[asterisk-bugs] [Asterisk 0019372]: Attended transfer - transfering phone left connected

[Asterisk Bug Tracker]

The following issue has been set as RELATED TO issue 0007784. 
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https://issues.asterisk.org/view.php?id=19372 
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Reported By:                jamicque
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19372
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.4.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-05-26 09:52 CDT
Last Modified:              2011-05-27 10:45 CDT
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Summary:                    Attended transfer - transfering phone left connected
Description: 
The issue 0015833 still exists in newest version of Asterisk.

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.



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Relationships       ID      Summary
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related to          0007784 Attended SIP Transfer Call Teardown Issue
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Issue History 
Date Modified    Username       Field                    Change               
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2011-05-27 10:45 lmadsen        Relationship added       related to 0007784  
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