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

[Asterisk Bug Tracker]

The following issue has been ASSIGNED. 
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http://bugs.digium.com/view.php?id=12713 
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Reported By:                davidw
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   12713
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.20 
Regression:                 No 
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:              2009-04-03 11:47 CDT
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Summary:                    SIP Protocol Violation when REFER rejected in
sip_transfer (Cisco CCM, post answer), and Transfer application misclaims
success.
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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 (0102682) svnbot (reporter) - 2009-04-03 11:47
 http://bugs.digium.com/view.php?id=12713#c102682 
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Repository: asterisk
Revision: 186382

U   trunk/channels/chan_iax2.c
U   trunk/channels/chan_sip.c
U   trunk/include/asterisk/frame.h
U   trunk/main/channel.c

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r186382 | file | 2009-04-03 11:47:28 -0500 (Fri, 03 Apr 2009) | 11 lines

Add better support for relaying success or failure of the ast_transfer()
API call.

This API call now waits for a special frame from the underlying channel
driver to
indicate success or failure. This allows the return value to truly convey
whether
the transfer worked or not. In the case of the Transfer() dialplan
application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.

(closes issue http://bugs.digium.com/view.php?id=12713)
Reported by: davidw
Tested by: file

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http://svn.digium.com/view/asterisk?view=rev&revision=186382 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-03 11:47 svnbot         Note Added: 0102682                          
2009-04-03 11:47 svnbot         Status                   ready for testing =>
assigned
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