0 is success. (issue #9205 reported by ballares)
	  ........

2007-03-05 15:20 +0000 [r57826]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57825 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
	  line Fixed a typo introduced via 9156 (either the gotos or their
	  doc strings are wrong) ........

2007-03-05 04:19 +0000 [r57768-57798]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Don't allow a NULL pointer to reach
	  ast_frdup. (issue #9155 reported by cmaj)

	* res/res_jabber.c: Don't reference a potentially NULL pointer.
	  (issue #9199 reported by klolik)

	* main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
	  reported by edgreenberg)

2007-03-03 15:31 +0000 [r57707]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
	  pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
	  Updated the regression tests

2007-03-03 06:45 +0000 [r57649]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
	  | 2 lines Memory leak of a list, if call recording was abandoned
	  ........

2007-03-03 00:59 +0000 [r57620]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/say.c: submitted patch for Georgian language, issue 9010,
	  submitted by Alexander Shaduri

2007-03-03 00:02 +0000 [r57591]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample: add missing configuration template.
	  Thanks to Lacy Moore on asterisk-users for pointing this out\!

2007-03-02  Russell Bryant  <russell@digium.com>

	* Asterisk 1.4.1 released.

2007-03-02 23:03 +0000 [r57556]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Update the check that is used to
	  determine whether zaptel transcoder support is present. The
	  interface has changed.

2007-03-02 17:06 +0000 [r57477]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
	  lines If a SIP message comes in and goes to a method handler that
	  requires additional values that may not be present then send back
	  an error. ........

2007-03-02 16:55 +0000 [r57426-57473]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57458 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
	  line further refinement in wording of goto documentation, as per
	  9156, goto not proceeding to next instruction ........

	* pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
	  right, but 9184 points out the problem-- the escape is removed by
	  pbx_config, and pbx_ael should also, before sending it down into
	  the pbx engine. Also, you have to insert it back in, if you are
	  generating extensions.conf code from the AEL.

2007-03-02 00:20 +0000 [r57364-57396]  Russell Bryant <russell@digium.com>

	* main/file.c: Return the correct digit that interrupted the
	  stream. This fixes exiting the Background application when using
	  the m option. (issue #9176, mjagdis)

	* configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
	  include/asterisk/channel.h: Merge changes from
	  svn/asterisk/team/russell/sla_updates * Originally, I put in the
	  documentation that only Zap interfaces would be supported on the
	  trunk side. However, after a discussion with Qwell, we came up
	  with a way to make IP trunks work as well, using some things
	  already in Asterisk. So, here it is, this now officially supports
	  IP trunks. * Update the SLA documentation to reflect how to setup
	  IP trunks. * Add a section in sla.txt that describes how to set
	  up an SLA system with voicemail. * Simplify the way DTMF
	  passthrough is handled in MeetMe. * Fix a bug that exposed itself
	  when using a Local channel on the trunk side in SLA. The
	  station's channel needs to be passed to the dial API when dialing
	  the trunk. * Change a WARNING message to DEBUG in channel.h. This
	  message is of no use to users.

2007-03-01 22:21 +0000 [r57318]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 57317 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
	  2007) | 2 lines Don't even attempt to optimize things when a
	  proxy channel is involved. It will just explode in weird and
	  unexplaineable ways. (issue #9175 reported by
	  clegall_proformatique) ........

2007-03-01 03:02 +0000 [r57263]  TransNexus OSP Development <support@transnexus.com>

	* doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.

2007-02-28 23:01 +0000 [r57144-57207]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
	  docs

	* configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
	  from svn/asterisk/team/russell/sla_updates * Add support for
	  private hold. By setting "hold=private" for a trunk, only the
	  station that put the call on hold will be able to retrieve it
	  from hold. Also, by setting "hold=private" for a station, any
	  call that station puts on hold can only be retrieved by that
	  station.

	* apps/app_meetme.c: Minor formatting change

	* configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
	  svn/asterisk/team/russell/sla_updates * Add support for the
	  "barge=no" option for trunks. If this option is set, then
	  stations will not be able to join in on a call that is on
	  progress on this trunk.

2007-02-28 19:23 +0000 [r57139]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 57118 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
	  line a small documentation update, to reflect reality in the goto
	  doc strings, as per 9156, Goto does not proceed to next prio if
	  jump fails ........

2007-02-28 18:57 +0000 [r57093]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
	  2007) | 2 lines Fix a few more issues with the agent logoff CLI
	  command. (issue #9123 reported by arbrandes) ........

2007-02-28 18:20 +0000 [r57089]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
	  changes from svn/asterisk/team/russell/sla_updates * Add support
	  for station ring delays. Ring delays can be set globally for a
	  station or for specific trunks on the station. * Fix a few bugs
	  in existing code. * Restructure and Reorganize code to improve
	  readability and maintainability. * Improve formatting of the "sla
	  show (trunks|stations)" CLI commands.

2007-02-28 17:55 +0000 [r57053-57055]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Picky compiler...

	* apps/app_speech_utils.c: Better handle timeouts when the
	  individual speaks after everything has been played but before the
	  timeout ends.

2007-02-28 17:15 +0000 [r57049]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: I was surprised that I had not yet downgraded
	  missing goto targets and macro call defs to a warning, in case
	  they are in extensions.conf; I rectified this problem. Also, A
	  goto in a macro to a target in a catch block was not being found;
	  I fixed this too; the cause was that I needed to treat catch
	  statements like an extension in the find_match code.

2007-02-27 17:36 +0000 [r56975]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Fix voicemail email attachments. I missed
	  the conversion of one of the line endings and there was an extra
	  one where it should not have been. (issue #9128)

2007-02-26 22:01 +0000 [r56922]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
	  picky... show deprecation warning in application help, too
	  (reported via list)

2007-02-26 20:42 +0000 [r56888]  Russell Bryant <russell@digium.com>

	* channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
	  if a device was not specified in alsa.conf, then we just use the
	  system default, instead of creating our own default of hw:0,0.
	  (issue #9139)

2007-02-26 20:07 +0000 [r56856]  Joshua Colp <jcolp@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
	  lines Obey the clearglobalvars option in extensions reload (or
	  dialplan reload depending on your version). (issue #9146 reported
	  by ramonpeek) ........

2007-02-26 20:04 +0000 [r56847]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a crash in my last change to
	  iax2_indicate(). (issue #9150)

2007-02-26 19:33 +0000 [r56805-56839]  Joshua Colp <jcolp@digium.com>

	* apps/app_record.c: Update app_record documentation to use new CLI
	  command, core show file formats. (issue #9151 reported by junky)

	* main/pbx.c: Use ast_strlen_zero to see if the language and/or
	  context argument is not present for Background instead of just
	  checking if it is NULL. (issue #9141 reported by mjagdis)

2007-02-26 16:51 +0000 [r56785]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Do more complete locking of the
	  chan_iax2_pvt struct in the indicate callback. (Problem brought
	  up by Ben Smithurst on the asterisk-dev list)

2007-02-26 16:36 +0000 [r56783]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c: Allow both of the show version files and core
	  show file versions CLI commands to work. (issue #9135 reported by
	  mvanbaak)

2007-02-26 01:04 +0000 [r56730-56740]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Move a comment to be in the correct struct.

2007-02-25 14:46 +0000 [r56685]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/channel.c, /: Merged revisions 56684 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
	  | 3 lines Issue 9130 - If prev is the last item on the channel
	  list, then evaluating additional conditions (e.g. name prefix)
	  will cause a NULL dereference. ........

2007-02-24 02:02 +0000 [r56569]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure to set a speeddials parent on
	  creation. Don't crash if hold is pressed when no call is active.
	  Don't return in places that we shouldn't..

2007-02-24 00:53 +0000 [r56548]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: update to match zaptel 1.4 API change that
	  was committed a few minutes ago

2007-02-23 23:24 +0000 [r56505]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, /: Merged revisions 56504 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
	  8 lines Fix up a couple more signal handlers to not do bad things
	  that could cause various undesirable results. The other day, I
	  made Asterisk deadlock by hitting Control-C because of a bad
	  signal handler. Now, signal handlers just set a flag and write to
	  an alert pipe for the flag to be handled. Then, there is another
	  thread that is monitoring for these flags. If being run in
	  console mode, it is just the main thread. If Asterisk is in the
	  background, a thread is created to do it. ........

2007-02-23 21:53 +0000 [r56457]  Joshua Colp <jcolp@digium.com>

	* main/sched.c: Change log notice to debug. It is possible for a
	  scheduled item to execute and be deleted at close to the same
	  time and unavoidable. If this happens this message creeps up.

2007-02-23 20:20 +0000 [r56407]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
	  4 lines Don't destroy mutexes before unregistering all of the
	  entry points from the core. Also, fix a potential memory leak
	  from not destroying the locks for all of the possible call
	  numbers (about 32k of them). ........

2007-02-23 18:59 +0000 [r56372]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_version_h: build special version strings for
	  AADK/S800i builds

2007-02-23 17:58 +0000 [r56341]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: The IMAP storage code uses the same code to
	  build the email that is used when voicemail is sent via email
	  using something like sendmail. In the patch from bug 8033 to fix
	  various IMAP storage problems, the line endings in the email file
	  were changed in the code from "\n" to "\r\n". However, this
	  breaks sending regular voicemail to email. So, this change
	  conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
	  enabled. (issue #9128, patch by jarjarbinks, modified by me to
	  not break IMAP storage)

2007-02-22 23:08 +0000 [r56277]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
	  doc/sla.txt: Merge changes from team/russell/sla_updates. This
	  batch of changes to the SLA code does a few different things. * I
	  made the SLA code event driven instead of having to act in a lot
	  of busy loops while dialing things to wait for state changes.
	  This makes the code more efficient and readable at the same time.
	  * I have implemented a couple of new features. The first is
	  inbound trunk ringing timeouts. This is an option that defines
	  how long to let an incoming call on a trunk to ring. * I have
	  also implemented ring timeouts for stations. They may be
	  specified for the entire station, meaning it is how long to let
	  the station ring before giving up. You can also specify a ring
	  timeout for a specific trunk on a station. So, you can say that
	  you only want a specific station to ring 5 seconds if it is line1
	  ringing, but otherwise, there is no timeout.

2007-02-22 18:49 +0000 [r56231]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
	  lines Only change the original or clone channel if it's the
	  channel behind the proxy channel, not if it's just a regular
	  bridged channel. ........

2007-02-22 14:06 +0000 [r56169]  TransNexus OSP Development <support@transnexus.com>

	* doc/osp.txt: Update OSP documentation for v1.4.

2007-02-22 10:33 +0000 [r56125]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Move message from verbose to debug

2007-02-22 02:39 +0000 [r56094]  Steve Murphy <murf@digium.com>

	* sounds/Makefile: updated the sound tarball versions in Makefile

2007-02-22 01:24 +0000 [r56011-56055]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Restructure a little bit of code to reduce
	  nesting. There is no functionality change here.

	* /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
	  3 lines If we receive a frame that is not in any of the
	  negotiated formats, then drop it. (potentially issue #8781 and
	  SPD-12) ........

2007-02-22 00:35 +0000 [r56008]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: Print out deprecation notice on usage output of CLI
	  commands. (issue #8925 reported by blitzrage)

2007-02-22 00:08 +0000 [r56006]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c: disable unloading of embedded modules... there is
	  a fundamental problem with doing so that will not be fixed in
	  this version of Asterisk due to its invasiveness

2007-02-21 20:35 +0000 [r55957]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
	  lines Change naughty warning message to provide useful
	  information. If a write now fails on a channel in meetme it will
	  tell you the channel name instead of spitting out the wrong error
	  message. ........

2007-02-21 20:27 +0000 [r55954]  Jason Parker <jparker@digium.com>

	* channels/chan_gtalk.c: Fix locking issue, and accept
	  "transport-accept" as a valid accept message. This should solve
	  issues 8970 and 8503.

2007-02-21 20:22 +0000 [r55951]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Simplify the last change to app_meetme, and
	  move the call to dispose_conf() up into the block where we know a
	  conf exists.

2007-02-21 20:16 +0000 [r55914-55949]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Only dispose of the conference if one was
	  created.

	* apps/app_speech_utils.c: Only start playing the next file if we
	  have not been quieted.

	* channels/chan_sip.c: Add a flag that indicates whether a SIP
	  dialog is an outgoing call or not. SIP_OUTGOING originally did it
	  but it was repurposed to the direction of the last transaction,
	  which can cause update_call_counter to falsely decrease the wrong
	  counters. (please don't hurt me oej) (issue #8943 reported by
	  mdu113)

2007-02-21 14:06 +0000 [r55869]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/make_version: Merged revisions 55868 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
	  Feb 2007) | 2 lines use new tag version script ........

2007-02-21 08:32 +0000 [r55834]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
	  after transfer (decrement inuse early on transferer's call leg)

2007-02-21 02:01 +0000 [r55799]  Jason Parker <jparker@digium.com>

	* channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
	  Issue 7764, patch by sailer

2007-02-21 01:03 +0000 [r55751-55758]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Improve the reference counting to fix bugs
	  where people report seeing conferences listed that have no
	  members. (issue #9073)

2007-02-21 00:11 +0000 [r55670-55741]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Better handle dropped IMAP connections.
	  (issue #9054 reported by bsmithurst)

	* channels/chan_sip.c: Return behavior I removed. I did not
	  remember that you could just add a localnet entry to make it
	  work.

	* channels/chan_sip.c: Don't test our own address against the
	  localnet settings. At least one person has had issues as a result
	  of this from #7051 so I'm reversing it. (issue #8821 reported by
	  kokoskarokoska)

	* /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
	  2007) | 2 lines Defer clearing callback information if channels
	  are up until they are hung up. This ensures the hangup process
	  goes smoothly and no channels get hung in limbo. (issue #8088
	  reported by kebl0155) ........

2007-02-20 20:26 +0000 [r55589-55634]  Russell Bryant <russell@digium.com>

	* main/http.c: Add the Asterisk version information to the Server
	  header in HTTP responses. (requested by Pari)

	* include/asterisk/manager.h: Increase the maximum number of
	  manager headers to 128, at the request of Pari.

2007-02-20 16:53 +0000 [r55555]  Jason Parker <jparker@digium.com>

	* channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
	  with strdupa (thanks file) 55555!

2007-02-20 16:41 +0000 [r55553]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample: Change the formatting of sla.conf.sample
	  to make it more readable. (issue #9112, blitzrage)

2007-02-19 21:12 +0000 [r55483]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: - Not sending arguments to an application is
	  not "out of memory" - Making error messages a bit more clear

2007-02-19 18:11 +0000 [r55435]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
	  | 2 lines forcename and forcegreetings options should check to
	  see if the recording already exists ........

2007-02-19 14:52 +0000 [r55397]  Doug Bailey <dbailey@digium.com>

	* channels/chan_iax2.c: Changed iax2 process thread to detached to
	  correct memory leak due to left over thread context on thread
	  exit. Modified module unload process to avoid deadlocks on
	  pthread cancels

2007-02-18 12:35 +0000 [r55250-55278]  Olle Johansson <oej@edvina.net>

	* /, apps/app_record.c: Merged revisions 55277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
	  lines Documentation update (#9053, jsmith) ........

	* /: Block patch that was made only for 1.2 (already implemented in
	  1.4 and trunk)

2007-02-17 17:39 +0000 [r55219]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Add missing membername option to AddQueueMember
	  documentation. (issue #9088 reported by seanbright)

2007-02-17 17:10 +0000 [r55217]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an issue where callerid would not be
	  displayed on some phones. Issue 8995, initial patch and research
	  done by wedhorn

2007-02-17 03:55 +0000 [r55086-55154]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
	  lines Answer the channel before recording privacy information.
	  (issue #8926 reported by lmamane) ........

	* apps/app_queue.c: Make the 'i' option of Queue actually work.
	  (issue #8986 reported by utis)

	* /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
	  lines Allow chan_sip to handle attended transfers from a SIP
	  phone that is sitting behind chan_agent. Yes folks, all it took
	  was one line of code. (issue #8784 reported by pzieba) ........

2007-02-17 00:40 +0000 [r55006-55052]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: If the
	  pg_config application is found, but there is probably executing
	  it, then consider postgres unavailable. (issue #8637)

	* codecs/gsm/Makefile: Filter out yet another architecture that
	  does not work with the optimizations in the built-in libgsm.
	  (issue 8637, ovi)

	* /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
	  revisions 55005 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
	  9 lines Revert the change I did in revisions 54955, 54969, and
	  54970, in 1.2, 1.4, and trunk. I decided that once a conference
	  is created from meetme.conf, it is acceptable behavior that the
	  pin can not be changed until the conference goes away. I also
	  added a note in meetme.conf to describe this behavior. We still
	  have another issue in 1.4 and trunk where some conferences with
	  no users don't go away. That is the real bug that needs to be
	  addressed here. ........

2007-02-16 22:18 +0000 [r55002]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
	  2007) | 2 lines Do not send indications through ast_indicate in
	  chan_agent but instead go directly to the technology. This way
	  when indications are emulated they happen on the Agent channel
	  and do not screw up formats on the channels. (issue #8439
	  reported by punkgode) ........

2007-02-16 21:12 +0000 [r54969]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
	  5 lines For conferences that are configured in meetme.conf, check
	  the configuration file every time someone joins the conference
	  instead of only when the conference is first created. This is to
	  ensure that changes to the pin numbers in the config file are
	  always honored. (issue #9073) ........

2007-02-16 18:51 +0000 [r54924]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Need to check macro extension as well as macro
	  context for directed pickup.

2007-02-16 18:03 +0000 [r54888-54898]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
	  default. It was set to enabled in pbx.c. However, if the option
	  was not present in extensions.conf, then pbx_config.c would set
	  it back to disabled.

	* res/res_features.c: Clean up a few coding guidelines issues -
	  spaces to tabs, use sizeof() to pass the size of a static buffer,
	  add spaces ...

2007-02-16 17:25 +0000 [r54886]  Jason Parker <jparker@digium.com>

	* main/asterisk.c: Clarify a restart message. It's silly, but the
	  reporter had a very valid point. Issue 9079

2007-02-16 17:02 +0000 [r54884]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Allow directed pickup to pick up the real
	  context instead of the macro context if a Macro is used. (issue
	  #8984 reported by jamesb63)

2007-02-16 12:06 +0000 [r54772-54787]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7541 - Handle multipart attachments
	  to SIP messages - even if boundary is quoted.

	* /, res/res_agi.c: Merged revisions 54771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
	  lines Issue #9069 - If we open with TH we should not close with
	  /TD. (seanbright) ........

2007-02-16 00:48 +0000 [r54481-54714]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Don't let dtmf leak over into the engine
	  and let it skew the results... also give DTMF results priority.
	  (issue #9014 reported by surftek)

	* apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
	  lines Use a separate variable to indicate execution should
	  continue instead of the return value. (issue #8842 reported by
	  pluto70) ........

	* apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
	  #9068 reported by mhardeman)

2007-02-14 18:44 +0000 [r54439]  Olle Johansson <oej@edvina.net>

	* /: Block patch only needed in 1.2

2007-02-14 16:56 +0000 [r54375]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
	  lines When handling glare on a PRI, move the requested channel
	  rather than hang up the old one. Fix for 8957 and 9011. ........

