<div dir="ltr"><br><div class="gmail_extra"><div class="gmail_quote">On Thu, Jun 27, 2013 at 8:36 AM, DadoMaker <span dir="ltr"><<a href="mailto:dadomaker@gmail.com" target="_blank">dadomaker@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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.<div>
<br></div><div><br></div><div>The server with the conference:</div>
<div><div>exten => 5777,1,GoTo(conf-confDemo,join,1)</div><div>[conf-confDemo]</div><div>exten => join,1,ConfBridge(confDemo/S/1)</div><div><br></div></div><div>The server from which some users dial in from:</div>
<div>exten => 5777,1,Dial(SIP/$EXTEN}@200_PBX)</div><div><br></div><div><br></div></div></blockquote><div><br></div><div style>Hi -</div><div style><br></div><div style>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.</div>
<div style><br></div><div style>Thanks</div><div style><br></div><div style>Matt </div></div><br clear="all"><div><br></div>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div>
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