<div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jun 27, 2013 at 9:53 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"><span style="font-family:arial,sans-serif;font-size:13px">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.</span><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">The server with the conference:</div><div style="font-family:arial,sans-serif;font-size:13px">
<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 style="font-family:arial,sans-serif;font-size:13px">The server from which some users dial in from:</div>
<div style="font-family:arial,sans-serif;font-size:13px">exten => 5777,1,Dial(SIP/$EXTEN}@200_PBX)</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">
Any insight appreciated.</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">Thanks,</div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px">Dado </div></div>
<br></blockquote><div><br></div><div style>Dado, subject sounds like a personal problem. Sorry couldn't resist.<br><br>How about some CLI debug info while trying a call?</div><div style><br></div><div style>Thanks,</div>
<div style>Steve T</div></div></div></div>