Summary: | ASTERISK-00176: WARNING[5126]: File chan_sip.c, Line 2216 (__transmit_response): Unable to determine sequence number from '' | ||
Reporter: | fhedberg (fhedberg) | Labels: | |
Date Opened: | 2003-08-26 15:20:33 | Date Closed: | 2011-06-07 14:10:11 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Receives the following error when originating a call from a sip ua and doing a hangup before the call is answered. Occurs only with a specific UA (www.i3micro.com) which is know to be pedantic in its sip parsing. The only problem is the warning, works perfectly anyhow... ****** ADDITIONAL INFORMATION ****** Sip read: INVITE sip:0703323033@62.209.162.162 SIP/2.0 Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392 Max-Forwards: 70 To: sip:0703323033@62.209.162.162 From: sip:462760000@213.65.16.89;tag=1031263061656357 Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 INVITE Contact: sip:462760000@213.65.16.97:6010 User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D) Content-Length: 157 Content-Type: application/sdp v=0 o=- 1031263061 1031263061 IN IP4 213.65.16.97 s=- c=IN IP4 213.65.16.97 t=0 0 m=audio 10002 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=ptime:30 a=sendrecv 11 headers, 9 lines Using latest request as basis request Sending to 213.65.16.97 : 6010 (non-NAT) Found audio format UNKN Found description format GSM Capabilities: us - 524558, them - 2/0, combined - 2 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 0703323033 in telavox list_route: hop: <sip:462760000@213.65.16.97:6010> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392;received=213.65.16.97 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as19495cff Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0703323033@62.209.162.162> Content-Length: 0 to 213.65.16.97:6010 -- Executing AGI("SIP/u2560099-944a", "std-outgoing.php") in new stack -- Launched AGI Script /usr/local/asterisk/var/lib/asterisk/agi-bin/std-outgoing.php -- AGI Script Executing Application: (Dial) Options: (Zap/g2/0703323033|180|Tr) -- Called g2/0703323033 Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392;received=213.65.16.97 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as19495cff Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0703323033@62.209.162.162> Content-Length: 0 to 213.65.16.97:6010 -- Zap/1-1 is ringing We're at 62.209.162.162 port 12776 Answering with preferred capability 2 Answering with non-codec capability 1 Transmitting (NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392;received=213.65.16.97 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as19495cff Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0703323033@62.209.162.162> Content-Type: application/sdp Content-Length: 192 v=0 o=root 24676 24676 IN IP4 62.209.162.162 s=session c=IN IP4 62.209.162.162 t=0 0 m=audio 12776 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 213.65.16.97:6010 Sip read: SIP_Call_A: NAT KEEP ALIVE 1 headers, 0 lines Sip read: CANCEL sip:0703323033@62.209.162.162 SIP/2.0 Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392 Max-Forwards: 70 To: sip:0703323033@62.209.162.162 From: sip:462760000@213.65.16.89;tag=1031263061656357 Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 CANCEL User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D) Content-Length: 0 9 headers, 0 lines Sending to 213.65.16.97 : 6010 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392;received=213.65.16.97 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as19495cff Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0703323033@62.209.162.162> Content-Length: 0 to 213.65.16.97:6010 Reliably Transmitting (NAT): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392;received=213.65.16.97 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as19495cff Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:0703323033@62.209.162.162> Content-Length: 0 to 213.65.16.97:6010 -- Hungup 'Zap/1-1' == Spawn extension (telavox, 0703323033, 1) exited non-zero on 'SIP/u2560099-944a' Sip read: ACK sip:0703323033@62.209.162.162 SIP/2.0 Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392 Max-Forwards: 70 To: sip:0703323033@62.209.162.162;tag=as19495cff From: sip:462760000@213.65.16.89;tag=1031263061656357 Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 ACK User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D) Content-Length: 0 9 headers, 0 lines Sip read: CANCEL sip:0703323033@62.209.162.162 SIP/2.0 Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392 Max-Forwards: 70 To: sip:0703323033@62.209.162.162 From: sip:462760000@213.65.16.89;tag=1031263061656357 Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 CANCEL User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D) Content-Length: 0 9 headers, 0 lines Sending to 213.65.16.97 : 6010 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 213.65.16.97:6010;branch=z9hG4bK1031263061667392 From: sip:462760000@213.65.16.89;tag=1031263061656357 To: sip:0703323033@62.209.162.162;tag=as507ed7ab Call-ID: 1031263061659077@213.65.16.97 CSeq: 20 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 213.65.16.97:6010 WARNING[5126]: File chan_sip.c, Line 2216 (__transmit_response): Unable to determine sequence number from '' | ||
Comments: | By: Mark Spencer (markster) 2003-08-27 10:26:28 I think Asterisk is confused because of the second CANCEL. Any idea why it's being sent? By: John Todd (jtodd) 2003-09-29 03:25:59 Is this still a problem? I may not understand this process correctly, but I think it's strange to see in the NAT KEEP ALIVE message a "CANCEL", which may be confusing asterisk. Is this normal? Do you have a "tethereal -V port 5060" example of this UA working correctly with some other SIP proxy/gateway? By: fhedberg (fhedberg) 2003-09-29 05:07:59 We have worked out the bug in the firmware of our TA. Please close bug, sorry for that... |