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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Feb 2 09:46:17 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14347 
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Reported By:                alesz
Assigned To:                putnopvut
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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:              
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Date Submitted:             2009-01-27 06:58 CST
Last Modified:              2009-02-02 09:46 CST
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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

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

---------------------------------------------------------------------- 
 (0099220) putnopvut (administrator) - 2009-02-02 09:46
 http://bugs.digium.com/view.php?id=14347#c99220 
---------------------------------------------------------------------- 
Getting a 603 back from the SIP peer on every call is definitely
problematic and needs to be dealt with, but it also is a completely
separate issue than the originally-reported one here.

Let's try to separate these out a little bit to keep things more
manageable. Could you please open a new bug report about this. On that
issue, it would be helpful to see SIP debug from a failed call, and it may
potentially be helpful to see sip.conf entries for the involved parties.

Also, since you stated that calls are being transferred to the correct
context in your latest note, would you be comfortable if I closed this
issue out as being fixed? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-02 09:46 putnopvut      Note Added: 0099220                          
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