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

Fri Apr 3 11:48:14 CDT 2009

A NOTE has been added to this issue. 
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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:                     closed
Asterisk Version:           1.4.20 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2008-05-23 12:59 CDT
Last Modified:              2009-04-03 11:48 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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 (0102683) svnbot (reporter) - 2009-04-03 11:48
 http://bugs.digium.com/view.php?id=12713#c102683 
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Repository: asterisk
Revision: 186383

_U  branches/1.6.2/

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r186383 | file | 2009-04-03 11:48:13 -0500 (Fri, 03 Apr 2009) | 17 lines

Blocked revisions 186382 via svnmerge

........
  r186382 | file | 2009-04-03 13:47:27 -0300 (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=186383 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-03 11:48 svnbot         Checkin                                      
2009-04-03 11:48 svnbot         Note Added: 0102683                          
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