<?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"/>
+ <!--Most channels to allow at once -->
+
<param name="max-sessions" value="1000"/>
</settings>
</configuration>
+
<configuration name="modules.conf" description="Modules">
<modules>
- <!-- Loggers (I'd load these first) -->
- <load module="mod_console"/>
- <!-- <load module="mod_syslog"/> -->
-
- <!-- XML Interfaces -->
- <!-- <load module="mod_xml_rpc"/> -->
-
- <!-- Event Handlers -->
- <!-- <load module="mod_event_multicast"/> -->
- <!-- <load module="mod_event_test"/> -->
- <!-- <load module="mod_zeroconf"/> -->
- <!-- <load module="mod_xmpp_event"/> -->
- <!-- <load module="mod_event_socket"/> -->
- <!-- <load module="mod_cdr"/> -->
-
- <!-- Directory Interfaces -->
- <!-- <load module="mod_ldap"/> -->
-
- <!-- Endpoints -->
- <load module="mod_sofia"/>
- <!--<load module="mod_iax"/>-->
- <load module="mod_portaudio"/>
- <!-- <load module="mod_woomera"/> -->
- <!-- <load module="mod_wanpipe"/> -->
- <!-- <load module="mod_dingaling"/> -->
-
- <!-- Applications -->
- <load module="mod_bridgecall"/>
- <load module="mod_echo"/>
- <load module="mod_dptools"/>
- <!-- <load module="mod_ivrtest"/> -->
- <load module="mod_playback"/>
- <load module="mod_commands"/>
- <!-- <load module="mod_commands"/> -->
-
- <!-- Dialplan Interfaces -->
- <load module="mod_dialplan_xml"/>
- <!-- <load module="mod_dialplan_directory"/> -->
-
- <!-- Codec Interfaces -->
- <load module="mod_g711"/>
- <load module="mod_gsm"/>
- <load module="mod_l16"/>
- <!-- <load module="mod_speex"/> -->
- <!-- <load module="mod_ilbc"/> -->
-
- <!-- File Format Interfaces -->
- <load module="mod_sndfile"/>
-
- <!-- Timers -->
- <load module="mod_softtimer"/>
-
- <!-- Languages -->
- <!-- <load module="mod_spidermonkey"/> -->
- <!-- <load module="mod_perl"/> -->
-
- <!-- ASR /TTS -->
- <!-- <load module="mod_cepstral"/> -->
- <!-- <load module="mod_rss"/> -->
-
- <!-- Conference Bridges -->
- <!--<load module="mod_conference"/>-->
+ <!-- Loggers (I'd load these first) -->
+ <load module="mod_console"/>
+ <!-- <load module="mod_syslog"/> -->
+
+ <!-- XML Interfaces -->
+ <!-- <load module="mod_xml_rpc"/> -->
+
+ <!-- Event Handlers -->
+ <!-- <load module="mod_cdr"/> -->
+ <!-- <load module="mod_event_multicast"/> -->
+ <!-- <load module="mod_event_socket"/> -->
+ <!-- <load module="mod_xmpp_event"/> -->
+ <!-- <load module="mod_zeroconf"/> -->
+
+ <!-- Directory Interfaces -->
+ <!-- <load module="mod_ldap"/> -->
+
+ <!-- Endpoints -->
+ <!-- <load module="mod_dingaling"/> -->
+ <!--<load module="mod_iax"/>-->
+ <load module="mod_portaudio"/>
+ <load module="mod_sofia"/>
+ <!-- <load module="mod_wanpipe"/> -->
+ <!-- <load module="mod_woomera"/> -->
+
+ <!-- Applications -->
+ <load module="mod_bridgecall"/>
+ <load module="mod_commands"/>
+ <!--<load module="mod_conference"/>-->
+ <load module="mod_dptools"/>
+ <load module="mod_echo"/>
+ <!--<load module="mod_park"/>-->
+ <load module="mod_playback"/>
+
+ <!-- Dialplan Interfaces -->
+ <!-- <load module="mod_dialplan_directory"/> -->
+ <load module="mod_dialplan_xml"/>
+
+ <!-- Codec Interfaces -->
+ <load module="mod_g711"/>
+ <load module="mod_gsm"/>
+ <!-- <load module="mod_ilbc"/> -->
+ <load module="mod_l16"/>
+ <!-- <load module="mod_speex"/> -->
+
+ <!-- File Format Interfaces -->
+ <load module="mod_sndfile"/>
+ <load module="mod_native_file"/>
+
+ <!-- Timers -->
+ <load module="mod_softtimer"/>
+
+ <!-- Languages -->
+ <!-- <load module="mod_spidermonkey"/> -->
+ <!-- <load module="mod_perl"/> -->
+
+ <!-- ASR /TTS -->
+ <!-- <load module="mod_cepstral"/> -->
+ <!-- <load module="mod_rss"/> -->
</modules>
</configuration>
<configuration name="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"/>
+
<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"/>
+
<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"/>
+
<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"/>
+
<!-- <param name="log_event" value="DEBUG"/> -->
+
<param name="all" value="DEBUG"/>
</mappings>
</configuration>
+
<configuration name="sofia.conf" description="sofia Endpoint">
<profiles>
- <profile name="test">
- <registrations>
