Hi everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2):
[Oct 3 10:08:55] WARNING[1650]: chan_sip.c:1981 retrans_pkt: Maximum retries exceeded on transmission 00215553- [email protected] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 3 10:08:55] WARNING[1650]: chan_sip.c:2003 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see doc/sip-retransmit.txt). -- Hungup 'Zap/1-1' == Spawn extension (my-phones, 917070, 1) exited non-zero on 'SIP/ 200-103fa658' Turning on SIP debugging shows that it tries to send the following data to the 7940 six times before giving up: <--- Reliably Transmitting (no NAT) to 172.16.3.245:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.3.245:5061;branch=z9hG4bK1d4425f3;received=172.16.3.245 From: "James" <sip: [email protected]>;tag=00215553ee040030116ccaac-32c56370 To: <sip:[email protected]>;tag=as680ce289 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 258 v=0 o=root 1622 1622 IN IP4 172.16.3.2 s=session c=IN IP4 172.16.3.2 t=0 0 m=audio 12388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv The following was observed on the 7940's telnet console: SIP Phone> Warning: Unrecognized attribute (silenceSupp) Warning: Unrecognized attribute (silenceSupp) sip_sm_ccb_match_branch_cseq: Method index not found SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message. As far as I can tell, the 'a=silenceSupp:off - - - -' header is not accepted by the 7940, which seems like a bug in the SIP image to me. However, I can't find a way to report this problem to Cisco without a support contract (which I do not have). Reverting to version 7.5 fixes the problem, but it is still present in 8.11. The problem is not present if the PSTN initiates the call, nor is it present if I allow the handsets to reinvite each other. Here's the sip.conf snippet if it helps: [general] port = 5060 bindaddr = 0.0.0.0 context = others localnet=172.16.3.0/24 [200] type=friend host=dynamic dtmfmode=rfc2833 nat=no username=200 secret=*removed* context=my-phones canreinvite=no Anyone else encountered this problem or have a workaround? Regards, James. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
