I would guess that you do not have voice when calling 200->201 because there is no Ack sent by "200-phone". And in the trace there is one way RTP. I would say that you should have "one way audio", not "no audio". I don't use polycom phones and I can't advice if you use appropriate firmware version.
I see another strange thing in the trace. Sipx always sets Record-route header, pointing to itself, so that all messages go through sipx. I do not see this header in the trace you provided, and I see Bye message sent directly from one phone to another. That is bad. There is something wrong in your setup. Search the wiki for topology description and setup examples... Your traces does not show full message flow. Did you take traces from the phones (mirrored ports)? I'd better just run "tcpdump -s 0 -w filename.cap" at the sixp server and then transfer filename.cap to your pc and view it with wireshark. You'll see full message flow... Phone1 <-> sipx <-> phone2. (Of course you'll see only packets that go through sipx). Or you may want to use sipviewer http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer It will show interprocess sipx communication, which is very usefull for troubleshooting. Rgds, Nikolay. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Claas Hilbrecht > Sent: Thursday, February 03, 2011 5:58 PM > To: Discussion list for users of sipXecs software > Subject: [sipx-users] phone INVITE -> TRYING -> RINGING -> > connect? but no audio/voice with PolyCom SoundPoint 650 > > Hello, > > I'm new to sipxecs (sipXconfig (4.2.1-018971.dhubler > 2010-08-21T04:59:18 > build34)) and have issues with my test setup. The phone (both > phones are PolyCom SoundPoint 650 with Firmware 3.1.3.0439) > 200 can call phone 201 but no audio/voice gets through. > Calling 200 from phone 201 works, audio/voice is ok. I > captured but calls with wireshark and the only difference I > see is a "SIP PRACK" request at the working capture. But both > phones are configured with a shared sipxecs phone group and > have no individual configuration beside that. I already tried > several time to reset the phones with "486*". I attached the > sipxecs configuration files for both phones in a zip archiv. > > Attached are two WireShark captures: > PolyCom_200_to_201-without-RTP.pcap: 200 -> 201 -> no audio/voice > PolyCom_201_to_200-without-RTP.pcap: 201 -> 200 -> audio/voice ok > > Any ideas what's going wrong or what I can test next? > > Thanks, > Claas > > Mit freundlichem Gruss > Claas Hilbrecht > > -- > http://www.linum.com mailto: [email protected] > Linum Software GmbH Langer Wall 5, 37574 Einbeck, Germany > Tel: +49-5561-926730 Fax: +49-5561-926750 Handelsregister > Amtsgericht Göttingen HRB 131128 Geschäftsführer Claas-Jörg Hilbrecht > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