- <registration name="asterlink">
- <param name="register-scheme" value="Digest"/>
- <param name="register-realm" value=""/>
- <param name="register-username" value="1001"/>
- <param name="register-password" value="nhy65tgb"/>
- <param name="register-from" value="sip:1001@208.64.200.40"/>
- <param name="register-to" value="sip:1001@66.250.68.194"/>
- <param name="register-proxy" value="sip:66.250.68.194:5060"/>
- <param name="register-frequency" value="20"/>
- </registration>
- </registrations>
- <settings>
- <param name="debug" value="1"/>
- <param name="rfc2833-pt" value="101"/>
- <param name="sip-port" value="5060"/>
- <param name="dialplan" value="XML"/>
- <param name="dtmf-duration" value="100"/>
- <param name="codec-prefs" value="PCMU@20i"/>
- <param name="codec-ms" value="20"/>
- <param name="use-rtp-timer" value="true"/>
- <param name="rtp-timer-name" value="soft"/>
- <param name="rtp-ip" value="192.168.1.20"/>
- <param name="sip-ip" value="192.168.1.20"/>
-
- <!-- 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>
+ <registrations>
+ <registration name="asterlink">
+
<param name="register-scheme" value="Digest"/>
+
<param name="register-realm" value=""/>
+
<param name="register-username" value="1001"/>
+
<param name="register-password" value="nhy65tgb"/>
+
<param name="register-from" value="sip:1001@208.64.200.40"/>
+
<param name="register-to" value="sip:1001@66.250.68.194"/>
+
<param name="register-proxy" value="sip:66.250.68.194:5060"/>
+
<param name="register-frequency" value="20"/>
+ </registration>
+ </registrations>
+ <settings>
+
<param name="debug" value="1"/>
+
<param name="rfc2833-pt" value="101"/>
+
<param name="sip-port" value="5060"/>
+
<param name="dialplan" value="XML"/>
+
<param name="dtmf-duration" value="100"/>
+
<param name="codec-prefs" value="PCMU@20i"/>
+
<param name="codec-ms" value="20"/>
+
<param name="use-rtp-timer" value="true"/>
+
<param name="rtp-timer-name" value="soft"/>
+
<param name="rtp-ip" value="192.168.1.20"/>
+
<param name="sip-ip" value="192.168.1.20"/>
+
+ <!-- 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 -->
<!-- info - informational message -->
<!-- debug - debug-level message -->
<settings>
- <param name="ident" value="freeswitch"/>
- <param name="facility" value="user"/>
- <param name="format" value="${time} - ${message}"/>
- <param name="level" value="debug,info,warning-alert"/>
+
<param name="ident" value="freeswitch"/>
+
<param name="facility" value="user"/>
+
<param name="format" value="${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"/>
+
<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"/>
+
<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"/>
+
<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"/>
+
<param name="span" value="1"/>
+
<param name="node" value="cpe"/>
+
<!-- <param name="switch" value="ni2"/> -->
+
<param name="switch" value="dms100"/>
+
<!-- <param name="switch" value="lucent5e"/> -->
+
<!-- <param name="switch" value="att4ess"/> -->
+
<!-- <param name="switch" value="euroisdn"/> -->
+
<!-- <param name="switch" value="gr303eoc"/> -->
+
<!-- <param name="switch" value="gr303tmc"/> -->
+
<param name="dp" value="national"/>
+
<!-- <param name="dp" value="international"/> -->
+
<!-- <param name="dp" value="local"/> -->
+
<!-- <param name="dp" value="private"/> -->
+
<!-- <param name="dp" value="unknown"/> -->
+
<param name="l1" value="ulaw"/>
+
<!-- <param name="l1" value="alaw"/> -->
+
<param name="bchan" value="1-23"/>
+
<param name="dchan" value="24"/>
+
<param name="dialplan" value="XML"/>
</span>
</configuration>
-
+
<configuration name="portaudio.conf" description="Soundcard Endpoint">
<settings>
- <param name="debug" value="2"/>
- <param name="dialplan" value="XML"/>
-
- <!-- partial string match on something in the name or the device # -->
- <param name="indev" value="USB"/>
- <param name="outdev" value="USB"/>
-
- <param name="cid-name" value="FreeSwitch"/>
- <param name="cid-num" value="5555551212"/>
+
<param name="debug" value="2"/>
+
<param name="dialplan" value="XML"/>
+
+ <!-- partial string match on something in the name or the device # -->
+
<param name="indev" value="USB"/>
+
<param name="outdev" value="USB"/>
