<!-- Call the FreeSWITCH conference via SIP -->
<!--<extension name="FreeSWITCH Conference SIP">-->
<!--<condition field="destination_number" expression="^888$">-->
- <!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
+ <!--<action application="bridge" data="sofia/$${domain}/888@conference.freeswitch.org"/>-->
<!--</condition>-->
<!--</extension> -->
<!-- Request a certain tone/file to be played while you wait for the call to be answered-->
<action application="set" data="ringback=${us-ring}"/>
<!--<action application="set" data="ringback=/home/ring.wav"/>-->
- <action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
+ <action application="bridge" data="sofia/$${domain}/1234@conference.freeswitch.org"/>
</condition>
</extension>
<!-- Start a dynamic conference and call someone at the same time -->
<extension name="2000">
<condition field="destination_number" expression="^2000$">
- <action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
+ <action application="conference" data="bridge:mydynaconf:sofia/$${domain}/1234@conference.freeswitch.org"/>
</condition>
</extension>
</settings>
<routes>
- <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
+ <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${domain}/$1"/>
<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
- <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
+ <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${domain}/$1"/>
</routes>
</configuration>
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
- <param name="dialplan" value="XML"/>
+ <param name="dialplan" value="enum,XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${default_codecs}"/>
<param name="codec-ms" value="20"/>