[asterisk-bugs] [Asterisk 0013440]: No TO tag after SIP INFO in early dialog

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13440 
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Reported By:                atca_pres
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13440
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 140115 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-07 15:29 CDT
Last Modified:              2008-09-09 15:27 CDT
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Summary:                    No TO tag after SIP INFO in early dialog
Description: 
A Calls Asterisk IVR
IVR answers
A enters B's ext
A pushes a DTMF while B is ringing
A hangs up

in SIP :
A INVITES Asterisk
Asterisk 200OK (IVR)
Asterisk INVITES B
B sends 180 Ringing
A sends SIP INFO 
Asterisk sends SIP INFO to B
A BYEs Asterisk
Asterisk sends CANCEL with no TO Tag

The UA cannot match the INVITE to the CANCEL because the CANCEL does not
have the TO tag that was in the provisional responses (100 trying and 180
ringing).

B only ack the CANCEL without sending the required 487 to cancel the
INVITE. B Stays ringing forever. 

Attached is the SIP debug + core and verbose 5 (asterisk -Tvvvvvdddddngc |
tee /tmp/verbosedebug.txt) and an ethereal capture for easier reading. 
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Relationships       ID      Summary
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related to          0013381 Wrong branch on CANCEL after SIP INFO i...
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---------------------------------------------------------------------- 
 (0092278) putnopvut (administrator) - 2008-09-09 15:27
 http://bugs.digium.com/view.php?id=13440#c92278 
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I have a new hypothesis on what may be causing the problem. I think the
problem is that the CANCEL has a lower Cseq number than the preceding INFO
request. If I'm correct, then there are several open issues all dealing
with a similar problem, the root of which is that Asterisk does not
gracefully handle multiple simultaneous transactions in a single dialog.

The basic issue is that if two outgoing transactions occur, then Asterisk
will ignore the response to the request with the lower Cseq number. To
confirm that this is the problem you are experiencing, if you can turn on
SIP history, you should see lines that say "Ignoring this retransmit" when
this failure occurs. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-09-09 15:27 putnopvut      Note Added: 0092278                          
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