+
+
<param name="cid-name" value="FreeSwitch"/>
+
<param name="cid-num" value="5555551212"/>
</settings>
</configuration>
-
+
<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
- <param name="publish" value="yes"/>
- <param name="browse" value="_sip._udp"/>
+
<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"/>
+
<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"/>
+
<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"/>
+
<param name="debug" value="0"/>
+
<param name="codec-prefs" value="PCMU"/>
</settings>
<!-- *NOTE* your resource (after the /) MUST contain the string "talk" (upper or lower case is ok) -->
<!-- *NOTE* as of May 2 2006 you must set"auto-login" to"true" if you want to be able to auto-login on startup"/> -->
<interface>
- <param name="name" value="jingle"/>
- <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"/>
- <!-- 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"/>
+ <param name="name" value="jingle"/>
+ <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"/>
+ <!-- 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"/>
</interface>
</configuration>
+
<configuration name="xml_rpc.conf" description="XML RPC">
<settings>
- <!-- The port where you want to run the http service (default 8080) -->
- <param name="http-port" value="8080"/>
- <!-- if all 3 of the following params exist all http traffic will require auth -->
- <param name="auth-realm" value="freeswitch"/>
- <param name="auth-user" value="freeswitch"/>
- <param name="auth-pass" value="works"/>
- <!-- The url to a gateway cgi that can generate xml similar to
- what's in this file only on-the-fly (leave it commented if you dont
- need it) -->
- <!-- one or more |-delim of configuration|directory|dialplan -->
- <!-- <param name="gateway-url" value="http://www.server.com/gateway.cgi" bindings="configuration"/> -->
+ <!-- The port where you want to run the http service (default 8080) -->
+ <param name="http-port" value="8080"/>
+ <!-- if all 3 of the following params exist all http traffic will require auth -->
+ <param name="auth-realm" value="freeswitch"/>
+ <param name="auth-user" value="freeswitch"/>
+ <param name="auth-pass" value="works"/>
+ <!-- The url to a gateway cgi that can generate xml similar to what's in -->
+ <!-- this file only on-the-fly (leave it commented if you dont need it)-->
+ <!-- one or more |-delim of configuration|directory|dialplan -->
+ <!-- <param name="gateway-url" value="http://www.server.com/gateway.cgi" bindings="configuration"/> -->
</settings>
</configuration>
<configuration name="rss.conf" description="RSS Parser">
<feeds>
- <!-- Just download the files to wherever and refer to them here -->
- <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
- <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
+ <!-- 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 -->
+ <!-- None of these paths are real if you want any of these options you need to really set them up -->
<configuration name="conference.conf" description="Audio Conference">
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
- <profile name="default">
- <!-- Sample Rate-->
- <param name="rate" value="8000"/>
- <!-- Number of milliseconds per frame -->
- <param name="interval" value="20"/>
- <!-- Energy level required for audio to be sent to the other users -->
- <param name="energy-level" value="300"/>
- <!-- TTS Engine to use -->
- <!--<param name="tts-engine" value="cepstral"/>-->
- <!-- TTS Voice to use -->
- <!--<param name="tts-voice" value="david"/>-->
-
- <!-- If TTS is enabled all audio-file params not beginning with '/'
- will be considered text to say with TTS -->
-
- <!-- File to play to acknowledge succees -->
- <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
- <!-- File to play to acknowledge failure -->
- <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
- <!-- File to play to acknowledge muted -->
- <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
- <!-- File to play to acknowledge unmuted -->
- <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
- <!-- File to play if you are alone in the conference -->
- <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
