    <chapter>
	<title>Server Operation</title>
	<section id="operationalpractices">
	    <title>Recommended Operational Practices</title>

	    <para>
		Operation of a SIP server is not always easy task.
		Server administrators are challenged by broken or
		misconfigured user agents, network and host failures,
		hostile attacks and other stress-makers. All such
		situations may lead to an operational failure. It is sometimes
		very difficult to figure out the root reason of
		a failure, particularly in a distributed environment
		with many SIP components involved.		
		In this section,
		we share some of our practices and refer to tools
		which have proven to
		make life of administrators easier
	    </para>

	<qandaset>
	    <qandaentry>
		<question>
		    <para>
			Keeping track of messages is good
		    </para>
		</question>
		<answer>
			<para>
			    Frequently, operational errors are discovered or reported
			    with a delay.
			    Users frustrated by an error
			    frequently approach administrators
			    and scream "even though my SIP requests were absolutely ok
			    yesterday, they were mistakenly denied by your server".
			    If administrators do not record all SIP traffic at
			    their site, they will be no more able to identify
			    the problem reason.
			    We thus recommend that site
			    operators record all messages passing their site and keep them
			    stored for some period of time.
			They may use utilities such as 
			<application>ngrep 
			</application> or 
			<application>tcpdump
			</application>.
			There is also a utility <application moreinfo="none">
			    scripts/harv_ser.sh</application> in <application moreinfo="none">
			ser</application> distribution for post-processing
			of captured messages. It summarizes messages captured
			by reply status and user-agent header field.
		    </para>
		</answer>
	    </qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Real-time Traffic Watching
		    </para>
		</question>
		<answer>
			<para>
		    Looking at SIP messages in real-time may help to gain
		    understanding of problems. Though there are commercial
		    tools available, using a simple, text-oriented tool
		    such as <application>ngrep</application> makes the job very well thanks to SIP's textual nature.
			</para>
		    <example id="usingngrep">
			<title>Using <application>ngrep</application>
			</title>
			<para>In this example, all messages at port 5060
			which include the string "bkraegelin" are captured
			and displayed</para>
			<programlisting format="linespecific">
[jiri@fox s]$ ngrep bkraegelin@ port 5060
interface: eth0 (195.37.77.96/255.255.255.240)
filter: ip and ( port 5060 )
match: bkraegelin@
#
U +0.000000 153.96.14.162:50240 -> 195.37.77.101:5060
REGISTER sip:iptel.org SIP/2.0.
Via: SIP/2.0/UDP 153.96.14.162:5060.
From: sip:bkraegelin@iptel.org.
To: sip:bkraegelin@iptel.org.
Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162.
Date: Thu, 26 Sep 2002 22:03:55 GMT.
CSeq: 101 REGISTER.
Expires: 10.
Content-Length: 0.
.

#
U +0.000406 195.37.77.101:5060 -> 153.96.14.162:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 153.96.14.162:5060.
From: sip:bkraegelin@iptel.org.
To: sip:bkraegelin@iptel.org.
Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162.
CSeq: 101 REGISTER.
WWW-Authenticate: Digest realm="iptel.org", nonce="3d9385170000000043acbf6ba9c9741790e0c57adee73812", algorithm=MD5.
Server: Sip EXpress router(0.8.8 (i386/linux)).
Content-Length: 0.
Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=31604 req_src_ip=153.96.14.162 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==1".

			</programlisting>
		    </example>
		</answer>
	    </qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Tracing Errors in Server Chains
		    </para>
		</question>
		<answer>
			<para>
			    A request may pass any number of proxy servers on
			    its path to its destination. If an error occurs
			    in the chain, it is difficult for upstream troubleshooters
			    and/or users complaining to administrators to learn 
			    more about error circumstances. 
			    <application moreinfo="none">ser
			    </application> does its best and displays extensive
			    diagnostics information in SIP replies. It allows 
			    troubleshooters and/or users who report to troubleshooters
			    to gain additional knowledge about request processing
			    status. 
			    This extended debugging information is part of the warning 
			    header field. See <xref linkend="usingngrep"> for an illustration
			    of a reply that includes such a warning header field. The header
			    field contains the following pieces of information:
			<itemizedlist>
			    <listitem>
				<para>
				Server's IP Address -- good to identify
				from which server in a chain the reply
				came.
				    </para>
			    </listitem>
			    <listitem>
				    <para>
					Incoming and outgoing URIs -- good to
					learn for which URI the reply was
					generated, as it may be rewritten
					many times in the path. Particularly
					useful for debugging of numbering plans.
				    </para>
			    </listitem>
			    <listitem>
				<para>
					Number of Via header fields in replied
					request -- that helps in assessment of
					request path length. Upstream clients would
					not know otherwise, how far away in terms
					of SIP hops their requests were replied.
				</para>
			    </listitem>
				<listitem>
				    <para>
					Server's process id. That is useful for
					debugging to discover situations when
					multiple servers listen at the same
					address.
				    </para>
				</listitem>
				<listitem>
				    <para>
					IP address of previous SIP hop as seen by
					the SIP server.
				    </para>
				</listitem>
			</itemizedlist>
		    </para>
			<para>
			    If server administrator is not comfortable with
			    disclosing all this information, he can turn them
			    off using the <varname>sip_warning</varname> configuration
			    option.
			</para>
		    <para>
			A nice utility for debugging server chains is
			<application moreinfo="none">sipsak</application>,
			SIP Swiss Army Knife, traceroute-like tool for SIP
			developed at iptel.org. It allows you to send
			OPTIONS request with low, increasing Max-Forwards 
			header-fields and follow how it propagates in
			SIP network. See its webpage at
			<ulink url="http://sipsak.berlios.de/">
			    http://sipsak.berlios.de/
			</ulink>.
		    </para>
		    <example>
			<title>Use of SIPSak for Learning SIP Path</title>
			<programlisting format="linespecific">
[jiri@bat sipsak]$ ./sipsak -T -s sip:7271@iptel.org
warning: IP extract from warning activated to be more informational
0: 127.0.0.1 (0.456 ms) SIP/2.0 483 Too Many Hops
1: ?? (31.657 ms) SIP/2.0 200 OK
	without Contact header

