[asterisk-bugs] [Asterisk 0013468]: IAX Transfer/releasing between 3 asterisk's are not working.

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Sep 12 08:54:52 CDT 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13468 
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Reported By:                nicox
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13468
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0-rc6 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-12 04:42 CDT
Last Modified:              2008-09-12 08:54 CDT
====================================================================== 
Summary:                    IAX Transfer/releasing between 3 asterisk's are not
working.
Description: 
Since the Asterisk Version 1.6.0-beta9 (the first 1.6.0 i tried) i the IAX
Releasing is not working on this machine.
I tried also 1.6.0-rc6 and the problem is still the same.

I have some asterisk 1.2.* and 1.4.* and SVN-trunk-r92976 running, and
with all machines calls are released after the call answered.

Only the machine with asterisk 1.6.0-* can't do this.

i don't know how to test things to help debugging it.

if you need anything to solve this problem, please let me know.

thanks a lot

Nico
====================================================================== 

---------------------------------------------------------------------- 
 (0092428) nicox (reporter) - 2008-09-12 08:54
 http://bugs.digium.com/view.php?id=13468#c92428 
---------------------------------------------------------------------- 
The call comes from Asterisk 1.4 or 1.2 to 1 "routing box" which handles
how to route the call next, the routing box is asterisk 1.4 
The routing box dials the call to "termination boxes" for this boxes we
using 1.2, SVN-trunk-r92976 and 1.6.0-*.

Where the calls are terminating via a box which is running 1.2 or the
svn-version the call is released and the media-path and anyting else is
handled without the "routing-box".

This scenario is not working with the 1.6 version of asterisk, in this
case the media patch is handled also via the "routing-boxes".


that means 
A (1.2) calls to B(1.4) calls to C(1.2)
the call will be released after answering and the media path and all
signalling is handled between A and C

A (1.2) calls to B(1.4) calls to D(1.6)
the call will not be released after answering, so the complete media path
is handled via C. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-12 08:54 nicox          Note Added: 0092428                          
======================================================================




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