- <!-- File to play when you join the conference -->
- <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
- <!-- File to play when you leave the conference -->
- <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
- <!-- File to play when you ae ejected from the conference -->
- <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
- <!-- File to play when the conference is locked -->
- <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
- <!-- File to play to prompt for a pin -->
- <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
- <!-- File to play to when the pin is invalid -->
- <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
- <!-- Conference pin -->
- <!--<param name="pin" value="12345"/>-->
- <!-- Default Caller ID Name for outbound calls -->
- <param name="caller-id-name" value="FreeSWITCH"/>
- <!-- Default Caller ID Number for outbound calls -->
- <param name="caller-id-number" value="8777423583"/>
- </profile>
+
<profile name="default">
+ <!-- Sample Rate-->
+
<param name="rate" value="8000"/>
+ <!-- Number of milliseconds per frame -->
+
<param name="interval" value="20"/>
+ <!-- Energy level required for audio to be sent to the other users -->
+
<param name="energy-level" value="300"/>
+ <!-- TTS Engine to use -->
+
<!--<param name="tts-engine" value="cepstral"/>-->
+ <!-- TTS Voice to use -->
+
<!--<param name="tts-voice" value="david"/>-->
+
+ <!-- If TTS is enabled all audio-file params not beginning with -->
+ <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
+
+ <!-- File to play to acknowledge succees -->
+
<!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
+ <!-- File to play to acknowledge failure -->
+
<!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
+ <!-- File to play to acknowledge muted -->
+
<!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
+ <!-- File to play to acknowledge unmuted -->
+
<!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
+ <!-- File to play if you are alone in the conference -->
+
<!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
+ <!-- File to play when you join the conference -->
+
<!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
+ <!-- File to play when you leave the conference -->
+
<!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
+ <!-- File to play when you ae ejected from the conference -->
+
<!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
+ <!-- File to play when the conference is locked -->
+
<!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
+ <!-- File to play to prompt for a pin -->
+
<!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
+ <!-- File to play to when the pin is invalid -->
+
<!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
+ <!-- Conference pin -->
+
<!--<param name="pin" value="12345"/>-->
+ <!-- Default Caller ID Name for outbound calls -->
+
<param name="caller-id-name" value="FreeSWITCH"/>
+ <!-- Default Caller ID Number for outbound calls -->
+
<param name="caller-id-number" value="8777423583"/>
+ </profile>
</profiles>
</configuration>
</section>
+
<section name="dialplan" description="Regex/XML Dialplan">
- <!-- Valid fields in conditions:
- "dialplan, caller_id_name, ani, ani2, caller_id_number,
- network_addr, rdnis, destination_number, uuid, source,
- context, chan_name" -->
+ <!-- Valid fields in conditions: -->
+ <!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
+ <!-- rdnis, destination_number, uuid, source, context, chan_name" -->
<!-- *NOTE* The special context name 'any' will match any context -->
<context name="default">
<extension name="tollfree">
- <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
- <action application="bridge" data="exosip/$1-freeswitch@voip.trxtel.com"/>
- </condition>
+ <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
+ <action application="bridge" data="sofia/test/$1-freeswitch@voip.trxtel.com"/>
+ </condition>
</extension>
- <!--<extension name="devconf">
- <condition field="destination_number" expression="^888$">
- <action application="bridge" data="exosip/888@66.250.68.194"/>
- </condition>
- </extension> -->
+ <!-- Call the FreeSWITCH conference via SIP -->
