[asterisk-bugs] [Asterisk 0007784]: Attended SIP Transfer Call Teardown Issue

[Asterisk Bug Tracker]

The following issue has been set as DUPLICATE OF issue 0015833. 
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https://issues.asterisk.org/view.php?id=7784 
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Reported By:                jcmoore
Assigned To:                
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Project:                    Asterisk
Issue ID:                   7784
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:            SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 43322 
Request Review:              
Resolution:                 unable to reproduce
Fixed in Version:           
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Date Submitted:             2006-08-22 20:45 CDT
Last Modified:              2009-09-16 10:47 CDT
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Summary:                    Attended SIP Transfer Call Teardown Issue
Description: 
When attempting an attended SIP transfer which spans multilple Asterisk
servers, ie. INVITE w/ Replaces, the call on one of the phones does not
receive a BYE.  

For example, in a call scenario from A->B->C, where A calls B, B then
attended transfers the call to C, the call on B is present even after the
call between A and C has been bridged.  The audio is correctly bridged
between A and C, there is no audio on B.

The issue is that instead of deferring the BYE, as is currently being done
in handle_invite_replaces(), the BYE needs to proceed as normal so the call
on B will hangup while preserving the sip_pvt structure so that the INVITE
w/ Replaces can have a basis from which to create the new outbound INVITE.
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Relationships       ID      Summary
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duplicate of        0015833 Transfering phone left connected
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Issue History 
Date Modified    Username       Field                    Change               
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2009-09-16 10:47 lmadsen        Relationship replaced    duplicate of 0015833
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