2007-02-14 01:09 +0000 [r54290]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
	  with it's placement, feel free to change it. (issue #9045
	  reported by gork)

2007-02-13 21:31 +0000 [r54204-54235]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove a couple of leftover debug messages

	* include/asterisk/devicestate.h: Fix the documentation on the
	  return values from device state provider registration and
	  deletion.

	* channels/chan_sip.c: If we fail to create the SIP socket, then
	  return -1 from reload_config() so that load_module() will return
	  AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
	  spammed with error messages every time chan_sip tries to send a
	  message.

2007-02-13 18:41 +0000 [r54180]  Olle Johansson <oej@edvina.net>

	* /: Blocking patch for 1.2 only

2007-02-12 19:17 +0000 [r54066-54103]  Russell Bryant <russell@digium.com>

	* main/dial.c, include/asterisk/dial.h: Change
	  ast_set_state_callback() to ast_dial_set_state_callback()

	* main/dial.c, apps/app_meetme.c, apps/app_page.c,
	  include/asterisk/dial.h: - Add the ability to register a callback
	  to monitor state changes in an asynchronous dial operation. -
	  Rename the various references to "status" to "state" in the dial
	  API

2007-02-12 16:34 +0000 [r54026]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Make the --without-oss argument work.
	  (issue #9026 reported by puzzled)

2007-02-12 15:38 +0000 [r54002]  Russell Bryant <russell@digium.com>

	* configs/users.conf.sample: Fix a typo where "vmpassword" should
	  be "vmsecret"

2007-02-10 09:09 +0000 [r53878-53881]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Fix VLDTMF reception

	* apps/app_echo.c: Much simpler than previous one ;-)

	* main/channel.c: Provide correct DTMF duration

	* main/cli.c: Bring deprecated 'debug channel <x|all>' command back

2007-02-10 06:06 +0000 [r53850]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac, acinclude.m4: don't display the
	  --with-imap message unless --with-imap was specified without a
	  path use '-n' instead of '! -z' for tests

2007-02-10 01:02 +0000 [r53783-53821]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Add some output for "show application
	  SLAStation/SLATrunk"

	* channels/chan_sip.c: Change some text to properly state "On
	  Hold", which was already done in trunk.

	* configs/sla.conf.sample, include/asterisk/app.h,
	  include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
	  channels/chan_sip.c, doc/sla.txt (added),
	  include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
	  team/russell/sla_rewrite This is a completely new implementation
	  of the SLA functionality introduced in Asterisk 1.4. It is now
	  functional and ready for testing. However, I will be adding some
	  additional features over the next week, as well. For information
	  on how to set this up, see configs/sla.conf.sample and
	  doc/sla.txt. In addition to the changes in app_meetme.c for the
	  SLA implementation itself, this merge brings in various other
	  changes: chan_sip: - Add the ability to indicate HOLD state in
	  NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
	  the channel is not bridged to another channel. linkedlists.h: -
	  Add support for rwlock based linked lists. dial.c: - Add the
	  ability to run ast_dial_start() without a reference channel to
	  inherit information from.

	* apps/app_echo.c: When the Echo() application receives the digit
	  '#', echo that back as well. Since we already sent the BEGIN
	  frame for that digit, it makes sense to send the END as well.

2007-02-09 23:52 +0000 [r53779-53781]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_gtalk.c: another dependency

	* apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
	  funcs/func_odbc.c, res/res_adsi.c: add some inter-module
	  dependencies

	* build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
	  scripts to work when both MODULEINFO and MAKEOPTS are present in
	  a source file

2007-02-09 19:33 +0000 [r53749]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Temporarily change musicclass on channel to one
	  specified in Dial so that the 'm' option functions properly.
	  (issue #8969 reported by christianbee)

2007-02-09 16:42 +0000 [r53715]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/imapstorage.txt, configure, configure.ac: clarify the fact
	  that voicemail IMAP storage cannot be built against a distro's
	  binary c-client library package (at least not at this time)

2007-02-08 23:18 +0000 [r53672]  Olle Johansson <oej@edvina.net>

	* main/acl.c: Don't output debug unless we asked for it

2007-02-08 17:54 +0000 [r53601]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Fix timeout issue when utterance is
	  longer then timeout itself.

2007-02-08 13:47 +0000 [r53530-53532]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/loader.c: Issue 9007 - Mutex not released on early return

	* apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
	  | 2 lines Issue 9003 - If fullname is empty, quote() passes back
	  "\"" ........

2007-02-07 23:52 +0000 [r53464-53497]  Russell Bryant <russell@digium.com>

	* main/db1-ast/Makefile: When building libdb1.a, put the additional
	  flags needed at the beginning of ASTCFLAGS, instead of at the
	  end. This way, we ensure that we find the local headers first
	  before accidentally trying to use headers that exist in locations
	  specified in the ASTCFLAGS passed from the main Makefile. (issue
	  #8637, ovi)

	* main/Makefile: The clean target actually needs to run "distclean"
	  on editline. This is because we need to make sure that its
	  configure script gets executed again, because the CFLAGS we want
	  to pass to editline may have changed.

2007-02-07 17:53 +0000 [r53434]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: We can not reliably do P2P bridging with DTMF passing
	  back with compensation if we need to listen for DTMF frames.
	  (issue #8962 reported by caio1982)

2007-02-07 17:39 +0000 [r53429]  Russell Bryant <russell@digium.com>

	* main/rtp.c: When parsing the NTP timestamp in a sender report
	  message, you are supposed to take the low 16 bits of the integer
	  part, and the high 16 bits of the fractional part. However, the
	  code here was erroneously taking the low 16 bits of the
	  fractional part. It then shifted the result 16 bits down, so the
	  result was always zero. This fix makes it grab the appropriate
	  high 16 bits, instead. (issue #8991, pointed out by
	  andre_abrantes)

2007-02-07 17:04 +0000 [r53358-53399]  Joshua Colp <jcolp@digium.com>

	* apps/app_playback.c: Directly load say.conf in load_module
	  instead of calling the reload function. (issue #8946 reported by
	  junky)

	* /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
	  lines Fix a few potential memory leaks with realtime users and
	  peers. (issue #8999 reported by bsmithurst) ........

2007-02-07 15:33 +0000 [r53355]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
	  | 2 lines Issue 7440 - Macro called from Macro from the h
	  extension exits prematurely ........

2007-02-07 09:22 +0000 [r53324]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 52843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
	  1 line fixed some possible segfaults. also fixed an very
	  important bug which occurs on high load (when calls are very fast
	  generated) ........

2007-02-07 05:24 +0000 [r53246-53294]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_jabber.c: Text fix for jabber reload command (reported by
	  bkruse via IRC)

	* main/manager.c, /: Merged revisions 53245 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
	  | 2 lines Issue 8987 - Status could return two responses
	  (mnicholson) ........

2007-02-05 23:43 +0000 [r53222]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Formatting

2007-02-05 17:06 +0000 [r53150-53152]  Joshua Colp <jcolp@digium.com>

	* apps/app_playback.c: Ensure say_cfg is NULL when the module is
	  loaded. (issue #8946 reported by junky)

	* apps/app_playback.c: Unregister Playback CLI commands as well as
	  dialplan application. (issue #8946 reported by junky)

2007-02-05 00:18 +0000 [r53143]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add some comments on queue system behaviour
	  and how it affects the SIP channel

2007-02-03 21:05 +0000 [r53138]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make SIPDtmfMode application work with
	  recent capability changes, and also fix an RTP stack issue when
	  the auto option was used. (issue #8972 reported by mdu113)

2007-02-03 20:44 +0000 [r53135-53136]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
	  4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when
	  the dial application exits early because of invalid arguments
	  instead of just leaving it empty. (issue #8975) ........

2007-02-03 10:02 +0000 [r53131]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
	  because due to compatibilities with CS1000 reported at
	  www.voip-info.org

2007-02-02 21:26 +0000 [r53129]  BJ Weschke <bweschke@btwtech.com>

	* UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
	  warning to the console that things might possibly be
	  misconfigured when queue member's states are still 'Not in Use'
	  when we're about to bridge them with a caller from queue. Also,
	  put some documentation quoted from oej's queues.txt efforts
	  started in /trunk today. This commit puts #7433 into feedback
	  state for 1.4, and pending no further negative feedback, it will
	  finally be closed.

2007-02-02 17:15 +0000 [r53114-53120]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Correct a copy/pasted error message line for RTCP.

	* main/config.c, /: Merged revisions 53117 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
	  lines Pass the glob expanded filename to process_text_line so
	  that error messages contain the actual filename, not the original
	  include one. (issue #8959 reported by tzafrir) ........

	* Makefile: Add systemname to asterisk.conf generation per recent
	  discussions about it. (issue #8968 reported by blitzrage)

2007-02-02 00:24 +0000 [r53109]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
	  p2p RTP call setup in SIP. You can enable it in sip.conf, but it
	  is now considered experimental until we solve the
	  AST_CONTROL_ANSWER with payload and videocaps stuff.

2007-02-01 22:24 +0000 [r53097-53104]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
	  lines Copy noncodeccapability over to the joint variable so that
	  telephone-event will get transmitted in the sent INVITE. ........

	* main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
	  here as well, but it apparently required both dev mode and no
	  optimizations to creep up.

	* /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
	  lines Don't negotiate RFC2833 when not configured to do so.
	  (issue #8799 reported by mdu113) ........

2007-02-01 21:24 +0000 [r53093]  Russell Bryant <russell@digium.com>

	* funcs/func_strings.c: Fix the FIELDQTY function to not crash.
	  (reported by blitzrage and Corydon on IRC)

2007-02-01 21:15 +0000 [r53091]  Olle Johansson <oej@edvina.net>

	* /: Going backwards, blame file.

2007-02-01 21:11 +0000 [r53086-53088]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
	  2007) | 2 lines Return previous behavior of having MOH pick up
	  where it was left off. (issue #8672 reported by sinistermidget)
	  ........

	* funcs/func_strings.c: Make func_strings build under dev mode.
	  Didn't I do this today already in the berkeley DB?

2007-02-01 21:05 +0000 [r53079-53085]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
	  call counter - If it's still set at time of dialog destruction,
	  make sure we decrement the device call counter properly before we
	  destroy the dialog

	* apps/app_queue.c: Change debug level for state change message
	  that is not really informative when debugging app_queue

	* channels/chan_sip.c: Cleaning up the devicestate callback
	  function

2007-02-01 20:13 +0000 [r53075-53077]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Oops.

	* /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
	  | 2 lines Bug 8965 ........

2007-02-01 19:33 +0000 [r53072]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c: Add missing 'F' letter to getopt so it magically
	  becomes a valid option. (issue #8960 reported by tzafrir)

2007-02-01 19:21 +0000 [r53070]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
	  | 2 lines No wonder FIELDQTY doesn't work with functions... the
	  documentation in pbx.c was wrong ........

2007-02-01 17:37 +0000 [r53064]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix silly logic. We really want to write
	  UDPTL frames out when the call is up.

2007-02-01 16:35 +0000 [r53062]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Add explanation of port= in combination
	  with defaultip= (thanks jsmith)

2007-02-01 13:17 +0000 [r53060]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: we update the name on any first reply of
	  our setup

2007-02-01 11:07 +0000 [r53057]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: chan_h323 is very stable, so let it built
	  by default

2007-02-01 00:24 +0000 [r53050-53052]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: When going on hold have the side that was put on hold
	  reinvite back to Asterisk. When going off hold have the side that
	  was taken off hold reinvited back to the other party.

	* main/rtp.c: Add more frame types to forward in the RTP bridge
	  loops.

2007-01-31 21:32 +0000 [r52859-53046]  Russell Bryant <russell@digium.com>

	* main/cdr.c, main/manager.c, pbx/pbx_spool.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
	  pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
	  channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
	  Merged revisions 53045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
	  3 lines Fix a bunch of places where pthread_attr_init() was
	  called, but pthread_attr_destroy() was not. ........

	* apps/app_userevent.c: Remove an extra \r\n from manager user
	  events. (issue #8955, mnicholson)

	* main/rtp.c, /: Merged revisions 53039 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
	  3 lines Use the proper format string to print unsigned values in
	  the rtp debug output. (issue #8954, wmis) ........

	* apps/app_queue.c: Only changed the paused status in an existing
	  queue member if the paused column exists.

	* apps/app_queue.c: Instead of always creating a realtime queue
	  member as unpaused, read the "paused" column and use that value
	  for the paused status of the member. (issue #8949, jmls)

	* contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
	  (issue #8363, johnlange)

	* doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
	  #8942, lters)

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  codecs/codec_gsm.c: When we are checking for a system installed
	  version of libgsm, we need to check for gsm.h as well.
	  Furthermore, when checking for this header, it may be located in
	  a gsm/ sub directory, so check for that, as well. (issue #8773)

	* channels/chan_sip.c: Only set the DTMF flag on the rtp structure
	  if the DTMF mode is actually RFC2833, not just that it is not
	  INFO. This makes it get set for inband DTMF as well, which is not
	  valid. (issue #8936)

	* main/asterisk.c, /: Merged revisions 52903 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
	  9 lines The SIGHUP handler was implemented to allow admins to
	  send SIGHUP to a running Asterisk process to reload the
	  configuration. However, doing the actual reload in the signal
	  handler itself is a very bad thing to do, because the reload
	  process includes calling non-reentrant functions such as
	  malloc/calloc/etc. If Asterisk is running in the background, then
	  the reload will happen immediately. However, if running in
	  console mode, the reload doesn't work until something is typed at
	  the console. That sort of defeats the purpose, but I don't see an
	  easy way to get around it at this point. ........

2007-01-30 15:29 +0000 [r52856]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Drop the deprecated show commands since the
	  original ones were changed back. (issue #8937 reported by
	  PCadach)

2007-01-30 08:46 +0000 [r52807-52809]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Revert reprecation of h.323 gk cycle
	  command from pre-1.4 version instead of duplicated h323 cycle gk

	* res/res_odbc.c: Don't play with free()'d pointers

	* configure, acinclude.m4: Handle non-standard OpenH323/PWLib
	  library names

2007-01-30 00:15 +0000 [r52763]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
	  5 lines Fix the extraction of the timestamp from video frames. It
	  was using the mapping for a mini-frame instead of a video-frame,
	  which caused it to get invalid data. (issue #8795, mihai)
	  ........

2007-01-29 23:43 +0000 [r52717]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
	  2007) | 2 lines Now that filename is part of the structure and
	  since it comes before postprocess... we have to add it to our
	  postprocess line. (reported on asterisk-dev by Boris Bakchiev)
	  ........

2007-01-29 22:58 +0000 [r52688-52695]  Russell Bryant <russell@digium.com>

	* main/Makefile: Add a missing quotation mark. This was pointed out
	  by jcmoore on #asterisk-dev.

	* main/manager.c: Remove a recursive lock of the manager session.
	  This was pointed out by zandbelt in issue #8711.

2007-01-29 22:12 +0000 [r52679]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_config.c: Argument number correction

2007-01-29 21:36 +0000 [r52611-52647]  Russell Bryant <russell@digium.com>

	* main/Makefile: ASTLDFLAGS needs to be passed to the editline
	  configure script as LDFLAGS. (issue #8928, zandbelt)

	* main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
	  mode translation. P2P bridging can only be used when the DTMF
	  modes don't match if the core is monitoring DTMF in both
	  directions. Then, the core will handle the translation.
	  Otherwise, this bridging method can not be used. (issue #8936)

	* main/manager.c: The session lock can not be held while calling
	  action callbacks. If so, then when the WaitEvent callback gets
	  called, then no event can happen because the session can't be
	  locked by another thread. Also, the session needs to be locked in
	  the HTTP callback when it reads out the output string. This fixes
	  the deadlock reported in both 8711 and 8934. Regarding issue
	  8711, there still may be an issue. If there is a second action
	  requested before the processing of the first action is finished,
	  there could still be some corruption of the output string buffer
	  used to build the result. (issue #8711, #8934)

2007-01-29 18:59 +0000 [r52572]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Use ast_calloc instead of malloc.

2007-01-29 17:57 +0000 [r52535]  Steve Murphy <murf@digium.com>

	* apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
	  backport to 1.4). It was committed to trunk via 7663. But it
	  wasn't so much an enhancement as a fix for the bad language
	  output for portuguese in Brazil, so, after a lot of prodding from
	  patient Brazilians, here is the same fix for 1.4

2007-01-29 17:33 +0000 [r52523]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Set quota information to 0 when creating a
	  vm_state. (issue #8924 reported by neutrino88)

2007-01-29 16:54 +0000 [r52506]  Russell Bryant <russell@digium.com>

	* main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
	  the last commit to the adaptive jitterbuffer code. - Specifically
	  indicate to the compiler that the "dropem" variable only needs
	  one but. - Change formatting to conform to coding guidelines.

2007-01-29 04:18 +0000 [r52494]  Jim Dixon <telesistant@hotmail.com>

	* main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
	  jitterbuf, whereas it would not complain about, and would allow
	  itself to be overfilled (per the max_jitterbuf parameter). Now it
	  rejects any data over and above that size, and complains about
	  it.

2007-01-28 05:15 +0000 [r52462]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* configure, configure.ac: Suggested change to fix normal usage of
	  --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
	  list)

2007-01-27 02:13 +0000 [r52335-52416]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
	  lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
	  follow documentation. (issue #7677 reported by amilcar) ........

	* main/manager.c: Have the manager interface send back an "Already
	  logged in" message instead of "Invalid/Unknown Command" when the
	  client authenticates for a second time. (issue #8509 reported by
	  pari)

	* /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
	  lines Make the last context entry read in the dominant one.
	  (issue #8918 reported by pj) ........

	* main/file.c: Fix core show file formats CLI command.

2007-01-25 19:18 +0000 [r52163-52265]  Joshua Colp <jcolp@digium.com>

	* /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
	  lines Allow dequeueing of frames with negative timestamp by
	  moving jitterbuffer frames check to jb_next. (issue #8546
	  reported by harmen) ........

	* channels/chan_sip.c: Drop out variables I accidentally put in.

	* channels/chan_sip.c: Decrement onHold count if we are hung up on
	  and still on hold. (issue #8909 reported by alexh42)

	* apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
	  2007) | 2 lines Add another note about audio files being played
	  back to each bridged party. (issue #8718 reported by ppyy)
	  ........

2007-01-25 01:37 +0000 [r52107-52160]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, configs/users.conf.sample: By suggestion
	  from kpfleming last week, change "vmpassword" to "vmsecret".

	* configure, configure.ac: Remove libnsl as a required lib for
	  libiksemel to work. This change was already made in the trunk.
	  (issue #8762)

	* include/asterisk/dial.h: Fix the formatting of doxygen comments
	  to properly indicate that the comment documents the previous
	  entity, as opposed to the next one.

2007-01-24 18:26 +0000 [r52052]  Steve Murphy <murf@digium.com>

	* utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
	  line updated check_expr via 8322 (refactoring of expression
	  checking impl); elfring contributed a nice code reorg, I
	  contributed some time to get it working again, better messages
	  ........

