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add a new software conference module (mod_conference)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@1770 d0543943-73ff-0310-b7d9-9358b9ac24b2
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Anthony Minessale committed
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@@ -7,7 +7,7 @@
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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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<!-- XML Interfaces -->
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<!-- <load module="mod_xml_rpc"/> -->
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<!-- Event Handlers -->
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@@ -59,6 +59,10 @@
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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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@@ -107,12 +111,12 @@
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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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<!-- 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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<!-- 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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@@ -228,8 +232,8 @@
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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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<!-- 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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@@ -243,16 +247,16 @@
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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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<!-- 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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<!-- 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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@@ -265,6 +269,48 @@
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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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<!-- 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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</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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@@ -274,17 +320,31 @@
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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="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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<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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@@ -294,30 +354,30 @@
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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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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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-->
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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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</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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</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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<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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