[asterisk-bugs] [Asterisk 0014347]: Transfer executes callers channel in wrong context

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jan 30 16:44:12 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14347 
====================================================================== 
Reported By:                alesz
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   14347
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.5 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-27 06:58 CST
Last Modified:              2009-01-30 16:44 CST
====================================================================== 
Summary:                    Transfer executes callers channel in wrong context
Description: 
hen call is transfered using blind or attended transfer using dtmf code
defined in features.conf or using phone transfer (tested on GXP2000 and
polycom 330) it appears callers channel goes to same context as callee's
last context instead of continuing on typed in extension in context defined
by TRANSFER_CONTEXT global variable.
Callee-s channel hangups ok.

Using Asterisk 1.6.0.5 & Freepbx 2.5.1.1. This started happening somewhere
around 1.6.0.3-rc1.

workaround: manually send call to correct context using goto. does not
work with consecutive transfers: first > second > third

====================================================================== 

---------------------------------------------------------------------- 
 (0099162) putnopvut (administrator) - 2009-01-30 16:44
 http://bugs.digium.com/view.php?id=14347#c99162 
---------------------------------------------------------------------- 
I re-ran my test using a fresh checkout of the 1.6.0 branch and found that
the problem is gone, now. For sanity's sake, I reverted back to 1.6.0.3 and
once again experienced the problem.

If possible, could you attempt to re-run your scenario with the 1.6.0
branch from subversion? I'd like confirmation that this issue has really
been fixed and won't continue to fail with a more complicated dialplan in
use. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-30 16:44 putnopvut      Note Added: 0099162                          
======================================================================




More information about the asterisk-bugs mailing list