424 lines
18 KiB
XML
424 lines
18 KiB
XML
<?xml version="1.0"?>
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<document type="freeswitch/xml">
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<section name="configuration" description="Various Configuration">
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<configuration name="modules.conf" description="Modules">
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<modules>
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<!-- Loggers (I'd load these first) -->
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<load module="mod_console"/>
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<!-- <load module="mod_syslog"/> -->
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<!-- XML Interfaces -->
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<!-- <load module="mod_xml_rpc"/> -->
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<!-- Event Handlers -->
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<!-- <load module="mod_event_multicast"/> -->
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<!-- <load module="mod_event_test"/> -->
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<!-- <load module="mod_zeroconf"/> -->
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<!-- <load module="mod_xmpp_event"/> -->
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<!-- Directory Interfaces -->
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<!-- <load module="mod_ldap"/> -->
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<!-- Endpoints -->
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<load module="mod_exosip"/>
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<!--<load module="mod_iax"/>-->
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<!-- <load module="mod_portaudio"/> -->
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<!-- <load module="mod_woomera"/> -->
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<!-- <load module="mod_wanpipe"/> -->
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<!-- <load module="mod_dingaling"/> -->
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<!-- Applications -->
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<load module="mod_bridgecall"/>
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<load module="mod_echo"/>
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<!-- <load module="mod_ivrtest"/> -->
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<load module="mod_playback"/>
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<load module="mod_commands"/>
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<!-- <load module="mod_commands"/> -->
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<!-- Dialplan Interfaces -->
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<load module="mod_dialplan_xml"/>
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<!-- <load module="mod_dialplan_directory"/> -->
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<!-- Codec Interfaces -->
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<load module="mod_g711"/>
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<load module="mod_gsm"/>
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<load module="mod_l16"/>
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<!-- <load module="mod_speex"/> -->
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<!-- <load module="mod_ilbc"/> -->
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<!-- File Format Interfaces -->
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<load module="mod_sndfile"/>
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<!-- Timers -->
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<load module="mod_softtimer"/>
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<!-- Languages -->
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<!-- <load module="mod_spidermonkey"/> -->
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<!-- <load module="mod_perl"/> -->
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<!-- ASR /TTS -->
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<!-- <load module="mod_cepstral"/> -->
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<!-- <load module="mod_rss"/> -->
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<!-- Conference Bridges -->
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<!--<load module="mod_conference"/>-->
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</modules>
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</configuration>
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<configuration name="iax.conf" description="IAX Configuration">
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<settings>
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<param name="debug" value="0"/>
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<!-- <param name="ip" value="1.2.3.4"> -->
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<param name="port" value="4569"/>
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<param name="dialplan" value="XML"/>
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<param name="codec-prefs" value="PCMU,PCMA,speex,L16"/>
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<param name="codec-master" value="us"/>
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<param name="codec-rates" value="8"/>
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</settings>
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</configuration>
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<configuration name="console.conf" description="Console Logger">
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<!-- pick a file name, a function name or 'all' -->
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<!-- map as many as you need for specific debugging -->
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<mappings>
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<!-- <param name="log_event" value="DEBUG"/> -->
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<param name="all" value="DEBUG"/>
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</mappings>
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</configuration>
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<configuration name="syslog.conf" description="Syslog Logger">
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<!-- SYSLOG -->
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<!-- emerg - system is unusable -->
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<!-- alert - action must be taken immediately -->
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<!-- crit - critical conditions -->
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<!-- err - error conditions -->
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<!-- warning - warning conditions -->
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<!-- notice - normal, but significant, condition -->
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<!-- info - informational message -->
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<!-- debug - debug-level message -->
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<settings>
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<param name="ident" value="freeswitch"/>
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<param name="facility" value="user"/>
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<param name="format" value="${time} - ${message}"/>
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<param name="level" value="debug,info,warning-alert"/>
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</settings>
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</configuration>
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<configuration name="exosip.conf" description="Exosip Endpoint">
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<settings>
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<param name="port" value="5060"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="PCMU,PCMA"/>
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<!-- Payload number to bind DTMF to-->
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<param name="rfc2833-pt" value="101"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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<!-- auto sense NAT issues and adjust accordingly -->
