[asterisk-bugs] [Asterisk 0012265]: SIP caller hanging up before answer does not stop Dial

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12265 
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Reported By:                kodomo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12265
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.18 
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:             03-20-2008 11:38 CDT
Last Modified:              03-28-2008 08:31 CDT
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Summary:                    SIP caller hanging up before answer does not stop
Dial
Description: 
If a SIP user hangs up before an answer, the Dial is not interrupted.
Instead, the following WARNING is issued:

[Mar 20 17:33:50] WARNING[20967]: app_dial.c:676 wait_for_answer: Unable
to forward voice frame

and the receiver side continues ringing (thereby blocking the channel and
causing costs, if the supposed call is picked up, eventually)



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---------------------------------------------------------------------- 
 kodomo - 03-28-08 08:31  
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Update:
I did some more testing today and found something interesting (maybe I was
wrong and it actually _is_ francesco_r's bug:
I tested three setups:
reroute: GXP2000<--SIP-->*1<--IAX2-->*2<--IAX2-->*1<--Srx-->ExternalPhone
(calls rerouted - i.e. looped - via a second system, currently not doing
anything, but logging)
When cancelling the call, *2 didn't get a notice about that cancelling
either.

direct: GXP2000<--SIP-->*1<--Srx-->ExternalPhone
Problem, as previously described.

private: GXP2000<--SIP-->*3<--IAX2-->*1<--Srx-->ExternalPhone
(calls placed via my own asterisk installation and routed over IAX2 to the
system in question)
No problems... calls were being hung up instantly...
[I previously assumed that *3 and *1 have the same * version, which is not
the case, so the fact that it worked there does not imply that it's not
happening with out:IAX2 - sorry, my fault - was too tired when I tested it
;) ]

So it seems, as if the point, where things are turning bad, is really when
the call is initiated via SIP and then is bridged to another channel type
(which would indicate the problem to be located in chan_sip, as I initially
suspected) 

Issue History 
Date Modified   Username       Field                    Change               
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03-28-08 08:31  kodomo         Note Added: 0084689                          
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