On Fri, Feb 4, 2011 at 5:51 AM, Claas Hilbrecht < [email protected]<claas.hilbrecht%[email protected]> > wrote:
> Hello Nikolay, > > thanks for the response. > > > 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 >> > > I omitted RTP in the trace at the first post to save space but here is the > full log. > > > don't use polycom phones and I can't advice if you use appropriate >> firmware version. >> > > The IRC (#sipxecs) folks told me to use this version. 3.1.3RevC if fine. > > > 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 >> > > Hmm, maybe this is because of HD audio? Reading < > http://wiki.sipfoundry.org/display/sipXecs/Polycom#Polycom-HDVoice> it > seems that the PolyCom phones talks directly to each other. HD audio or not, if the phones are on the same LAN segment (or sipx is told to treat the media as local via intranet subnets) all media between handsets is peer to peer. This is not unique to polycom, nor G722 (HD) codecs). > > > 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... >> > > I suspect there were configs on the phone(s) previous to using them with sipx. If this is the case, please confirm. If you did use these elsewhere but did not do a "reset local settings" via the admin/reset menu, you should do so. > I think my setup is the most simple one. I have LAN 192.0.2.0/24 dedicated > to VoiP. DNS and DHCP are handeled by sipXecs. A SmartNode 4638 is used to > connect sipXecs to our PSTN lines. Only VoIP devices are connected to the > lan. > > > Your traces does not show full message flow. >> Did you take traces from the phones (mirrored ports)? >> > > For debugging cases like this one I use a plain old 100 MBit/ HUB (yes a > HUB, no switch). So all traffic is "mirroed" to all ports. Makes debugging > with tcpdump/wireshark much more easier. But as I said before I use > WireShark to filter only the call. > > 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 >> useful for troubleshooting. >> > > Attached you will find a "no-audio.xml" for the sipXviewer. > > Thanks for your help > > > 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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