			</programlisting>
			<para>
			    Note that in this example, the second hop
			    server does not issue any warning header fields
			    in replies and it is thus impossible to display 
			    its IP address in <application moreinfo="none">
			    SIPsak</application>'s output.
			</para>
		    </example>
		</answer>
	    </qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Watching Server Health
		    </para>
		</question>
		<answer>
		    <para>
			Watching Server's operation status in real-time may
			also be a great aid for trouble-shooting. 
			<application>ser</application> has an excellent 
			facility, a FIFO server, which allows UNIX
			tools to access server's internals. (It is 
			similar to how Linux tool access Linux kernel
			via the proc file system.) The FIFO server
			accepts commands via a FIFO (named pipe) and
			returns data asked for. Administrators do not
			need to learn details of the FIFO communication
			and can serve themselves using a front-end
			utility <application moreinfo="none">serctl</application>.
			Of particular interest for 
			monitoring server's operation are 
			<application moreinfo="none">serctl</application>
			commands
			<command moreinfo="none">ps</command> and
			<command moreinfo="none">moni</command>.
			The former displays running 
			<application moreinfo="none">ser</application>
			processes, whereas the latter shows statistics.
		    </para>
		    <example>
			<title>serctl ps command</title>
			<para>
			    This example shows 10 processes running at a host.
			    The process 0, "attendant" watches child processes
			    and terminates all of them if a failure occurs in
			    any of them. Processes 1-4 listen at local
			    interface and processes 5-8 listen at Ethernet
			    interface at port number 5060. Process number
			    9 runs FIFO server, and process number 10
			    processes all server timeouts.
			</para>
			<programlisting format="linespecific">
[jiri@fox jiri]$ serctl ps
0	31590	attendant
1	31592	receiver child=0 sock=0 @ 127.0.0.1::5060
2	31595	receiver child=1 sock=0 @ 127.0.0.1::5060
3	31596	receiver child=2 sock=0 @ 127.0.0.1::5060
4	31597	receiver child=3 sock=0 @ 127.0.0.1::5060
5	31604	receiver child=0 sock=1 @ 195.37.77.101::5060
6	31605	receiver child=1 sock=1 @ 195.37.77.101::5060
7	31606	receiver child=2 sock=1 @ 195.37.77.101::5060
8	31610	receiver child=3 sock=1 @ 195.37.77.101::5060
9	31611	fifo server
10	31627	timer
			  
			</programlisting>
		    </example>
		</answer>
	    </qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Is Server Alive
		    </para>
		</question>
		<answer>
		    <para>
			It is essential for solid operation to know
			continuously that server is alive. We've been
			using two tools for this purpose. 
			<application moreinfo="none">sipsak</application>
			does a great job of "pinging" a server, which
			may be used for alerting on unresponsive servers.
		    </para>
		    <para>
			<application moreinfo="none">monit</application> is
			a server watching utility which alerts when
			a server dies.
		    </para>
		</answer>
	    </qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Dealing with DNS
		    </para>
		</question>
		<answer>
		    <para>
			SIP standard leverages DNS. Administrators of
			<application moreinfo="none">ser</application> should
			be aware of impact of DNS on server's operation.
			Server's attempt to resolve an unresolvable address
			may block a server process in terms of seconds. To be
			safer that the server doesn't stop responding
			due to being blocked by DNS resolving, we recommend
			the following practices:
			<itemizedlist>
			    <listitem>
				<para>
				    Start a sufficient number of children processes.
				    If one is blocked, the other children will
				    keep serving.
				</para>
			    </listitem>
			    <listitem>
				<para>
				    Use DNS caching. For example, in Linux,
				    there is an <application moreinfo="none">
				    nscd</application> daemon available for
				    this purpose.
				</para>
			    </listitem>
			    <listitem>
				<para>
				    Process transactions statefully if memory
				    allows. That helps to absorb retransmissions
				    without having to resolve DNS for each of
				    them.
				</para>
			    </listitem>
			</itemizedlist>
		    </para>
		</answer>
	    </qandaentry>
		<qandaentry>
			<question>
				<para>
					Logging
				</para>
			</question>
			<answer>
			<anchor id="logging">
			<para>
			    <application>ser</application> by default logs
			    to <application>syslog</application> facility.
			    It is very useful to watch log messages for
			    abnormal behavior. Log messages, subject to
			    <application>syslog</application> configuration
			    may be stored at different files, or even at remote
			    systems. A typical location of the log file is
			    <filename>/var/log/messages</filename>.
			</para>
			<note>
			    <para>
				One can also use other <application>syslogd</application>
				implementation. <application>metalog</application>
				(<ulink url="http://metalog.sourceforge.net/">
				    http://metalog.sourceforge.net/
				</ulink>)
				features regular expression matching that enables
				to filter and group log messages.
			    </para>
			</note>
			<para>
			    For the purpose of debugging configuration scripts, one may
			    want to redirect log messages to console not to pollute
			    syslog files. To do so configure <application moreinfo="none">ser</application>
			    in the following way:
			    <itemizedlist>
				<listitem>
				    <para>
					Attach ser to console by setting <varname>fork=no</varname>.
				    </para>
				</listitem>
				<listitem>
				    <para>
					Set explicitly at which address 
					<application moreinfo="none">ser</application>
					should be listening, e.g., <varname>listen=192.168.2.16</varname>.
				    </para>
				</listitem>
				<listitem>
				    <para>
					Redirect log messages to standard error by setting
					<varname>log_stderror=yes</varname>
				    </para>
				</listitem>
				<listitem>
				    <para>
					Set appropriately high log level. (Be sure that you redirected logging
					to standard output. Flooding system logs with many detailed messages
					would make the logs difficult to read and use.) You can set the global
					logging threshold value with the option <varname>debug=nr</varname>,
					where the higher <varname>nr</varname> the more detailed output.
					If you wish to set log level only for some script events, include
					the desired log level as the first parameter of the
					<command moreinfo="none">log</command> action in your script.
					The messages will be then printed if <command moreinfo="none">log</command>'s
					level is lower than the global threshold, i.e., the lower the more
					noisy output you get.
					<example>
					    <title>Logging Script</title>
					    <programlisting format="linespecific">
&logging;
					    </programlisting>
					    <para>
						The following SIP message causes then logging output as shown
						bellow.
					    </para>
					    <programlisting format="linespecific">
REGISTER sip:192.168.2.16 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.33:5060
From: sip:113311@192.168.2.16
To: sip:113311@192.168.2.16
Call-ID: 00036bb9-0fd305e2-7daec266-212e5ec9@192.168.2.33
Date: Thu, 27 Feb 2003 15:10:52 GMT
CSeq: 101 REGISTER
User-Agent: CSCO/4
Contact: sip:113311@192.168.2.33:5060
Content-Length: 0
Expires: 600                                 
					    </programlisting>
					    <programlisting format="linespecific">
[jiri@cat sip_router]$ ./ser -f examples/logging.cfg 
Listening on 
              192.168.2.16 [192.168.2.16]::5060
Aliases: cat.iptel.org:5060 cat:5060 
WARNING: no fork mode 
 0(0) INFO: udp_init: SO_RCVBUF is initially 65535
 0(0) INFO: udp_init: SO_RCVBUF is finally 131070
 0(17379) REGISTER received
 0(17379) request for other domain received					
					    </programlisting>
					</example>
				    </para>
				</listitem>
			    </itemizedlist>
			</para>
			</answer>
		</qandaentry>
	    <qandaentry>
		<question>
		    <para>
			Labeling Outbound Requests
		    </para>
		</question>
		<answer>
		    <para>
		    Without knowing, which pieces of script code a relayed
		    request visited, trouble-shooting would be difficult.
		    Scripts typically apply different processing to
		    different routes such as to IP phones and PSTN
		    gateways. We thus recommend to label outgoing
		    requests with a label describing the type of processing
		    applied to the request.
			</para>
		    <para>
			Attaching "routing-history" hints to relayed
			requests is as easy as using the 
			<command moreinfo="none">append_hf</command>
			action exported by textops module. The following
			example shows how different labels are attached
			to requests to which different routing logic
			was applied.
			<example>
			    <title>"Routing-history" labels</title>
			    <programlisting format="linespecific">
# is the request for our domain?
# if so, process it using UsrLoc and label it so.
if (uri=~[@:\.]domain.foo") {
   if (!lookup("location")) {
    sl_send_reply("404", "Not Found");
    break;
   };
   # user found -- forward to him and label the request
   append_hf("P-hint: USRLOC\r\n");
} else {
# it is an outbound request to some other domain --
# indicate it in the routing-history label
   append_hf("P-hint: OUTBOUND\r\n");
};
t_relay();
			    </programlisting>
			    <para>
				This is how such a labeled requests looks
				like. The last header field includes
				a label indicating the script processed
				the request as outbound.
			    </para>
			    <programlisting format="linespecific">
#
U 2002/09/26 02:03:09.807288 195.37.77.101:5060 -> 203.122.14.122:5060
SUBSCRIBE sip:rajesh@203.122.14.122 SIP/2.0.
Max-Forwards: 10.
Via: SIP/2.0/UDP 195.37.77.101;branch=53.b44e9693.0.
Via: SIP/2.0/UDP 203.122.14.115:16819.
From: sip:rajeshacl@iptel.org;tag=5c7cecb3-cfa2-491d-a0eb-72195d4054c4.
To: sip:rajesh@203.122.14.122.
Call-ID: bd6c45b7-2777-4e7a-b1ae-11c9ac2c6a58@203.122.14.115.
CSeq: 2 SUBSCRIBE.
Contact: sip:203.122.14.115:16819.
User-Agent: Windows RTC/1.0.
Proxy-Authorization: Digest username="rajeshacl", realm="iptel.org", algorithm="MD5", uri="sip:rajesh@203.122.14.122", nonce="3d924fe900000000fd6227db9e565b73c465225d94b2a938", response="a855233f61d409a791f077cbe184d3e3".
Expires: 1800.
Content-Length: 0.
P-hint: OUTBOUND.			    </programlisting>
			</example>
		</para>
		</answer>
	    </qandaentry>
	</qandaset>
	</section> <!-- operational practises -->

