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

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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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:                     closed
Asterisk Version:           1.4.26.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-09-04 12:57 CDT
Last Modified:              2010-05-17 10:45 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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Relationships       ID      Summary
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has duplicate       0007784 Attended SIP Transfer Call Teardown Issue
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 (0121983) lmadsen (administrator) - 2010-05-17 10:45
 https://issues.asterisk.org/view.php?id=15833#c121983 
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Suspending this issue. If this continues to be a problem please open a new
issue reported against the latest release in the branch you're using in
production (i.e. latest 1.4.x, or 1.6.2.x) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-17 10:45 lmadsen        Note Added: 0121983                          
2010-05-17 10:45 lmadsen        Status                   feedback => closed  
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