[asterisk-bugs] [Asterisk 0016701]: Attended transfer broken in 1.6.1.13

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16701 
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Reported By:                mtryfoss
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16701
Category:                   PBX/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.13 
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:             2010-01-26 01:08 CST
Last Modified:              2010-01-26 07:58 CST
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Summary:                    Attended transfer broken in 1.6.1.13
Description: 
Error description: 

1. A calls B
2. B picks up and talks to A
3. B does attended transfer to C
4. C picks up, but B still hears ringing
5. A and B are connected again (AT timeout exceeded on console) 

https://issues.asterisk.org/view.php?id=16513 is a similar problem, but I
was told to file a different ticket since this is version 1.6.1.
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 (0117185) lmadsen (administrator) - 2010-01-26 07:58
 https://issues.asterisk.org/view.php?id=16701#c117185 
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1) Is this a built-in transfer via features.conf, or is this a SIP
transfer?

2) you need to provide console output showing the problem, along with
debugging enabled, and the SIP console trace 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-26 07:58 lmadsen        Note Added: 0117185                          
2010-01-26 07:58 lmadsen        Status                   new => feedback     
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