	<section>
	    <title>HOWTOs</title>
	    <para>
		This section is a "cookbook" for dealing with common tasks,
		such as user management or controlling access
		to PSTN gateways.
	    </para>
	    <section>
		<title>User Management</title>

			<para>
			    There are two tasks related to management of SIP users:
			    maintaining user accounts and maintaining user contacts.
			    Both these jobs can be done using the 
			    <application moreinfo="none">serctl</application>
			    command-line tool. Also, the complimentary web
			    interface, <application moreinfo="none">serweb</application>,
			    can be used for this purpose as well.
			</para>
			<para>
			    If user authentication is turned on, which is a highly
			    advisable practice, user account must be created before
			    a user can log in. To create a new user account, call the
			    <command moreinfo="none">serctl add</command> utility
			    with username, password and email as parameters. It
			    is important that the environment <varname>SIP_DOMAIN</varname>
			    is set to your realm and matches realm values used in
			    your script. The realm value is used for calculation
			    of credentials stored in subscriber database, which are
			    bound permanently to this value.
			    <screen format="linespecific">
[jiri@cat gen_ha1]$ export SIP_DOMAIN=foo.bar
[jiri@cat gen_ha1]$ serctl add newuser secret newuser@foo.bar
MySql Password: 
new user added
			    </screen>
			</para>
			<para><application moreinfo="none">serctl</application> can
			    also change user's password or remove existing accounts
			    from system permanently.
			    <screen format="linespecific">
[jiri@cat gen_ha1]$ serctl passwd newuser newpassword
MySql Password: 
password change succeeded
[jiri@cat gen_ha1]$ serctl rm newuser                
MySql Password: 
user removed
			    </screen>
			</para>
			<para>
			    User contacts are typically automatically uploaded by SIP phones
			    to server during registration process and administrators do not
			    need to worry about them. However, users
			    may wish to append permanent contacts to PSTN gateways
			    or to locations in other administrative domains. 
			    To manipulate the contacts in such cases, use
			    <application moreinfo="none">serctl ul</application>
			    tool. Note that this is the only correct way
			    to update contacts -- direct changes to back-end
			    MySql database do not affect server's memory. Also note,
			    that if persistence is turned off (usrloc "db_mode"
			    parameter set to "0"), all contacts are gone on server
			    reboot. Make sure that persistence is enabled if you
			    add permanent contacts.
			</para>
			<para>
			    To add a new permanent contact for a user, call 
			    <application moreinfo="none">serctl ul add &lt;username&gt
			    &lt;contact&gt;</application>. To delete 
			    all user's contacts, call 
			    <application>serctl ul rm &lt;username&gt;</application>.
			    <application moreinfo="none">serctl ul show &lt;username&gt;</application>
			    prints all current user's contacts.
			    <screen format="linespecific">
[jiri@cat gen_ha1]$ serctl ul add newuser sip:666@gateway.foo.bar
sip:666@gateway.foo.bar
200 Added to table
('newuser','sip:666@gateway.foo.bar') to 'location'
[jiri@cat gen_ha1]$ serctl ul show newuser
&lt;sip:666@gateway.foo.bar&gt;;q=1.00;expires=1073741812
[jiri@cat gen_ha1]$ serctl ul rm newuser  
200 user (location, newuser) deleted
[jiri@cat gen_ha1]$ serctl ul show newuser
404 Username newuser in table location not found
			    </screen>
			</para>
	    </section> <!-- user management -->
	    <section>
		<title>User Aliases</title>

