<?xml version="1.0"?>
<document type="freeswitch/xml">
<section name="configuration" description="Various Configuration">
<configuration name="switch.conf" description="Modules">
<settings>
<!--Most channels to allow at once -->
<param name="max-sessions" value="1000"/>
</settings>
</configuration>
<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<!-- <load module="mod_syslog"/> -->
<!-- XML Interfaces -->
<!-- <load module="mod_xml_rpc"/> -->
<!-- Event Handlers -->
<!-- <load module="mod_event_multicast"/> -->
<!-- <load module="mod_event_test"/> -->
<!-- <load module="mod_zeroconf"/> -->
<!-- <load module="mod_xmpp_event"/> -->
<!-- <load module="mod_event_socket"/> -->
<!-- Directory Interfaces -->
<!-- <load module="mod_ldap"/> -->
<!-- Endpoints -->
<load module="mod_exosip"/>
<!--<load module="mod_iax"/>-->
<load module="mod_portaudio"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_wanpipe"/> -->
<!-- <load module="mod_dingaling"/> -->
<!-- Applications -->
<load module="mod_bridgecall"/>
<load module="mod_echo"/>
<load module="mod_dptools"/>
<!-- <load module="mod_ivrtest"/> -->
<load module="mod_playback"/>
<load module="mod_commands"/>
<!-- <load module="mod_commands"/> -->
<!-- Dialplan Interfaces -->
<load module="mod_dialplan_xml"/>
<!-- <load module="mod_dialplan_directory"/> -->
<!-- Codec Interfaces -->
<load module="mod_g711"/>
<load module="mod_gsm"/>
<load module="mod_l16"/>
<!-- <load module="mod_speex"/> -->
<!-- <load module="mod_ilbc"/> -->
<!-- File Format Interfaces -->
<load module="mod_sndfile"/>
<!-- Timers -->
<load module="mod_softtimer"/>
<!-- Languages -->
<!-- <load module="mod_spidermonkey"/> -->
<!-- <load module="mod_perl"/> -->
<!-- ASR /TTS -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Conference Bridges -->
<!--<load module="mod_conference"/>-->
</modules>
</configuration>
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="listen-ip" value="127.0.0.1"/>
<param name="listen-port" value="8021"/>
<param name="password" value="ClueCon"/>
</settings>
</configuration>
<configuration name="iax.conf" description="IAX Configuration">
<settings>
<param name="debug" value="0"/>
<!-- <param name="ip" value="1.2.3.4"> -->
<param name="port" value="4569"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
<param name="codec-master" value="us"/>
<param name="codec-rates" value="8"/>
</settings>
</configuration>
<configuration name="console.conf" description="Console Logger">
<!-- pick a file name, a function name or 'all' -->
<!-- map as many as you need for specific debugging -->
<mappings>
<!-- <param name="log_event" value="DEBUG"/> -->
<param name="all" value="DEBUG"/>
</mappings>
</configuration>
<configuration name="sofia.conf" description="sofia Endpoint">
<profile name="test">
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="PCMU@20i"/>
<param name="codec-ms" value="20"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-ip" value="192.168.1.20"/>
<param name="sip-ip" value="192.168.1.20"/>
<!-- optional ; -->
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
</profile>
</configuration>
<configuration name="syslog.conf" description="Syslog Logger">
<!-- SYSLOG -->
<!-- emerg - system is unusable -->
<!-- alert - action must be taken immediately -->
<!-- crit - critical conditions -->
<!-- err - error conditions -->
<!-- warning - warning conditions -->
<!-- notice - normal, but significant, condition -->
<!-- info - informational message -->
<!-- debug - debug-level message -->
<settings>
<param name="ident" value="freeswitch"/>
<param name="facility" value="user"/>
<param name="format" value="${time} - ${message}"/>
<param name="level" value="debug,info,warning-alert"/>
</settings>
</configuration>
<configuration name="exosip.conf" description="Exosip Endpoint">
<settings>
<param name="port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<!-- the @20 is optional number of ms you want to use. Use it only
if you know the codec supports it -->
<param name="codec-prefs" value="PCMU@20i,PCMA@20i"/>
<!-- Example to call for speex in wideband 16k mode
you can have up to 2 '@; after the codec name followed by either
'i' (interval eg 20i for 20ms) or 'k' (kilohertz eg 16000k for 16khz)-->
<!--<param name="codec-prefs" value="SPEEX@16000k"/>-->
<!-- Payload number to bind DTMF to-->
<param name="rfc2833-pt" value="101"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- auto sense NAT issues and adjust accordingly -->
<param name="use-rtp-auto-adjust" value="true"/>
<!-- pick one (default if not specified is 'guess'); -->
<param name="rtp-ip" value="guess"/>
<!-- <param name-"rtp-ip" value="10.0.0.1"/> -->
<!-- leave commented or 0.0.0.0 for all ip -->
<!-- <param name="sip-ip" value="127.0.0.1"/> -->
<!-- optional ; -->
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
<!-- specify 'myrealm' with certian key -->
<!-- use !myrealm! at beginning of url to activate -->
<!-- exosip/!myrealm!1000@dest -->
<!-- srtp:<param name="myrealm" value="ffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
</settings>
</configuration>
<configuration name="woomera.conf" description="Woomera Endpoint">
