reformat sample conf, remove exosip references
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2906 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Michael Jerris
parent
0fcde9649c
commit
efc894b6dd
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@ -1,12 +1,15 @@
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<?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="switch.conf" description="Modules">
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<settings>
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<!--Most channels to allow at once -->
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<param name="max-sessions" value="1000"/>
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</settings>
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</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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@ -17,46 +20,46 @@
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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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<!-- <load module="mod_event_socket"/> -->
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<!-- <load module="mod_cdr"/> -->
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<!-- <load module="mod_event_multicast"/> -->
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<!-- <load module="mod_event_socket"/> -->
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<!-- <load module="mod_xmpp_event"/> -->
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<!-- <load module="mod_zeroconf"/> -->
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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_sofia"/>
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<!-- <load module="mod_dingaling"/> -->
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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_sofia"/>
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<!-- <load module="mod_wanpipe"/> -->
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<!-- <load module="mod_dingaling"/> -->
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<!-- <load module="mod_woomera"/> -->
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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_dptools"/>
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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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<!--<load module="mod_conference"/>-->
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<load module="mod_dptools"/>
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<load module="mod_echo"/>
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<!--<load module="mod_park"/>-->
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<load module="mod_playback"/>
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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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<load module="mod_dialplan_xml"/>
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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_ilbc"/> -->
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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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<load module="mod_native_file"/>
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<!-- Timers -->
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<load module="mod_softtimer"/>
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@ -69,9 +72,6 @@
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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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@ -111,6 +111,7 @@
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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="sofia.conf" description="sofia Endpoint">
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<profiles>
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<profile name="test">
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@ -159,6 +160,7 @@
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</profile>
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</profiles>
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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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@ -278,8 +280,7 @@
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<param name="auto-login" value="true"/>
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<!-- SASL "plain" or "md5" -->
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<param name="sasl" value="plain"/>
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<!-- if the server where the jabber is hosted is not the same
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as the one in the jid -->
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<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
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<!--<param name="server" value="alternate.server.com"/>-->
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<!-- Enable TLS or not -->
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<param name="tls" value="true"/>
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@ -296,6 +297,7 @@
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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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@ -304,9 +306,8 @@
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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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<!-- The url to a gateway cgi that can generate xml similar to what's in -->
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<!-- this file only on-the-fly (leave it commented if you dont 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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@ -320,8 +321,7 @@
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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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<!-- None of these paths are real if you want any of these options 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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@ -338,8 +338,8 @@
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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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<!-- If TTS is enabled all audio-file params not beginning with -->
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<!-- '/' or with drive: (i.e. c:) 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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@ -373,29 +373,37 @@
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</profiles>
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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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<!-- Valid fields in conditions: -->
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<!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
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<!-- rdnis, destination_number, uuid, source, 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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<action application="bridge" data="sofia/test/$1-freeswitch@voip.trxtel.com"/>
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</condition>
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</extension>
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<!--<extension name="devconf">
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<condition field="destination_number" expression="^888$">
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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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<!-- 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$">-->
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<!--<action application="bridge" data="sofia/test/888@66.250.68.194"/>-->
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<!--</condition>-->
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<!--</extension> -->
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<!-- Call the FreeSWITCH conference via IAX -->
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<!--<extension name="FreeSWITCH Conference IAX">-->
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<!--<condition field="destination_number" expression="^8888$">-->
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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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<extension name="testmusic">
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<condition field="destination_number" expression="^1234$">
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<action application="bridge" data="exosip/1234@66.250.68.194"/>
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<action application="bridge" data="sofia/test/1234@66.250.68.194"/>
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</condition>
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</extension>
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@ -409,37 +417,33 @@
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<!-- Start a dynamic conference and call someone at the same time -->
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<extension name="2000">
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<condition field="destination_number" expression="^2000$">
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<action application="conference" data="bridge:mydynaconf:exosip/1234@66.250.68.194"/>
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<action application="conference" data="bridge:mydynaconf:sofia/test/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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<!-- 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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<!-- 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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<extension name="playsound1" continue="true">
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<condition field="source" expression="mod_sofia"/>
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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_sofia"/>
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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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@ -447,6 +451,7 @@ could have been under the same <extension> tag as well.
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<action application="bridge" data="wanpipe/a/a/$1"/>
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</condition>
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</extension>
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<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
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<extension name="9999">
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<condition field="source" expression="mod_iax"/>
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@ -454,18 +459,7 @@ could have been under the same <extension> tag as well.
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<action application="playback" data="/var/sounds/beep.gsm"/>
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</condition>
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</extension>
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<!-- Call the FreeSWITCH conference via SIP -->
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<extension name="FreeSWITCH Conference SIP">
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<condition field="destination_number" expression="^888$">
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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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