+ <!--<extension name="FreeSWITCH Conference SIP">-->
+ <!--<condition field="destination_number" expression="^888$">-->
+ <!--<action application="bridge" data="sofia/test/888@66.250.68.194"/>-->
+ <!--</condition>-->
+ <!--</extension> -->
+
+ <!-- Call the FreeSWITCH conference via IAX -->
+ <!--<extension name="FreeSWITCH Conference IAX">-->
+ <!--<condition field="destination_number" expression="^8888$">-->
+ <!--<action application="bridge" data="iax/guest@66.250.68.194/888"/>-->
+ <!--</condition>-->
+ <!--</extension>-->
<extension name="testmusic">
- <condition field="destination_number" expression="^1234$">
- <action application="bridge" data="exosip/1234@66.250.68.194"/>
- </condition>
+ <condition field="destination_number" expression="^1234$">
+ <action application="bridge" data="sofia/test/1234@66.250.68.194"/>
+ </condition>
</extension>
<!-- Enter an existing conference -->
<extension name="1000">
- <condition field="destination_number" expression="^1000$">
- <action application="conference" data="freeswitch"/>
- </condition>
+ <condition field="destination_number" expression="^1000$">
+
<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$">
- <action application="conference" data="bridge:mydynaconf:exosip/1234@66.250.68.194"/>
- </condition>
+ <condition field="destination_number" expression="^2000$">
+ <action application="conference" data="bridge:mydynaconf:sofia/test/1234@66.250.68.194"/>
+ </condition>
</extension>
- <!-- if the destination is an exact match on the extension name
- you do not need any regex in the condition
- <extension name="999">
- <condition><action application="bridge" data="exosip/888@66.250.68.194"/></condition>
- </extension>-->
- <!-- extensions starting with 4, all the numbers after 4 form a numeric filename
- continue=true means keep looking for more extensions to match
- *NOTE* The entire dialplan is parsed ONCE when the call starts
- so any call info acquired after the various actions cannot
- be taken into consideration.
-
-The first match will play a beep and the second one plays
-the desired file. This is for demo purposes both actions
-could have been under the same <extension> tag as well.
- -->
+ <!-- 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>
+ <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>
+ <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>
+ <condition field="rdnis" expression="8881231234"/>
+ <condition field="destination_number" expression="(.*)">
+
<action application="bridge" data="wanpipe/a/a/$1"/>
+ </condition>
</extension>
+
<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
<extension name="9999">
- <condition field="source" expression="mod_iax"/>
- <condition field="destination_number" expression="9999">
- <action application="playback" data="/var/sounds/beep.gsm"/>
- </condition>
- </extension>
- <!-- Call the FreeSWITCH conference via SIP -->
- <extension name="FreeSWITCH Conference SIP">
- <condition field="destination_number" expression="^888$">
- <action application="bridge" data="exosip/888@66.250.68.194"/>
- </condition>
- </extension>
- <!-- Call the FreeSWITCH conference via IAX -->
- <extension name="FreeSWITCH Conference IAX">
- <condition field="destination_number" expression="^8888$">
- <action application="bridge" data="iax/guest@66.250.68.194/888"/>
- </condition>
+ <condition field="source" expression="mod_iax"/>
+ <condition field="destination_number" expression="9999">
+ <action application="playback" data="/var/sounds/beep.gsm"/>
+ </condition>
</extension>
+
</context>
</section>
<domain name="mydomain.com">
<!--the user id (the left hand side of the @ in the addr-->
<user id="1000">
- <!-- omit password for authless registration -->
- <param name="password" value="mypass"/>
- <!--various endpoints and application will look for user specific settings here -->
- <param name="mypref" value="myval"/>
+ <!-- omit password for authless registration -->
+
<param name="password" value="mypass"/>
+ <!--various endpoints and application will look for user specific settings here -->
+
<param name="mypref" value="myval"/>
</user>
</domain>
</section>
projects / thirdparty / freeswitch.git / commitdiff
