Zhou Yu - 2015-07-31

It is basically two steps, first is you want to end the call if the first person leaves the call,
The second step is you want to end the bridge to sip when the conference is over.

Now. after you hangup, you still need to wait 30 second to put in another call, due to network traffic 😊 You know haha. I also updated in sourceforge the solution.

So I changed
on tjr209
/usr/local/freeswitch/conf/dialplan/default.xml

<extension name="cdquality_conferences">
  <condition field="destination_number" expression="^(37\d{2})$">
    <action application="answer"/>
    <!-- auto dial extension when conference starts -->
    <action application="set" data="hangup_after_bridge=true"/>
    <!-- action application="set" data="conference_auto_outcall_caller_id_name=halef"/ -->
    <!-- action application="set" data="conference_auto_outcall_flags=vmute"/ -->
    <action application="conference_set_auto_outcall" data="sofia/gateway/it-tjr61.dhbw-stuttgart.de/7709@it-tjr61.dhbw-stuttgart.de:5060"/>
    <!-- action application="conference_set_auto_outcall" data="sofia/gateway/it-tjr61.dhbw-stuttgart.de/7709@it-tjr61.dhbw-stuttgart.de:5060"/-->
    <!--<action application="conference_set_auto_outcall" data="sofia/gateway/141.31.8.69/7709@141.31.8.69:5060"/>-->

    <!-- endconf flag is attempt to stop conference after caller hangs up, but Halef is not done with dialog-->
    <action application="conference" data="$1-${domain_name}@video-mcu-stereo+flags{endconf}"/>
    <!-- action application="conference" data="$1-${domain_name}@video-mcu-stereo"/> -->
  </condition>
</extension>
 

Last edit: Zhou Yu 2015-07-31