<settings>
<param name="debug" value="0"/>
</settings>
</configuration>
<configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
<settings>
<param name="debug" value="1"/>
<param name="dialplan" value="XML"/>
<param name="mtu" value="320"/>
<param name="dtmf-on" value="800"/>
<param name="dtmf-off" value="100"/>
<param name="supress-dtmf-tone" value="yes"/>
</settings>
<span>
<param name="span" value="1"/>
<param name="node" value="cpe"/>
<!-- <param name="switch" value="ni2"/> -->
<param name="switch" value="dms100"/>
<!-- <param name="switch" value="lucent5e"/> -->
<!-- <param name="switch" value="att4ess"/> -->
<!-- <param name="switch" value="euroisdn"/> -->
<!-- <param name="switch" value="gr303eoc"/> -->
<!-- <param name="switch" value="gr303tmc"/> -->
<param name="dp" value="national"/>
<!-- <param name="dp" value="international"/> -->
<!-- <param name="dp" value="local"/> -->
<!-- <param name="dp" value="private"/> -->
<!-- <param name="dp" value="unknown"/> -->
<param name="l1" value="ulaw"/>
<!-- <param name="l1" value="alaw"/> -->
<param name="bchan" value="1-23"/>
<param name="dchan" value="24"/>
<param name="dialplan" value="XML"/>
</span>
</configuration>
<configuration name="portaudio.conf" description="Soundcard Endpoint">
<settings>
<param name="debug" value="2"/>
<param name="dialplan" value="XML"/>
<!-- partial string match on something in the name or the device # -->
<param name="indev" value="USB"/>
<param name="outdev" value="USB"/>
<param name="cid-name" value="FreeSwitch"/>
<param name="cid-num" value="5555551212"/>
</settings>
</configuration>
<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
<param name="publish" value="yes"/>
<param name="browse" value="_sip._udp"/>
</settings>
</configuration>
<configuration name="xmpp_event.conf" description="XMPP Event Handler">
<settings>
<param name="#debug" value="1"/>
<param name="jid" value="freeswitch@my.jabber.com/me"/>
<param name="passwd" value="mypass"/>
<param name="target-jid" value="freeswitch@reader.org/him"/>
</settings>
</configuration>
<configuration name="dialplan_directory.conf" description="Dialplan Directory">
<settings>
<param name="directory-name" value="ldap"/>
<param name="host" value="ldap.freeswitch.org"/>
<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
<param name="pass" value="test"/>
<param name="base" value="dc=freeswitch,dc=org"/>
</settings>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<!-- *NOTE* your resource (after the /) MUST contain the string "talk" (upper or lower case is ok) -->
<!-- *NOTE* as of May 2 2006 you must set"auto-login" to"true" if you want to be able to auto-login on startup"/> -->
<interface>
<param name="name" value="jingle"/>
<param name="login" value="myjid@myserver.com/talk"/>
<param name="password" value="mypass"/>
<param name="dialplan" value="XML"/>
<param name="message" value="Jingle all the way"/>
<param name="rtp-ip" value="10.0.0.1"/>
<param name="auto-login" value="true"/>
<!-- SASL "plain" or "md5" -->
<param name="sasl" value="plain"/>
<!-- if the server where the jabber is hosted is not the same
as the one in the jid -->
<!--<param name="server" value="alternate.server.com"/>-->
<!-- Enable TLS or not -->
<param name="tls" value="true"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- or -->
<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
<!-- default extension (if one cannot be determined) -->
<param name="exten" value="888"/>
<!-- VAD choose one -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<param name="vad" value="both"/>
</interface>
</configuration>
<configuration name="xml_rpc.conf" description="XML RPC">
<settings>
<!-- The port where you want to run the http service (default 8080) -->
<param name="http-port" value="8080"/>
<!-- if all 3 of the following params exist all http traffic will require auth -->
<param name="auth-realm" value="freeswitch"/>
<param name="auth-user" value="freeswitch"/>
<param name="auth-pass" value="works"/>
<!-- The url to a gateway cgi that can generate xml similar to
what's in this file only on-the-fly (leave it commented if you dont
need it) -->
<!-- one or more |-delim of configuration|directory|dialplan -->
<!-- <param name="gateway-url" value="http://www.server.com/gateway.cgi" bindings="configuration"/> -->
</settings>
</configuration>
<configuration name="rss.conf" description="RSS Parser">
<feeds>
<!-- Just download the files to wherever and refer to them here -->
<!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
<!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
</feeds>
</configuration>
<!-- None of these paths are real if you want any of these options
you need to really set them up -->
<configuration name="conference.conf" description="Audio Conference">
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
<profile name="default">
<!-- Sample Rate-->
<param name="rate" value="8000"/>
<!-- Number of milliseconds per frame -->
<param name="interval" value="20"/>
<!-- Energy level required for audio to be sent to the other users -->
<param name="energy-level" value="300"/>
<!-- TTS Engine to use -->
