Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and 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 [email protected] - no reply to our critical packet (see doc/sip-retransmit.txt).".
I thought the issue had cleared itself up, but it has resurfaced. We're running 1.4.22 on built from source, using Cisco 7961G phones with the SIP firmware image (and I've tried most of the recent firmware versions for the Cisco phones to see if that would change things at all). 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 <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK1971acea;received=10.2.0.203 From: "Jim Felderman" <sip:[email protected]>;tag=001d45b61d4906943a6bb290-9435a462 To: <sip:[email protected]>;tag=as00aa5042 Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1db91b00" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis request - [email protected] 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 10.2.0.203:29422 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 10.2.0.203:29422 Looking for Voicemail in internal (domain 10.2.0.2) list_route: hop: <sip:[email protected]: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=z9hG4bKddc0a0b8;received=10.2.0.203 From: "Jim Felderman" <sip:[email protected]>;tag=001d45b61d4906943a6bb290-9435a462 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 <------------> Audio is at 10.2.0.2 port 12088 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=z9hG4bKddc0a0b8;received=10.2.0.203 From: "Jim Felderman" <sip:[email protected]>;tag=001d45b61d4906943a6bb290-9435a462 To: <sip:[email protected]>;tag=as58c8eec4 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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 12088 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 14:33:01] NOTICE[15644]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? Sending to 10.2.0.203 : 5060 (no NAT) cworks-phones1*CLI> <--- 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=z9hG4bK8a220b9f;received=10.2.0.203 From: "Jim Felderman" <sip:[email protected]>;tag=001d45b61d4906943a6bb290-9435a462 To: <sip:[email protected]>;tag=as58c8eec4 Call-ID: [email protected] CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 <------------> Really destroying SIP dialog '[email protected]' Method: BYE Sending to 10.2.0.203 : 5060 (no NAT) cworks-phones1*CLI> <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK51921801;received=10.2.0.203 From: "Jim Felderman" <sip:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as537c53ce Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20c9330e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis request - [email protected] 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:[email protected]: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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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 '[email protected]' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.2.0.203:5060: NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport From: "asterisk" <sip:[email protected]>;tag=as73ca9f87 To: <sip:[email protected]:5060;transport=udp> Contact: <sip:[email protected]> Call-ID: [email protected] 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:[email protected] Voice-Message: 3/5 --- Really destroying SIP dialog '[email protected]' 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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 [email protected] 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 [email protected] - no reply to our critical packet (see doc/sip-retransmit.txt). Really destroying SIP dialog '[email protected]' 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:[email protected]>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:[email protected]>;tag=as53449c29 Call-ID: [email protected] 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 '[email protected]' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.2.0.203:5060: NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport From: "asterisk" <sip:[email protected]>;tag=as0b88d5a9 To: <sip:[email protected]:5060;transport=udp> Contact: <sip:[email protected]> Call-ID: [email protected] 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:[email protected] Voice-Message: 2/6 --- Really destroying SIP dialog '[email protected]' 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
