[asterisk-bugs] [Asterisk 0015833]: Transfering phone left connected

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15833 
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Reported By:                viraptor
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15833
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.26.1 
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:             2009-09-04 12:57 CDT
Last Modified:              2009-09-07 05:54 CDT
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Summary:                    Transfering phone left connected
Description: 
When doing a remote attended transfer in one of these 2 setups:

phones A,B,C --- proxy --- asterisks Z,X
when A->B call is on Z and B->C is on X, or:

phones A,B (with identity B1,B2), C --- asterisks Z,X
(A,B1 register on Z; B2,C on X)
when A->B1 call is on Z and B2->C is on X

In both scenarios Z and X are friends with no authentication needed.

The B phone doesn't get properly disconnected. asterisks invite/replace
each other properly and the audio channel is ok. B itself drops one of the
calls. But Z is not disconnecting B's call at all. You can replicate that
scenario with minimalistic dialplan - _X.,Dial(SIP/${EXTEN}) in default on
both sides.

If you do the same transfer, but on a single asterisk (local attended
transfer), then the transferring phone will drop one of the call legs
itself (like above) and asterisk will additionally drop the second one.

Tested with Snom 1XX, 3XX and GXP phones - problem always appears and the
call is never dropped if the transfer is remote.
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 (0110273) viraptor (reporter) - 2009-09-07 05:54
 https://issues.asterisk.org/view.php?id=15833#c110273 
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I believe this problem is the same as issue#7784. Using the patch included
there (applied to 1.6.1.6) resolved the problem for me (as in - the
transferring party hangs up - not sure if there are any pvt or channel
leaks)

I'll attach my sip debug logs (asked for in issue#7784). 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-07 05:54 viraptor       Note Added: 0110273                          
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