[asterisk-bugs] [Asterisk 0016185]: Crash of Outgoing Call
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Nov 30 10:54:25 CST 2009
The following issue has been CLOSED
======================================================================
https://issues.asterisk.org/view.php?id=16185
======================================================================
Reported By: alex70
Assigned To: tilghman
======================================================================
Project: Asterisk
Issue ID: 16185
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: SVN
JIRA: SWP-365
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2
SVN Revision (number only!): 227947
Request Review:
Resolution: no change required
Fixed in Version:
======================================================================
Date Submitted: 2009-11-05 06:26 CST
Last Modified: 2009-11-30 10:54 CST
======================================================================
Summary: Crash of Outgoing Call
Description:
Everyday our asterisk system updates itself with the most recent svn
branch.
Last night asterisk update to revision r227947 and every outbound call
falls as soon as the called pick up the phone.
Our suspect is there is something wrong with the patch uploaded by
mnicholson to resolve issue https://issues.asterisk.org/view.php?id=16005 with:
R227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17
lines - "This patch modifies the Dial application to monitor the calling
channel for hangups while playing back announcements"
Please see also attached verbosity file
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0016238 sip calls drop because of BYE's
======================================================================
----------------------------------------------------------------------
(0114371) tilghman (administrator) - 2009-11-30 10:54
https://issues.asterisk.org/view.php?id=16185#c114371
----------------------------------------------------------------------
As evidenced by the SIP dialog, "ignoresdpversion" is indeed the correct
fix, as we would otherwise ignore SDP offers where the version has not
changed. A correctly implemented client should increment the version
number each time there is a change in the SDP offer.
Issue History
Date Modified Username Field Change
======================================================================
2009-11-30 10:54 tilghman Note Added: 0114371
2009-11-30 10:54 tilghman Status assigned => closed
2009-11-30 10:54 tilghman Resolution open => no change
required
======================================================================
More information about the asterisk-bugs
mailing list