2007-01-24 18:20 +0000 [r52016-52049]  Joshua Colp <jcolp@digium.com>

	* main/dial.c (added), apps/app_page.c, main/Makefile,
	  include/asterisk/dial.h (added): Merge in dialing API and the
	  app_page that uses it. (issue #BE-118)

	* channels/chan_sip.c: Fix changing channel formats when joint
	  capability changes and there are no audio formats... I didn't
	  break it originally! (issue #8535 reported by ivoc)

2007-01-24 17:14 +0000 [r52000]  Russell Bryant <russell@digium.com>

	* configure: rebuild configure script to reflect last chan_h323
	  related changes.

2007-01-24 12:57 +0000 [r51979-51989]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: added fix from #8899

	* channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
	  Jan 2007) | 1 line fixed the busy problem (dialstatus was not
	  busy when we called a busy extension) ........

2007-01-24 09:30 +0000 [r51931]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Show capabilities *and* preference in
	  general settings in "sip show settings" (reported by Clona/Telio
	  - Thanks!)

2007-01-24 08:04 +0000 [r51895]  Paul Cadach <paul@odt.east.telecom.kz>

	* acinclude.m4: Allow x64 builds of H.323 (please, rebuild
	  configure)

2007-01-24 00:59 +0000 [r51829-51848]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 51843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
	  6 lines Fix an issue related to synchronization of recordings
	  when using Monitor(). The bug is a miscalculation of the amount
	  to seek the stream for writing to disk when the number of samples
	  coming in and out of a channel do not match up. (issue #8298,
	  #8887, report and patch by guillecabeza, patch files created and
	  testing done by whoiswes) ........

	* apps/app_while.c, /: Merged revisions 51828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
	  4 lines Don't set a new value for the END_ variable on the
	  channel before using the old value. If you do, it will lead to
	  accessing a memory address that has been free()'d. (issue #8895,
	  arkadia) ........

2007-01-23 22:46 +0000 [r51788]  Joshua Colp <jcolp@digium.com>

	* channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_alsa.c,
	  channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
	  drivers to use module referencing so that unloading them while in
	  use will not result in crashes. (issue #8897 reported by junky)

2007-01-23 22:04 +0000 [r51750-51781]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix some bugs in process_message(). The manager
	  session lock needs to be held when sending some sort of response,
	  or calling one of the manager action callbacks. This resolves an
	  issue where people using the GUI would get random crashes when
	  they start clicking around a lot. (issue #8711, reported and
	  debugged by zandbelt)

	* main/http.c: Fix setting the default port of 8088 on 64-bit or
	  big-endian machines.

	* main/manager.c: When traversing the list of manager actions, the
	  iterator needs to be initialized to the list head *after* locking
	  the list. Also, lock the actions list in one place it is being
	  accessed where it was not being done.

2007-01-23 20:32 +0000 [r51683-51716]  Steve Murphy <murf@digium.com>

	* res/res_features.c: this mod from 8593 (dstchannel in cdr is
	  empty when transfer call).

	* main/callerid.c: via 8748 (callerid.c loses name when returning
	  PRIVATE_NUMBER flag), the user suggested this mod, saying it
	  would allow 'WITHHELD' to appear in the name field, which would
	  be useful

2007-01-23 10:28 +0000 [r51648-51649]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
	  6 lines * more additions to make the RESTART message work * added
	  fix for misdn_call to allow SETUPs with empty extensions,
	  replaced the strtok_r functions with strsep for that (inspired by
	  Sandro Cappellazzo, thanks) ........ r50506 | crichter |
	  2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
	  L2 UP, the L1 is UP definitely too, so we set the L1 state up as
	  well. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c: manually merged r49922 and r50335, because
	  of conflicts. this commint includes addition of the ISDN RESTART
	  Message

2007-01-23 06:51 +0000 [r51615]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
	  startup if h323 configuration file not found (reported by
	  mithraen)

2007-01-23 03:00 +0000 [r51513-51558]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only change audio formats on the channel if
	  we have an audio format to change to. (issue #8535 reported by
	  ivoc)

	* /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
	  2007) | 2 lines Yield before reading from zaptel timing source
	  under Solaris so that other threads get a chance to do things.
	  (issue #7875 reported by bob) ........

2007-01-22 19:28 +0000 [r51409]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This fixes 8836, according to dnatural

2007-01-22 19:13 +0000 [r51360-51407]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
	  2007) | 2 lines Move filestream creation to Mixmonitor loop. This
	  will prevent a blank file from being created if no frames ever
	  pass through to be recorded. (issue #7589 reported by
	  steve_mcneil) ........

2007-01-20 06:53 +0000 [r51348-51350]  Jason Parker <jparker@digium.com>

	* configs/say.conf.sample: Fix Italian numeral support in say.conf
	  for "_[2-9]00" case. "2131" would've translated to something
	  along the lines of (pardon my..Italian {or lack thereof})
	  "duecentocentotrentuno", which makes no sense at all.

	* configs/say.conf.sample: Fix German language support in say.conf
	  Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
	  einundzwanzig has the same format as zweiundzwanzig (as do all
	  other "_ZX" spoken numerals) Fix support for numbers in the
	  10,000,000 to 99,999,999 range. Add support for numbers in the
	  100,000,000 to 999,999,999 range.

2007-01-20 00:13 +0000 [r51302-51343]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Remove an unused instance of an unnamed enum.

	* apps/app_meetme.c: Remove another duplicated definition

	* apps/app_meetme.c: Remove a variable that was declared twice.

	* codecs/gsm/Makefile: Add a couple more processors that need
	  optimizations excluded. (issue #8637)

	* channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
	  AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
	  thing. So, a digit would have been interpreted incorrectly here.
	  Since the channel driver will always have the begin and end
	  callbacks called for a digit, only support the button-down and
	  button-up messages.

	* .cleancount: Bump the cleancount since my last commit changed the
	  channel structure.

	* channels/chan_oss.c, main/rtp.c, main/channel.c,
	  channels/chan_phone.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, channels/chan_features.c,
	  channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
	  channels/chan_zap.c, channels/chan_local.c, main/frame.c,
	  channels/chan_sip.c, channels/chan_agent.c,
	  include/asterisk/channel.h, channels/chan_gtalk.c,
	  channels/chan_iax2.c: Merge the changes from the
	  /team/group/vldtmf_fixup branch. The main bug being addressed
	  here is a problem introduced when two SIP channels using SIP INFO
	  dtmf have their media directly bridged. So, when a DTMF END frame
	  comes into Asterisk from an incoming INFO message, Asterisk would
	  try to emulate a digit of some length by first sending a DTMF
	  BEGIN frame and sending a DTMF END later timed off of incoming
	  audio. However, since there was no audio coming in, the DTMF_END
	  was never generated. This caused DTMF based features to no longer
	  work. To fix this, the core now knows when a channel doesn't care
	  about DTMF BEGIN frames (such as a SIP channel sending INFO
	  dtmf). If this is the case, then Asterisk will not emulate a
	  digit of some length, and will instead just pass through the
	  single DTMF END event. Channel drivers also now get passed the
	  length of the digit to their digit_end callback. This improves
	  SIP INFO support even further by enabling us to put the real
	  digit duration in the INFO message instead of a hard coded 250ms.
	  Also, for an incoming INFO message, the duration is read from the
	  frame and passed into the core instead of just getting ignored.
	  (issue #8597, maybe others...)

	* main/asterisk.c: Merged revisions 51300 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
	  4 lines Fix a memory leak on command line tab completion. The
	  container for the matches was freed, but the individual matches
	  themselves were not. (issue #8851, arkadia) ........

2007-01-19 00:17 +0000 [r51272-51274]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c: chan_zap compiles without libpri after
	  committing 7877 patch

	* channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
	  | 3 lines issue 7877: chan_zap module reload does not use
	  default/initialized values on subsequent loads. Reset
	  configuration variables to default values prior to parsing
	  configuration file. ........

2007-01-18 23:36 +0000 [r51270]  Kevin P. Fleming <kpfleming@digium.com>

	* /: block this patch since it is already here

2007-01-18 22:50 +0000 [r51265]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, main/channel.c, main/pbx.c,
	  funcs/func_strings.c, main/app.c: Add some more checks for
	  option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
	  patch(es) by tgrman

2007-01-18 21:54 +0000 [r51262]  Russell Bryant <russell@digium.com>

	* Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
	  Ensure that the locations given to the Asterisk configure script
	  for ncurses, curses, termcap, or tinfo are further passed along
	  to the editline configure script. This fixes some
	  cross-compilation environments. (issue #8637, reported by ovi,
	  patch by me)

2007-01-18 21:14 +0000 [r51256]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
	  Jan 2007) | 2 lines If a timezone is not specified, assume
	  localtime (instead of gmtime) (Issue #7748) ........

2007-01-18 19:17 +0000 [r51251]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Only start timeout once we reach the end
	  of the files to play back.

2007-01-18 18:42 +0000 [r51245]  Jason Parker <jparker@digium.com>

	* main/cli.c: Fix an issue with file name completion in "module
	  load" and "load". Issue 8846

2007-01-18 18:36 +0000 [r51243]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Copy MOH settings when calling a peer so
	  that if they put someone on hold or get put on hold themselves
	  they get the right music class. (issue #8840 reported by mdu113)

2007-01-18 18:28 +0000 [r51241]  Jason Parker <jparker@digium.com>

	* main/channel.c: Fix an issue with deprecated commands

2007-01-18 17:49 +0000 [r51236]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
	  Jan 2007) | 2 lines Document all the fields, including the
	  indication that "uniqueid" should not be renamed. ........

2007-01-18 17:18 +0000 [r51233]  Russell Bryant <russell@digium.com>

	* main/manager.c: Make the "hasmanager" option in users.conf
	  actually have an effect. (issue #8740, LnxPrgr3)

2007-01-18 00:48 +0000 [r51211-51213]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Build the IMAP remote directory string
	  better and properly. Fix an issue with encoding the GSM voicemail
	  when attaching to the voicemail. (issue #8808 reported by
	  akohlsmith)

	* main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
	  (issue #8840 reported by mdu113)

2007-01-17 23:31 +0000 [r51198-51205]  Russell Bryant <russell@digium.com>

	* funcs/func_odbc.c: Fix some instances where when loading
	  func_odbc, a double-free could occur. Also, remove an unneeded
	  error message. If the failure condition is actually a memory
	  allocation failure, a log message will already be generated
	  automatically.

	* channels/chan_zap.c: Instead of dividing the offset by 2
	  directly, make it more clear that the offset is being scaled by
	  the size of the elements in the buffer. (Inspired by a discussing
	  on the asterisk-dev list about this code)

	* /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
	  3 lines Move the check for a failure of ast_channel_alloc() to
	  before locking the pvt structure again. Otherwise, on a failure,
	  this will cause a deadlock. ........

2007-01-17 20:56 +0000 [r51195]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, main/utils.c: Merged revisions 51194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
	  | 4 lines When ast_strip_quoted was called with a zero-length
	  string, it would treat a NULL as if it were the quoting character
	  (and would thus return the string in memory immediately following
	  the passed-in string). ........

2007-01-17 17:36 +0000 [r51186]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: re-add "password" for realtime voicemail

2007-01-17 06:36 +0000 [r51182]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Return the correct result when directly writing out a
	  packet so that the core doesn't then decide to handle it the
	  regular way again. (issue #8833 reported by rcourtna)

2007-01-17 01:29 +0000 [r51176]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: a few more coding style cleanups and one
	  bug fix (from AnthonyL)

2007-01-17 00:46 +0000 [r51172]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
	  scheduler callback.

2007-01-17 00:20 +0000 [r51165-51170]  Jason Parker <jparker@digium.com>

	* main/rtp.c: Fix issue with dtmf continuation packets when the
	  dtmf digit is 0... Issue 8831

	* apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
	  IMAP storage and realtime voicemail. Also update the vmdb sql
	  script for IMAP specific options. Issue 8819, initial patches by
	  bsmithurst (slightly modified by me)

	* doc/voicemail_odbc_postgresql.txt: change documentation to
	  reflect new procedure in 1.4/trunk

2007-01-16 21:51 +0000 [r51159-51162]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
	  51161 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
	  | 2 lines Add documentation walkthrough on getting Postgres to
	  work with voicemail (from Issue 8513) ........

	* apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
	  | 2 lines Postgres driver doesn't like a NULL pointer when
	  retrieving the length (Bug 8513) ........

2007-01-16 17:46 +0000 [r51150]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: minor things i missed before i get jumped
	  on

2007-01-16 17:39 +0000 [r51148]  Joshua Colp <jcolp@digium.com>

	* /, res/res_features.c: Merged revisions 51145 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
	  lines Return previous behavior. ParkedCalls will be able to do
	  DTMF based transfers again. trunk however will get an option to
	  allow this to be set on/off. (issue #8804 reported by nortex)
	  ........

2007-01-16 17:36 +0000 [r51146]  Jason Parker <jparker@digium.com>

	* main/file.c: Display more useful output when streaming files.
	  Include the channel name to which the file is being played. Issue
	  8828, patch by junky.

2007-01-16 05:55 +0000 [r51087]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
	  lines Add none as a valid callgroup/pickupgroup option. I
	  consider it a bug that it would inherit it all the way down and
	  not have any way to reset it to nothing - so that's why it is in
	  1.2. (issue #8296 reported by gkloepfer) ........

2007-01-16 01:15 +0000 [r51057]  Russell Bryant <russell@digium.com>

	* main/config.c: It is possible for the config pointer to be NULL
	  here, so it needs to be checked before dereferencing it.

2007-01-16 00:22 +0000 [r51030]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
	  changing voicemail password in users.conf from voicemail main,
	  written by AnthonyL bug #8436

2007-01-15 23:49 +0000 [r50994]  Russell Bryant <russell@digium.com>

	* Makefile.rules: Filter out a few CFLAGS that are not valid
	  CXXFLAGS.

2007-01-15 21:08 +0000 [r50957]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
	  | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
	  lines Solves issue with forwarding voicemails from folders other
	  than inbox. patch by anthonyl. ........

2007-01-15 18:23 +0000 [r50921]  Jason Parker <jparker@digium.com>

	* main/asterisk.c: re-add deprecated "show version" CLI command.

2007-01-15 16:36 +0000 [r50895]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Move event processing into do_message so that it
	  gets executed again when events are tripped.

2007-01-15 15:03 +0000 [r50867]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in, main/Makefile,
	  configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
	  ACX_PTHREAD macro from the Autoconf macro archive for setting up
	  compiler pthreads support... should improve portability to
	  platforms with unusual pthreads requirements

2007-01-14 21:59 +0000 [r50820]  Joshua Colp <jcolp@digium.com>

	* main/astmm.c: Add missing newlines for two memory CLI commands.

2007-01-14 05:13 +0000 [r50782]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
	  main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
	  main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
	  main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
	  main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
	  main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
	  main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
	  main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
	  main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
	  main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
	  main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
	  main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
	  main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
	  main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
	  main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
	  main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
	  Jan 2007) | 2 lines Bug 8814 - db should look for its header
	  using a relative path, instead of the system path (Fixes FreeWRT)
	  ........

2007-01-13 16:45 +0000 [r50754]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, build_tools/make_sample_voicemail (added): when
	  building the sample greetings for maibox 1234@default during
	  'make samples', build a greeting for each language and file
	  format the user selected to install with menuselect (reported by
	  Brian Capouch on asterisk-dev)

2007-01-13 06:00 +0000 [r50674-50727]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Only write a frame out to the channel if one
	  exists. There are cases where one may not and would therefore
	  cause the channel driver to segfault. (issue #8434 reported by
	  slimey)

	* res/res_snmp.c: Only join the snmp thread on an unload if the
	  thread is actually running. (issue #8810 reported by junky)

2007-01-12 19:24 +0000 [r50647]  Jason Parker <jparker@digium.com>

	* configs/voicemail.conf.sample: Update documentation to state that
	  you shouldn't use realtime static with voicemail.conf

2007-01-12 16:42 +0000 [r50602]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: We need to check for res being 0 in do_message
	  itself, otherwise our headers will get lost.

2007-01-12 14:42 +0000 [r50562]  Kevin P. Fleming <kpfleming@digium.com>

	* main/pbx.c, /: Merged revisions 50561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
	  | 2 lines minor documentation clarification ........

2007-01-11 05:53 +0000 [r50377-50468]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove check for channel state as it can
	  definitely be something other then ring, and also clean up the
	  code a bit. This should solve the parking issues and maybe some
	  attended transfer issues people have been seeing.

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
	  support to see whether NAT was detected (yay symmetric RTP) and
	  also add a check in chan_sip so that if NAT has been detected and
	  the reinvite behind nat option has been turned off, then just do
	  partial bridge. (issue #8655 reported by mnicholson)

	* apps/app_speech_utils.c: Merge speech-multi branch which adds
	  support for joining multiple sound files together to be played
	  one after another in SpeechBackground.

	* main/config.c: Fix parsing when using something like ldap
	  settings. (done by anthonyl)

	* channels/chan_sip.c: Fix chan_sip not working issue. Let's not
	  prematurely return 0. (issue #8783 reported by st41ker)

2007-01-10 16:45 +0000 [r50346]  Jason Parker <jparker@digium.com>

	* cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
	  it fail to load if the config file existed. Issue 8777

2007-01-10 04:55 +0000 [r50266-50298]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
	  lines Add another return value to dial_exec_full that indicates
	  execution is going to continuing at a new
	  extension/context/priority and to just let it slide. (issue #8598
	  reported by jon) ........

	* main/pbx.c: Ensure data's existence before trying to access it.
	  (issue #8774 reported by rcourtna)

2007-01-10 02:17 +0000 [r50228]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 50227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
	  6 lines Make the number that represents the major version number
	  a single digit instead of 2. Using two digits makes it an octal
	  number when put into version.h, which breaks the compilation of
	  any out of tree module that checks the version for any version
	  after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev
	  mailing list, who gave credit to vihai for pointing it out)
	  ........

2007-01-09 17:11 +0000 [r50186]  Jason Parker <jparker@digium.com>

	* main/cli.c: Re-add CLI command that should have only been
	  deprecated in 1.4. Thanks kshumard! (reported in person, so no
	  associated issue #)

2007-01-09 13:40 +0000 [r50151]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
	  | 4 lines The advent of realtime has enabled people to use commas
	  in the fullname field. This could cause an issue with sending
	  voicemails, when the field is unquoted. (Issue 8595) ........

2007-01-09 11:25 +0000 [r50124]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - handle re-invites properly in sip_hangup()
	  - Add some invitestate status changes just to be sure

2007-01-08 23:39 +0000 [r50098]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
	  where it wouldn't ever parse a password, since it was using
	  "secret" instead of "password" Issue 8761, reported by and patch
	  suggestion from ssokol.

2007-01-08 21:11 +0000 [r50073]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_senddtmf.c: we can't unlock a channel if we cant find
	  it. - AnthonyL bug #8741

2007-01-08 18:21 +0000 [r50032]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Disable the more intense packet2packet bridging until
	  the bugs can be worked out.

2007-01-08 14:26 +0000 [r49925-50006]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8677 - Handle failure of T.38
	  re-invite This is not a fix, but adding an error message to tell
	  the admin that we have a bad configuration. We should not send
	  T.38 re-invites to devices that can't handle it (with the current
	  architecture where you have to hard-code t.38 support per
	  device). To really fix this, we need to figure out a way to tell
	  the incoming call that the re-invite failed, so we can signal
	  failure on that end and go back to the original call.

	* channels/chan_sip.c: Issue #8524, support multiple via header
	  values (tardieu) Thanks!

