[asterisk-bugs] [Asterisk 0016179]: Asterisk Crash after SIP Transfer

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Nov 4 22:47:52 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16179 
====================================================================== 
Reported By:                xinyer
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16179
Category:                   Channels/chan_sip/Transfers
Reproducibility:            have not tried
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.26.2 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-11-04 02:31 CST
Last Modified:              2009-11-04 22:47 CST
====================================================================== 
Summary:                    Asterisk Crash after SIP Transfer
Description: 
Asterisk crashes without any error, the last two lines of log before crash
are: 
VERBOSE[2998] logger.c:     -- SIP/8083-08e16f60 is ringing
DEBUG[1891] chan_sip.c: SIP transfer: Succeeded to masquerade channels.

Similar error was reported in the following thread:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13256#93893

but it is closed. This happened to me twice today, but it works fine for
the last one week. I couldn't reproduce the crash.
====================================================================== 

---------------------------------------------------------------------- 
 (0113173) xinyer (reporter) - 2009-11-04 22:47
 https://issues.asterisk.org/view.php?id=16179#c113173 
---------------------------------------------------------------------- 
Dear Imadsen,

1) My server did not generate any core file, and I'm using safe_asterisk.
Do I need to reinstall asterisk with DONT_OPTIMIZE flag?

2) The last output at the console show that there was an attended
transfer. The call was from one SIP extension to another SIP extension,
then when one of the user tried to transfer to a third SIP extension the
server crash. There is no error message left in the log. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-04 22:47 xinyer         Note Added: 0113173                          
======================================================================




More information about the asterisk-bugs mailing list