[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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