[asterisk-bugs] [Asterisk 0017284]: SIP attended transfer broken

Asterisk Bug Tracker noreply at bugs.digium.com
Wed May 5 14:13:57 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
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:                     assigned
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:              
====================================================================== 
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                          
======================================================================




More information about the asterisk-bugs mailing list