[asterisk-bugs] [Asterisk 0018745]: calls not routing to polycom handset after transfer using softkeys

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18745 
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Reported By:                vipkilla
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18745
Category:                   Resources/res_agi
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.16.1 
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:             2011-02-03 15:53 CST
Last Modified:              2011-02-04 08:42 CST
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Summary:                    calls not routing to polycom handset after transfer
using softkeys
Description: 
We are running asterisk 1.6 and AGI scripts for call flow. Because Polycom
handset handles most of transfer when using soft keys, the AGI script is
never notified of the handset hanging up after transfer is complete.
Therefore AGI script believes handset is busy (487 Busy Here is sent) until
the transferred call is terminated
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 (0131496) davidw (reporter) - 2011-02-04 08:42
 https://issues.asterisk.org/view.php?id=18745#c131496 
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I imagine that other readers will find it much easier to read if you set up
logger.conf to log to a file in sufficient detail, and then add the file as
an attachment.  Do not zip it!

However, it is not clear to me what evidence you are providing that
anything has gone wrong - the SIP trace looks correct.  It looks like a
valid Refer/replaces transfer. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-04 08:42 davidw         Note Added: 0131496                          
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