[asterisk-bugs] [Asterisk 0010696]: Doing an attended transfer, asterisk tries multiple times a native bridge

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10696 
====================================================================== 
Reported By:                tbelder
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10696
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             09-11-2007 03:28 CDT
Last Modified:              10-08-2007 06:24 CDT
====================================================================== 
Summary:                    Doing an attended transfer, asterisk tries multiple
times a native bridge
Description: 
When doing an attended transfer, the cli shows 'Native bridging
SIP/403-090e0088 and SIP/401-090e1790' more then 200 times. You can see it
in verbose3.txt
Although the transfer is successful.

Using Thomson 2030st and hanging up the transferred call, both phones
rings again. You can see this in thomson.cap
Using Snoms, this isn't happening. You can see this in snom.cap

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

---------------------------------------------------------------------- 
 ramonpeek - 10-08-07 06:24  
---------------------------------------------------------------------- 
Hello everyone!
I'm a colleague of the reporter of this issue (tbelder)
And I've looked into this issue as well

It seems the CLI showing; 'Native bridging SIP/403-090e0088 and
SIP/401-090e1790' over more then 200 times is nearly a side effect of what
is really going wrong.

What is really happening is that Thomson, but also Polycom, phones put the
active call leg on hold first and then transfer the call by sending the
REFER request.
Snom phones on the other hand send the REFER immediately and by doing that
they skip a problem that exists in Asterisk.

It seems Asterisk send 2 re-invites in order to put the call back on hold,
where there should actually be send only one. 
(I spoke to Olle Johannson about this and he somehow noticed this, thanks
Olle!)

There are another issues in the bugtracker that actually are the same as
this issue.. see issue 10868 for a more detailed explaination. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
10-08-07 06:24  ramonpeek      Note Added: 0071622                          
======================================================================