			<para>
			    Frequently, it is desirable for a user to have multiple
			    addresses in a domain. For example, a user with username "john.doe" wants to be
			    reachable at a shorter address "john" or at a numerical address
			    "12335", so that PSTN callers with digits-only key-pad can reach
			    him too.
			</para>
			<para>
			    With <application moreinfo="none">ser</application>, you can maintain
			    a special user-location table and translate existing aliases to canonical
			    usernames using the <command moreinfo="none">lookup</command>
			    action from usrloc module. The following script fragment demonstrates
			    use of <command moreinfo="none">lookup</command> for this purpose.
			    <example>
				<title>Configuration of Use of Aliases</title>
				<programlisting format="linespecific">
if (!uri==myself) { # request not for our domain...
  route(1); # go somewhere else, where outbound requests are processed
  break;
};
# the request is for our domain -- process registrations first
if (method=="REGISTER") { route(3); break; };

# look now, if there is an alias in the "aliases" table; don't care
# about return value: whether there is some or not, move ahead then
lookup("aliases");

# there may be aliases which translate to other domain and for which
# local processing is not appropriate; check again, if after the
# alias translation, the request is still for us
if (!uri==myself) { route(1); break; };

# continue with processing for our domain...
...
  
				</programlisting>
			    </example>
			</para>
			<para>
			    The table with aliases is updated using the
			    <application moreinfo="none">serctl</application>
			    tool. <application moreinfo="none">
			    serctl alias add &lt;alias&gt; &lt;uri&gt;</application>
			    adds a new alias, 
			    <application moreinfo="none">serctl alias show &lt;user&gt;</application>
			    prints an existing alias, and
			    <application moreinfo="none">serctl alias rm &lt;user&gt;</application>
			    removes it.
			    <screen format="linespecific">
[jiri@cat sip_router]$ serctl alias add 1234 sip:john.doe@foo.bar
sip:john.doe@foo.bar
200 Added to table
('1234','sip:john.doe@foo.bar') to 'aliases'
[jiri@cat sip_router]$ serctl alias add john sip:john.doe@foo.bar
sip:john.doe@foo.bar
200 Added to table
('john','sip:john.doe@foo.bar') to 'aliases'
[jiri@cat sip_router]$ serctl alias show john                    
&lt;sip:john.doe@foo.bar&gt;;q=1.00;expires=1073741811
[jiri@cat sip_router]$ serctl alias rm john  
200 user (aliases, john) deleted				
			    </screen>
			</para>
			<para>
			    Note that persistence needs to be turned on in usrloc
			    module. All changes to aliases will be otherwise lost
			    on server reboot. To enable persistence, set the
			    db_mode usrloc parameter to a non-zero value.
			    <programlisting format="linespecific">
# ....load module ...
loadmodule "modules/usrloc/usrloc.so"
# ... turn on persistence -- all changes to user tables are immediately
# flushed to mysql
modparam("usrloc", "db_mode",   1)
# the SQL address:
modparam("usrloc", "db_url","mysql://ser:secret@dbhost/ser")
			    </programlisting>
			</para>
	    </section> <!-- user aliases -->
	    <section id=acl>
		<title>Access Control (PSTN Gateway)</title>
			<para>
			    It is sometimes important to exercise some sort of
			    access control. A typical use case is when 
			    <application moreinfo="none">ser</application> is used
			    to guard a PSTN gateway. If a gateway was not well guarded,
			    unauthorized users would be able to use it to terminate calls in PSTN,
			    and cause high charges to its operator.
			</para>
			<para>
			    There are few issues you need to understand when
			    configuring <application moreinfo="none">ser</application>
			    for this purpose. First, if a gateway is built or configured to
			    accept calls from anywhere, callers may easily bypass your
			    access control server and communicate with the gateway
			    directly. You then need to enforce at transport layer
			    that signaling is only accepted if coming via
			    <application moreinfo="none">ser</application> and
			    deny SIP packets coming from other hosts and port numbers.
			    Your network must be configured not to allow forged
			    IP addresses. Also, you need to turn on record-routing
			    to assure that all session requests will travel via 
			    <application moreinfo="none">ser</application>.			    
			    Otherwise, caller's devices would send subsequent SIP requests 
			    directly to your gateway, which would fail because of transport 
			    filtering.
			</para>
			<para>
			    Authorization (i.e., the process of determining who may call where)
			    is facilitated in <application moreinfo="none">ser</application>
			    using <emphasis>group membership</emphasis> concept. Scripts make 
			    decisions on whether a caller is authorized to make a call to
			    a specific destination based on user's membership in a group.
			    For example a policy may be set up to allow calls to international
			    destinations only to users, who are members of an "int" group.			    
			    Before user's group membership is checked, his identity
			    must be verified first. Without cryptographic verification of user's
			    identity, it would be impossible to assert that a caller really
			    is who he claims to be.
			</para>
			<para>
			    The following script demonstrates, how to configure <application moreinfo="none">ser</application>
			    as an access control server for a PSTN gateway. The script verifies user
			    identity using digest authentication, checks user's privileges,
			    and forces all requests to visit the server.
			    <example>
				<title>Script for Gateway Access Control</title>
				<programlisting format="linespecific">
&gatewayacl;
				</programlisting>
			    </example>
			</para>
			<para>
			    Use the <application moreinfo="none">serctl</application> tool to
			    maintain group membership. 
			    <application moreinfo="none">serctl acl grant &lt;username&gt; &lt;group&gt;</application>
			    makes a user member of a group, 
			    <application>serctl acl show &lt;username&gt;</application> shows groups
			    of which a user is member, and
			    <application>serctl acl revoke &lt;username&gt; [&lt;group&gt;]</application>
			    revokes user's membership in one or all groups.
			    <screen format="linespecific">
[jiri@cat sip_router]$ serctl acl grant john int
MySql Password: 
+------+-----+---------------------+
| user | grp | last_modified       |
+------+-----+---------------------+
| john | int | 2002-12-08 02:09:20 |
+------+-----+---------------------+
			    </screen>
			</para>
	    </section> <!-- access control -->
	    <section>
		<title>Accounting</title>
			<para>
			    In some scenarios, like termination of calls in PSTN, SIP administrators
			    may wish to keep track of placed calls. <application moreinfo="none">ser</application>
			    can be configured to report on completed transactions. Reports are sent
			    by default to <application moreinfo="none">syslog</application> facility.
			    Support for RADIUS and mysql accounting exists as well.
			</para>
			<para>
			    Note that <application moreinfo="none">ser</application> is no way 
			    call-stateful. It reports on completed transactions, i.e., after 
			    a successful call set up is reported, it drops any call-related 
			    state. When a call is terminated, transactional state for BYE request
			    is created and forgotten again after the transaction completes.
			    This is a feature and not a bug -- keeping only transactional
			    state allows for significantly higher scalability. It is then
			    up to the accounting application to correlate call initiation
			    and termination events.
			</para>
			<para>
			    To enable call accounting, tm and acc modules need to be loaded,
			    requests need to be processed statefully and labeled for
			    accounting. That means, if you want a transaction to be reported,
				the initial request must have taken the path 
				"<command>setflag(X)</command>, <command>t_relay</command>"
				in <application>ser</application> script. X must have the
				value configured in <varname>acc_flag</varname>
				configuration option.
			</para>
			<para>
				Also note, that by default only transactions that initiate
				a SIP dialog (typically INVITE) visit a proxy server.
				Subsequent transactions are exchanged directly between
				end-devices, do not visit proxy server and cannot be
				reported. To be able to report on subsequent transactions,
				you need to force them visit proxy server by turning 
				record-routing on. 
			</para>
			<para>
				
