+++ /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"/>
- <!--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>
-
-
projects / thirdparty / freeswitch.git / commitdiff