	* channels/chan_sip.c: We only need one forward declaration

	* channels/chan_sip.c: Issue 8735: Terminate state when extension
	  is unavailable for subscription

2007-01-08 05:11 +0000 [r49890]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
	  lines Ensure we use the default refresh value of 60 if the remote
	  server does not send one. (issue #8746 reported by maethor)
	  ........

2007-01-08 03:53 +0000 [r49866]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: since we use AC_PATH_TOOL to find tools,
	  we should use the results it provides for us (reported by Brian
	  Capouch on the asterisk-dev list)

2007-01-07 21:44 +0000 [r49831-49834]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
	  | 2 lines If openstream fails, then we crash (Issue 8564)
	  ........

	* channels/chan_sip.c: Second condition was a subset of the first,
	  so hold was never decremented, thus hint stayed stuck (Issue
	  8747)

2007-01-06 00:24 +0000 [r49742]  Jason Parker <jparker@digium.com>

	* main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
	  byte of allocated memory! This looks like it may have been a
	  chicken/egg scenario.. You had to call a cleanup func, because
	  everything was allocated. Then since you had to call a cleanup
	  func, you were forced to allocate - ie; strdup("").

2007-01-05 23:51 +0000 [r49710-49715]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, acinclude.m4: one more time...

	* configure, acinclude.m4: proper fix for r49712

	* configure, acinclude.m4: if --with-foo=<path> is specific for a
	  configure option, ensure that it is used for header file checking
	  as well

	* main/manager.c: ast_func_read() needs a writable copy of the
	  function name to be passed

2007-01-05 23:16 +0000 [r49705]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
	  chan_zap also depend on zaptel. This fixes an issue (8727) with
	  zaptel being in a different directory, using --with-zaptel.

2007-01-05 22:52 +0000 [r49676-49680]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: don't 'consume' the params list before we try to
	  use it again

	* res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
	  main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
	  main/db.c, channels/chan_zap.c, channels/chan_sip.c,
	  apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
	  utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
	  apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
	  AMI and AMI/HTTP requests by nearly 20K in most cases

2007-01-05 22:14 +0000 [r49675]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Don't keep repeating the warning over and over
	  when the end of the call is reached. (issue #8724 reported by
	  xrg)

2007-01-05 17:09 +0000 [r49581-49636]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
	  | 2 lines ensure that threads which are supposed to be detached
	  (because we aren't going to wait on them) are created properly
	  ........

	* channels/chan_iax2.c: revert the dynamic_list insertion change...
	  that was not the right thing to do

	* channels/chan_iax2.c: create the IAX2 processing threads as
	  background threads so they will use smaller stacks when we create
	  a dynamic thread, put it on the dynamic_list right away so we
	  don't lose track of it

2007-01-04 23:00 +0000 [r49568]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: It's possible for the iax2 pvt to
	  disappear, so if it has... don't bother looking for dpentries.

2007-01-04 22:51 +0000 [r49553]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/threadstorage.h, main/asterisk.c,
	  build_tools/cflags.xml, include/asterisk.h, main/Makefile,
	  main/threadstorage.c (added), main/utils.c: add support for
	  tracking thread-local-storage objects that exist via
	  'threadstorage' CLI commands

2007-01-04 22:28 +0000 [r49551]  Joshua Colp <jcolp@digium.com>

	* main/config.c: Only free comments and line buffer once we reach
	  the first level. (issue #8678 reported by ssokol, fixed by
	  anthonyl)

2007-01-04 21:58 +0000 [r49460-49536]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/iax2-parser.c, main/frame.c: don't mark these
	  allocations as 'cache' allocations when caching has been disabled

	* channels/iax2-parser.c: if we're going to decrement the frame
	  count when we free a frame, we should inrement it when we create
	  one :-)

	* channels/iax2-parser.c, channels/iax2-parser.h,
	  channels/chan_iax2.c: only do IAX2 frame caching for voice and
	  video frames

	* main/frame.c: don't do frame header caching in the core if
	  LOW_MEMORY is defined

	* channels/iax2-parser.c: don't define this type either if
	  LOW_MEMORY is enabled

2007-01-04 18:11 +0000 [r49459]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
	  | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
	  lines converted a lot of 256 to PATH_MAX and some white space
	  fixes. ........

2007-01-04 18:06 +0000 [r49457-49458]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode

	* codecs/Makefile: make building of codec_gsm against the system
	  GSM library actually work

2007-01-04 16:50 +0000 [r49413]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
	  | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
	  lines good catch russell sorry i missed that. fix magic number
	  with proper sizeof ........

2007-01-04 04:33 +0000 [r49388]  Russell Bryant <russell@digium.com>

	* funcs/func_realtime.c: Fix the REALTIME() dialplan function.
	  ast_build_string() advances the string pointer to the position to
	  begin the next write into the buffer. So, this pointer can not be
	  used to copy the contents of the string later. The beginning of
	  the buffer must be saved. Interestingly enough, this code could
	  not have ever worked. (Pointed out by Sebb on IRC, thanks!)

2007-01-03 23:32 +0000 [r49355]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
	  https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
	  | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
	  lines When using ODBC_STORAGE VoicemailMain doesn't create the
	  subdirectories for a mailbox such as the INBOX directory. this
	  patch solves that problem, was written by anthony be-125 ........

2007-01-03 09:06 +0000 [r49313]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  configs/misdn.conf.sample: Merged revisions
	  48319,48321,48467,48552,48576,49135,49303 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
	  1 line changed a few debugs to higher debug levels ........
	  r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
	  1 line added the export and import of the MISDN_ADDRESS_COMPLETE
	  Variable to inidcate wether the extension is already completely
	  dialed or if there might come additional digits by information
	  elements. also added some docs for that. ........ r48467 |
	  crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
	  removed FIXUP state. added check for channel allocation conflict
	  when we create a setup while the other site creates a setup on
	  the same channel, besides the check we resolve this conflict.
	  ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
	  Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
	  preselected channel we just accept it, even when we're NT. added
	  some checks for segfaults. ........ r48576 | crichter |
	  2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
	  reject a channel, because it's in use already, we shouldn't
	  process the setup anymore. made the channel allocation a bit
	  easier and more understandable, removed a few unused lines
	  ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
	  Jan 2007) | 1 line added check for channel ranges in the
	  set/empty channel functions. set pmp_l1_check default to no.
	  added misdn restart pid cli command. added cleaning of channel
	  when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
	  2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
	  check for bridging in misdn_call to avoid setting
	  echocancellation when 2 mISDN channels are involved and when
	  bridging is set. That lead to a kernel panic before under
	  different situations, because we switched about 2 times between
	  hardware bridging and echocancelation * readded MISDN_URATE
	  variable which got lost before, this should make app_v110 work
	  again * fixed typo ........

2007-01-03 03:21 +0000 [r49282]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, Makefile.rules: various Makefile improvements to get
	  chan_vpb (and any other C++ modules) to build properly

2007-01-03 01:19 +0000 [r49259]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Check pvt structure presence before passing
	  to send_command. This gets rid of the irritating message about a
	  packet without pvt structure. This happens because the scheduled
	  item is getting cancelled at almost the exact moment it is
	  getting executed.

2007-01-02 22:30 +0000 [r49237]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
	  pbx/ael/ael.flex: This is a slight modification to Josh's edits
	  for #8579; both files edited were the produced by flex; so the
	  source files need to be changed instead, and the generated files
	  regenerated.

2007-01-02 19:58 +0000 [r49212]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
	  enabled at that point in the code

2007-01-02 17:33 +0000 [r49189]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Allow fractions of a second in the Wait()
	  application, like it says it allows.

2007-01-02 13:59 +0000 [r49165]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: remove comment that is unrelated to this
	  function

2007-01-02 12:08 +0000 [r49145]  Olle Johansson <oej@edvina.net>

	* configs/features.conf.sample: Adding note on effect of
	  applicationmap features on re-invites

2007-01-01 23:34 +0000 [r49098-49102]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
	  configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
	  and transcoding support in the system's Zaptel installation, and
	  make only the modules that need those features dependent on them
	  (this will allow building the other Zaptel-using parts of
	  Asterisk against older versions of Zaptel or those on other
	  platforms that haven't caught up yet to the Linux version)

	* Makefile: use a simpler (and portable) method to ensure that
	  menuselect is built as a host binary

	* Makefile: revert this change until a better solution can be
	  found... 'env -i' was not being used properly, but even when
	  changed to do so, this process fails during cross-compilation
	  because the menuselect build still sees 'CC' as set to the
	  cross-compiler

2007-01-01 20:14 +0000 [r49096]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: remove incomplete implementation of dnsmgr.
	  Let's fix this in trunk.

2006-12-30 18:31 +0000 [r49063-49073]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: IAX has been deprecated for quite some time so
	  we had better use IAX2 when creating the dial string for users.
	  (issue #8697 reported by ssokol)

	* channels/chan_zap.c: Use asprintf to build the channel names
	  instead of custom function. I believe the custom function is
	  doing some things that are not portable across all
	  implementations. (issue #8570 reported by hterag & issue #8692
	  reported by nicolasg)

	* main/rtp.c: If the Packet2Packet bridge is being broken because
	  of a masquerade then attempt to read a frame in so the masquerade
	  actually happens. Otherwise weirdness will occur. (issue #8696
	  reported by kjotte)

	* channels/chan_iax2.c: Initialize the packet queue in load_module
	  instead of just declaring the list with the default value. (issue
	  #8695 reported by ssokol)

2006-12-30 00:40 +0000 [r49061]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
	  comma args converted to vertical bars. I hope this change does
	  little harm.

2006-12-29 00:50 +0000 [r49042-49048]  Kevin P. Fleming <kpfleming@digium.com>

	* /: put this value into the correct property

	* /, BUGS: Merged revisions 49045 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
	  | 2 lines location of the bug posting guidelines has changed
	  ........

	* sample.call: simple commit to test CIA integration

2006-12-28 21:26 +0000 [r49032-49035]  Jason Parker <jparker@digium.com>

	* main/cli.c: Fix some deprecated commands. Issue 8682, patch by me

	* main/http.c: saw this in passing... fix a small typo

2006-12-28 20:08 +0000 [r49028]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: new versions of sounds

2006-12-28 19:52 +0000 [r49024]  Jason Parker <jparker@digium.com>

	* main/http.c: make the uris_lock a rwlock instead of a mutex lock
	  - needs to be forward ported to trunk

2006-12-28 19:43 +0000 [r49022]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Backport support for read/write locks.

2006-12-28 19:21 +0000 [r49020]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
	  pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
	  pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
	  pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
	  from the ael stuff. Also, threw in a minor fix to frame.c to
	  avoid build-killing compiler warnings.

2006-12-27 22:28 +0000 [r49009]  Joshua Colp <jcolp@digium.com>

	* main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
	  available when LOW_MEMORY is used and things are being built in
	  the utils directory, so we need to resort to the old method of
	  strncpy. (issue #8579 reported by mottano)

2006-12-27 22:06 +0000 [r48998-49006]  Kevin P. Fleming <kpfleming@digium.com>

	* main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
	  main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
	  main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
	  main/http.c, main/logger.c: since these variables all have static
	  duration, none of them need initializers (they default to zero
	  anyway)

	* include/asterisk/options.h, main/asterisk.c, main/file.c: move
	  extern declaration for this option to a header file where it
	  belongs provide an initial value for 'languageprefix' option,
	  instead of relying on randomness to provide a useful value

2006-12-27 21:06 +0000 [r48993-48997]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Only include acl.h and lock.h once

	* channels/chan_sip.c: Only set rfc2833compensate flag once
	  (handle_common_options)

	* channels/chan_sip.c: - Remove checking for T38 options twice.
	  Keeping them in handle_common_options

2006-12-27 18:33 +0000 [r48987-48988]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: make the option actually match the
	  documentation

	* channels/iax2-parser.c, include/asterisk/utils.h,
	  include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
	  memory' and 'show memory summary' to distinguish memory
	  allocations that were done for caching purposes, so they don't
	  look like memory leaks

2006-12-27 17:59 +0000 [r48975-48985]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
	  politically correct

	* channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
	  cisco MWI support. Normally we try not to change our software for
	  bugs in other devices. But in this case, the Cisco phones are so
	  widespread so we try to implement a fix while waiting for a
	  bugfix from Cisco.

	* channels/chan_sip.c: - Make sure handle_common_options return 1
	  when we found a common option - Move uncommon (only global)
	  option away from handle_common_options Reported by rizzo. Thanks!

	* channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
	  re-sending invite with auth.

	* /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
	  (rizzo, #8600)

2006-12-26 05:20 +0000 [r48960-48966]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Get rid of a needless memory allocation and
	  only create a conference structure in find_conf_realtime if data
	  was read from realtime. (issue #8669 reported by robl)

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
	  API call that initializes an RTP structure. We need this because
	  chan_sip is cheeky and uses a temporary RTP structure for codec
	  purposes, and the API calls that are used rely on the lock.
	  (Pointed out on asterisk-dev by Andy Wang)

	* configure, configure.ac: Clean up autoconf file (gets rid of
	  warnings seen when rebuilding configure) and rebuild configure.

2006-12-25 05:21 +0000 [r48931-48956]  Russell Bryant <russell@digium.com>

	* /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
	  6 lines Fix an error introduced by copying and pasting the
	  handling of the >= operator for the MATH function. If a single
	  equal sign was used as an operator, the function would treat it
	  is as if it were the >= operator. Now, it properly handles it as
	  an invalid operator. (issue #8665, patch by tempest1) ........

	* channels/chan_oss.c: Fix a typo in an error message that
	  indicated that the MGCP channel type could not be registered,
	  instead of the correct type, OSS.

	* /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
	  3 lines Check for the proper return value on an error in a call
	  to mmap(). This was reported by Andy Wang on the asterisk-dev
	  list. Thanks! ........

	* /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
	  3 lines Remove a couple of misplaced dots in log messages. This
	  was reported by Andrea Spadaccini on the asterisk-dev mailing
	  list. ........

	* main/http.c: Implement locking for the list of URI handlers to
	  make it thread-safe.

2006-12-23  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0 released.

2006-12-22 22:33 +0000 [r48870-48906]  Jason Parker <jparker@digium.com>

	* Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.

	* channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia

2006-12-21 20:26 +0000 [r48783]  Joshua Colp <jcolp@digium.com>

	* /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
	  lines Add new silence sound files to the spec for Redhat. (issue
	  #8652 reported by alvaro_palma_aste) ........

2006-12-20 02:56 +0000 [r48592-48637]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
	  builds.

	* apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
	  it is then passed to the IMAP store file function. (issue #8614
	  reported by punknow)

	* doc/snmp.txt: find is not the same as bind when it comes to
	  documentation. (issue #8626 reported by johann8384)

2006-12-19 21:28 +0000 [r48586]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/Makefile: suppress compiler warnings in this module
	  until it can be improved

2006-12-19 21:12 +0000 [r48585]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
	  lines Free localuser structure when we fail to dial (issue #8612
	  reported by rizzo) ........

2006-12-19 21:03 +0000 [r48583]  Luigi Rizzo <rizzo@icir.org>

	* apps/app_sms.c: fix a bogus datalen in the frames generated by
	  app_sms (causing noisy output if you listen to the output!) This
	  affects trunk as well, whereas 1.2 is ok.

2006-12-19 14:57 +0000 [r48577]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
	  type for these unixODBC API calls, eliminating warnings on 64-bit
	  platforms that use the 'new' 64-bit types for ODBC API calls

2006-12-19 03:46 +0000 [r48571]  Joshua Colp <jcolp@digium.com>

	* Makefile: Use env -i to start a fresh environment when going to
	  build menuselect. This is more portable then using unset. (issue
	  #8543 reported by jtodd)

2006-12-18 17:23 +0000 [r48566]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/channel.h: unbreak the macro used for
	  incrementing the frame counters. I don't know when the bug was
	  introduced, but with the typical usage c->fin =
	  FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
	  trunk as well (fix coming).

2006-12-18 17:15 +0000 [r48564]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Put thread into proper list if we abort
	  handling due to an error, and also hold the lock while putting it
	  back into the proper idle list so we don't prematurely get a
	  signal. (issue #8604 reported by arkadia)

2006-12-18 11:59 +0000 [r48513-48554]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
	  utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
	  utils/ael_main.c: remove some now-unnecessary explicit includes
	  of autoconfig.h clean up per-file dependencies during 'make
	  clean'

	* build_tools/prep_tarball: need an additional argument here to
	  make the downloads actually occur

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
	  these calls from thinking they have multiple arguments

	* codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
	  funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
	  main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
	  main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
	  simplify dependency tracking system, using the compiler's
	  built-in method for generating them, and only doing dependency
	  tracking if developer mode is enabled via the configure script

	* Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
	  really, really have to have autoconfig.h included before all
	  other headers (especially system headers), the Makefile will now
	  force it to happen (this will fix build problems with files like
	  ast_expr2f.c, where we can't control the inclusion order in the
	  file itself)

	* funcs/func_curl.c: instead of initializing the curl library every
	  time the CURL() function is invoked, do it only once per thread
	  (this allows multiple calls to CURL() in the dialplan for a
	  channel to run much more quickly, and also to re-use connections
	  to the server) (thanks to JerJer for frequently complaining about
	  this performance problem)

2006-12-15 19:55 +0000 [r48502-48506]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Turn payload_lock into bridge_lock and make it
	  encompass all RTP structure contents that may relate to bridge
	  information, including who we are bridged to.

	* channels/chan_iax2.c: Hold call structure lock in places where a
	  qualify or peer action can destroy it.

	* channels/chan_iax2.c: Lock network retransmission queue in all
	  places that it is used.

2006-12-15 10:55 +0000 [r48481-48487]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
	  from 1.2)

	* channels/chan_sip.c: Update to latest IANA spec

2006-12-15 06:28 +0000 [r48461-48478]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Use a wakeup variable so that we don't wait
	  on IO indefinitely if packets need to be retransmitted.

	* main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
	  structure can change AFTER a bridge has started. This comes from
	  the packet handling of the SIP response when indication that it
	  was answered has been sent. Therefore we need to protect this
	  data with a lock when we read/write. (issue #8232 reported by
	  tgrman)

	* main/rtp.c: Remove direct RTCP bridging. I've come to the
	  conclusion that we should handle this through the core and not
	  just forward it on. Should solve a few bugs.

2006-12-12  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta4 released.

2006-12-12 04:13 +0000 [r48401]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
	  is the way it should have been done.

2006-12-11 23:02 +0000 [r48396-48399]  Matt O'Gorman <mogorman@digium.com>

	* sounds/Makefile: new sounds package with 100% more silence

	* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.2 ........
	  r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
	  | 4 lines app_externalivr needs a real silence file, and
	  additional changes to add silence files into core instead of
	  extra patch provided by bug 8177 with minor additions. ........

2006-12-11 21:31 +0000 [r48391]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Return non-existant callerid handling to
	  that which it was before. In 1.4 and trunk callerid can be
	  allocated but not have any contents so we have to use
	  ast_strlen_zero before passing it to the relevant functions.
	  (issue #8567 reported by pabelanger)

2006-12-11 05:37 +0000 [r48382]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: STRFTIME() does not actually require an
	  argument (issue 8540)

2006-12-11 05:36 +0000 [r48377-48381]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
	  RTCP callback functions so they no longer share a common one.

