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"/> -->

	<!-- Multi-Faceted -->
	<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
	<load module="mod_enum"/>

	<!-- XML Interfaces -->
	<!-- <load module="mod_xml_rpc"/> -->
	<!-- <load module="mod_xml_curl"/> -->

	<!-- 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"/>
	<!--<load module="mod_spidermonkey_odbc"/>-->
      </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@conference.freeswitch.org"/>
		 <param name="register-proxy" value="sip:conference.freeswitch.org: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="auto"/>
	    <param name="sip-ip" value="auto"/>

	    <!--Uncomment to set all inbound calls to no media mode-->
	    <!--<param name="inbound-no-media" value="true"/>-->

	    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
	    <!--<param name="inbound-late-negotiation" value="true"/>-->

	    <!-- 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>
	<!-- indev, outdev, ringdev: 
	     partial case sensitive string match on something in the name 
	     or the device number prefixed with # eg "#1" (or blank for default) -->

	<!-- device to use for input -->
	<param name="indev" value=""/>
	<!-- device to use for output -->
	<param name="outdev" value=""/>

	<!--device to use for inbound ring -->
	<!--<param name="ringdev" value=""/>-->
	<!--File to play as the ring sound -->
	<!--<param name="ring-file" value="/sounds/ring.wav"/>-->
	<!--Number of seconds to pause between rings -->
	<!--<param name="ring-interval" value="5"/>-->

	<!--file to play when calls are on hold-->
	<!--<param name="hold-file" value="/sounds/holdmusic.wav"/>-->
	<!--Timer to use for hold music (i'd leave this one commented)-->
	<!--<param name="timer-name" value="soft"/>-->

	<!--Default dialplan and caller-id info -->
	<param name="dialplan" value="XML"/>
	<param name="cid-name" value="FreeSwitch"/>
	<param name="cid-num" value="5555551212"/>

	<!--audio sample rate and interval -->
	<param name="sample-rate" value="8000"/>
	<param name="codec-ms" value="20"/>
      </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="auto"/>
	<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="auto"/> -->
	<!-- <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="auto"/>
	<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_curl.conf" description="cURL XML Gateway">
      <settings>
	<!-- 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.mydomain.com/test.cgi" bindings="dialplan"/>-->
	<!-- set this to provide authentication credentials to the server -->
	<!--<param name="gateway-credentials" value="muser:mypass"/>-->
      </settings>
    </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"/>
      </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">
      <!-- Advertise certian presence on startup . -->
      <advertise>
	<room name="888@sub.mydomain.com" status="FreeSWITCH"/>
      </advertise>

<!-- These are the default keys that map when you do not specify a caller control group -->	
<!-- Note: none and default are reserved names for group names -->	
	<caller-controls>
	  <group name="default">
	   <control action="mute" digits="0"/>
	   <control action="deaf mute" digits="*"/>
	   <control action="energy up" digits="9"/>
	   <control action="energy equ" digits="8"/>
	   <control action="energy dn" digits="7"/>
	   <control action="vol talk up" digits="3"/>
	   <control action="vol talk zero" digits="2"/>
	   <control action="vol talk dn" digits="1"/>
	   <control action="vol listen up" digits="6"/>
	   <control action="vol listen zero" digits="5"/>
	   <control action="vol listen dn" digits="4"/>
	   <control action="hangup" digits="#"/>
	  </group>
	</caller-controls>

