--- /dev/null
+<?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="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/$1"/>
+ </routes>
+ </configuration>
+
+ <configuration name="ivr.conf" description="IVR menus">
+ <menus>
+ <menu name="main"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
+ <entry action="menu-exit" digits="*"/>
+ <entry action="menu-sub" digits="2" param="menu2"/>
+ <entry action="menu-exec-api" digits="3" param="api arg"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-back" digits="5"/>
+ <entry action="menu-call-transfer" digits="7" param="888"/>
+ <entry action="menu-sub" digits="8" param="menu8"/>>
+ </menu>
+ <menu name="menu8"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav"
+ timeout ="15"
+ max-failures="3">
+ <entry action="menu-back" digits="#"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-top" digits="*"/>
+ </menu>
+ <menu name="menu2"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav"
+ timeout ="15"
+ max-failures="3">
+ <entry action="menu-back" digits="#"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-top" digits="*"/>
+ </menu>
+ </menus>
+ </configuration>
+
+ </section>
+
+ <section name="dialplan" description="Regex/XML Dialplan">
+ <!-- Valid fields in conditions: -->
+ <!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
+ <!-- rdnis, destination_number, uuid, source, context, chan_name" -->
+
+ <!-- *NOTE* The special context name 'any' will match any context -->
+ <context name="default">
+ <extension name="556"> <!-- demo phrases -->
+ <condition field="destination_number" expression="^556$">
+ <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>
+
+
projects / thirdparty / freeswitch.git / commitdiff