	* apps/app_meetme.c: Use the correct API call to say a device state
	  changed. (Yes, I'm a nub.)

	* apps/app_meetme.c: Don't access the conference structure after it
	  has been freed.

2006-12-11 00:47 +0000 [r48375]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
	  res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
	  apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
	  | 5 lines When doing a fork() and exec(), two problems existed
	  (Issue 8086): 1) Ignored signals stayed ignored after the exec().
	  2) Signals could possibly fire between the fork() and exec(),
	  causing Asterisk signal handlers within the child to execute,
	  which caused nasty race conditions. ........

2006-12-10 03:04 +0000 [r48372]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
	  line This version applies the patch suggested by stevens in bug
	  7836 (make inbound channel RINGING state consistent with other
	  channels). ........

2006-12-09 15:59 +0000 [r48362-48363]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use locking when accessing the
	  registrations list. This list is not actually used very often, so
	  the likelihood of there being a problem is pretty small, but
	  still possible. For example, if the CLI command to list the
	  registrations was called at the same time that a reload was
	  occurring and the registrations list was getting destroyed and
	  rebuilt, a crash could occur. In passing, go ahead and convert
	  this list to use the linked list macros.

2006-12-07 18:17 +0000 [r48357]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
	  Dec 2006) | 3 lines Ensure that the file position is not
	  incremented beyond the total number of files available for
	  playback. (issue #8539, ulogic) ........

2006-12-07 15:33 +0000 [r48349]  Steve Murphy <murf@digium.com>

	* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
	  killed bug 8423 -- OriginateSuccess and OriginateError incomplete
	  channel name. May it rest in peace.

2006-12-06 16:25 +0000 [r48326]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
	  retransmitted to Asterisk

2006-12-06 16:15 +0000 [r48323]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
	  Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
	  in the sample configuration file. (issue #8526, arkadia) ........

2006-12-06 12:27 +0000 [r48316-48317]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't send Contact on MESSAGE

2006-12-05 20:42 +0000 [r48279]  Jason Parker <jparker@digium.com>

	* configure.ac: Fix curl version number testing to be much more
	  friendly to non-bash shells. Issue 8508, patch by me. This
	  *SHOULD* be POSIX compliant now..

2006-12-05 17:29 +0000 [r48264-48270]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Merging the invitestate-1.4 branch after
	  successful testing. Will check if I can solve this with less
	  changes in 1.2.

	* configs/sip.conf.sample: Add missing s from another repository.
	  (thanks jcmoore!)

	* configs/sip.conf.sample: Updating sip.conf.sample with
	  information about T38 not working when chan_local or chan_agent
	  is involved in the call. I don't know how big a fix that would be
	  to solve, but this is the current state of affairs. (Chan_sip
	  currently checks if the other side of the bridge has a SIP tech.
	  We could/should implement another check, possibly for udptl_write
	  or some flag in the ast_channel structure).

2006-12-05 01:41 +0000 [r48252-48254]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Oops, forgot to release the odbc handle

	* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
	  | 6 lines If the recording in the database is too large, it will
	  fail to retrieve with an mmap error. Not too sure why this
	  doesn't happen when we put it in the database, also, but since
	  that doesn't seem to be broken, I'm not going to fix it (at least
	  until someone reports it). Solution is to ask for the file in
	  smaller chunks. (Bug 8385) ........

2006-12-04 21:48 +0000 [r48237-48248]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix an issue which didn't allow
	  unavail/greet/busy/etc messages from being saved into ODBC (and
	  probably IMAP).

2006-12-04 17:54 +0000 [r48228-48230]  Jason Parker <jparker@digium.com>

	* configs/voicemail.conf.sample: Add documentation to
	  voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
	  blitzrage.

	* doc/snmp.txt: Attempt to document some of the dependencies that
	  are needed for net-snmp Issue 8499 - initial patch by blitzrage.

2006-12-03 06:34 +0000 [r48223]  Russell Bryant <russell@digium.com>

	* sounds/Makefile: When "fetch" is in use, instead of "wget",
	  --continue is not a valid option. (issue #8451)

2006-12-02 21:45 +0000 [r48199-48219]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Removing one of two pieces of code to
	  handle 481 response on INVITE - Move handling of REFER response
	  to handle_response_refer()

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
	  configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
	  transmission happens - Encapsulate RTP timers in the rtp
	  structure so we have one for video and one for audio The video
	  one is not used in 1.4, really. Will be used for RTP keepalives
	  when we can send something that video phones support in the RTP
	  stream. I now this is a big architectual change at this stage for
	  1.4, but decided it was needed to avoid future bug reports. -
	  Document the RTP NAT keepalive option in sip.conf.sample Issue
	  7679 in the bug tracker. Please test.

2006-12-02 03:50 +0000 [r48195]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h: Backport the comment containing the
	  warning regarding the limitations on the usage of this function.
	  It is thread safe, but not technically reentrant.

2006-12-01 23:37 +0000 [r48193]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
	  | 2 lines if Dial() is going to send music-on-hold to the calling
	  party, it has to send PROGRESS first to ensure that the reverse
	  audio path has been setup first (BE-106) ........

2006-12-01 23:16 +0000 [r48190]  Russell Bryant <russell@digium.com>

	* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
	  FreeBSD 6.1 does not include wget by default. However, it has
	  fetch which will work just fine for our purposes of downloading
	  the sounds packages. So, check for both wget and fetch and the
	  configure script and use what was found to download them. If
	  neither one was found, and sound packages are selected that must
	  be downloaded, the install process will print out an informative
	  error message indicating the situation. Also, fix a couple places
	  where "make" was hard coded into some output messages by
	  replacing them with the $(MAKE) variable. (issue #8451, initial
	  patch by pabelanger, with additional modifications by me)

2006-12-01 20:25 +0000 [r48184-48186]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 48183 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
	  lines Fix a small typo - issue 8848, reported by pabelanger
	  ........

2006-12-01 19:38 +0000 [r48179]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/cli.c: Double-unlock error (reported by blitzrage on IRC)

2006-12-01 17:41 +0000 [r48177]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
	  "limitonpeers" patch from trunk, to fix a lot of issues with
	  queues and SIP device states - Remove support for T.38 early
	  media, since it's impossible. (Two patches in one - extra friday
	  evening offer due to being off line from svn today... :-)

2006-11-30 21:18 +0000 [r48168]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
	  do a partial bridge for Google Talk since we need to handle STUN.
	  (issue #8448 reported by phsultan)

2006-11-30 20:51 +0000 [r48166]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue 8319 - change noncecount before
	  using it.

2006-11-30 20:28 +0000 [r48143-48162]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
	  lines Only print out debug message if bridged channel is not
	  NULL. (issue #8412 reported by jubilex) ........

	* /, res/res_features.c: Merged revisions 48154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
	  lines Do not listen for DTMF on the bridge that comes into
	  existence when ParkedCall is executed. This means native bridging
	  can now occur for this. (issue #8406 reported by kebl0155)
	  ........

	* main/cdr.c, /: Merged revisions 48151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
	  lines Print certain CDR messages out at the NOTICE level versus
	  WARNING since they can occur when used with the CDR applications
	  and are perfectly fine. (issue #8367 reported by dartvader)
	  ........

	* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
	  2006) | 2 lines Document 'port' for SIP peers, came up because of
	  the current mailing list thread. (issue #8450 reported by
	  blitzrage) ........

2006-11-30 14:29 +0000 [r48129-48135]  Olle Johansson <oej@edvina.net>

	* doc/manager.txt: Explain status reports and make codefreeze more
	  happy :-)

	* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
	  GS 487 adapter without CSEQ on separate line in the REGISTER
	  request. Imported from 1.2.

2006-11-29 21:05 +0000 [r48115]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
	  mm_login. (issue #8420 reported by slimey)

2006-11-29 19:56 +0000 [r48113]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Explain the use device status system
	  implemented in SIP for subscriptions, queues and manager a bit
	  better. Like in 1.2, you will get more detailed information if
	  you set a call limit for a device. When the call limit is
	  reached, the status system will report a device as busy. For
	  queues, setting a call limit per SIP device is propably a
	  requirement. In most cases, it will work much better if you only
	  use type=peer and not type=friend. We might decide to backport
	  the new setting from trunk to apply all call limits to the peer
	  part of a friend only.

2006-11-29 16:50 +0000 [r48107]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 48106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
	  lines If the frame was duplicated before writing out then we need
	  to free it. (issue #8429 reported by edguy3) ........

2006-11-29 08:03 +0000 [r48105]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.

2006-11-29 04:26 +0000 [r48101]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Don't crash if the mailstream was not
	  created.

2006-11-28 18:26 +0000 [r48095]  Jason Parker <jparker@digium.com>

	* Makefile: Export several more variables in top level Makefile.
	  Inspired by issue 8438.

2006-11-28 16:57 +0000 [r48054-48088]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
	  2006) | 2 lines According to the research I have done we never
	  needed to include compiler.h in the first place so let's not!
	  (issue #8430 reported by edguy3) ........

	* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
	  lines Use the proper function to get the new message count
	  instead of always using the filesystem. (issue #8421 reported by
	  slimey) ........

2006-11-27 17:20 +0000 [r48049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
	  Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
	  ........

2006-11-27 17:17 +0000 [r48046]  Russell Bryant <russell@digium.com>

	* main/manager.c: Remove a couple of unused variables (issue #8380,
	  casper)

2006-11-27 15:32 +0000 [r48038]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
	  lines Do not reference the freed outgoing structure in the debug
	  message. (issue #8425 reported by arkadia) ........

2006-11-27 06:41 +0000 [r48031]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change logging message

2006-11-26 00:26 +0000 [r48015-48017]  Steve Murphy <murf@digium.com>

	* funcs/func_cdr.c: might as well also document the raw values of
	  the flag vars

	* /, funcs/func_cdr.c: A little bit of func_cdr documentation
	  upgrade-- no bug# involved, although 8221 may have inspired it.

2006-11-25 09:28 +0000 [r48002]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
	  and future releases, you can disable subscription support totally
	  or per peer in sip.conf with allowsubscribe = yes | no

2006-11-24 17:17 +0000 [r47992]  Steve Murphy <murf@digium.com>

	* main/translate.c: bug 8189 posted this fix for main/translate.c
	  for PLC

2006-11-24 15:46 +0000 [r47989]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
	  Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
	  beatufied some logs, changed some loglevels. changed the default
	  value of block_on_alarm ........

2006-11-23 11:01 +0000 [r47959]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't allocate unused variable.

2006-11-22 21:47 +0000 [r47944]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Video will never reach Packet2Packet bridging and can
	  do more harm then good.

2006-11-21 17:32 +0000 [r47897]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If we have the non standard G726-32 setting turned on
	  we want to return G726-32 to the SDP, not our AAL2 string. (issue
	  #8330 reported by voipgate)

2006-11-21 15:20 +0000 [r47892]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
	  provisional response. Let's not treat that as early media.
	  (discovered at the AVTF meeting in Paris).

2006-11-20 20:01 +0000 [r47863-47864]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Oops, merge missed release of odbc object

	* apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
	  | 2 lines Failing to trap -1 error from mmap causes segfault
	  (Issue 8385) ........

2006-11-20 19:51 +0000 [r47850-47860]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 47859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
	  lines Don't forget to byte swap if we are exiting the smoother
	  feed early. (issue #8287 reported by arturs) ........

2006-11-16 23:00 +0000 [r47777]  Kevin P. Fleming <kpfleming@digium.com>

	* /, doc/billing.txt: update documentation regarding IAX2 transfers
	  and CDRs Merged revisions 47776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
	  | 2 lines update clearly wrong documentation regarding cdr_custom
	  ........

2006-11-16 21:11 +0000 [r47762-47764]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Compare technology using the pointers
	  instead of a straight comparison based on name. (issue #8228
	  reported by dean bath)

2006-11-16 20:09 +0000 [r47758]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: check for pre-1.4 versions of Zaptel and
	  abort the configure script if found with an appropriate error
	  message

2006-11-16 19:24 +0000 [r47755]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
	  notification optional, in order to avoid a lot of extra database
	  lookups for all those realtime users out there.

2006-11-16 18:29 +0000 [r47748-47751]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47750 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
	  2006) | 2 lines Because of the way chan_local is written we
	  should be extra careful and make sure our callback functions have
	  a tech_pvt. (issue #8275 reported by mflorell) ........

	* apps/app_meetme.c: Don't unreference the SLA object if there is
	  no SLA object in the devicestate callback. (issue #8354 reported
	  by loloski)

2006-11-16 16:51 +0000 [r47733-47744]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't fixup if there's nothing to fixup

	* UPGRADE.txt: Warn users about change in canreinvite

	* channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
	  authenticated (according to the RFC) - Update docs on
	  canreinvite. "nonat" is the recommended setting for most users
	  with phones behind a NAT.

2006-11-15 22:31 +0000 [r47712]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47711 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
	  2006) | 2 lines Make sure that the pvt structure exists before
	  trying to do fixup on Local channels. (issue #7937 reported by
	  mada123, fix by alamantia with mods by me) ........

2006-11-15 21:56 +0000 [r47709]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL

2006-11-15 21:33 +0000 [r47707]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: We need to ensure timelimit stuff is included as
	  well so warnings get played. (issue #8050 reported by KNK)

2006-11-15 20:50 +0000 [r47701]  Kevin P. Fleming <kpfleming@digium.com>

	* main/file.c: don't try to call fclose() if fopen() failed

2006-11-15 20:31 +0000 [r47698]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Improve SIP history - Never send reply to
	  ACK (again...)

2006-11-15 20:31 +0000 [r47684-47697]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
	  | 4 lines ensure that message duration is included in email
	  notifications for forwarded messages (BE-96, fix by me after
	  corydon used his clue-bat on me) ensure that duration in the
	  message metadata is updated if prepending is done during
	  forwarding (related to BE-96) remove prototype for API call that
	  does not exist ........

	* main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
	  Nov 2006) | 2 lines clear the category's variable tail pointer as
	  well when variables are detached from it ........ r47688 |
	  kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
	  lines when appending a list of variable to a category, ensure the
	  tail pointer points to the last variable in the list ........
	  r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
	  | 2 lines when re-writing the config file, don't repeat the path
	  if it hasn't changed ........

	* main/config.c, /: Merged revisions 47682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
	  | 2 lines ouch... don't use printf, use ast_log/ast_verbose
	  ........

2006-11-15 17:46 +0000 [r47672]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: fix longest match search in find_cli. Trunk already
	  fixed. 1.2 not affected (well, i have no idea, the code is
	  totally different there).

2006-11-15 15:25 +0000 [r47649-47656]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Send error message when we can't allocate
	  SIP dialog, possibly due to limitation of file descriptors.
	  (imported from 1.2)

2006-11-15 04:45 +0000 [r47645]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If NAT detection is turned on or already detected
	  then say NAT is active when setting the remote RTP peer when
	  doing early bridging. (issue #8365 reported by marcelbarbulescu)

2006-11-15 00:19 +0000 [r47641]  Kevin P. Fleming <kpfleming@digium.com>

	* main/term.c: more formatting cleanup, and avoid running off the
	  end of the string

2006-11-15 00:14 +0000 [r47639]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Turn notice about unknown RTCP packet type into a
	  debug message instead.

2006-11-15 00:05 +0000 [r47635]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
	  platforms (this variable is an 'int' anyway, comparing it to
	  'signed long' is not useful)

2006-11-14 22:17 +0000 [r47625-47632]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
	  lines Update copyright information in the ADSI logo blob.
	  ........

	* channels/chan_sip.c: Only keep the video RTP structure around if
	  1. Video support is enabled and 2. A video codec is enabled on
	  the dialog

	* funcs/func_uri.c: Small documentation clarification for
	  URIENCODE. (issue #8294 reported by salaud)

2006-11-14 18:54 +0000 [r47621]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Conversion of res_odbc API to include ast_
	  prefix did not completely transition app_voicemail when
	  ODBC_STORAGE is used (reported on IRC by caio1982, not in
	  bugtracker)

2006-11-14 16:45 +0000 [r47617]  Joshua Colp <jcolp@digium.com>

	* apps/app_amd.c: Use LOG_DEBUG to print out the indication that
	  app_amd is using default settings instead of using LOG_NOTICE.
	  This stops needless logging of this information under normal
	  circumstances. (issue #8361 reported by Seb7)

2006-11-14 16:22 +0000 [r47597-47613]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Update documentation to fit the
	  implementation...

	* /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
	  retransmission system if it's an OPTION packet from peerpoke

2006-11-13 21:28 +0000 [r47584]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
	  lines Initialize global pointers for connection and result to
	  NULL. (issue #8356 reported by james) ........

2006-11-13 20:20 +0000 [r47581]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
	  | 2 lines Having more than 255 old messages caused corruption in
	  the new/old count ........

2006-11-13 19:15 +0000 [r47576]  Steve Murphy <murf@digium.com>

	* main/config.c: This solves bug 8342, whereby a crash occurs under
	  certain circumstances while reading a config file with comments--
	  a call to CB_ADD shouldn't happen if withcomments is zero

2006-11-13 19:11 +0000 [r47573]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/cli.c, channels/chan_sip.c: Re-enable old deprecated
	  commands

2006-11-13 19:10 +0000 [r47572]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: - Don't reply to INVITE already replied
	  to when we get BYE - Declare errmsg as int. Oops.

2006-11-13 18:18 +0000 [r47564]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
	  the messed if, but we all forgot to update the regressions. Until
	  now.

2006-11-13 17:13 +0000 [r47553]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
	  found... just confuses users

2006-11-13 17:08 +0000 [r47542-47551]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
	  lines When sending an SMS with a user data header properly set
	  the UDH flag in the first byte. (issue #8347 reported by
	  hoffmeis) ........

	* main/cli.c: Free full command string upon unregistering of CLI
	  command. Backported from revision 47536 from rizzo.

2006-11-13 16:00 +0000 [r47540]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Only produce error message about sip history
	  once

2006-11-13 05:48 +0000 [r47527]  Russell Bryant <russell@digium.com>

	* configure, acinclude.m4: AC_PROG_SED is included in autoconf
	  2.60, but apparently it is not included in 2.59. So, to maintain
	  compatability with 2.59 since it is a small change, copy this
	  macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
	  #8345)

2006-11-13 05:46 +0000 [r47523-47526]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
	  | 2 lines If the execute fails a second time, make sure that we
	  don't pass back a stale handle ........

	* channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
	  | 2 lines Don't play dialtone if the seizing the channel fails
	  (Bug 7754) ........

2006-11-12 16:12 +0000 [r47507-47513]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
	  DEA!!!)

	* channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
	  UDPTL in sdp...

	* channels/chan_sip.c: - Don't destroy SIP dialog because of a
	  failed T.38 re-invite. Wait for a bye. Final response to a
	  re-invite does not mean that the session dies, only that the
	  re-invite fails. - Keep RTP active during processing of T.38
	  re-invite. If the re-invite fails, RTP needs to remain as before
	  the re-invite. Issue 8338 - darren1713. Please test.

	* channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
	  -Add some comments to t.38 code

2006-11-12 06:23 +0000 [r47492-47497]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
	  4 lines Only do the check to determine whether the channel
	  calling this function is an IAX2 channel when getting the IP
	  address using the special argument, CURRENTCHANNEL. (issue #8341,
	  jcovert) ........