			    <example>
				<title>Configuration with Enabled Accounting</title>
				<programlisting format="linespecific">
&accountingexample;
				</programlisting>
			    </example>
			</para>
	    </section> <!-- accounting -->
	    <section>
		<title>Reliability</title>

			<para>
			    It is essential to guarantee continuous
			    service operation even under erroneous conditions, 
			    such as host or network failure. The major issue in such
			    situations is transfer of operation to a backup
			    infrastructure and making clients use it.
			</para>
			<para>
			    The SIP standard's use of DNS SRV records has been
			    explicitly constructed to handle with server failures.
			    There may be multiple servers responsible for a domain
			    and referred to by DNS. If it is impossible to communicate
			    with a primary server, a client can proceed to another one.
			    Backup servers may be located in a different geographic
			    area to minimize risk caused by areal operational
			    disasters: lack of power, flooding, earthquake, etc.
			    <note>
				<sidebar>
				    <para>Unless there are redundant DNS
				    servers, fail-over capability cannot be guaranteed.
				    </para>
				</sidebar>
			    </note>
			    Unfortunately, at the moment of writing this documentation
			    (end of December 2002) only very few SIP products
			    actually implement the DNS fail-over mechanism. Unless
			    networks with SIP devices supporting this mechanism are
			    built, alternative mechanisms must be used to force 
			    clients to use backup servers. Such a mechanism is
			    disconnecting primary server and replacing it with
			    a backup server locally.
			    It unfortunately precludes geographic dispersion and
			    requires network multihoming to avoid dependency on
			    single IP access. Another method is to update DNS
			    when failure of the primary server is detected.
			    The primary drawback of this method is its latency:
			    it may take long time until all clients learn to use
			    the new server.
			</para>
			<para>
			    The easier part of the redundancy story is replication of 
			    <application moreinfo="none">ser</application>
			    data. <application moreinfo="none">ser</application>
			    relies on replication capabilities of its back-end database.
			    This works with one exception: user location database.
			    User location database is a frequently accessed table,
			    which is thus cached in server's memory to improve
			    performance. Back-end replication does not affect
			    in-memory tables, unless server reboots. To facilitate
			    replication of user location database, 
			    server's SIP replication feature must be enabled
			    in parallel with back-end replication.
			</para>
			<para>
			    The design idea of replication of user location database
			    is easy: Replicate any successful REGISTER requests to
			    a peer server. To assure that digest credentials can
			    be properly verified, both servers need to use the same
			    digest generation secret and maintain synchronized time.
			    A known limitation of this method is it does not replicate
			    user contacts entered in another way, for example using
			    web interface through FIFO server.
			    The following script example shows configuration of
			    a server that replicates all REGISTERs.
			    <example>
				<title>Script for Replication of User Contacts</title>
				<programlisting format="linespecific">
&replicateexample;				    
				</programlisting>
			    </example>
			</para>
	    </section> <!-- reliability -->
	    <section>
		<title>Stateful versus Stateless Forwarding</title>
		<para>
		    <application moreinfo="none">ser</application> allows both stateless
		    and stateful request processing. This memo explains what are pros and cons of
		    using each method. The rule of thumb is "stateless for scalability,
		    stateful for services". If you are unsure which you need, stateful
		    is a safer choice which supports more usage scenarios.
		</para>
			<para>
			    Stateless forwarding with the
			    <command moreinfo="none">forward(uri:host, uri:port)</command> action
			    guarantees high scalability. It withstands high load and
			    does not run out of memory. A perfect use of stateless forwarding
			    is load distribution.
			</para>
			<para>
			    Stateful forwarding using the <command moreinfo="none">t_relay()</command>
			    action is known to scale worse. It can quickly run out of memory and
			    consumes more CPU time. Nevertheless, there are scenarios which are
			    not implementable without stateful processing. In particular:
			    <itemizedlist>
				<listitem>
				    <para>
					<emphasis>Accounting</emphasis> requires stateful processing
					to be able to collect transaction status and issue a single
					report when a transaction completes.
				    </para>
				</listitem>
				<listitem>
				    <para>
					<emphasis>Forking</emphasis> only works with stateful forwarding.
					Stateless forwarding only forwards to the default URI out of the
					whole destination set.
				    </para>
				</listitem>
				<listitem>
				    <para>
					<emphasis>DNS resolution</emphasis>. DNS resolution may be
					better served with stateful processing. If a request is forwarded
					to a destination whose address takes long time to resolve,
					a server process is blocked and unresponsive. Subsequent 
					request retransmissions from client will cause other processes
					to block too if requests are processed statelessly. As a result,
					<application moreinfo="none">ser</application> will quickly
					run out of available processes. With stateful forwarding,
					retransmissions are absorbed and do not cause blocking of
					another process.
				    </para>
				</listitem>
				<listitem>
				    <para>
					<emphasis>Forwarding Services</emphasis>. All sort of services 
					with the "forward_on_event" logic, which rely on 
					<command moreinfo="none">t_on_failure</command> tm
					action must be processed statefully.
				    </para>
				</listitem>
			<listitem>
			    <para>
				<emphasis>
				    Fail-over.
				</emphasis>
				If you wish to try out another destination, after a primary destination
				failed you need to use stateful processing. With stateless processing
				you never know with what status a forwarded request completed downstream
				because you immediately release all processing information after the 
				request is sent out. 