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<param name="use-rtp-auto-adjust" value="true"/>
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<!-- pick one (default if not specified is 'guess'); -->
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<param name="rtp-ip" value="guess"/>
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<!-- <param name-"rtp-ip" value="10.0.0.1"/> -->
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<!-- leave commented or 0.0.0.0 for all ip -->
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<!-- <param name="sip-ip" value="127.0.0.1"/> -->
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<!-- optional ; -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
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<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
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<!-- specify 'myrealm' with certian key -->
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<!-- use !myrealm! at beginning of url to activate -->
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<!-- exosip/!myrealm!1000@dest -->
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<!-- srtp:<param name="myrealm" value="ffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff"/> -->
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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</settings>
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</configuration>
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<configuration name="woomera.conf" description="Woomera Endpoint">
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<settings>
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<param name="debug" value="0"/>
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</settings>
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</configuration>
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<configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
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<settings>
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<param name="debug" value="1"/>
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<param name="dialplan" value="XML"/>
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<param name="mtu" value="320"/>
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<param name="dtmf-on" value="800"/>
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<param name="dtmf-off" value="100"/>
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<param name="supress-dtmf-tone" value="yes"/>
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</settings>
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<span>
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<param name="span" value="1"/>
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<param name="node" value="cpe"/>
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<!-- <param name="switch" value="ni2"/> -->
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<param name="switch" value="dms100"/>
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<!-- <param name="switch" value="lucent5e"/> -->
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<!-- <param name="switch" value="att4ess"/> -->
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<!-- <param name="switch" value="euroisdn"/> -->
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<!-- <param name="switch" value="gr303eoc"/> -->
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<!-- <param name="switch" value="gr303tmc"/> -->
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<param name="dp" value="national"/>
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<!-- <param name="dp" value="international"/> -->
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<!-- <param name="dp" value="local"/> -->
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<!-- <param name="dp" value="private"/> -->
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<!-- <param name="dp" value="unknown"/> -->
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<param name="l1" value="ulaw"/>
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<!-- <param name="l1" value="alaw"/> -->
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<param name="bchan" value="1-23"/>
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<param name="dchan" value="24"/>
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<param name="dialplan" value="XML"/>
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</span>
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</configuration>
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<configuration name="portaudio.conf" description="Soundcard Endpoint">
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<settings>
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<param name="debug" value="2"/>
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<param name="dialplan" value="XML"/>
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<!-- partial string match on something in the name or the device # -->
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<param name="indev" value="USB"/>
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<param name="outdev" value="USB"/>
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<param name="cid-name" value="FreeSwitch"/>
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<param name="cid-num" value="5555551212"/>
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</settings>
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</configuration>
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<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
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<settings>
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<param name="publish" value="yes"/>
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<param name="browse" value="_sip._udp"/>
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</settings>
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</configuration>
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<configuration name="xmpp_event.conf" description="XMPP Event Handler">
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<settings>
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<param name="#debug" value="1"/>
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<param name="jid" value="freeswitch@my.jabber.com/me"/>
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<param name="passwd" value="mypass"/>
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<param name="target-jid" value="freeswitch@reader.org/him"/>
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</settings>
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</configuration>
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<configuration name="dialplan_directory.conf" description="Dialplan Directory">
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<settings>
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<param name="directory-name" value="ldap"/>
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<param name="host" value="ldap.freeswitch.org"/>
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<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
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<param name="pass" value="test"/>
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<param name="base" value="dc=freeswitch,dc=org"/>
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</settings>
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</configuration>
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<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
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<settings>
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<param name="debug" value="0"/>
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<param name="codec-prefs" value="PCMU"/>
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</settings>
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<!-- *NOTE* your resource (after the /) MUST contain the string "talk" (upper or lower case is ok) -->
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<!-- *NOTE* as of May 2 2006 you must set"auto-login" to"true" if you want to be able to auto-login on startup"/> -->