	* Makefile: Add the target "menuconfig" as an alias for the
	  "menuselect" target. This is just a favor to users so that if you
	  accidentally type "make menuconfig" instead of "make menuselect",
	  it still works. (inspired by a comment on IRC from wangster
	  calling me an "especially devious asterisk developer" for having
	  it be menuselect instead of menuconfig. :) )

	* main/term.c: Tweak the formatting of this new function to better
	  conform to coding guidelines.

2006-11-11 02:04 +0000 [r47490]  Matt O'Gorman <mogorman@digium.com>

	* main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
	  safe output!

2006-11-10 22:23 +0000 [r47480]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we don't use 32 bits when we only
	  need one bit.

2006-11-10 21:42 +0000 [r47463-47476]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: ...and make sure that the dialog is
	  destroyed, even if we don't get any answer on the bye... This is
	  the channel that remains dead after the SIP transfer

	* channels/chan_sip.c: Add debug output while trying to trace bug
	  in bug report

	* channels/chan_sip.c: Make sure we destroy dialog...

	* /, channels/chan_sip.c: Small cleanup of handle_request_invite()
	  - imported from 1.2 with changes

2006-11-10 19:47 +0000 [r47462]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
	  callerid name for switches that bork on it.

2006-11-10 18:56 +0000 [r47454]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
	  SDP (alphaque)

2006-11-10 17:13 +0000 [r47444]  Luigi Rizzo <rizzo@icir.org>

	* build_tools/prep_moduledeps: grep -m is not available on BSD, so
	  use head -1 instead

2006-11-10 16:53 +0000 [r47437]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Only split up extension and context if a
	  value exists. (issue #8332 reported by loloski)

2006-11-10 16:51 +0000 [r47436]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Discussion of these CLI changes resulted in
	  more consistency (Bug 8236)

2006-11-10 16:36 +0000 [r47432-47433]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
	  removing them should be LOG_NOTICE, not LOG_DEBUG

	* apps/app_queue.c: reflect addition/removal of dynamic queue
	  members in queue_log, so that people using dialplan replacement
	  for AgentCallbackLogin can still track login/logout (issue #7736,
	  reported/patched by whoiswes but this commit was written by me
	  and covers all three paths for AQM/RQM)

2006-11-10 13:04 +0000 [r47414-47418]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Rip out half implementation of 491 response
	  support, since it wasn't implemented properly and caused memory
	  leaks in the case of us getting 491's, which Asterisk actually
	  sends... Since it is a bit too complicated to fix this, I'll rip
	  it out of 1.4 and put it on the to-do-list for future releases.
	  Now, we handle this as congestion, which it really is. Issue
	  #8331

	* channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
	  Thanks fenlander!

2006-11-10 03:44 +0000 [r47398-47405]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Fix building of chan_h323 by completeing
	  some structure definitions. (issue #8327 reported by Mithraen)

	* apps/app_voicemail.c: Do conversion in a more easier to read and
	  working way for \r, \n, and \t. (issue #8324 reported by
	  johnlange)

2006-11-09 21:26 +0000 [r47391]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, channels/chan_zap.c,
	  build_tools/prep_moduledeps: Work around an issue that caused
	  menuselect to display a bogus description for app_voicemail and
	  chan_zap. These modules use some preprocessor directives to
	  determine what it will report to Asterisk as its description.
	  However, the way we extract this information from the source
	  files for menuselect is not smart enough to figure this out.
	  (issue #8326, #8328)

2006-11-09 16:53 +0000 [r47380]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
	  2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
	  higher as, well, it's apparently going to be removed. This should
	  make all you FC6 fans happy as your Asterisk will now build
	  without any mods. ........

2006-11-09 16:28 +0000 [r47352-47377]  Russell Bryant <russell@digium.com>

	* main/cli.c: fix tab completion for "core debug channel" and "core
	  no debug channel"

	* main/cli.c: Fix "core show channel". Also, fix tab completion for
	  both "core show channel" and "core show channels".

	* main/cli.c: Fix "core debug channel <whatever>". I guess someone
	  needs to go through and audit every CLI command that changed
	  number of arguments ...

	* main/asterisk.c: revert the previous change, which actually
	  modified the deprecated command, "show profile". Now, actually
	  apply the change to "core show profile".

	* main/asterisk.c: Fix argument parsing for the "core show profile"
	  CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)

	* main/cli.c: Fix another CLI command, "core show uptime" ...
	  (issue #8323, reported by johnlange, fixed by myself)

	* main/asterisk.c: fix "core show version" to reflect the new
	  number of arguments for this CLI command (issue #8316, kshumard)

2006-11-08 23:14 +0000 [r47344-47348]  Steve Murphy <murf@digium.com>

	* main/channel.c: This update fixes 7531

	* channels/chan_skinny.c: Committed in behalf of 8190.

2006-11-08 21:46 +0000 [r47333-47338]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: the battle over CLI command formats has broken
	  stuff...

	* channels/chan_sip.c: add simple fix for SDP to report proper
	  sample rate for G.722 media sessions

2006-11-08 17:03 +0000 [r47323-47331]  Russell Bryant <russell@digium.com>

	* utils/streamplayer.c: I occasionally get email from users that
	  are trying to figure out what this does, or due to some
	  misunderstanding as to what it is supposed to do, can't get it to
	  work. So, I have added some text here to hopefully explain what
	  this application does and does not do.

	* channels/chan_gtalk.c: Make this module build again

	* configure, configure.ac, acinclude.m4: Copy the macros from
	  libtool.m4 to our own acinclude.m4 such that libtool is no longer
	  required to be installed to be able to generated the configure
	  script.

2006-11-08 07:43 +0000 [r47309-47310]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)

2006-11-07 23:46 +0000 [r47303]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
	  include/asterisk/stringfields.h, apps/app_voicemail.c,
	  main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
	  channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
	  channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
	  solve the problem in bug 7506. It's a lot of rework to solve a
	  fairly small problem... such is life.

2006-11-07 20:14 +0000 [r47284-47287]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Make MOH work as it did before in
	  chan_local, without this then it can go funky when transfers and
	  MOH are involved. (issue #7671 reported by jmls)

2006-11-07 18:56 +0000 [r47279]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/musiconhold.conf.sample: clean up sample config, and make
	  native file playback the more obvious default choice

2006-11-07 18:38 +0000 [r47275]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: large overhaul to voicemail imap support.
	  Allows support for more imap servers, also a better
	  implementation of several parts of the original work. patch
	  provided by 8033 with major upgrades.

2006-11-07 17:30 +0000 [r47268]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
	  continue.

2006-11-07 13:13 +0000 [r47250]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Fixing the attack shield so it doesn't
	  produce attacks... Issue 8265 - never reply to an ACK

2006-11-07 01:25 +0000 [r47239]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
	  Nov 2006) | 5 lines If random order is enabled for files mode
	  music on hold, set a random initial position, instead of always
	  starting at the first file, and doing the random operation only
	  when switching to the next file. (bug reported by John Lange on
	  the asterisk-dev mailing list) ........

2006-11-04 18:32 +0000 [r47199]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
	  transfer from "john" Thank you!

2006-11-04 18:10 +0000 [r47192-47196]  Russell Bryant <russell@digium.com>

	* main/cli.c: Fix another bug in "core set debug" ...

	* main/asterisk.c, main/cli.c: Really fix the "core set debug" and
	  "core set verbose" CLI commands.

	* main/cli.c: fix the "atleast" option to the "core set verbose"
	  and "core set debug" CLI commands

2006-11-03 23:17 +0000 [r47176]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This fix introduced via bug 8233

2006-11-03 17:53 +0000 [r47107-47108]  Luigi Rizzo <rizzo@icir.org>

	* bootstrap.sh: align bootstrap.sh with the version in trunk (needs
	  to be blocked as it is already in trunk)

	* configure.ac: add proper environment vars to detect modules on
	  freebsd. (already applied to trunk so it needs to be blocked
	  there)

2006-11-02 23:49 +0000 [r47051-47053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
	  channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
	  changes making the CLI more consistent with "category verb
	  arguments" (continuation of issue 8236)

	* main/config.c, main/cli.c, main/channel.c, main/manager.c,
	  channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
	  main/http.c, main/file.c, main/logger.c, main/image.c,
	  res/res_indications.c, main/asterisk.c, res/res_odbc.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
	  channels/chan_local.c, main/frame.c, channels/chan_sip.c,
	  res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
	  res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
	  Reverse change of "show" to "list" and make several other
	  commands more consistent with "category verb arguments"

2006-11-02 19:56 +0000 [r46992-47015]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Move check for codec translation to
	  sip_call() instead of in add_sdp. No one bothers with the result
	  of add_sdp anyway... Yet...

	* channels/chan_sip.c: Disable code for T38 over TCP and RTP since
	  there's no trace of actual functionality for it :-)

2006-11-02 17:49 +0000 [r46965]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
	  Nov 2006) | 3 lines ignore files in a music on hold directory
	  that begin with '.' (issue #8249, cboie) ........

2006-11-02 17:17 +0000 [r46963]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix

2006-11-02 16:45 +0000 [r46937]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't send INVITE when we have determined
	  that we can't offer any audio formats due to lack of transcoding
	  support (or incorrect configuration)

2006-11-02 16:06 +0000 [r46930]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
	  lines Repeat after me oej: I will at least make sure my code
	  compiles before I commit it. ........

2006-11-02 15:24 +0000 [r46901]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)

2006-11-02 14:02 +0000 [r46845-46883]  Russell Bryant <russell@digium.com>

	* /, main/callerid.c: Add the missing call to free described in
	  issue #8268. Also, add a bunch of missing calls to free in
	  callerid_feed_jp().

	* main/say.c: fix saying one hundred and two hundred in hebrew
	  (issue #7810, eldadran)

	* Makefile, configure, codecs/gsm/Makefile, configure.ac,
	  build_tools/strip_nonapi, makeopts.in: Fixes for
	  cross-compilation on mips (issue #8058, ywalther, with some
	  modifications)

	* aclocal.m4, build_tools/menuselect-deps.in, configure,
	  build_tools/embed_modules.xml, configure.ac: Add a check in the
	  configure script to determine whether ld is GNU ld or not. This
	  is needed because module embedding only works for gnu ld. GNU ld
	  is now listed as a dependency for all of the module embedding
	  options in menuselect. (issue #8143)

2006-11-01 20:35 +0000 [r46822]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c: bind address support from bug 8164

2006-11-01 19:49 +0000 [r46802]  Steve Murphy <murf@digium.com>

	* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
	  accept longer strings or mass confusion and a lot of lost time is
	  the result

2006-11-01 18:39 +0000 [r46780]  Joshua Colp <jcolp@digium.com>

	* main/Makefile: Force poll() emulation for Darwin to always be on.
	  It's too broken to consider being used. This resolves the console
	  issue OSX users have been seeing. I would have liked to autoconf
	  this but I haven't been able to come up with a test case that
	  works. Que sera.

2006-11-01 18:26 +0000 [r46778]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
	  9 lines soxmix and Asterisk expect different file extensions for
	  certain formats. This was already handled for the wav49 format.
	  However, it was not handled for ulaw and alaw. I fixed this in
	  such a way that using the alternate extensions for ulaw and alaw
	  will only happen if we know we're calling soxmix, and not a
	  custom script defined using the MONITOR_EXEC variable. The wav49
	  processing was left alone so that external scripts will see no
	  behavior change. (issue #7550, reported by mnicholson, proposed
	  patch by junky, committed fix is a bit different) ........

2006-11-01 18:21 +0000 [r46775]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: It's another round of chan_iax2 fixes!
	  Should hopefully fix the deadlock issues people have been
	  reporting. IAXtel now has qualify turned on for 800 peers and it
	  is handling it fine.

2006-11-01 17:48 +0000 [r46760]  Steve Murphy <murf@digium.com>

	* main/config.c: Cleanups suggested by Russell.

2006-11-01 16:39 +0000 [r46744]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Prevent an infinite loop when config
	  processing gets to a jitterbuffer option

2006-10-31 22:02 +0000 [r46716]  Jason Parker <jparker@digium.com>

	* main/translate.c: Fix "core show translation" output. Issue
	  #8243, patch by Damin.

2006-10-31 21:47 +0000 [r46711-46714]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/translate.h, main/translate.c: add an API so
	  that translators can activate/deactivate themselves when needed

	* include/asterisk/translate.h, main/translate.c: revert changes
	  that were the wrong way to address this... proper fix coming

	* main/translate.c: let's set the seen flag early enough to
	  actually make a difference...

	* include/asterisk/translate.h, main/translate.c: don't re-do setup
	  operations for translators that can dynamically register
	  themselves

2006-10-31 10:56 +0000 [r46583-46631]  Olle Johansson <oej@edvina.net>

	* main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
	  #8089 - Fix the ENUM support (picking one record by number).
	  Thanks otmar!

	* /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
	  when we're supposed to support ;rport. Issue #7473.

	* /, channels/chan_sip.c: If peer fails ACL check, fail peer at
	  REGISTER

	* channels/chan_sip.c: Fix T38 too. Thanks, tgrman !

2006-10-31 06:30 +0000 [r46554-46563]  Russell Bryant <russell@digium.com>

	* contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
	  boot process to ensure it starts after stuff like MySQL (issue
	  #8253, Alric)

	* /, main/utils.c: Merged revisions 46560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
	  3 lines When handling the case where the hostname is just an IPV4
	  numeric address, be sure to set the address type. (issue #8247,
	  alexr) ........

	* /, res/res_agi.c: Merged revisions 46557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
	  3 lines fix some copy/paste bugs in the checking of arguments for
	  the "control stream file" AGI command (issue #8255, mnicholson)
	  ........

	* main/translate.c: Add a small tweak to the code that checks to
	  see whether destination formats are translatable based on the
	  source format. If we have already determined that there is no
	  translation path in one direction, don't bother checking the
	  other direction.

2006-10-30 22:19 +0000 [r46511-46526]  Kevin P. Fleming <kpfleming@digium.com>

	* main/translate.c: when unregistering a translator, don't rebuild
	  the translation matrix unless needed when filtering formats out
	  of an offer, ensure we check for translation ability in both
	  directions

	* include/asterisk/linkedlists.h: ensure that items removed from a
	  list are always unlinked from the list (next pointer set to NULL)

2006-10-30 21:09 +0000 [r46474-46506]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Don't explicitly link in crypt as it is
	  not used on some platforms.

	* channels/chan_iax2.c: We need to lock the pvt structure during
	  retransmission as another worker thread may be doing something as
	  well.

2006-10-30 16:27 +0000 [r46382-46433]  Olle Johansson <oej@edvina.net>

	* main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
	  include/asterisk/doxyref.h, channels/chan_sip.c,
	  main/ast_expr2f.c, include/asterisk/module.h,
	  formats/format_ogg_vorbis.c, main/app.c,
	  include/asterisk/channel.h, include/asterisk/lock.h,
	  include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
	  kshumard. An extra big thankyou is given to everyone that
	  contributes to doxygen! THANK YOU!

	* main/rtp.c, /: Bind RTCP to the same IP as RTP

	* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
	  redirects (imported from 1.2)

	* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
	  wrong content-type (imported from 1.2, modified)

	* channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
	  was reported for trunk, but obviously exists in 1.4 too.

	* channels/chan_sip.c: Restoring the old logic, since working
	  around it and fixing it seemed too complicated. - The
	  SIP_OUTGOING flag indicates the direction of the last transaction
	  in the dialog. - The initreq stores the last request in the
	  dialog, the request that opened the latest transaction. Please
	  now retry all the 1.4 bug reports with mixed to/from headers,
	  tags etc in ACK, BYE, CANCEL. Thanks!

	* channels/chan_sip.c: Accepting a message twice may be
	  misinterpreted...

	* channels/chan_sip.c: - 183 is not reliable message... - Error
	  should not have SDP

2006-10-28 16:37 +0000 [r46377]  Joshua Colp <jcolp@digium.com>

	* utils/Makefile: Don't build muted on OpenBSD, it is not
	  supported.

2006-10-27 19:03 +0000 [r46370]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: move the copy of the default settings to the
	  global settings back out of process_zap, so that they aren't
	  overwritten when process_zap is called multiple times

2006-10-27 18:29 +0000 [r46367]  Olle Johansson <oej@edvina.net>

	* contrib/asterisk-ng-doxygen: Put some doxygen pressure on
	  Christian :-)

2006-10-27 17:39 +0000 [r46358-46363]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
	  res/res_musiconhold.c: We should always be using _exit() after a
	  fork() or vfork() instead of exit(). This is because exit() does
	  some extra cleanup which in some implementations of vfork(), for
	  example, can actually modify the state of the parent process,
	  causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)

	* channels/chan_zap.c: Instead of iterating all of the options once
	  to look for jitterbuffer options, and then again for everything
	  else, move the processing of jitterbuffer options into the main
	  loop so that there are no erroneous messages about ignoring
	  unknown options. (issue #8226)

2006-10-27 10:03 +0000 [r46351-46353]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 46350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
	  1 line fixed a bug which caused chan_misdn to try to allocate 2
	  times the same channel on high load, which then caused
	  instability of mISDN. removed a useless function from isdn_lib.c
	  ........

	* channels/misdn_config.c: fixed not compile issue, which was just
	  introduced

	* channels/misdn_config.c, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  Merged revisions 46176 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
	  1 line added nttimeout option to configure wether we disconnect
	  calls on NT timeouts or not during an overlapdial session
	  ........

2006-10-26 17:57 +0000 [r46335-46340]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
	  lines oops - somebody forgot to change this - long ago, probably.
	  ........

	* CHANGES: grammar check

2006-10-26 16:38 +0000 [r46331]  Olle Johansson <oej@edvina.net>

	* CHANGES: Corrections to changes (Multiparking is not included)

2006-10-26 16:31 +0000 [r46329]  Russell Bryant <russell@digium.com>

	* main/translate.c: - If the source has no audio or no video
	  portion, do not call powerof() to get the format index. - Don't
	  run through the audio and video loops if there is no audio or
	  video portion of the source If 0 is passed to powerof, it will
	  return -1. This value of -1 was then being used as an array index
	  in these loops, which caused a crash on some systems. Other than
	  this issue, this code works as we expected it to. If a format is
	  not in the source, and we have to translation path to it, it is
	  not offered in the list of acceptable destination formats. (fixes
	  issue #8231)

2006-10-26 12:15 +0000 [r46317]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: update to reflect G.722 addition

2006-10-26 04:18 +0000 [r46298]  Russell Bryant <russell@digium.com>

	* doc/backtrace.txt: update backtrace documentation to reflect
	  changes in 1.4 (issue #8230, kshumard)

2006-10-26 01:37 +0000 [r46287]  Mark Spencer <markster@digium.com>

	* main/config.c, main/manager.c: Fix config comment code
	  preservation code (thanks murf!)

2006-10-25 20:14 +0000 [r46276]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Old todo note - Don't add Contact header on
	  BYE and Cancel

2006-10-25 19:24 +0000 [r46253-46255]  Russell Bryant <russell@digium.com>

	* configure.ac: fix error output when checking for openh323 to
	  refer to openh323 instead of pwlib (issue #8222, misaksen)

2006-10-25 19:16 +0000 [r46252]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Somewhat ugly code to try to fix issue
	  #7608. Since the problem was not very well defined, the fix is a
	  bit fuzzy too... Thanks to Luigi for accidentally spotting the
	  possible problem!

