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>
      <!--Any variables defined here will be available in every channel, in the dialplan etc -->
      <variables>
	<variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/>
	<variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/>
	<variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/>
      </variables>
    </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_cdr"/> -->
        <!-- <load module="mod_event_multicast"/> -->
        <!-- <load module="mod_event_socket"/> -->
        <!-- <load module="mod_xmpp_event"/> -->
        <!-- <load module="mod_zeroconf"/> -->

        <!-- Directory Interfaces -->
        <!-- <load module="mod_ldap"/> -->

        <!-- Endpoints -->
        <!-- <load module="mod_dingaling"/> -->
        <!--<load module="mod_iax"/>-->
        <load module="mod_portaudio"/>
        <load module="mod_sofia"/>
        <!-- <load module="mod_wanpipe"/> -->
        <!-- <load module="mod_woomera"/> -->

        <!-- Applications -->
        <load module="mod_bridgecall"/>
        <load module="mod_commands"/>
        <!--<load module="mod_conference"/>-->
        <load module="mod_dptools"/>
        <load module="mod_echo"/>
        <!--<load module="mod_park"/>-->
        <load module="mod_playback"/>

        <!-- Dialplan Interfaces -->
        <!-- <load module="mod_dialplan_directory"/> -->
        <load module="mod_dialplan_xml"/>

        <!-- Codec Interfaces -->
        <load module="mod_g711"/>
        <load module="mod_gsm"/>
        <!-- <load module="mod_ilbc"/> -->
        <load module="mod_l16"/>
        <!-- <load module="mod_speex"/> -->

        <!-- File Format Interfaces -->
        <load module="mod_sndfile"/>
        <load module="mod_native_file"/>

        <!-- Timers -->
        <load module="mod_softtimer"/>

        <!-- Languages -->
        <!-- <load module="mod_spidermonkey"/> -->
        <!-- <load module="mod_perl"/> -->

        <!-- ASR /TTS -->
        <!-- <load module="mod_cepstral"/> -->
        <!-- <load module="mod_rss"/> -->

      </modules>
    </configuration>

    <configuration name="spidermonkey.conf" description="Spider Monkey JavaScript Plug-Ins">
      <modules>
	<load module="mod_spidermonkey_teletone"/>
	<load module="mod_spidermonkey_core_db"/>
      </modules>
    </configuration>

    <configuration name="event_multicast.conf" description="Multicast Event">
      <settings>
        <param name="address" value="225.1.1.1"/>
        <param name="port" value="4242"/>
        <param name="bindings" value="all"/>
      </settings>
    </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">
      <profiles>
        <profile name="mydomain1.com">
          <registrations>
            <!-- <registration name="asterlink">
              <param name="register-scheme" value="Digest"/>
              <param name="register-realm" value=""/>
              <param name="register-username" value="1001"/>
              <param name="register-password" value="nhy65tgb"/>
              <param name="register-from" value="sip:1001@208.64.200.40"/>
              <param name="register-to" value="sip:1001@66.250.68.194"/>
              <param name="register-proxy" value="sip:66.250.68.194:5060"/>
              <param name="register-frequency" value="20"/>
            </registration> -->
          </registrations>
          <settings>
            <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-timer-name" value="soft"/>
            <param name="rtp-ip" value="192.168.1.20"/>
            <param name="sip-ip" value="mydomain1.com"/>

            <!-- this lets anything register -->
            <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
            <param name="accept-blind-reg" value="true"/>

            <!--<param name="auth-calls" value="true"/>-->
            <!-- on authed calls, authenticate *all* the packets not just invite -->
            <!--<param name="auth-all-packets" value="true"/>-->

            <!-- 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"/>-->
          </settings>
        </profile>
      </profiles>
    </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="woomera.conf" description="Woomera Endpoint">
      <settings>
        <param name="debug" value="0"/>
      </settings>
      <interface>
        <param name="host" value="localhost"/>
        <param name="port" value="42420"/>
        <param name="audio-ip" value="127.0.0.1"/>
        <param name="dialplan" value="XML"/>
      </interface>
    </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* change <x-profile></x-profile> to <profile></profile> to enable -->

      <!-- Client Profile (Original mode) -->
      <x-profile type="client">
        <param name="name" value="mydomain.com"/>
        <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"/>
	<param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/>
        <!-- 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"/>
      </x-profile>