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<interface>
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<param name="name" value="jingle"/>
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<param name="login" value="myjid@myserver.com/talk"/>
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<param name="password" value="mypass"/>
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<param name="dialplan" value="XML"/>
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<param name="message" value="Jingle all the way"/>
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<param name="rtp-ip" value="10.0.0.1"/>
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<param name="auto-login" value="true"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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<!-- or -->
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<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
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<!-- default extension (if one cannot be determined) -->
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<param name="exten" value="888"/>
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<!-- VAD choose one -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<param name="vad" value="both"/>
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</interface>
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</configuration>
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<configuration name="xml_rpc.conf" description="XML RPC">
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<settings>
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<!-- The port where you want to run the http service (default 8080) -->
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<param name="http-port" value="8080"/>
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<!-- if all 3 of the following params exist all http traffic will require auth -->
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<param name="auth-realm" value="freeswitch"/>
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<param name="auth-user" value="freeswitch"/>
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<param name="auth-pass" value="works"/>
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<!-- The url to a gateway cgi that can generate xml similar to
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what's in this file only on-the-fly (leave it commented if you dont
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need it) -->
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<!-- one or more |-delim of configuration|directory|dialplan -->
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<!-- <param name="gateway-url" value="http://www.server.com/gateway.cgi" bindings="configuration"/> -->
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</settings>
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</configuration>
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<configuration name="rss.conf" description="RSS Parser">
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<feeds>
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<!-- Just download the files to wherever and refer to them here -->
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<!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
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<!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
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</feeds>
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</configuration>
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<!-- None of these paths are real if you want any of these options
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you need to really set them up -->
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<configuration name="conference.conf" description="Audio Conference">
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<!-- Profiles are collections of settings you can reference by name. -->
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<profiles>
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<profile name="default">
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<!-- Sample Rate-->
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<param name="rate" value="8000"/>
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<!-- Number of milliseconds per frame -->
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<param name="interval" value="20"/>
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<!-- Energy level required for audio to be sent to the other users -->
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<param name="energy_level" value="300"/>
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<!-- TTS Engine to use -->
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<!--<param name="tts_engine" value="cepstral"/>-->
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<!-- TTS Voice to use -->
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<!--<param name="tts_voice" value="david"/>-->
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<!-- If TTS is enabled all audio-file params not beginning with '/'
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will be considered text to say with TTS -->
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<!-- File to play to acknowledge succees -->
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<!--<param name="ack_sound" value="/soundfiles/beep.wav"/>-->
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<!-- File to play to acknowledge failure -->
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<!--<param name="nack_sound" value="/soundfiles/beeperr.wav"/>-->
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<!-- File to play to acknowledge muted -->
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<!--<param name="muted_sound" value="/soundfiles/muted.wav"/>-->
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<!-- File to play to acknowledge unmuted -->
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<!--<param name="unmuted_sound" value="/soundfiles/unmuted.wav"/>-->
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<!-- File to play if you are alone in the conference -->
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<!--<param name="alone_sound" value="/soundfiles/yactopitc.wav"/>-->
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<!-- File to play when you join the conference -->
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<!--<param name="enter_sound" value="/soundfiles/welcome.wav"/>-->
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<!-- File to play when you leave the conference -->
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<!--<param name="exit_sound" value="/soundfiles/exit.wav"/>-->
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<!-- File to play when you ae ejected from the conference -->
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<!--<param name="kicked_sound" value="/soundfiles/kicked.wav"/>-->
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<!-- File to play when the conference is locked -->
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<!--<param name="locked_sound" value="/soundfiles/locked.wav"/>-->
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<!-- File to play to prompt for a pin -->
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<!--<param name="pin_sound" value="/soundfiles/pin.wav"/>-->
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<!-- File to play to when the pin is invalid -->
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<!--<param name="bad_pin_sound" value="/soundfiles/invalid-pin.wav"/>-->
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<!-- Conference pin -->
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<!--<param name="pin" value="12345"/>-->
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<!-- Default Caller ID Name for outbound calls -->
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<param name="caller_id_name" value="FreeSWITCH"/>
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<!-- Default Caller ID Number for outbound calls -->
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<param name="caller_id_number" value="8777423583"/>
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</profile>
