[asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

[Robert McGilvray]

Hello,

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.

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.

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.

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

https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM&usp=sharing

Can anyone offer some insight?

Regards,

BobM
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