[asterisk-bugs] [Asterisk 0012713]: [patch] SIP Protocol Violation when REFER rejected in sip_transfer (Cisco CCM, post answer), and Transfer application misclaims

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=12713 
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Reported By:                davidw
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12713
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.20 
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:             2008-05-23 12:59 CDT
Last Modified:              2009-12-02 09:38 CST
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Summary:                    [patch] SIP Protocol Violation when REFER rejected
in sip_transfer (Cisco CCM, post answer), and Transfer application misclaims 
Description: 
A call from Cisco CCM was answered and then transferred back to a different
extension on the Cisco.

Asterisk first reported a successful TRANSFERSTATUS, then tried to use a
REFER method.  This was rejected as unsupported.  Asterisk responded with
BYE, which the Cisco accepted.  However it continued to try to send REFERs,
which the Cisco, of course, rejected, because the session reference had
been deleted by the BYE!
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Relationships       ID      Summary
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related to          0010052 NOTIFY race condition when state change...
related to          0011848 Incorrect dialog matching and requests ...
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 (0114516) davidw (reporter) - 2009-12-02 09:38
 https://issues.asterisk.org/view.php?id=12713#c114516 
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487 Session Cancelled needs handling, but I'm not quite sure how yet.  If
you cancel the transferred call on a Polycom
(PolycomSoundPointIP-SPIP_330-UA/2.2.2.0084), before answer, it will keep
the original SIP session up, but you cannot access it from the Polycom user
interface, and it won't send RTP, even when connected.

One could argue that this is a success, because the transfer didn't fail
because of a party B problem, but I wonder if there are any phones that
don't close the call, and do allow it to be retrieved from the user
interface.

I wonder if it requires a third state. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-02 09:38 davidw         Note Added: 0114516                          
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