      <!-- Profiles are collections of settings you can reference by name. -->
      <profiles>
	<!--If no profile is specified it will default to "default"-->
	<profile name="default">
	  <!-- Domain (for presence) -->
	  <param name="domain" value="sub.mydomain.com"/>
	  <!-- 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"/>
	  <!-- Name of the caller control group to use for this profile -->
	  <!-- <param name="caller-controls" value="some name"/> -->
	  <!-- 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 beginning with -->
	  <!-- 'say:' will be considered text to say with TTS -->
	  <!-- Set a default path here so you can use relative paths in the other sound params-->
	  <!--<param name="sound-prefix" value="/soundfiles"/>-->
	  <!-- File to play to acknowledge succees -->
	  <!--<param name="ack-sound" value="beep.wav"/>-->
	  <!-- File to play to acknowledge failure -->
	  <!--<param name="nack-sound" value="beeperr.wav"/>-->
	  <!-- File to play to acknowledge muted -->
	  <!--<param name="muted-sound" value="muted.wav"/>-->
	  <!-- File to play to acknowledge unmuted -->
	  <!--<param name="unmuted-sound" value="unmuted.wav"/>-->
	  <!-- File to play if you are alone in the conference -->
	  <!--<param name="alone-sound" value="yactopitc.wav"/>-->
	  <!-- File to play when you join the conference -->
	  <!--<param name="enter-sound" value="welcome.wav"/>-->
	  <!-- File to play when you leave the conference -->
	  <!--<param name="exit-sound" value="exit.wav"/>-->
	  <!-- File to play when you ae ejected from the conference -->
	  <!--<param name="kicked-sound" value="kicked.wav"/>-->
	  <!-- File to play when the conference is locked -->
	  <!--<param name="locked-sound" value="locked.wav"/>-->
	  <!-- File to play when the conference is locked during the call-->
	  <!--<param name="is-locked-sound" value="is-locked.wav"/>-->
	  <!-- File to play when the conference is unlocked during the call-->
	  <!--<param name="is-unlocked-sound" value="is-unlocked.wav"/>-->
	  <!-- File to play to prompt for a pin -->
	  <!--<param name="pin-sound" value="pin.wav"/>-->
	  <!-- File to play to when the pin is invalid -->
	  <!--<param name="bad-pin-sound" value="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>

    <configuration name="enum.conf" description="ENUM Module">
      <settings>
	<param name="default-root" value="e164.org"/>
      </settings>

      <routes>
	<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
	<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
	<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
      </routes>
    </configuration>

    <configuration name="ivr.conf" description="IVR menus">
      <menus>
	<menu name="main"
	      greet-long="/soundfiles/greet-long.wav" 
	      greet-short="/soundfiles/greet-short.wav"
	      invalid-sound="/soundfiles/invalid.wav"
	      exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
	  <entry action="menu-exit" digits="*"/>
	  <entry action="menu-sub" digits="2" param="menu2"/>
	  <entry action="menu-exec-api" digits="3" param="api arg"/>
	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
	  <entry action="menu-back" digits="5"/>
	  <entry action="menu-call-transfer" digits="7" param="888"/>
	  <entry action="menu-sub" digits="8" param="menu8"/>>
	</menu>
	<menu name="menu8"
	      greet-long="/soundfiles/greet-long.wav"
	      greet-short="/soundfiles/greet-short.wav"
	      invalid-sound="/soundfiles/invalid.wav"
	      exit-sound="/soundfiles/exit.wav"
	      timeout ="15"
	      max-failures="3">
	  <entry action="menu-back" digits="#"/>
	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
	  <entry action="menu-top" digits="*"/>
	</menu>
	<menu name="menu2"
	      greet-long="/soundfiles/greet-long.wav"
	      greet-short="/soundfiles/greet-short.wav"
	      invalid-sound="/soundfiles/invalid.wav"
	      exit-sound="/soundfiles/exit.wav"
	      timeout ="15"
	      max-failures="3">
	  <entry action="menu-back" digits="#"/>
	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
	  <entry action="menu-top" digits="*"/>
	</menu>
      </menus>
    </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="556"> <!-- demo phrases -->
        <condition field="destination_number" expression="^556$">
          <action application="answer"/>
          <action application="sleep" data="1000"/>
          <action application="phrase" data="spell,${caller_id_name}"/>
          <action application="phrase" data="spell-phonetic,${caller_id_name}"/>
          <action application="phrase" data="timespec,12:45:15"/>
          <action application="phrase" data="saydate,0"/>
          <action application="phrase" data="msgcount,130"/>
          <action application="phrase" data="ip-addr,66.250.68.194"/>
          <action application="phrase" data="saydate,$strepoch(2006-03-23 7:23)"/>
          <!--<action application="phrase" data="timeleft,3:30"/>-->
	</condition>
      </extension>