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</profiles>
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<rooms>
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<room name="freeswitch" profile="default"/>
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</rooms>
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</configuration>
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</section>
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<section name="dialplan" description="Regex/XML Dialplan">
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<!-- Valid fields in conditions:
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"dialplan, caller_id_name, ani, ani2, caller_id_number,
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network_addr, rdnis, destination_number, uuid, source,
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context, chan_name" -->
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<!-- *NOTE* The special context name 'any' will match any context -->
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<context name="default">
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<extension name="tollfree">
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<condition field="destination_number" expression="^(18[0{2}8{2}7{2}6{2}]\d{7})">
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<action application="bridge" data="exosip/$1-freeswitch@voip.trxtel.com"/>
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</condition>
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</extension>
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<extension name="devconf">
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<condition field="destination_number" expression="^888$">
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<action application="bridge" data="exosip/888@66.250.68.194"/>
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</condition>
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</extension>
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<!-- Enter an existing conference -->
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<extension name="1000">
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<condition field="destination_number" expression="^1000$">
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<action application="conference" data="1000"/>
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</condition>
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</extension>
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<!-- Start a dynamic conference and call someone at the same time -->
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<extension name="2000">
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<condition field="destination_number" expression="^2000$">
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<action application="conference" data="bridge:mydynaconf:exosip/1234@66.250.68.194"/>
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</condition>
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</extension>
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<!-- if the destination is an exact match on the extension name
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you do not need any regex in the condition -->
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<extension name="999">
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<condition><action application="bridge" data="exosip/888@66.250.68.194"/></condition>
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</extension>
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<!-- extensions starting with 4, all the numbers after 4 form a numeric filename
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continue=true means keep looking for more extensions to match
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*NOTE* The entire dialplan is parsed ONCE when the call starts
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so any call info acquired after the various actions cannot
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be taken into consideration.
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The first match will play a beep and the second one plays
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the desired file. This is for demo purposes both actions
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could have been under the same <extension> tag as well.
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-->
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<extension name="playsound1" continue="true">
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<condition field="source" expression="mod_exosip"/>
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<condition field="destination_number" expression="^4(\d+)">
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<action application="playback" data="/var/sounds/beep.gsm"/>
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</condition>
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</extension>
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<extension name="playsound2">
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<condition field="source" expression="mod_exosip"/>
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<condition field="destination_number" expression="^4(\d+)">
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<action application="playback" data="/root/$1.raw"/>
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</condition>
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</extension>
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<!-- send everything with a certian RDNIS to Wanpipe ISDN -->
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<extension name="To PRI">
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<condition field="rdnis" expression="8881231234"/>
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<condition field="destination_number" expression="(.*)">
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<action application="bridge" data="wanpipe/a/a/$1"/>
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</condition>
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</extension>
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<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
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<extension name="9999">
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<condition field="source" expression="mod_iax"/>
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<condition field="destination_number" expression="9999">
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<action application="playback" data="/var/sounds/beep.gsm"/>
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</condition>
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</extension>
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<!-- Call the FreeSWITCH conference via SIP -->
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<extension name="FreeSWITCH Conference SIP">
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<condition field="destination_number" expression="^888$">
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<action application="bridge" data="exosip/888@66.250.68.194"/>
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</condition>
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</extension>
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<!-- Call the FreeSWITCH conference via IAX -->
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<extension name="FreeSWITCH Conference IAX">
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<condition field="destination_number" expression="^8888$">
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<action application="bridge" data="iax/guest@66.250.68.194/888"/>
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</condition>
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</extension>
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</context>
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</section>
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<section name="directory" description="User Directory">
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</section>
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</document>
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