[asterisk-dev] Asterisk mishandling user busy isdn releases
Ken Leland III
k3leland at monmouth.com
Tue Feb 12 08:35:26 CST 2008
This issue has been solved in the dialplan:)
Thanks to johan's post in the bug reort:
http://bugs.digium.com/view.php?id=11918
----------------------------------------------------------------------
johan - 02-11-08 17:48
----------------------------------------------------------------------
Could you try:
Dial(SIP/1234,,g)
.. and then do the logic to play tones ...
>From the help: "g: When the called party hangs up, exit to execute more
commands in the current context."
Or would you consider that to be a work around?
----------------------------------------------------------------------
> > Hello All,
> > I have found that Asterisk does not play a busy tone when it receives a
> > USER BUSY ISDN RELEASE messages following an ISDN CONNECT message.
> >
> > Details:
> > 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.
> >
> > Questions:
> > Is there a way to configure this behavior in the dial plan?
> >
> > If not does anyone think that adding the following logic to Asterisk
> > would be a useful patch?
> >
> > If a zap channel is bridged with a sip channel, and the zap channel
> > receives a user busy RELEASE, play a busy tone inband on the sip
channel.
> >
> > Cheers,
-
--
Ken W. Leland III
Engineering
k3leland at monmouth.com
Monmouth Telecom
10 Drs. James Parker Blvd., Suite 110
Red Bank, NJ 07701
732-704-1000 X283
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