[asterisk-bugs] [Asterisk 0015598]: SIP crash on ACK

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=15598 
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Reported By:                jln17
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15598
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Target Version:             1.6.2.1
Asterisk Version:           1.6.2.0-beta3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-07-28 09:56 CDT
Last Modified:              2009-08-31 08:26 CDT
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Summary:                    SIP crash on ACK
Description: 
At the end of the registration process crash just when the connection is
established on sending ACK.

==========
[Jul 27 17:18:10] DEBUG[15417]: channel.c:2986
ast_internal_timing_enabled: Internal timing is disabled
(option_internal_timing=0 chan->timingfd=34)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:9651 add_sdp: Done building
SDP. Settling with this capability: 0x100 (g729)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:3016 initialize_initreq:
Initializing
already initialized SIP dialog
47599549603048a3397eaaf11e4565a1 at 212.68.197.108(pr esumably reinvite)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:3289 __sip_xmit: Trying to put
'INVITE sip' onto UDP socket destined for 77.72.169.129:5060
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3754 __sip_ack: Acked pending
invite104
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3791 __sip_ack: Stopping
retransmission on '47599549603048a3397eaaf11e4565a1 at 212.68.197.108' of
Request 104: Match Found
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:16796 handle_response_invite:
SIP response 200 to RE-invite on outgoing call
47599549603048a3397eaaf11e4565a1 at 212.68.197.108
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:7625 process_sdp: SDP version
number same as previous SDP. Not parsing this SDP.
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:5282 update_call_counter:
Updating call counter for outgoing call
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3289 __sip_xmit: Trying to put
'ACK sip:+3' onto UDP socket destined for 77.72.169.129:5060
ns*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root at ns asterisk]# 
==========
Nothing happend if I set canreinvite to no.
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---------------------------------------------------------------------- 
 (0109830) lmadsen (administrator) - 2009-08-31 08:26
 https://issues.asterisk.org/view.php?id=15598#c109830 
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Can you please re-test with the latest 1.6.2 branches from subversion?

(svn co http://svn.asterisk.org/svn/asterisk/branches/1.6.2
asterisk-1.6.2-vanilla)

Or you can wait for Asterisk-1.6.2.0-rc1 which should be released early
this week.

If you get the same crash issue, please attach a backtrace per the
backtrace.txt file in the doc/ subdirectory of your Asterisk source.
Additionally, make sure you have enabled DONT_OPTIMIZE in the Compiler
Flags of menuselect.

Because this is SIP related, you need to provide the following
information:

* history of the call (recordhistory=yes, dumphistory=yes, in sip.conf)
* sip debug (which you've provided)
* relative configuration and dialplan snippet in order to reproduce the
issue
* backtrace of the crash

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-08-31 08:26 lmadsen        Note Added: 0109830                          
2009-08-31 08:26 lmadsen        Status                   new => feedback     
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