      <extension name="tollfree">
	<condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
	  <action application="enum" data="$1"/>
	  <action application="bridge" data="${enum_auto_route}"/>
	</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@conference.freeswitch.org"/>-->
      <!--</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@conference.freeswitch.org/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@conference.freeswitch.org"/>
	</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@conference.freeswitch.org"/>
	</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>

  <!-- phrases section (under development still) -->
  <section name="phrases" description="Speech Phrase Management">
    <macros>
      <language name="en" sound_path="/snds" tts_engine="cepstral" tts_voice="david">
	<macro name="msgcount">
	  <input pattern="(.*)">
	    <match>
	      <action function="execute" data="sleep(1000)"/>
	      <action function="play-file" data="vm-youhave.wav"/>
	      <action function="say" data="$1" method="pronounced" type="items"/>
	      <action function="play-file" data="vm-messages.wav"/>
	      <!-- or -->
	      <!--<action function="speak-text" data="you have $1 messages"/>-->
	    </match>
	  </input>
	</macro>
	<macro name="saydate">
	  <input pattern="(.*)">
	    <match>
	      <action function="say" data="$1" method="pronounced" type="current_date_time"/>
	    </match>
	  </input>
	</macro>
	<macro name="timespec">
	  <input pattern="(.*)">
	    <match>
	      <action function="say" data="$1" method="pronounced" type="time_measurement"/>
	    </match>
	  </input>
	</macro>
	<macro name="ip-addr">
	  <input pattern="(.*)">
	    <match>
	      <action function="say" data="$1" method="iterated" type="ip_address"/>
	      <action function="say" data="$1" method="pronounced" type="ip_address"/>
	    </match>
	  </input>
	</macro>
	<macro name="spell">
	  <input pattern="(.*)">
	    <match>
	      <action function="say" data="$1" method="pronounced" type="name_spelled"/>
	    </match>
	  </input>
	</macro>
	<macro name="spell-phonetic">
	  <input pattern="(.*)">
	    <match>
	      <action function="say" data="$1" method="pronounced" type="name_phonetic"/>
	    </match>
	  </input>
	</macro>
	<macro name="tts-timeleft">
	  <!-- The parser will visit each <input> tag and execute the actions in <match> or <nomatch> depending on the pattern param -->
	  <!-- If the function "break" is encountered all parsing will cease -->
	  <input pattern="(\d+):(\d+)">
	    <match>
	      <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/>
	      <action function="break"/>
	    </match>
	    <nomatch>
	      <action function="speak-text" data="That input was invalid."/>
	    </nomatch>
	  </input>
	  <input pattern="(\d+) min (\d+) sec">
	    <match>
	      <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/>
	    </match>
	    <nomatch>
	      <action function="speak-text" data="That input was invalid."/>
	    </nomatch>
	  </input>
	</macro>
      </language>
      <language name="fr" sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral" tts_voice="jean-pierre">
	<macro name="msgcount">
	  <input pattern="(.*)">
	    <match>
	      <action function="play-file" data="tuas.wav"/>
	      <action function="say" data="$1" method="pronounced" type="items"/>
	      <action function="play-file" data="messages.wav"/>
	    </match>
	  </input>
	</macro>
	<macro name="timeleft">
	  <input pattern="(\d+):(\d+)">
	    <match>
	      <action function="speak-text" data="il y a $1 minutes et de $2 secondes de restant"/>
	    </match>
	  </input>
	</macro>
      </language>
    </macros>
  </section>
</document>