<!--<param name="tts-engine" value="cepstral"/>-->
<!-- TTS Voice to use -->
<!--<param name="tts-voice" value="david"/>-->
<!-- If TTS is enabled all audio-file params not beginning with '/'
will be considered text to say with TTS -->
<!-- File to play to acknowledge succees -->
<!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
<!-- File to play to acknowledge failure -->
<!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
<!-- File to play to acknowledge muted -->
<!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
<!-- File to play to acknowledge unmuted -->
<!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
<!-- File to play if you are alone in the conference -->
<!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
<!-- File to play when you join the conference -->
<!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
<!-- File to play when you leave the conference -->
<!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
<!-- File to play when you ae ejected from the conference -->
<!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
<!-- File to play when the conference is locked -->
<!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
<!-- File to play to prompt for a pin -->
<!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
<!-- File to play to when the pin is invalid -->
<!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="FreeSWITCH"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="8777423583"/>
</profile>
</profiles>
</configuration>
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<!-- Valid fields in conditions:
"dialplan, caller_id_name, ani, ani2, caller_id_number,
network_addr, rdnis, destination_number, uuid, source,
context, chan_name" -->
<!-- *NOTE* The special context name 'any' will match any context -->
<context name="default">
<extension name="tollfree">
<condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
<action application="bridge" data="exosip/$1-freeswitch@voip.trxtel.com"/>
</condition>
</extension>
<extension name="devconf">
<condition field="destination_number" expression="^888$">
<action application="bridge" data="exosip/888@66.250.68.194"/>
</condition>
</extension>
<extension name="testmusic">
<condition field="destination_number" expression="^1234$">
<action application="bridge" data="exosip/1234@66.250.68.194"/>
</condition>
</extension>
<!-- Enter an existing conference -->
<extension name="1000">
<condition field="destination_number" expression="^1000$">
<action application="conference" data="freeswitch"/>
</condition>
</extension>
<!-- Start a dynamic conference and call someone at the same time -->
<extension name="2000">
<condition field="destination_number" expression="^2000$">
<action application="conference" data="bridge:mydynaconf:exosip/1234@66.250.68.194"/>
</condition>
</extension>
<!-- if the destination is an exact match on the extension name
you do not need any regex in the condition
<extension name="999">
<condition><action application="bridge" data="exosip/888@66.250.68.194"/></condition>
</extension>-->
<!-- extensions starting with 4, all the numbers after 4 form a numeric filename
continue=true means keep looking for more extensions to match
*NOTE* The entire dialplan is parsed ONCE when the call starts
so any call info acquired after the various actions cannot
be taken into consideration.
The first match will play a beep and the second one plays
the desired file. This is for demo purposes both actions
could have been under the same <extension> tag as well.
-->
<extension name="playsound1" continue="true">
<condition field="source" expression="mod_exosip"/>
<condition field="destination_number" expression="^4(\d+)">
<action application="playback" data="/var/sounds/beep.gsm"/>
</condition>
</extension>
<extension name="playsound2">
<condition field="source" expression="mod_exosip"/>
<condition field="destination_number" expression="^4(\d+)">
<action application="playback" data="/root/$1.raw"/>
</condition>
</extension>
<!-- send everything with a certian RDNIS to Wanpipe ISDN -->
<extension name="To PRI">
<condition field="rdnis" expression="8881231234"/>
<condition field="destination_number" expression="(.*)">
<action application="bridge" data="wanpipe/a/a/$1"/>
</condition>
</extension>
<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
<extension name="9999">
<condition field="source" expression="mod_iax"/>
<condition field="destination_number" expression="9999">
<action application="playback" data="/var/sounds/beep.gsm"/>
</condition>
</extension>
<!-- Call the FreeSWITCH conference via SIP -->
<extension name="FreeSWITCH Conference SIP">
<condition field="destination_number" expression="^888$">
<action application="bridge" data="exosip/888@66.250.68.194"/>
</condition>
</extension>
<!-- Call the FreeSWITCH conference via IAX -->
<extension name="FreeSWITCH Conference IAX">
<condition field="destination_number" expression="^8888$">
<action application="bridge" data="iax/guest@66.250.68.194/888"/>
</condition>
</extension>
</context>
</section>
<section name="directory" description="User Directory">
</section>
</document>