Hi Lincoln, Asterisk was expecting ACK after sending the 200 OK message. After repeated attempts at sending the 200 OK message and not receiving ACK, it terminated the call. Are you able to do a packet capture on the phone end? Mostly likely the phone is sending the ACK, but its either sent to somewhere else or your firewall is blocking it (not likely since you are able to receive the call in the first place). The packet capture on the phone end will probably show you the smoking gun.
Regards, Steve Lincoln King-Cliby wrote: > Hi All, > > I posted this a couple weeks ago with no response, I'm hoping that someone > will see it this time around and be so kind as to offer advice for resolving > this issue (or point me in the direction of a better place to ask) > > "Some" (but not all) calls on one of our Asterisk boxes are being dropped in > Voicemail -- only in voicemail -- after about 20 seconds with the error > logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: > Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 - no reply to > our critical packet (see doc/sip-retransmit.txt).". > > We're running Asterisk 1.4.22 built from source and Cisco 7961G phones with > the SIP firmware image. I've tried most of the recent firmware versions for > the phones with no real impact on the issue. Strange thing is that while all > of the phones use a variation on the same config file (with the only changes > being the SIP account details and speed dial keys) but one user in particular > seems to suffer the issue far more frequently. > > I would appreciate any assistance since I'm stumped. The output of SIP DEBUG > for the extension most frequently affected by the issue is below; starting > with one call to voicemail that was successfully completed, followed by a 2nd > call that was dropped after approximately 18 seconds. > > The issue is consistently inconsistent - it doesn't happen on every call to > Voicemail, but those that it does happen on it's always within the first > approximately 20 seconds of the call; once you pass the 25 second mark you're > free and clear for that call-it will not be dropped. It also seems like it's > possible to reproduce the issue by making several calls to Voicemail in short > order, but this isn't the only trigger as sometimes the first call to > voicemail in 12+ hours will also trigger it. > > I'm also baffled by the fact that this ONLY crops up on calls to Voicemail on > the local box; SIP to SIP calls on the same Asterisk box, SIP to SIP calls > from this Asterisk box to an Asterisk Appliance at a remote site, SIP to > POTS, and POTS to SIP calls are completely unaffected. > > Again, any advice/suggestions/things to look at/etc are greatly appreciated! > > Thanks in advance, > > Lincoln > > <------------> > Scheduling destruction of SIP dialog > '001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203' in 32000 ms (Method: INVITE) > Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis request - > 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 18 > Found RTP audio format 101 > Peer audio RTP is at port 10.2.0.203:24394 Found audio description format > PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio > description format G729 for ID 18 Found audio description format > telephone-event for ID 101 > Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec > capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port > 10.2.0.203:24394 Looking for Voicemail in internal (domain 10.2.0.2) > list_route: hop: <sip:1...@10.2.0.203:5060;transport=udp> > cworks-phones1*CLI> > <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2> > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Length: 0 > > > <------------> > Audio is at 10.2.0.2 port 13256 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x8 (alaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > > <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > <------------> > Retransmitting #1 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > Retransmitting #2 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > Retransmitting #3 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > Scheduling destruction of SIP dialog > '44b7d5c43fff7c0567e6c3be3d7d6...@10.2.0.2' in 32000 ms (Method: NOTIFY) > Reliably Transmitting (no NAT) to 10.2.0.203:5060: > NOTIFY sip:1...@10.2.0.203:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport > From: "asterisk" <sip:aster...@10.2.0.2>;tag=as73ca9f87 > To: <sip:1...@10.2.0.203:5060;transport=udp> > Contact: <sip:aster...@10.2.0.2> > Call-ID: 44b7d5c43fff7c0567e6c3be3d7d6...@10.2.0.2 > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Max-Forwards: 70 > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 83 > > Messages-Waiting: yes > Message-Account: sip:aster...@10.2.0.2 > Voice-Message: 3/5 > > --- > Really destroying SIP dialog '44b7d5c43fff7c0567e6c3be3d7d6...@10.2.0.2' > Method: NOTIFY Retransmitting #4 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > Retransmitting #5 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > Retransmitting #6 (no NAT) to 10.2.0.203:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:voicem...@10.2.0.2> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 27452 27452 IN IP4 10.2.0.2 > s=session > c=IN IP4 10.2.0.2 > t=0 0 > m=audio 13256 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 > > --- > [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1958 retrans_pkt: Maximum > retries exceeded on transmission > 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 for seqno 102 (Critical > Response) -- See doc/sip-retransmit.txt. > [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up > call 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 - no reply to our > critical packet (see doc/sip-retransmit.txt). > Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203' > Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT) cworks-phones1*CLI> > <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call > leg/transaction does not exist > Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203 > From: "Jim Felderman" > <sip:1...@10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b > To: <sip:voicem...@10.2.0.2>;tag=as53449c29 > Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 > CSeq: 103 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog > '0d32ee8515d9f6dc4439002f1d601...@10.2.0.2' in 32000 ms (Method: NOTIFY) > Reliably Transmitting (no NAT) to 10.2.0.203:5060: > NOTIFY sip:1...@10.2.0.203:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport > From: "asterisk" <sip:aster...@10.2.0.2>;tag=as0b88d5a9 > To: <sip:1...@10.2.0.203:5060;transport=udp> > Contact: <sip:aster...@10.2.0.2> > Call-ID: 0d32ee8515d9f6dc4439002f1d601...@10.2.0.2 > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Max-Forwards: 70 > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 83 > > Messages-Waiting: yes > Message-Account: sip:aster...@10.2.0.2 > Voice-Message: 2/6 > > --- > Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d601...@10.2.0.2' > Method: NOTIFY cworks-phones1*CLI> > > > -- > Lincoln King-Cliby, CTS > Applications Engineer > ControlWorks Consulting, LLC > V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ > Crestron Authorized Independent Programmer > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
