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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Dec 12 11:57:32 CST 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:                russell
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Project:                    Asterisk
Issue ID:                   13468
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.1-beta3 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-12 04:42 CDT
Last Modified:              2008-12-12 11:57 CST
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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
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---------------------------------------------------------------------- 
 (0096330) abutton (reporter) - 2008-12-12 11:57
 http://bugs.digium.com/view.php?id=13468#c96330 
---------------------------------------------------------------------- 
We are experiencing the same issue between two of our asterisk machines.
Both are running 1.6.x, we have tryed a couple different versions of 1.6
with no change.

We are also getting the lagid.

Basically, we have our IVR for the entire company, on Site A. So when you
call to Site B, it swings you to site A for the IVR, then when you select
someone from Site B, it swings back. I was hoping this would release the
call, as to not have the entire call tromboned from one site to the other
for the duration of the call.

We a call attempts to transfer here, we see 4 'unable to transfer'
messages on Site A's console.

I can gather more information if needed, just let me know what would
help.

Thanks. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-12 11:57 abutton        Note Added: 0096330                          
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