				<note>
				    <para>
					Positive return value of stateless
					<command moreinfo="none">forward</command> action only indicates that
					a request was successfully sent out, and does not gain any knowledge
					about whether it was successfully received or replied. Neither does
					the return value of
					the stateful <command moreinfo="none">t_relay</command> action family
					gain you this knowledge. However, these actions store transactional
					context with which includes original request and allows you to
					take an action when a negative reply comes back or a timer strikes.
					See <xref linkend="replyprocessingsection"> for an example script 
					which launches another
					branch if the first try fails.
				    </para>
				</note>

			    </para>
			</listitem>
			    </itemizedlist>
			</para>
	    </section> <!-- stateful vs. stateless -->
	    <section>
		<title>Serving Multiple Domains</title>
			<para>
			    <application moreinfo="none">ser</application> can be configured to
			    serve multiple domains. To do so, you need to take the following steps:
			    <orderedlist>
				<listitem id="createtable">
				    <para>
					Create separate subscriber and location database table
					for each domain served and name them uniquely.
				    </para>
				</listitem>
				<listitem>
				    <para>
					Configure your script to distinguish between multiple
					served domains. Use regular expressions for domain
					matching as described in <xref linkend="redomainmatching">.
				    </para>
				</listitem>
				<listitem>
				    <para>
					Update table names in usrloc and auth actions to reflect
					names you created in <xref linkend="createtable">.
				    </para>
				</listitem>
				
			    </orderedlist>
			</para>
			<para>
				The latest <application>SER</application> release includes automated
				multidomain management which greatly automates maintenance of multiple	
				domains. Ask our technical support for more help.
			</para>
	    </section> <!-- multiple domains -->
	    <section id="missedcalls">
		<title>Reporting Missed Calls</title>
			<para>
			    <application moreinfo="none">ser</application> can report missed
			    calls via <application moreinfo="none">syslog</application> facility
			    or to mysql. Mysql reporting can be utilized by 
			    <application moreinfo="none">ser</application>'s 
			    complementary web-interface, <application moreinfo="none">serweb</application>.
			    (See more in <xref linkend="serweb">).
			</para>
			<para>
			    Reporting on missed calls is enabled by acc module.
			    There are two cases, on which you want to report. The first
			    case is when a callee is off-line. The other case is when
			    a user is on-line, but call establishment fails. There
			    may be many failure reasons (call cancellation, inactive phone,
			    busy phone, server timer, etc.), all of them leading to
			    a negative (>=300) reply sent to caller. The acc module
			    can be configured to issue a missed-call report whenever
			    a transaction completes with a negative status. Two following
			    script fragment deals with both cases.
			</para>
			<para>
			    First, it reports
			    on calls missed due to off-line callee status
			    using the <command moreinfo="none">acc_request</command>
			    action. The action is wrapped in transactional
			    processing (<command moreinfo="none">t_newtran</command>)
			    to guarantee that reports are not
			    duplicated on receipt of retransmissions.
			    </para>
			<para>
			    Secondly, transactions to on-line users are marked
			    to be reported on failure. That is what the 
			    <command moreinfo="none">setflag(3)</command> action
			    is responsible for, along with the configuration option
			    "log_missed_flag". This option configures <application moreinfo="none">ser</application>
			    to report on all transactions, which were marked
			    with flag 3.			   
			    <programlisting format="linespecific">
loadmodule("modules/tm/tm.so");
loadmodule("modules/acc/acc.so");
....
# if a call is labeled using setflag(3) and is missed, it will
# be reported
...
modparam("acc", "log_missed_flag", 3 );
if (!lookup("location")) {
     # call invitations to off-line users are reported using the
     # acc_request action; to avoid duplicate reports on request
     # retransmissions, request is processed statefully (t_newtran,
     # t_reply)
     if ((method=="INVITE" || method=="ACK") && t_newtran() ) {
          t_reply("404", "Not Found");
	  acc_request("404 Not Found");
          break;
     };
     # all other requests to off-line users are simply replied
     # statelessly and no reports are issued
    sl_send_reply("404", "Not Found");
    break;
} else {
     # user on-line; report on failed transactions; mark the
     # transaction for reporting using the same number as 
     # configured above; if the call is really missed, a report
     # will be issued
     setflag(3);
     # forward to user's current destination
     t_relay();
     break;
};
			    </programlisting>
			    
			</para>
	    </section> <!-- missed calls -->
	    <section>
		<title>NAT Traversal</title>
		<para>
		    NATs are worst things that ever happened to SIP. These devices
		    are very popular because they help to conserve IP address space
		    and save money charged for IP addresses. Unfortunately, they
		    translate addresses in a way which is not compatible with SIP.
		    SIP advertises receiver addresses in its payload. The advertised
		    addresses are invalid out of NATed networks. As a result,
		    SIP communication does not work across NATs without extra
		    effort.
		</para>
		<para>
		    There are few methods that may be deployed to traverse NATs.
		    How proper their use is depends on the deployment scenario.
		    Unfortunately, all the methods have some limitations and
		    there is no straight-forward solution addressing all
		    scenarios. Note that none of these methods takes explicit
		    support in <application moreinfo="none">ser</application>.
		</para>
		<para>
		    The first issue is whether SIP users are in control of 
		    their NATs. If not (NATs are either operated by ISP or
		    they are sealed to prevent users setting them up), the
		    only method is use of a STUN-enabled phone. STUN is 
		    a very simple protocol used to fool NAT in such a way,
		    they permit SIP sessions. Currently, we are aware of
		    one softphone (kphone) and one hardphone (snom) with
		    STUN support, other vendors are working on STUN support
		    too. Unfortunately, STUN gives no NAT traversal
		    guarantee -- there are types of NATs, so called
		    symmetric NATs, over which STUN fails to work.
		    <note>
			<para>
			    There is actually yet another method to address
			    SIP-unaware, user-uncontrolled NATs. It is based
			    on a proxy server, which relays all signaling and
			    media and mangles packets to make them more
			    NAT-friendly. The very serious problem with this
			    method is it does not scale.
			</para>
		    </note>
		</para>
		<para>
		    If users are in control of their own NAT, as typically residential
		    users are, they can still use STUN. However, they may use other
		    alternatives too. One of them is to replace their NAT with
		    a SIP-aware NAT. Such NATs have built-in SIP awareness,
		    that patches problems caused by address translations. Prices
		    of such devices are getting low and there are available
		    implementations (Intertex, Cisco/PIX). No special support
		    in phones is needed.
		</para>
		<para>
		    Other emerging option is UPnP. UPnP is a protocol that allows
		    phones to negotiate with NAT boxes. You need UPnP support in
		    both, NAT and phones. As UPnP NATs are quite affordable,
		    costs are not an obstacle. Currently, we are aware of one
		    SIP phone (SNOM) with UPnP support.
		</para>
		<para>
		    Geeks not wishing to upgrade their firewall to a SIP-aware or
		    UPnP-enabled one may try to configure static address translation.
		    That takes phones with configuration ability to use fixed port
		    numbers and advertise outside address in signaling. Cisco phones
		    have this capability, for example. The NAT devices need to
		    be configured to translate outside port ranges to the 
		    ranges configured in phones.		    
		</para>
	    </section> <!-- NAT traversal -->

