freeswitch/conf/freeswitch.xml

<?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>