<?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"/>
<!--RTP port range -->
<!--<param name="rtp-start-port" value="16384"/>-->
<!--<param name="rtp-end-port" value="32768"/>-->
</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="192.168.1.20"/>
<param name="sip-ip" value="mydomain1.com"/>
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-no-media" 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="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_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>
<configuration name="mod_cdr.conf" description="CDR Configuration">
<pddcdr>
<param name="path" value="/work/temp/pddcdr"/>
<!-- <param name="chanvars" value="username,accountcode"/> -->
<!-- <param name="chanvars_fixed" value="foo"/> -->
</pddcdr>
<csvcdr>
<param name="path" value="/work/temp/csvcdr"/>
<param name="size_limit" value="25"/>
<!-- <param name="chanvars_fixed" value="username"/> -->
<!-- <param name="chanvars_supp" value="*"/> -->
<!-- <param name="repeat_fixed_in_supp" value="0"/> -->
</csvcdr>
<mysqlcdr>
<param name="hostname" value="10.0.0.1"/>
<param name="username" value="test"/>
<param name="password" value="test"/>
<param name="dbname" value="testing"/>
<!-- This following line logs username as a varchar, and The_Kow as a tiny -->
<!-- <param name="chanvars_fixed" value="username=s,The_Kow=t"/> -->
<!-- If you do not want to log any and all chanvars to the chanvar table, comment the next 2 lines out -->
<param name="chanvars_supp" value="*"/>
<param name="chanvars_supp_repeat_fixed" value="0"/>
</mysqlcdr>
</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>