      <!-- Component (Server to Server Login) -->
      <x-profile type="component">
	<!-- All traffic for *@sub.mydomain.com will come to you -->
	<param name="name" value="sub.mydomain.com"/>
        <param name="password" value="secret"/>
	<param name="dialplan" value="XML"/>
	<param name="rtp-ip" value="208.64.200.42"/>
	<param name="server" value="jabber.server.org:5347"/>
        <!-- disable to trade async for more calls -->
        <param name="use-rtp-timer" value="true"/>
	<!-- "_auto_" means the extension will be automaticly set to the called jid -->
        <param name="exten" value="_auto_"/>
        <!--<param name="vad" value="both"/>-->
      </x-profile>

    </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 -->
          <!-- '/' or with drive: (i.e. c:) 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, -->
    <!-- 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="sofia/test/$1-freeswitch@voip.trxtel.com"/>
        </condition>
      </extension>

      <!-- Call the FreeSWITCH conference via SIP -->
        <!--<extension name="FreeSWITCH Conference SIP">-->
          <!--<condition field="destination_number" expression="^888$">-->
            <!--<action application="bridge" data="sofia/test/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>-->

      <extension name="testmusic">
        <condition field="destination_number" expression="^1234$">
	  <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
	  <action application="set" data="ringback=${us-ring}"/>
	  <!--<action application="set" data="ringback=/home/ring.wav"/>-->
          <action application="bridge" data="sofia/test/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:sofia/test/1234@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_sofia"/>
        <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_sofia"/>
        <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>

    </context>
  </section>

  <section name="directory" description="User Directory">
    <!--the domain or ip (the right hand side of the @ in the addr-->
    <domain name="jabber.org">
      <!--the user id (the left hand side of the @ in the addr-->
      <user id="stpeter">
	<params>
	  <!-- omit password for authless registration -->
	  <param name="password" value="mypass"/>
	</params>
	
	<vcard xmlns='vcard-temp'>
	  <FN>Peter Saint-Andre</FN>
	  <N>
	    <FAMILY>Saint-Andre</FAMILY>
	    <GIVEN>Peter</GIVEN>
	    <MIDDLE/>
	  </N>
	  <NICKNAME>stpeter</NICKNAME>
	  <URL>http://www.jabber.org/people/stpeter.php</URL>
	  <BDAY>1966-08-06</BDAY>
	  <ORG>
	    <ORGNAME>Jabber Software Foundation</ORGNAME>
	    <ORGUNIT>Jabber Software Foundation</ORGUNIT>
	  </ORG>
	  <TITLE>Executive Director</TITLE>
	  <ROLE>Patron Saint</ROLE>
	  <TEL><WORK/><VOICE/><NUMBER>303-308-3282</NUMBER></TEL>
	  <TEL><WORK/><FAX/><NUMBER/></TEL>
	  <TEL><WORK/><MSG/><NUMBER/></TEL>
	  <ADR>
	    <WORK/>
	    <EXTADD>Suite 600</EXTADD>
	    <STREET>1899 Wynkoop Street</STREET>
	    <LOCALITY>Denver</LOCALITY>
	    <REGION>CO</REGION>
	    <PCODE>80202</PCODE>
	    <CTRY>USA</CTRY>
	  </ADR>
	  <TEL><HOME/><VOICE/><NUMBER>303-555-1212</NUMBER></TEL>
	  <TEL><HOME/><FAX/><NUMBER/></TEL>
	  <TEL><HOME/><MSG/><NUMBER/></TEL>
	  <ADR>
	    <HOME/>
	    <EXTADD/>
	    <STREET/>
	    <LOCALITY>Denver</LOCALITY>
	    <REGION>CO</REGION>
	    <PCODE>80209</PCODE>
	    <CTRY>USA</CTRY>
	  </ADR>
	  <EMAIL><INTERNET/><PREF/><USERID>stpeter@jabber.org</USERID></EMAIL>
	  <JABBERID>stpeter@jabber.org</JABBERID>
	  <DESC>
	    More information about me is located on my 
	    personal website: http://www.saint-andre.com/
	  </DESC>
	</vcard>

      </user>
    </domain>
  </section>
</document>