2006-10-25 19:08 +0000 [r46249]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: update warning message to include "agi" option
	  (issue #8225, jmls)

2006-10-25 18:13 +0000 [r46237-46248]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: use 1.4.3 extra sounds with corrected silence
	  files

	* sounds/sounds.xml, sounds/Makefile: add support for prebuilt
	  G.722 prompts and music on hold files

2006-10-25 15:56 +0000 [r46214-46216]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: show settings doesn't produce a list of
	  similar objects, it should stay a "show"

2006-10-25 14:32 +0000 [r46200]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
	  channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
	  pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
	  main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
	  cdr/cdr_custom.c, channels/chan_mgcp.c,
	  apps/app_parkandannounce.c, apps/app_voicemail.c,
	  channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
	  res/res_adsi.c, main/utils.c, apps/app_ices.c,
	  pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
	  apps/app_getcpeid.c: apparently developers are still not aware
	  that they should be use ast_copy_string instead of strncpy... fix
	  up many more users, and fix some bugs in the process

2006-10-25 04:58 +0000 [r46165]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/pbx.c: WaitExten truncates decimals of times to wait,
	  instead of accepting them (Bug 8208)

2006-10-25 00:26 +0000 [r46152-46154]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
	  channels/chan_h323.c, channels/chan_iax2.c,
	  include/asterisk/frame.h: add passthrough and file format support
	  for G.722 16KHz audio (issue #5084, original patch by andrew,
	  updated by mithraen)

	* channels/chan_sip.c, main/translate.c: code zone experiment:
	  don't offer formats in the outbound INVITE that aren't either
	  passthrough or translatable

	* main/translate.c: if multiple translators are registered for the
	  same source/dest combination, ensure that the lowest-cost one is
	  always inserted earlier in the list

2006-10-24 20:30 +0000 [r46142]  Mark Spencer <markster@digium.com>

	* res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
	  #8147)

2006-10-24 19:29 +0000 [r46130]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: We need to initialize our scheduler pthread
	  condition... yes.

2006-10-24 08:34 +0000 [r46114-46117]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: merge 45152 don't leak descriptors in http.c

	* channels/chan_sip.c: merge 45966 refer_to_domain potentially
	  containing options

	* channels/chan_sip.c: merge 46026 improper checks on get_header()
	  return values

	* channels/chan_sip.c: merge 46045 prevent NULL args to
	  ast_strdupa() in chan_sip.c

2006-10-24 05:23 +0000 [r46093]  Russell Bryant <russell@digium.com>

	* Makefile: Restore the ability to remove the firmware directory
	  without causing the installation to fail (issue #8111)

2006-10-24 03:53 +0000 [r46080-46083]  Kevin P. Fleming <kpfleming@digium.com>

	* main/translate.c: ensure that the translation matrix is properly
	  lock-protected every place it is used

	* include/asterisk/translate.h, main/translate.c: add an API call
	  to allow channel drivers to determine which media formats are
	  compatible (passthrough or transcode) with the format an existing
	  channel is already using

	* doc/imapstorage.txt: simplify and correct voicemail IMAP storage
	  build instructions

2006-10-24 03:01 +0000 [r46078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/channel.c: Pass through a frame if we don't know what it is,
	  rather than trying to pass a NULL, which will segfault a channel
	  driver (Bug 8149)

2006-10-24 01:27 +0000 [r45999-46067]  Russell Bryant <russell@digium.com>

	* utils/muted.c, utils/ael_main.c: In muted.c, check the return
	  value of strdup. In ael_main.c, check the return value of calloc.
	  (issue #8157) In passing fix a few minor bugs in ael_main.c. The
	  last argument to strncpy() was a hard-coded 100, where it should
	  have been 99. I changed this to use sizeof() - 1.

	* apps/app_meetme.c: Fix the descriptions of some of the
	  MeetMeAdmin options (issue #8098, mflorell)

	* res/res_jabber.c: don't crash when an incoming message has no
	  "from" (issue #8205, jmls)

2006-10-23 00:27 +0000 [r45928]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
	  lines Don't leak memory mmmk? ........

2006-10-22 21:44 +0000 [r45916]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
	  Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
	  couldn't be initialized it would cause a segfault after 'reload'.
	  Reported by Drew/Matt thx. ........

2006-10-21 18:49 +0000 [r45818]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c: Add a couple missing unregistrations of
	  manager actions and remove duplicate unregistrations of
	  applications. (issue #8194, jmls)

2006-10-21 18:48 +0000 [r45775-45817]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: Don't use promotion on Darwin because it doesn't
	  seem to work quite right in all cases, this should solve the
	  unresolved symbol issue people have been seeing.

	* Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
	  installed in the proper location (reported on asterisk-dev
	  mailing list)

2006-10-20 07:44 +0000 [r45741]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Let's understand SIP: - REFER can create
	  dialog, Asterisk does not support it yet - NOTIFY can create
	  dialog in Asterisk's implementation (voicemail) even though we
	  don't support the server side of it. In this case, the standard
	  is a side issue ;-) - Added extened functionality for unsupported
	  methods (PING, PUBLISH) so we don't create PVT's for those
	  either. Russellb needs to judge what to do with this in 1.2, but
	  I think the current implementation n 1.2 is a bug since we're
	  sending bad replies to NOTIFY and REFER outside of dialogs

2006-10-19 17:24 +0000 [r45678-45694]  Joshua Colp <jcolp@digium.com>

	* res/res_jabber.c: Let's remember to unregister JabberStatus too
	  (issue #8184 reported by jmls)

	* /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
	  2006) | 2 lines Respect language selection when seeing if the
	  file exists (issue #8178 reported by mnicholson) ........

	* channels/chan_sip.c: If the jitterbuffer is forced on then we
	  can't partially bridge (reported by wangster on #asterisk-dev)

2006-10-19 00:59 +0000 [r45622]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Don't leak the actual thread-specific
	  sip_pvt struct

2006-10-18 23:49 +0000 [r45621]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't leak memory when a chan_sip thread is
	  destroyed that has a thread-local temp_pvt allocated

2006-10-18 21:03 +0000 [r45595]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c: Don't modify things if we are using vfork as
	  this is very bad and may cause unexpected behavior (issue #7970
	  reported by Nick Gavrikov)

2006-10-18 11:54 +0000 [r45517]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: remove duplicate declarations

2006-10-18 04:09 +0000 [r45464]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: merge from trunk: move ast_variables_destroy() to a
	  better place in handle_uri() to avoid leaking memory on non
	  existing files.

2006-10-18 03:02 +0000 [r45452]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Don't segfault if you're using a channel driver that
	  doesn't turn RTCP on

2006-10-18 02:41 +0000 [r45439-45441]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't attempt to access private data members of
	  the pthread_mutex_t object, because this does not work on all
	  linux systems. Instead, just access the reentrancy field in the
	  ast_mutex_info struct when DEBUG_THREADS is enabled. If
	  DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
	  DEBUG_THREADS on as well. (issue #8139, me)

	* configs/sip_notify.conf.sample: update entry to reboot a snom
	  phone (issue #7850, pnlarsson)

2006-10-17  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta3 released.

2006-10-17 22:31 +0000 [r45408-45410]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/stringfields.h, main/ast_expr2.c,
	  main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
	  optimize the 'quick response' code a bit more... no more malloc()
	  or memset() for each response expand stringfields API a bit to
	  allow reusing the stringfield pool on a structure when needed,
	  and remove some unnecessary code when the structure was being
	  freed

2006-10-17 20:38 +0000 [r45378-45381]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't create a "real" pvt structure for
	  requests that shouldn't be able to create one. Instead use a
	  temporary pvt and fill it with enough information so we can send
	  a reply.

2006-10-17 17:39 +0000 [r45329]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Adding information about Marks
	  direct-RTP hack to the docs...

2006-10-17 17:22 +0000 [r45327]  Kevin P. Fleming <kpfleming@digium.com>

	* LICENSE: provide licensing language for IAXy firmware file

2006-10-16 20:06 +0000 [r45246-45280]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
	  directed pickup (BE-85).

2006-10-16 13:59 +0000 [r45196-45213]  Olle Johansson <oej@edvina.net>

	* CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
	  your support!

	* channels/chan_sip.c: Don't destroy dialog for unexpected REFER
	  response...

2006-10-14 04:38 +0000 [r45143]  Steve Murphy <murf@digium.com>

	* funcs/func_rand.c: update the doc string for both AEL and
	  extensions.conf users.

2006-10-13 23:02 +0000 [r45125]  Kevin P. Fleming <kpfleming@digium.com>

	* main/acl.c don't drop the entire permit/deny list when an attempt
	  is made to add an invalid entry (BE-92)

2006-10-13 21:06 +0000 [r45104-45106]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c: Clear the quiet flag too since we are
	  restarting a recognition again (reported on -dev by Stephan
	  Edelman)

	* res/res_speech.c: Check return value from engine in case of
	  failure (ie: out of licenses) (reported on -dev mailing list)

2006-10-13 20:52 +0000 [r45103]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest17 (added),
	  pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest17 (added),
	  pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
	  this release via these changes

2006-10-13 19:19 +0000 [r45088]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: avoiding warning, fixing potential bug

2006-10-13 18:42 +0000 [r45051-45079]  Joshua Colp <jcolp@digium.com>

	* codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
	  codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
	  codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
	  codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
	  codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
	  codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
	  codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
	  codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
	  codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
	  codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
	  codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
	  codecs/lpc10/analys.c, codecs/lpc10/onset.c,
	  codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
	  codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
	  codecs/lpc10/median.c, codecs/lpc10/encode.c,
	  codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
	  codecs/lpc10/invert.c: And file said... let the compiler warnings
	  STOP!

	* apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
	  reported by mnicholson)

	* apps/app_playback.c: Move say.conf existence check to do_say
	  function since it is called from multiple places (issue #8144
	  reported by kshumard)

2006-10-13 16:19 +0000 [r45049]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
	  we have multiple bindings (reported on asterisk-dev)

2006-10-13 16:01 +0000 [r45031-45040]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Complete merging in RPID screen changes
	  (issue #8101 reported by hristo, patch by oej in revision 44757)

	* main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
	  the background refresh item back into the scheduler if enabled
	  since it is deleted during reload. (issue #8142 reported by
	  p_lindheimer)

2006-10-13 15:41 +0000 [r45027]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: use a configure script test for PMTU discovery
	  control instead of just assuming it's available on Linux

2006-10-13 14:45 +0000 [r44994-45026]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
	  echocandisable issues when bridged. this caused a kernel panic
	  sometimes.. also some minor formatting fixes

	* channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
	  got a wrong isdn cause at RELEASE_COMPLETE

2006-10-12 22:07 +0000 [r44992]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: merge formatting and minor code
	  simplifications from trunk

2006-10-12 20:34 +0000 [r44982]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c: fix for bug 7764.

2006-10-12 19:14 +0000 [r44956-44971]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: we can only send one 'a=ptime' attribute per
	  media session, not one for each format

	* main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
	  main/utils.c: ensure that IAX2 and SIP sockets allow UDP
	  fragmentation when running on Linux (thanks to Brian Candler on
	  the asterisk-dev list for the tip)

2006-10-12 16:56 +0000 [r44945]  Russell Bryant <russell@digium.com>

	* main/manager.c: fix a silly typo in a comment that I saw while
	  reading the commit list

2006-10-12 16:08 +0000 [r44942]  Joshua Colp <jcolp@digium.com>

	* Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
	  #8135 reported by ssokol)

2006-10-12 12:55 +0000 [r44921]  Nadi Sarrar <ns@beronet.com>

	* main/manager.c: append_event must be called while holding the
	  session lock

2006-10-12 10:24 +0000 [r44911]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: change some debug output to use LOG_DEBUG
	  instead of verbose output

2006-10-11 16:57 +0000 [r44888]  Jason Parker <jparker@digium.com>

	* main/db1-ast/Makefile: These are already set by the parent
	  Makefile.. There is no need to have this here (it doesn't
	  actually work anyways).

2006-10-11 09:18 +0000 [r44854]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: removed warning because of missing
	  prototype declaration

2006-10-10 19:23 +0000 [r44830]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Do not set default/global values in the
	  variable declaration, set it in reload_config()

2006-10-10 17:21 +0000 [r44819]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Move some stuff around so that a NOTIFY
	  dialog won't hang around until the end of the world under certain
	  circumstances

2006-10-10 16:44 +0000 [r44809]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
	  CHANNEL() function sometime mix parameter and value

2006-10-10 16:42 +0000 [r44808]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_logic.c: Lost of a bit of logic when this was
	  simplified between 1.2 and 1.4 (Bug 8117)

2006-10-10 16:30 +0000 [r44806]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Bail out if we have no refer structure and
	  we get a refer response

2006-10-10 16:21 +0000 [r44805]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: more merge from trunk (comments and change a
	  static function name)

2006-10-10 15:23 +0000 [r44788]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only set DTMF information if an RTP
	  structure exists

2006-10-10 13:50 +0000 [r44786]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
	  support of dynamically enabling hdlc on bchannels

2006-10-10 08:25 +0000 [r44776-44777]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: whitespace changes related to previous
	  commit

	* channels/chan_sip.c: merge a few code simplifications that have
	  gone into trunk during last week, to reduce differences between
	  the two branches and make porting fixes easier.

2006-10-09 16:12 +0000 [r44764]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix a problem where phones that go
	  "missing" never got unregistered. Issue #8067, reported by pj,
	  patch by Anthony LaMantia (with minor whitespace modifications)

2006-10-09 15:46 +0000 [r44759-44760]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
	  the deadlock

	* channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
	  (issue #8115 reported by vazir)

2006-10-08 14:14 +0000 [r44746]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: do not dereference p if we
	  know it is NULL

2006-10-07 14:39 +0000 [r44684]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
	  caller's transfer capability too

2006-10-07 11:37 +0000 [r44650-44665]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: put common code in a
	  function to avoid repetitions.

	* channels/chan_sip.c: remove hardwired usage of 5060, use
	  DEFAULT_SIP_PORT instead

	* channels/chan_sip.c: option_debug checking
	  before printing to debug channel.

	* channels/chan_sip.c: backport simplifications on sip_register,
	  usage of ast_set2_flag(), and fixes to the handling of failed
	  module loading.

	* channels/chan_sip.c: improve and document function
	  get_in_brackets(), introducing a helper function
	  find_closing_quote() of more general use.

2006-10-06 21:28 +0000 [r44629-44631]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/linkedlists.h: ensure that mutex locks inside
	  list heads are initialized properly on platforms that require
	  constructor initialization (issue #8029, patch from timrobbins)

	* CHANGES: remove Jingle as per mog

2006-10-06 21:08 +0000 [r44628]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Remove the seqno check for RFC2833, the handler is
	  smart enough to not need it.

2006-10-06 21:07 +0000 [r44627]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: various cleanups

2006-10-06 18:46 +0000 [r44581-44605]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: When the sequence number rolls over then reset the
	  recorded sequence number for DTMF (issue #8106 reported by
	  bungalow)

	* main/file.c: Even more frames to treat as though the remote side
	  disappeared (issue #8097 reported by eldadran)

2006-10-06 15:59 +0000 [r44567]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c, main/http.c: make sure sockets are blocking when
	  they should be blocking.

2006-10-06 12:53 +0000 [r44559-44563]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed segfault which happens during
	  hold/transfer action

	* channels/chan_misdn.c: if INFORMATION Message come with keypad
	  instead of called party number, we just use the keypad as called
	  party number.

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  added the option 'reject_cause' to make it possible to set
	  the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
	  which is automatically rejected because chan_misdn does not
	  support that kind of callwaiting. Therefore chan_misdn supports
	  now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
	  now gets the info if the requested channel is incoming or
	  outgoing to make the 3. channel possible

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
	  removed a useless bc field, added setting of frame.delivery fields,
	  some minor code cleanups

2006-10-05 19:57 +0000 [r44502]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Treat busy control frames as hangup in the file streaming
	  core (issue #8097 reported by eldadran)

2006-10-05 18:21 +0000 [r44488]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
	  Many thanks to Doug!

2006-10-05 18:01 +0000 [r44486]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
	  hanging by a thread if the other side is already setup with T.38

2006-10-05 16:10 +0000 [r44476]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: don't segfault when an argument without a close
	  parenthesis is found stop parsing as soon as that situation
	  occurs

2006-10-05 15:22 +0000 [r44465-44466]  Steve Murphy <murf@digium.com>

	* CHANGES: I put the accumulated changes from the commit logs and
	  inspection, into CHANGES. Hope everyone approves!

	* configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
	  install process sticks muted.conf in /etc/asterisk, so that's
	  where muted should look for it, right?

2006-10-05 02:40 +0000 [r44450]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't totally bail out if T.38 was
	  negotiated

2006-10-05 01:42 +0000 [r44433-44436]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix Polycom presence notification again

2006-10-04 22:52 +0000 [r44407-44409]  Luigi Rizzo <rizzo@icir.org>

	* utils/Makefile: as far as i can tell astman only uses newt...

	* Makefile: put linker flags in ASTLDFLAGS where they belong

2006-10-04 21:17 +0000 [r44390-44393]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
	  requests add workaround for new Polycom firmware SUBSCRIBE
	  requests (bug is known to exist in 2.0.1 firmware)

	* include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
	  work

2006-10-04 19:57 +0000 [r44380]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
	  pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
	  pbx/ael/ael-test/ael-test16/extensions.ael (added),
	  pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
	  pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
	  pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
	  problems reported in bug 8090

2006-10-04 19:47 +0000 [r44378]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
	  main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
	  channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
	  main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
	  include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
	  channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
	  main/devicestate.c, main/utils.c, res/res_musiconhold.c,
	  channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
	  thread creation code a bit reduce standard thread stack size
	  slightly to allow the pthreads library to allocate the stack+data
	  and not overflow a power-of-2 allocation in the kernel and waste
	  memory/address space add a new stack size for 'background'
	  threads (those that don't handle PBX calls) when LOW_MEMORY is
	  defined

2006-10-04 17:04 +0000 [r44337-44365]  Steve Murphy <murf@digium.com>

	* configs/muted.conf.sample: I've been meaning to add some
	  explanation about muted... here it is

	* configs/manager.conf.sample: CLI reverbification update to this
	  config file

	* apps/app_macro.c: In response to bug 7776, a Warning has been
	  added to the doc string for Macro().

2006-10-04 00:25 +0000 [r44322]  Kevin P. Fleming <kpfleming@digium.com>

	* main/asterisk.c, main/loader.c, main/term.c, Makefile,
	  include/asterisk.h: ensure that local include files are always
	  used avoid a duplicate function name (term_init())

2006-10-03 22:35 +0000 [r44312]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
	  client without resource.

