[asterisk-bugs] [Asterisk 0017284]: SIP attended transfer broken
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed May 5 14:13:58 CDT 2010
The following issue has been RESOLVED.
======================================================================
https://issues.asterisk.org/view.php?id=17284
======================================================================
Reported By: dvossel
Assigned To: dvossel
======================================================================
Project: Asterisk
Issue ID: 17284
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: high
Status: resolved
Asterisk Version: SVN
JIRA: SWP-1407
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 260805
Request Review:
Resolution: fixed
Fixed in Version:
======================================================================
Date Submitted: 2010-05-04 11:08 CDT
Last Modified: 2010-05-05 14:13 CDT
======================================================================
Summary: SIP attended transfer broken
Description:
SIP attended transfers are broken in trunk. If an attended transfer is
attempted, right as the transferer hangs up to connect the two calls all
the calls are terminated. It doesn't matter if it is a semi-attended
transfer or not.
So A calls B. A transfers B to C. right as A hangs up to connect B and C
(regardless if C has picked up yet or not) all the calls terminate. This is
easy to reproduce.
======================================================================
----------------------------------------------------------------------
(0121433) svnbot (reporter) - 2010-05-05 14:13
https://issues.asterisk.org/view.php?id=17284#c121433
----------------------------------------------------------------------
Repository: asterisk
Revision: 261316
U trunk/channels/chan_sip.c
------------------------------------------------------------------------
r261316 | dvossel | 2010-05-05 14:13:57 -0500 (Wed, 05 May 2010) | 10
lines
fixes sip native transfer
The Refer-To header field containing the Replaces header in the URI
was not being decoded properly. This caused invalid parsing between
the caller id field and the domain resulting in a failed transfer.
(closes issue https://issues.asterisk.org/view.php?id=17284)
Reported by: dvossel
------------------------------------------------------------------------
http://svn.digium.com/view/asterisk?view=rev&revision=261316
Issue History
Date Modified Username Field Change
======================================================================
2010-05-05 14:13 svnbot Checkin
2010-05-05 14:13 svnbot Note Added: 0121433
2010-05-05 14:13 svnbot Status assigned => resolved
2010-05-05 14:13 svnbot Resolution open => fixed
======================================================================
More information about the asterisk-bugs
mailing list