Re: Asterisk mishandling user busy isdn releases

I tried using the Progress command in the dialplan.
I also tried setting progressinband=yes in sip.conf.
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 actually
inserting 
audio.  The problem I am having requires Asterisk to replace
a BYE 
message with a busy tone.

I have updated Bug Report 0011918 with all of our tests so
far.

Cheers
-- 
Ken W. Leland  III
Engineering
k3lelandmonmouth.com
Monmouth Telecom

> The below approach does not work because Asterisk
treats the call as an 
> answered call not a busy call.
> It seems as thought once Asterisk has received an ISDN
CONNECT message 
> the call is bridged and control will not be returned to
the dialplan 
> until the special hangup extension at which point it is
already too late 
> as the audio bridge is turned down.
> 
> I have opened a bug report on this:
> http://bugs.
digium.com/view.php?id=11918
> 
> Quoted from original post:
> "Asterisk does not play a busy tone when it
receives a USER BUSY ISDN 
> RELEASE messages following an ISDN CONNECT
message"
> > I would try this:
> > 
> > exten => 123,1,Dial(.....)
> > exten =>
123,2,Gotoif($["$" ==
"BUSY"]?3:4)
> > exten => 123,3,Busy()
> > exten => 123,4,Hangup()

Try this instead (not tested, but I remember using the same
approach):

exten => 123,1,Progress
exten => 123,2,Dial(.....)

and then nothing further. If Asterisk falls off the dialplan
after a Dial,
it should return the status of the dial back to the calling
channel. The
preceding call to Progress instructs Asterisk to play the
appropriate
progress tones in-band as required.

Cheers
Tony
-- 
Tony Mountifield
Work: tonysoftins.co.uk - http://www.softins.co.uk

Play: tonymountifield.org - http://tony.mountifield.o
rg


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev