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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13545 
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Reported By:                davidw
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13545
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:              2008-09-23 12:39 CDT
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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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 (0092801) davidw (reporter) - 2008-09-23 12:39
 http://bugs.digium.com/view.php?id=13545#c92801 
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The problem doesn't happen if the call is actually answered (note, because
it is PSTN, there is no reverse clear and the call has to be forced down
from the dialplan):

exten => 6183,1,Answer
exten => 6183,n,Dial(SIP/988888888888 at 192.168.10.10,10,gS(5))
exten => 6183,n,Read(variable,enter-ext-of-person,,,100)
exten => 6183,n,Noop(${variable})

it also doesn't happen when there isn't an early SDP (although I think
there were more re-invites than I expected!  The g is a hangover, this call
was allowed to timeout ringing):

exten => 6183,1,Answer
exten => 6183,n,Dial(SIP/6906 at 192.168.10.10,10,g)
exten => 6183,n,Read(variable,enter-ext-of-person,,,100)
exten => 6183,n,Noop(${variable}) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-23 12:39 davidw         Note Added: 0092801                          
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