2006-10-03 20:18 +0000 [r44298]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix a logic error in my previous fix to the queue
	  reload code

2006-10-03 18:42 +0000 [r44286]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Change default presentation indicator
	  to "user provided not screened" if octet 3a missed in
	  CallingPartyNumber IE

2006-10-03 18:35 +0000 [r44284]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Use VideoSupport instead so it is considered
	  a valid XML attribute name. (issue #8075 reported by renemendoza)

2006-10-03 18:30 +0000 [r44283]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Fix preparation of type and
	  presentation of calling number

2006-10-03 00:01 +0000 [r44240]  Matt O'Gorman <mogorman@digium.com>

	* doc/jingle.txt, channels/chan_jingle.c (removed),
	  include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
	  res/res_jabber.c: updated res_jabber for even better component
	  support, soon will be jep-0100 compliant. also removed
	  chan_jingle and infromed info from jingle.txt, chan_gtalk still
	  works and should be used in this version.

2006-10-02 20:11 +0000 [r44199-44215]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Change the fd on the I/O context in case it
	  changed during the reload, which is indeed possible. (issue #7943
	  reported by eclubb)

	* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
	  instead of hardcoding the path for the error message (issue #7942
	  reported by eclubb)

2006-10-02 18:52 +0000 [r44186]  Paul Cadach <paul@odt.east.telecom.kz>

	* configs/users.conf.sample, pbx/pbx_config.c: Missed part of
	  userconf functionality for chan_h323

2006-10-02 17:25 +0000 [r44169]  Joshua Colp <jcolp@digium.com>

	* main/io.c: Shrink when current_ioc is unused. It is set to -1 when
	  unused, not 0. (issue #7941 reported by eclubb)

2006-10-02 17:16 +0000 [r44166-44167]  Paul Cadach <paul@odt.east.telecom.kz>

	* doc/realtime.txt: Typo fix

	* channels/chan_h323.c: Optimization of oh323_indicate(): less
	  locks - less problems, plus single exit point

2006-10-02 02:38 +0000 [r44146]  Mark Spencer <markster@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
	  you're not talking about a channel :)

2006-10-01 19:32 +0000 [r44135]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Do not simulate any audio tones if we got
	  PROGRESS message

2006-10-01 18:30 +0000 [r44111-44125]  Russell Bryant <russell@digium.com>

	* Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
	  be empty. The cause is that since ASTDATADIR is explicitly
	  exported using "export ASTDATADIR" at the top of the Makefile,
	  make no longer considers the variable "undefined", so the
	  Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
	  #8063, reported by akohlsmith, fixed by me)

	* configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
	  option in the sample queues.conf (issue #8065, adamg)

2006-10-01 15:01 +0000 [r44109]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: sync with trunk - move variable declarations
	  to the beginning of a block.

2006-09-30 19:20 +0000 [r44090]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/rtp.c: Allow one-way RTP streams (device->Asterisk)

2006-09-30 16:28 +0000 [r44080]  Luigi Rizzo <rizzo@icir.org>

	* codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
	  build problems: - with AST_DEVMODE, building codecs/lpc10 fails
	  because of lots of warnings, and the configure step in editline
	  fails as well. Fix this by removing the -Werror in these steps. -
	  on FreeBSD (but probably on other platforms as well), the final
	  link of asterisk fails because AST_LIBS was not exported to the
	  subdirs Makefiles. Add a proper fix in the top-level Makefile (a
	  possible alternative way is to add "export AST_LIBS" near the
	  beginning of the file). With this fix, i believe that some of the
	  platform-specific conditionals in main/Makefile are redundant
	  (because they should be already dealt with in the top level
	  Makefile) but i don't have a platform to check. Merging to head
	  will happen in a moment.

2006-09-30 16:12 +0000 [r44068-44078]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
	  of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
	  by phsultan with a small fix by me, myself or I. Thanks,
	  Philippe! (This was caused by my changes to the transaction
	  handling)

	* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
	  VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
	  sends ACK not on OK message only (when remote party answers) but
	  on RINGING message too, so when we send 200 OK message, we get
	  unidentified ACK message (because INVITE acknowledged on RINGING
	  message already), so 200 OK retransmits within its retransmission
	  interval then call gets dropped. If someone else knows how to
	  provide workaround for such cases, please, fix it in correct way.
	  Thanks to ssh from #asteriskru for provide access to his box to
	  study and fix this case.

2006-09-29 22:51 +0000 [r44055-44057]  Kevin P. Fleming <kpfleming@digium.com>

	* agi, utils: ignore temporary files made by the Makefiles during a
	  build

	* codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
	  codecs/Makefile, utils/Makefile, configure,
	  build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
	  Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
	  pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
	  system bugs, and convert Makefiles to be compatible with GNU make
	  3.80

2006-09-29 22:35 +0000 [r44053]  Jason Parker <jparker@digium.com>

	* main/asterisk.c, main/cli.c: Fix a bug with the removal of
	  'atleast' argument to 'core verbose' and 'core debug'. Add that
	  argument back in.

2006-09-29 21:09 +0000 [r44022-44043]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
	  carefully when no CallingNumber IE available

	* channels/h323/ast_h323.cxx: Fake display name by called number on
	  incoming calls (until passing connected number/connected name is
	  not implemented)

	* channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
	  includes

	* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
	  pass TON/PRESENTATION information - original
	  H323Connection::SendSignalSetup() destroys Q.931 fields.

2006-09-29 18:49 +0000 [r44011-44012]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile: yet another place where we were not using the
	  correct CFLAGS by default

	* main/Makefile: missed one conversion to ASTCFLAGS

2006-09-29 18:30 +0000 [r44009]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
	  TON/PRESENTATION information too

2006-09-29 18:25 +0000 [r43952-44008]  Kevin P. Fleming <kpfleming@digium.com>

	* main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
	  main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
	  Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
	  CFLAGS and LDFLAGS for build of Asterisk components, because they
	  are also then used for non-Asterisk components (like menuselect);
	  use our own variables instead

	* configure, configure.ac: support --without-curl in configure
	  script

	* Makefile.rules: another cross-compile fix

	* Makefile: a couple more environment settings that can't leak into
	  the menuselect build

	* main/cli.c: proper fix for ast_group_t change

	* include/asterisk/lock.h: eliminate compiler warning when
	  DEBUG_CHANNEL_LOCKS is enabled and users of this header file
	  don't also include channel.h

2006-09-28 20:11 +0000 [r43944]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Fix incorrect argument order for member names,
	  on persisted members. Issue 8047, patch by jmls.

2006-09-28 18:05 +0000 [r43932-43933]  Joshua Colp <jcolp@digium.com>

	* apps/app_playback.c, res/res_monitor.c,
	  include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
	  channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  main/udptl.c, main/frame.c, funcs/func_timeout.c,
	  channels/chan_sip.c, apps/app_festival.c,
	  channels/iax2-provision.c, apps/app_alarmreceiver.c,
	  res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
	  Put in missing \ns on the end of ast_logs (issue #7936 reported
	  by wojtekka)

2006-09-28 17:35 +0000 [r43919]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix buggy (and overly complex) loop used during reload
	  of app_queue for static member list updating

2006-09-28 17:34 +0000 [r43918]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Extend call establishment timeout

2006-09-28 17:31 +0000 [r43913-43915]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make sure the pvt exists before accessing
	  it again as it may have gone away (issue #7562 reported by Seb7
	  and issue #7939 reported by sorg)

	* main/cli.c: Warning be gone!

2006-09-28 16:41 +0000 [r43899]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: app_queue is comparing the device names incorrectly
	  while checking their statuses. It's internal list of interfaces
	  includes the dial string, while the argument passed to this
	  function does not have the dial string (/n for a local channel).
	  This causes it to ignore the device state changes because it
	  thinks it belongs to none of its members. (#8040 reported and
	  patch by tim_ringenbach)

2006-09-28 16:17 +0000 [r43893]  Joshua Colp <jcolp@digium.com>

	*  apps/app_meetme.c: Stop the stream after waitstream returns so that our
	  formats get restored. (issue #7370 reported by kryptolus)

2006-09-28 15:56 +0000 [r43877]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Fix compiler warning

2006-09-28 15:29 +0000 [r43864-43873]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
	  tim_ringenbach reported and patched)

	* apps/app_queue.c: Autopause not working for queue members. (#8042
	  - jmls reported and patch)

2006-09-28 12:58 +0000 [r43861-43862]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
	  remote side to start media on outgoing PROGRESS message

	* include/asterisk/compiler.h: Put attribute tag at correct place

2006-09-28 11:03 +0000 [r43852]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
	  when the call could not be properly established in misdn_call.
	  also removed the ACK_HDLC stuff which is not really needed.

2006-09-28 10:51 +0000 [r43843-43846]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Do not open transmit channel until
	  TCS is received

	* main/file.c: Don't warn on HOLD/UNHOLD control frames

	* main/file.c: Don't treat unknown control frames as voice

2006-09-27 20:21 +0000 [r43816]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Avoid inability to lock directory log message by
	  creating the directory ahead of time. (Issue 7631)

2006-09-27 19:44 +0000 [r43801-43803]  Jason Parker <jparker@digium.com>

	* apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
	  not being set under certain circumstances. Fix a minor issue, to
	  make it use the filenames that were parsed, instead of the entire
	  argument string. Fix Background() to return -1 like Playback(),
	  if no args are specified.

2006-09-27 19:10 +0000 [r43783-43798]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Compensate for out of order packets better if RFC2833
	  compensation is turned on.

	* channels/chan_iax2.c: Get rid of two functions from a time now
	  past (we THINK these are from pre-recursive lock time) that may
	  be contributing to two open issues on the bug tracker (7562/7939)
	  and that has the potential to just make bad things happen if the
	  timing is right.

2006-09-27 16:55 +0000 [r43779]  Russell Bryant <russell@digium.com>

	* main/channel.c,res/res_features.c: Fix a problem that occurred if
	  a user entered a digit
	  that matched a bridge feature that was configured using multiple
	  digits, and the digit that was pressed timed out in the feature
	  digit timeout period. For example, if blind transfer is
	  configured as '##', and a user presses just '#'. In this
	  situation, the call would lock up and no longer pass any frames.
	  (issue #7977 reported by festr, and issue #7982 reported by
	  michaels and valuable input provided by mneuhauser and kuj. Fixed
	  by me, with testing help and peer review from Joshua Colp). There
	  are a couple of issues involved in this fix: 1) When
	  ast_generic_bridge determines that there has been a timeout, it
	  returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
	  this result, it calls ast_generic_bridge over again with the same
	  timestamp for the next event. This results in an endless loop of
	  nothing until the call is terminated. This is resolved by simply
	  changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
	  sees a timeout. 2) I also changed ast_channel_bridge such that if
	  in the process of calculating the time until the next event, it
	  knows a timeout has already occured, to immediately return
	  AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
	  anyway. 3) In the process of testing the previous two changes, I
	  ran into a problem in res_features where ast_channel_bridge would
	  return because it determined that there was a timeout. However,
	  ast_bridge_call in res_features would then determine by its own
	  calculation that there was still 1 ms before the timeout really
	  occurs. It would then proceed, and since the bridge broke out and
	  did *not* return a frame, it interpreted this as the call was
	  over and hung up the channels. The reason for this was because
	  ast_bridge_call in res_features and ast_channel_bridge in
	  channel.c were using different times for their calculations.
	  channel.c uses the start_time on the bridge config, which is the
	  time that the feature digit was recieved. However, res_features
	  had another time, 'start', which was set right before calling
	  ast_channel_bridge. 'start' will always be slightly after
	  start_time in the bridge config, and sometimes enough to round up
	  to one ms. This is fixed by making ast_bridge_call use the same
	  time as ast_channel_bridge for the timeout calculation. ........

2006-09-27 16:24 +0000 [r43775]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
	  versioning, since Asterisk has it's own

2006-09-27 16:23 +0000 [r43774]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make rfc2833compensate a global option.

2006-09-27 04:35 +0000 [r43756]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Backport revision 43754 from the trunk,
	  which removes an unused buffer from mm_login to close bug 8038,
	  as well as addresses some formatting and coding guidelines issues
	  in passing. Originally, I did not commit this to 1.4 since it is
	  not necessarily fixing a bug. However, since the IMAP storage
	  code is brand new, I decided it would be better to make the
	  change here as well, in case someone has to work on this code to
	  address issues in the very near future. I don't want to make
	  unnecessary merge problems going to the trunk.

2006-09-27 02:32 +0000 [r43739]  Steve Murphy <murf@digium.com>

	* configs/extensions.ael.sample: This change to extensions.ael was
	  to fix bug 8031; the install scripts are causing it to be copied
	  to /etc/asterisk/extensions.ael, and because it is a fairly
	  direct conversion of the original extensions.conf, the macro and
	  context names clash with the existing extensions.conf. So, I put
	  an ael- in front of all macros and contexts, and checked every
	  goto and macro call. Also, this file compiles under aelparse.

2006-09-26 20:56 +0000 [r43710]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Back in revision 4798, this message was changed from
	  using ast_cli() to directly calling write(). During this change,
	  checking if this was a remote console was removed. This caused
	  this message about using "exit" or "quit" to exit an Asterisk
	  console to come up in times where it did not make sense. This
	  change restores the check to see if this is a remote console
	  before printing the message. (fixes BE-65)

2006-09-26 20:47 +0000 [r43707]  Joshua Colp <jcolp@digium.com>

	* .cleancount, main/cli.c, channels/chan_sip.c,
	  include/asterisk/channel.h: Use proper type to represent the group variable
	  (issue #8025 reported by makoto)

2006-09-26 20:30 +0000 [r43700-43703]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Add missing newline character in the warning
	  message about deprecated TOS values in configuration.

	* apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
	  mailbox definitions, don't introduce a length limit on the
	  definition by using a 256 byte temporary storage buffer. Instead,
	  make the temporary buffer just as big as it needs to be to hold
	  the entire mailbox definition. (fixes BE-68)

2006-09-26 20:19 +0000 [r43695-43697]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Strip options off the argument passed for
	  devicestate in chan_local. (issue #8034 reported by pcardozo)

	* apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
	  overhaul of the whisper support. 1. We need to duplicate the
	  frame from ast_translate 2. We need to ensure we always have
	  signed linear coming in for signed linear combining. 3. We need
	  to ensure we are always feeding signed linear out. 4. Properly
	  store and restore write format when beeping on the channel we are
	  whispering on. 5. Properly discontinue the stream on the channel
	  for the beep. (issue #8019 reported by timkelly1980)

2006-09-26 18:34 +0000 [r43676]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: update to use 1.4.3 core sounds, with corrected
	  beep/beeperr/tt-monkeys files

2006-09-26 18:08 +0000 [r43650-43674]  Jason Parker <jparker@digium.com>

	* doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
	  Dan Austin. Maximum values were incorrect, which is why this is
	  being put in 1.4

	* channels/chan_skinny.c: Add proper codec support to chan_skinny.
	  Works with at least ulaw, alaw, and g729a. This is technically a
	  "new feature", but there are justifications for it. I found a bug
	  with the recent rtp packetization changes, which caused the media
	  setup to fail under certain circumstances, particularly when
	  using allow=all, or having no allow= statements (globally or on
	  the device). I could have either removed the rtp packetization
	  features, or I could add proper codec support (which, without, I
	  think most people would consider to be a bug anyways).

2006-09-25 22:07 +0000 [r43640-43642]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Should have moved these lines up in the
	  merge, instead of removing them

	* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
	  delete=yes was ignored 2) maxmessages was ignored

2006-09-25 21:26 +0000 [r43626-43635]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
	  channels/h323/cisco-h225.asn: Fix ASN1 description of
	  non-standard Cisco extensions

	* channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
	  changes of trunk: 1) r43540: Avoid possible deadlock on channel
	  destruction 2) r43590: Disable fastStart if requested by remote
	  side

2006-09-25 15:23 +0000 [r43616]  Jason Parker <jparker@digium.com>

	* sounds/Makefile: One more fix for sounds installation - this time
	  for portability. Reported to asterisk-dev mailing list.

2006-09-25 14:52 +0000 [r43605]  Steve Murphy <murf@digium.com>

	* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
	  crashing if trying to play an OGG moh file.

2006-09-25 06:15 +0000 [r43582]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
	  channels/chan_h323.c: Merged revisions 43472,43495 from trunk

2006-09-24 14:58 +0000 [r43553-43564]  Russell Bryant <russell@digium.com>

	* channels/iax2-provision.c: Fix a CLI command registration issue
	  where an erroneous message claiming that "iax2 show provisioning"
	  was already registered. This was because this command was
	  registering itself as both the command, as well as the command it
	  is deprecating. (issue #8022, reported by bjweeks, fixed by
	  myself)

	* channels/chan_iax2.c:Check to see if the channel that is activating the
	  IAXPEER function is actually an IAX2 channel before proceeding to
	  process it to avoid crashing. (issue #8017, reported by admott,
	  fixed by myself)

2006-09-22 23:44 +0000 [r43524]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: don't output the 'build complete' message when the
	  target being run is already going to do an installation

2006-09-22 22:12 +0000 [r43518]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
	  properly. Remove reload support, since it doesn't
	  actually...work.

2006-09-22 21:36 +0000 [r43505-43508]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This commits a change to return
	  MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
	  goes well for bug 8004

	* pbx/pbx_ael.c: If the extensions.ael file not found, or
	  unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.

2006-09-22 17:25 +0000 [r43492]  Jason Parker <jparker@digium.com>

	* main/cli.c: Make sure we explicitly set the CLI command to not be
	  deprecated, if it isn't.

2006-09-22 16:42 +0000 [r43486-43489]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: use rebuilt extra sounds

	* main/channel.c: all the Linux systems I have don't use
	  '__m_count' for this field, so I don't know where this came
	  from...

2006-09-22 15:47 +0000 [r43477-43484]  Russell Bryant <russell@digium.com>

	* include/asterisk/threadstorage.h: backport the compatability fix
	  to use attribute_malloc instaed of __attribute__ ((malloc))

	* channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
	  could not be configured (issue #8006, Mithraen)

	* main/frame.c: Suppress a compiler warning about the use of a
	  potentially uninitialized variable. It couldn't actually happen,
	  though.

2006-09-22 03:01 +0000 [r43469]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: First shot at unload_module in
	  chan_skinny.. More to come.

2006-09-21 23:50 +0000 [r43466]  Matt O'Gorman <mogorman@digium.com>

	* include/asterisk/jabber.h, channels/chan_gtalk.c,
	  res/res_jabber.c: updates for better compontent support

2006-09-21 23:24 +0000 [r43464]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
	  actually documented how the new features in res_odbc actually
	  work. (Oops)

2006-09-21 22:21 +0000 [r43454-43456]  Joshua Colp <jcolp@digium.com>

	* channels/chan_oss.c: Some more clean up in the load function for
	  chan_oss (issue #8002 reported by Mithraen with minor mods by
	  moi)

	* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
	  (issue #8001 reported by Mithraen with mods by moi)

2006-09-21 21:21 +0000 [r43450]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile, build_tools/strip_nonapi (added): add another
	  attempt to strip non-API symbols from the final binary... script
	  will need to be extended to work on non-Linux systems

2006-09-21 20:22 +0000 [r43410-43445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_url.c: Fix documentation to reflect how Url() really
	  works

	* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates

2006-09-21  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta2 released.

2006-09-21 16:08 +0000 [r43404-43405]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile: remove this change... it requires binutils 2.17

2006-09-20 23:19 +0000 [r43396]  Jason Parker <jparker@digium.com>

	* build_tools/make_version: fix minor typo in the way version is
	  handled

2006-09-20  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta1 released.
