<div dir="ltr">The cogerence works but doesnt go over my trunk. Its bypassing and the codec is PCM of phone. But in phone to phone call, the rtp traverses the trunk and I capture gsm packets to verify.<div><br></div><div style>
The sip debug for conf call setup and leave:</div><div style><div>*CLI> == Using SIP RTP CoS mark 5</div><div> -- Executing [5777@public:1] Goto("SIP/127.0.0.1-00000012", "conf-confDemo,join,1") in new stack</div>
<div> -- Goto (conf-confDemo,join,1)</div><div> -- Executing [join@conf-confDemo:1] ConfBridge("SIP/127.0.0.1-00000012", "1") in new stack</div><div> > 0x7f006c015150 -- Probation passed - setting RTP source address to <a href="http://192.168.100.100:4002">192.168.100.100:4002</a></div>
<div> -- <SIP/127.0.0.1-00000012> Playing 'conf-onlyperson.ulaw' (language 'en')</div><div> -- <SIP/127.0.0.1-00000012> Playing 'confbridge-join.ulaw' (language 'en')</div>
<div> -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-join.slin' (language 'en')</div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [5777@default:1] Goto("SIP/5700-00000013", "conf-confDemo,join,1") in new stack</div>
<div> -- Goto (conf-confDemo,join,1)</div><div> -- Executing [join@conf-confDemo:1] ConfBridge("SIP/5700-00000013", "1") in new stack</div><div> > 0x7f006c031d90 -- Probation passed - setting RTP source address to <a href="http://127.0.0.1:4004">127.0.0.1:4004</a></div>
<div> -- <SIP/5700-00000013> Playing 'confbridge-join.ulaw' (language 'en')</div><div> > 0x7f006c031d90 -- Switching RTP source address to <a href="http://192.168.1.10:4004">192.168.1.10:4004</a></div>
<div> -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-join.slin' (language 'en')</div><div> -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-leave.slin' (language 'en')</div>
<div> -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-leave.slin' (language 'en')</div><div><br></div><div style>Thanks,</div><div style>Dado</div></div></div><div class="gmail_extra"><br><br>
<div class="gmail_quote">On Thu, Jun 27, 2013 at 10:36 AM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.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"><br><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">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></div><div>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><br></div><div>Thanks,</div>
<div>Steve T</div></div></div></div>
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