[asterisk-bugs] [Asterisk 0014496]: IMAP crash multiple callers / callers hangup at beep
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon May 11 10:23:44 CDT 2009
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=14496
======================================================================
Reported By: vbcrlfuser
Assigned To: lmadsen
======================================================================
Project: Asterisk
Issue ID: 14496
Category: Applications/app_voicemail/IMAP
Reproducibility: sometimes
Severity: crash
Priority: normal
Status: assigned
Asterisk Version: 1.4.22
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-02-17 22:10 CST
Last Modified: 2009-05-11 10:23 CDT
======================================================================
Summary: IMAP crash multiple callers / callers hangup at beep
Description:
A multiple CPU machine + asterisk-1.4.22 + imap2004g, with more than one
user in voicemail (not the same mailbox) disconnecting right after or close
to beep + 3 seconds timeout for abandon causes, is a bad combination.
In same cases IMAP warnings about parsing response and in most cases a
segfault. I'd like to know what the root cause is and what fixes it? I can
produce any backtraces / scenarios anyone needs to lend a hand here.
======================================================================
----------------------------------------------------------------------
(0104526) lmadsen (administrator) - 2009-05-11 10:23
http://bugs.digium.com/view.php?id=14496#c104526
----------------------------------------------------------------------
I realize it's not a PJSUA problem as I can reproduce this with just 3-4
lines on a softphone.
My problem is that I can't seem to get any sort of valid debugging
information, and I was hoping that the script you were using to hit the
server would help me reproduce this easier and faster than with a softphone
and some lines which didn't seem to be consistant in reproducing (as I have
to setup all the lines manually).
Issue History
Date Modified Username Field Change
======================================================================
2009-05-11 10:23 lmadsen Note Added: 0104526
======================================================================
More information about the asterisk-bugs
mailing list