		<section>
		<title>Using Only Latest User's Contact for Forwarding
		</title>
			<para>
				In some scenarios, it may be beneficial only to use only one
				registered contact per user. If that is the case, setting
				registrar module's parameter <varname>append_branches</varname>
				to 1 will eliminate forking and forward all requests only
				to a single contact. If there are multiple contacts, a contact
				with highest priority is chosen. This can be changed to
				the "freshest" contact by setting module parameter's
				<varname>desc_time_order</varname> to 1.
			</para>

		</section>

	    <section>
		<title>Authentication Policy: Prevention of Unauthorized Domain 
		    Name Use in From and More</title>
		<para>
		    Malicious users can claim a name of domain, to which they do 
		    not administratively belong, in From header field. This
		    behavior cannot be generally prevented. The reason is
		    that requests with such a faked header field do not need
		    to visit servers of the domain in question. However, if they
		    do so, it is desirable to assure that users claiming
		    membership in a domain are actually associated with it.
		    Otherwise the faked requests would be relayed and appear
		    as coming from the domain, which would increase
		    credibility of the faked address and decrease credibility of
		    the proxy server.
		</para>
		<para>
		    Preventing unauthorized domain name use in relayed requests 
		    is not difficult.
		    One needs to authenticate each request with name of the
		    served domain in From header field. To do so, one can
		    search for such a header field using <command moreinfo="none">search</command>
		    action (textops module) and force authentication if the
		    search succeeds.
		    <note>
			<para>
			    A straight-forward solution might be to authenticate
			    ALL requests. However, that only works in closed
			    networks in which all users have an account in the
			    server domain. In open networks, it is desirable to permit
			    incoming calls from callers from other domains without
			    any authentication. For example, a company may wish
			    to accept calls from unknown callers who are
			    new prospective customers.
			    
			</para>
		    </note>
		    <programlisting format="linespecific">
# does the user claim our domain "foo.bar" in From?
if (search("^(f|From):.*foo.bar")) {
        # if so, verify credential
	if (!proxy_authorize("foo.bar", "subscriber")) { 
              # don't proceed if credentials broken; challenge
	      proxy_challenge("foo.bar", "0");
	      break;
        };
};
		    </programlisting>
		</para>
		<para>
		    In general, the authentication policy may be very rich. You may not
		    forget each request deserves its own security and you need to 
		    decide whether it shall be authenticated or not. As mentioned
		    above, in closed networks, you may want to authenticate absolutely 
		    every request. That however prohibits traffic from users from
		    other domains. A pseudo-example of a reasonable policy is attached:
		    it looks whether a request is registration, it claims to originate
		    from our domain in From header field, or is a local request to
		    another domain.
		    <programlisting format="linespecific">
# (example provided by Michael Graff on [serusers] mailing list
if (to me):
    if register
          www_authorize or fail if not a valid register
          done
    if claiming to be "From" one of the domains I accept registrations for
          proxy_authorize
          done
    if not to me (I'm relaying for a local phone to an external address)
          proxy_authorize
          done
		    </programlisting>
		</para>
		<para>
		    You also may want to apply additional restriction to how
		    digest username relates to usernames claimed in From and
		    To header fields. For example, the <command moreinfo="none">check_to</command>
		    action enforces the digest id to be equal to username
		    in To header fields. That is good in preventing someone
		    with valid credentials to register as someone else
		    (e.g., sending a REGISTER with valid credentials of
		    "joe" and To belonging to "alice"). Similarly,
		    <command moreinfo="none">check_from</command> is used
		    to enforce username in  from to equal to digest id.
		    <note>
			<para>
			    There may be a need for a more complex relationship
			    between From/To username and digest id. For example,
			    providers with an established user/password database
			    may wish to keep using it, whereas permitting users
			    to claim some telephone numbers in From. To address
			    such needs generally, there needs to be a 1:N mapping
			    between digest id and all usernames that are acceptable
			    for it. This is being addressed in a newly contributed
			    module "domain", which also addresses more generally
			    issues of domain matching for multidomain scenarios.
			</para>
		    </note>
		</para>
		<para>
		    Other operational aspect affecting the authentication policy
		    is guarding PSTN gateways (see <xref linkend="acl">). There
		    may be destinations that are given away for free whereas
		    other destinations may require access control using
		    group membership, to which  authentication is a prerequisite.
		</para>

