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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 15:20:33Date Closed:2011-06-07 14:10:11
Priority:MinorRegression?No
Status:Closed/CompleteComponents: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...