[asterisk-bugs] [Asterisk 0017324]: Asterisk crash after Hangup app

Thu May 13 09:22:21 CDT 2010

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17324 
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Reported By:                corruptor
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17324
Category:                   Core/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.28-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-05-12 07:46 CDT
Last Modified:              2010-05-13 09:22 CDT
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Summary:                    Asterisk crash after Hangup app
Description: 
This is probably related to astobj issue 0017321
To reproduce create simple context.

[testcon]                                                                 
     exten => _1X.,1,Noop()                 
exten => _1X.,n,Hangup 

 -- Executing [111 at testcon:1] NoOp("SIP/1102-00000000", "") in new stack
 -- Executing [111 at testcon:2] Hangup("SIP/1102-00000000", "") in new stack

  == Spawn extension (testcon, 111, 2) exited non-zero on
'SIP/1102-00000000'
[May 12 16:20:59] ERROR[5000]:
/usr/src/asterisk-1.6.0.28-rc1/include/asterisk/lock.h:357
__ast_pthread_mutex_destroy: astobj2.c line 268 (ao2_ref): Error destroying
mutex &obj->priv_data.lock: Device or resource busy
[May 12 16:20:59] ERROR[5000]: astobj2.c:115 INTERNAL_OBJ: bad magic
number 0x2e303132 for 0x8a145d0                                
[May 12 16:20:59] ERROR[5000]: utils.c:523 lock_info_destroy: Thread
'pbx_thread           started at [ 3993] pbx.c ast_pbx_start()' still has a
lock! - 'p' (0x8a145d0) from 'sip_hangup' in chan_sip.c:5152!

Then crash. Backtrace is attached.
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Relationships       ID      Summary
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related to          0017321 [regression] asterisk sip becomes unres...
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 (0121863) lmadsen (administrator) - 2010-05-13 09:22
 https://issues.asterisk.org/view.php?id=17324#c121863 
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Can you also reproduce this on the latest 1.6.2 release?

---

Per the Asterisk maintenance timeline page at
http://www.asterisk.org/asterisk-versions maintenance (bug) support for the
1.6.0 and 1.6.1 branches has ended. For continued maintenance support
please move to the 1.6.2 branch.

More information on this change can be found in the release announcement:
http://www.asterisk.org/node/49924 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-13 09:22 lmadsen        Note Added: 0121863                          
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