[asterisk-bugs] [Asterisk 0018254]: Attended transfer failure

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18254 
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Reported By:                kwemheuer
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18254
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-2562 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.8 
SVN Revision (number only!): 293886 
Request Review:              
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Date Submitted:             2010-11-04 13:57 CDT
Last Modified:              2010-11-19 13:07 CST
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Summary:                    Attended transfer failure
Description: 
I set up an asterisk system (1.8 SVN revision 293886 checked out today) and
do some testing in a SIP only environment.

With three phones doing attended transfer I have some weird behaviour.

phone1 calls phone2, phone2 sets call on hold and calls phone3. Then
phone2 is doing an attended transfer. With Snom phones there seems to be no
problem. With aastra phones the call is terminated on one side only. In the
described scenario phone3 seems to be connected, whereas phone1 is idle
(see issue-att-transfer.log). From Asterisk's point of view both legs are
terminated.

It doesn't matter, who is doing the transfer. If the original call (in the
above senario phone1) is doing the transfer, there is also one party hung
up while the other seems to be connected (on the phone side).

This might be related to https://issues.asterisk.org/view.php?id=18185, but the
behaviour is different, so I make
up a new ticket. The workaround fix from
https://issues.asterisk.org/view.php?id=18185 doesn't work (the logfile
attached is from an original svn checkout, no patches applied).
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Relationships       ID      Summary
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related to          0018185 Blind transfer failure, A calls B, B tr...
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 (0128994) schmidts (manager) - 2010-11-19 13:07
 https://issues.asterisk.org/view.php?id=18254#c128994 
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what i see in the log is that directmedia in used so rtp data from phone A
to phone B doesnt pass asterisk. Maybe thats why there is something strange
going on.

Please try to turn of directmedia and tell us what happens.

thanks!

Stefan 

Issue History 
Date Modified    Username       Field                    Change               
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2010-11-19 13:07 schmidts       Note Added: 0128994                          
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