[asterisk-bugs] [Asterisk 0013545]: Channel re-invited on destination ringing not re-invited back if ringing abandoned.

[Asterisk Bug Tracker]

The following issue has been ASSIGNED. 
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http://bugs.digium.com/view.php?id=13545 
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Reported By:                davidw
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   13545
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-23 08:25 CDT
Last Modified:              2008-12-18 11:12 CST
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Summary:                    Channel re-invited on destination ringing not
re-invited back if ringing abandoned.
Description: 
An incoming SIP call is answered by an agent and then AMI transferred to a
PSTN line on Cisco CCM.  The Cisco provides SDP on the Ringing response and
Asterisk re-invites the incoming call immediately it gets that response.

The Dial command times out and cancels the outgoing call, but at no time
does the re-invite get undone, even when the dialplan eventually
successfully returns the call to the agent.  The result is a silent call.

The un-re-invite can be forced by parking the call and then unparking it
(in this case with an AMI Originate which queues it back to an agent).

This is a big problem for us as it is important for our application that
as many calls as possible have their speech path removed from the Asterisk
system.

I am also concerned that specifying multiple destinations in the Dial
command, may not inhibit the re-invite, leading to conflicting re-invites,
in the order of the Ringing events.  However, I haven't confirmed that this
is the case.
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 (0096628) svnbot (reporter) - 2008-12-18 11:12
 http://bugs.digium.com/view.php?id=13545#c96628 
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Repository: asterisk
Revision: 165591

U   branches/1.4/main/rtp.c

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r165591 | file | 2008-12-18 11:12:03 -0600 (Thu, 18 Dec 2008) | 4 lines

Only care about a compatible codec for early bridging if we are actually
bridging to another channel. If we are not we actually want to bring the
audio back to us.
(closes issue http://bugs.digium.com/view.php?id=13545)
Reported by: davidw

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http://svn.digium.com/view/asterisk?view=rev&revision=165591 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-18 11:12 svnbot         Checkin                                      
2008-12-18 11:12 svnbot         Note Added: 0096628                          
2008-12-18 11:12 svnbot         Status                   acknowledged =>
assigned
2008-12-18 11:12 svnbot         Assigned To              putnopvut => file   
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