[asterisk-bugs] [Asterisk 0011918]: Asterisk not playing busy after media bridge

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11918 
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Reported By:                remiq
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11918
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
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:             02-04-2008 08:08 CST
Last Modified:              02-08-2008 09:42 CST
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Summary:                    Asterisk not playing busy after media bridge
Description: 
We are connecting our Asterisk PBX to a Lucent telephone switch via ISDN
PRIs.  The telephone switch provides an authorization code feature where
every call originated from asterisk is immediately answered by the
telephone switch, and prompted for an authorization code, before completing
the call.  The sip phone registered with Asterisk dials a 10 digit number,
hears the prompt, and dials a 4-digit code.  The telephone switch
authenticates the code and attempts to complete the call.  In the event
that the far end is available the telephone switch passes the ringing tones
inband over the pri.  In the event that the far end is busy, the telephone
switch sends an ISDN RELEASE message with the cause: USERBUSY.  Asterisk
receives the RELEASE message and sends a BYE message to the sip phone and a
RELEASE COMPLETE back to the switch. This does NOT result in the user
hearing a busy signal.  Instead, the user hears a click and dead air as if
they were hung up on.  Since there is already a media bridge, once the
Asterisk receives the USERBUSY, Asterisk should play an inband busy tone to
the sip channel.  
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---------------------------------------------------------------------- 
 kenlee - 02-08-08 09:42  
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I have been following this issue on the mailing list.  Here is a list of
things we tried to resolve this issue through configuration:
1) setting progressinband=yes in sip.conf
2) using Progress before Dial() in the dialplan

Neither of these work, I suspect because they are both used to send "early
audio" before the call setup is completed.  I also think that by setting
this you are simply allowing Asterisk to pass early audio between the two
channels as opposed to Asterisk acutally inserting audio.  The problem we
are having requires Asterisk to replace a BYE message with a busy tone.

3)
_X.,1,Dial()
_X.,2,Busy()
This doesn't work because the ISDN CONNECT causes Dial to behave as if the
call was successful.
4)
_X.,1,Dial()
h,1,Playtones(busy)
This doesn't work because (i presume) that the audio path has already been
cut down once the hangup extension is given control in the dial plan. 

Issue History 
Date Modified   Username       Field                    Change               
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02-08-08 09:42  kenlee         Note Added: 0081916                          
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