Good evening, I’m working with a heavily edited version of Freeswitch and I’m trying to get a multicast paging group added to a conference such that any audio in the conference is broadcast to the paging group as well.  I’ve successfully gotten a multicast paging group working in the dialplan as follows:

    <extension name="rtp_multicast_page">

       <condition field="destination_number" expression="^3456$">

          <action application="set" data="absolute_codec_string=PCMU@20i"/>

          <action application="set" data="jitterbuffer_msec=500:1000:20"/>

          <action application="answer"/>

          <action application="esf_page_group" data="234.5.6.7 23458 30"/>

       </condition>

    </extension>

            When I use X-Lite registered to FS and call “3456”, I get the spoken audio out the paging group correctly.  Now I’m trying to link the paging group to a conference.  I’ve tried experimenting with adding it to an existing conference by using “conference_set_auto_outcall”, but that’s failing:

<action application="conference_set_auto_outcall" data="3456@$${domain}"/>

            And bridge attempts aren’t working as well:

      <action application="bridge" data="sofia/internal/3456@$${domain}"/>

            I’m new to FS and have been scouring the Wiki and mailing lists for clues on how to accomplish this.  Is this something that is not possible or are I missing a configuration step?  Any clues would be appreciated, thanks…