I have been having this same issue for weeks with sipxecs 4.4 and Polycom 330, 335, 560 and 670's. I have been working with polycom on the issue as it might be a polycom issue. What polycom is saying this the the phones are not sending the ACK after the MOH thus breaking the whole chain. I have gone back in firmware to the earliest supported for each device as well as the newest 3.2.5 SIP and 4.3.0 ROM. Polycom has even got me to customize a config for 3.3.1f SIP software. Same issue. I also have been working with Cisco on this issue as it was first noticed in my environment on inbound/outbound only. But now the issue seems to be between polycoms also.
For me, I can only put people on hold once and then resume. After that, I can't put them on hold again, transfer, conference or anything else. I have 225 phones doing this. I don't know if this was an issue in 4.2.1. The one thing I do see in SipX is the codec error in the as soon as I hit the hold and later come the sipxproxy.log errors. See below. I have set the system to g711u, g722 with the same results. The SDP looks like they are correct. Codec's are not conflicting. My environment is a cisco 2811 gateway to PRI and two SipXecs servers all with polycom phones. Are there more having this issue with Polycom? This is from the freeswitch log. 2011-04-14 14:03:33.608713 [ERR] sofia.c:5135 Codec Error! v=0 o=- 1302815009 1302815010 IN IP4 10.1.128.18 s=Polycom IP Phone c=IN IP4 10.1.128.18 t=0 0 a=sendonly m=audio 0 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 I would see these in the sipxproxy.log file "2011-04-07T19:39:00.095535Z":3278:SIP:ERR:pbx01.uws.edu:SipUserAgent-2:41EBB940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction" "2011-04-07T19:39:00.102472Z":3280:NAT:WARNING:pbx01.uws.edu:SipRouter-15:41FBC940:SipXProxy:"'ytXjrZaKNyQjp': Received unexpected event UpdateRequest while in state 'WaitingForInvite'" "2011-04-07T19:39:17.401042Z":3358:SIP:WARNING:pbx01.uws.edu:SipClientUdp-12:40F82940:SipXProxy:"SipTransaction::handleChildIncoming duplicate 2xx response received on UAC for INVITE with no ACK" -- Paul Curtis From: "Matthew Kitchin (public/usenet)" <mkitchin.pub...@gmail.com<mailto:mkitchin.pub...@gmail.com>> Reply-To: Discussion list for users of sipXecs software <sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org>> Date: Wed, 20 Apr 2011 16:13:34 -0500 To: <sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org>> Subject: Re: [sipx-users] issue with calls on hold with Polycom 335 What codecs (and order of preference) are you using? What version of Sipx? On 4/20/2011 4:02 PM, Paul Stulac wrote: I am currently testing sipx with just three phones, 1 polycom soundpoint IP 335 and 2 Bria softphones. With the polycom every time I put a call on hold and press the resume softkey I can’t get connected back to the call; the phone display says “connecting….” and I have to disconnect the call. With the bria softphones I can put calls on hold and resume them without issue. Any ideas where I should start looking? Thanks, Paul _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/
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