	    </section> <!-- authentication policy, faked froms -->
	    <section>
		<title>Connecting to PBX Voicemail Using a Cisco Gateway</title>
		<para>
		    In some networks, administrators may wish to utilize their
		    PBX voicemail systems behind PSTN gateways. There is a practical problem
		    in many network settings: it is not clear for whom a call to
		    voicemail is. If voicemail is identified by a single number,
		    which is then put in INVITE's URI, there is no easy way to
		    learn for whom a message should be recorded. PBX voicemails
		    utilize that PSTN protocols signal the number of originally
		    called party. If you wish to make the PBX voicemail work,
		    you need to convey the number in SIP and translate it in
		    PSTN gateways to its PSTN counterpart.
		</para>
		<para>
		    There may be many different ways to achieve this scenario. Here
		    we describe the proprietary mechanism Cisco gateways use and how to 
		    configure <application moreinfo="none">ser</application> to
		    make the gateways happy. Cisco gateways expect the number
		    of originally called party to be located in proprietary
		    <varname>CC-Diversion</varname> header field. When a SIP 
		    INVITE sent via a PSTN gateway to PBX voicemail has number
		    of originally called party in the header field, the voicemail
		    system knows for whom the incoming message is. That is at least
		    true for AS5300/2600 with Cisco IOS 12.2.(2)XB connected to
		    Nortel pbxs via PRI. (On the other hand, 12.2.(7b) is known
		    not to work in this scenario.)
		</para>
		<para>
		    <application moreinfo="none">ser</application> needs then to
		    be configured to append the <varname>CC-Diversion</varname>
		    header field name for INVITEs sent to PBX voicemail.
		    The following script shows that: when initial forwarding
		    fails (nobody replies, busy is received, etc.), a new branch
		    is initiated to the pbx's phone number. 
		    <command moreinfo="none">append_urihf</command> is used to
		    append the <varname>CC-Diversion</varname> header field. It
		    takes two parameters: prefix, which includes header name,
		    and suffix which takes header field separator. 
		    <command moreinfo="none">append_urihf</command> inserts
		    original URI between those two.
		    <example>
			<title>Forwarding to PBX/Voicemail via Cisco Gateways</title>
			<programlisting format="linespecific">
&ccdiversion;
			</programlisting>
		    </example>
		    
		</para>
	    </section>
	</section> <!-- howtos -->

	<section>
	    <title>Troubleshooting</title>
	    <para>
		This section gathers practices how to deal with errors
		known to occur frequently. To understand how to watch
		SIP messages, server logs, and in general how to
		troubleshoot, read also <xref linkend="operationalpractices">. 
	    </para>
	    <qandaset>
		<qandaentry>
		    <question>
			<para>
			SIP requests are replied by <application>ser</application> with
			"483 Too Many Hops" or "513 Message Too Large"
		        </para>
		    </question>

		    <answer>
			<para>
			    In both cases, the reason is probably an error in
			    request routing script which caused an infinite loop.
			    You can easily verify whether this happens by
			    watching SIP traffic on loopback interface. A typical
			    reason for misrouting is a failure to match local
			    domain correctly. If a server fails to recognize
			    a request for itself, it will try to forward it
			    to current URI in believe it would forward them
			    to a foreign domain. Alas, it forwards the request
			    to itself again. This continues to happen until
			    value of max_forwards header field reaches zero
			    or the request grows too big. Solutions is easy:
			    make sure that domain matching is correctly
			    configured. See <xref linkend="domainmatching">
			    for more information how to get it right.
			</para>
		    </answer>		    
		</qandaentry>
		<qandaentry>
			
		    <question>
			        
			<para>
			
			    Windows Messenger authentication fails.
			</para>
		    </question>
		    <answer>
			<anchor id="msmbug">
			<para>
			    The most likely reason for this problem is a bug
			    in Windows Messenger. WM only authenticates if
			    server name in request URI equals authentication
			    realm. After a challenge is sent by SIP server,
			    WM does not resubmit the challenged request at all
			    and pops up authentication window again.
			    If you want to authenticate WM, you need to
			    set up your realm value to equal server name.
			    If your server has no name, IP address can be used
			    as realm too. The realm value is configured in
				scripts as the first parameter of all
				<command>{www|proxy}_{authorize|challenge}</command>
				actions.
			</para>
		    </answer>
		</qandaentry>
		<qandaentry>
		    <question>
			<para>
			    On a multihomed host, forwarded messages carry other 
			    interface in Via than used for sending, or messages 
			    are not sent and an error log is issued "invalid 
			    sendtoparameters one possible reason is the server 
			    is bound to localhost".
			</para>
		    </question>
		    <answer>
			<anchor id="mhomed">
			<para>
			    Set the configuration option <varname>mhomed</varname>
			    to "1". <application moreinfo="none">ser</application>
			    will then attempt to calculate the correct interface.
			    It's not done by default as it degrades performance
			    on single-homed hosts or multi-homed hosts that are
			    not set-up as routers.
			</para>
		    </answer>
		</qandaentry>
		<qandaentry>
		    <question>
			<para>
			    I receive "ERROR: t_newtran: transaction already in process" in my logs.
			</para>
		    </question>
		    <answer>
			<para>
			    That looks like an erroneous use of tm module in script.
			    tm can handle only one transaction per request. If you
			    attempt to instantiate a transaction multiple times,
			    <application moreinfo="none">ser</application> will complain.
			    Anytime any of <command moreinfo="none">t_newtran</command>,
			    <command moreinfo="none">t_relay</command> or 
			    <command moreinfo="none">t_relay_to_udp</command> actions is
			    encountered, tm attempts to instantiate a transaction.
			    Doing so twice fails. Make sure that any of this
			    commands is called only once during script execution.
			</para>
		    </answer>
		</qandaentry>
		<qandaentry>
		    <question>
			<para>
			    I try to add an alias but 
			    <command moreinfo="none">serctl</command>
			    complains that table does not exist.
			</para>
		    </question>
		    <answer>
			<para>
			    You need to run <application moreinfo="none">ser</application>
			    and use the command
			    <command moreinfo="none">lookup("aliases")</command>
			    in its routing script. That's because the table 
			    of aliases is
			    stored in cache memory for high speed. The cache
			    memory is only set up when the 
			    <application moreinfo="none">ser</application>
			    is running and configured to use it. If that is
			    not the case, 
			    <application moreinfo="none">serctl</application>
			    is not able to manipulate the aliases table.
			</para>
		    </answer>
		</qandaentry>

	    <qandaentry>
		<question>
		    <para>I started <application>ser</application> with
			<varname>children=4</varname> but many more processes
			were started. What is wrong?
			</para>
		    </question>
		<answer>
		    <para>
			That's ok. The <varname>children</varname> parameter defines
			how many children should process each transport protocol in
			parallel. Typically, the server listens to multiple protocols
			and starts other supporting processes like timer or FIFO
			server too. Call <application>serctl ps</application> to watch
			running processes.
			</para>
		    </answer>
		</qandaentry>
	    <qandaentry>
		<question>
		    <para>
			I decided to use a compiled version of <application>ser</application>
			but it does not start any more.
			</para>
		    </question>
		<answer>
		    <para>
			You probably kept the same configuration file, which tries to load modules
			from the binary distribution you used previously. Make sure that modules
			paths are valid and point to where you compiled <application>ser</application>.
			Also, watch logs for error messages "ERROR: load_module: could not open 
			module".
			</para>
		    </answer>
		</qandaentry>
	    
	    </qandaset>
	</section> <!-- troubleshooting -->
    </chapter> <!-- operation -->
