[asterisk-dev] Conference wont traverse my SIP trunk

DadoMaker dadomaker at gmail.com
Thu Jun 27 08:36:45 CDT 2013


My conference call wont go thru my SIP trunk.  I may be missing a dialplan
configuration setting as my PCM phone to phone calls go over the (GSM) tunk.


The server with the conference:
exten => 5777,1,GoTo(conf-confDemo,join,1)
[conf-confDemo]
exten => join,1,ConfBridge(confDemo/S/1)

The server from which some users dial in from:
exten => 5777,1,Dial(SIP/$EXTEN}@200_PBX)

Any insight appreciated.

Thanks,

Dado
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