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<p class="MsoNormal">Hello, <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from
moving forward. Everything works fine until a call is placed on hold, after resuming the call the user cannot hear audio from the bridge. The same thing works perfectly with 1.8.20.1.
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">The scenario is: Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed offer -> ConfBridge.
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we’re dealing with a bug / interop issue with the culprit possibly being a=inactive lines in the SDP.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I’ve included a link (on drive) to two separate SIP traces, one using ngrep and the other is the output of pjsip logging and the relevant sections of my pjsip.conf<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="color:#1F497D"><a href="https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM&usp=sharing">https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM&usp=sharing</a><o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Can anyone offer some insight?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Regards,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">BobM<o:p></o:p></p>
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