It looks to me like you are mixing up the ideas of user vs. extension in a dialplan. When you call bridge with data="sofia/" you are telling freeswitch to use sofia to set up the call. Sofia is trying to initiate a SIP call based on the other arguments, which will try to do a SIP invite to 3456@$${domain} - probably not what you want. Multicast paging is not actually SIP, nor is it even a phone call as far as set up, caller ID, etc. That's why it's done as an application, just like recording or any other audio stream output. You could try a loopback bridge - that might do what you want. This will originate a call back into the dialplan, using the dialplan parts you already have working. <action application="bridge" data="loopback/3456/YOUR_CONTEXT/XML"/> Drew Terenzini wrote:
Good evening, I’m working with a heavily edited version of Freeswitch and I’m trying to get a multicast paging group added to a conference such that any audio in the conference is broadcast to the paging group as well. I’ve successfully gotten a multicast paging group working in the dialplan as follows:
<extension name="rtp_multicast_page">
<condition field="destination_number" expression="^3456$">
<action application="set" data="absolute_codec_string=PCMU@20i"/>
<action application="set" data="jitterbuffer_msec=500:1000:20"/>
<action application="answer"/>
<action application="esf_page_group" data="234.5.6.7 23458 30"/>
</condition>
</extension>
When I use X-Lite registered to FS and call “3456”, I get the spoken audio out the paging group correctly. Now I’m trying to link the paging group to a conference. I’ve tried experimenting with adding it to an existing conference by using “conference_set_auto_outcall”, but that’s failing:
<action application="conference_set_auto_outcall" data="3456@$${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…
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org