[asterisk-dev] Conference wont traverse my SIP trunk

[Matthew Jordan]

On Thu, Jun 27, 2013 at 8:36 AM, DadoMaker <dadomaker at gmail.com> wrote:

> 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)
>
>
>
Hi -

This question isn't really appropriate for the asterisk-dev mailing list.
The asterisk-dev mailing list is used to discuss Asterisk development and
code, not questions about configuration, deployment or usage problems.
You'll get better answers faster if you ask this question on the
asterisk-users mailing list.

Thanks

Matt


-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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