[asterisk-bugs] [Asterisk 0017284]: SIP attended transfer broken

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17284 
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Reported By:                dvossel
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   17284
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   high
Status:                     assigned
Asterisk Version:           SVN 
JIRA:                       SWP-1407 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 260805 
Request Review:              
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Date Submitted:             2010-05-04 11:08 CDT
Last Modified:              2010-05-04 12:22 CDT
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Summary:                    SIP attended transfer broken
Description: 
SIP attended transfers are broken in trunk.  If an attended transfer is
attempted, right as the transferer hangs up to connect the two calls all
the calls are terminated.  It doesn't matter if it is a semi-attended
transfer or not.

So A calls B.  A transfers B to C.  right as A hangs up to connect B and C
(regardless if C has picked up yet or not) all the calls terminate. This is
easy to reproduce.



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 (0121345) davidw (reporter) - 2010-05-04 12:22
 https://issues.asterisk.org/view.php?id=17284#c121345 
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If you hangup in a SIP attended transfer, Asterisk has no way of knowing
that it is a transfer.  You definitely need the SIP debugging output to
prove that the the phone converts a hangup into REFER/Replaces.

Until the transfer is completed with an explicit REFER/Replaces, it will
look like two different calls to Asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-04 12:22 davidw         Note Added: 0121345                          
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