On Mon, Feb 2, 2009 at 12:39 PM, Lincoln King-Cliby <linc...@controlworks.com> 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 >
I have a customer with the same complaint and I am trying to figure it out as well. I have not caught the debug action yet though. First, are you using FreePBX? Second, are you using the "announce" feature. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _______________________________________________ -- 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
