[asterisk-dev] Asterisk mishandling user busy isdn releases

[Klaus Darilion]

I would try this:

exten => 123,1,Dial(.....)
exten => 123,2,Gotoif($["${DIALSTATUS}" == "BUSY"]?3:4)
exten => 123,3,Busy()
exten => 123,4,Hangup()

regards
klaus


Ken Leland III schrieb:
>  > > 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?
> 
>  > You can check DIALSTATUS after Dial() and play Busy() if the DIALSTATUS
>  > is busy.
>  > klaus
> Thanks klaus,  I gave this a try, with no luck, using the following 
> hangup extensions:
> 
> exten => h,1,Gotoif($[${HANGUPCAUSE} = 17]?BUSY:HANGUP)
> exten => h,n(BUSY),Playtones(busy)
> exten => h,n(HANGUP),Hangup
> 
> Here is some partial console output + SIP debug:
> 
> -- Executing [h at test:1] GotoIf("SIP/testuser-10685f60", "1?BUSY:HANGUP") 
> in new stack
> -- Goto (test,h,2)
> -- Executing [h at test:2] PlayTones("SIP/testuser-10685f60", "busy") in 
> new stack
> -- Executing [h at test:3] Hangup("SIP/testuser-10685f60", "") in new stack
> -- Asterisk sends SIP BYE message.
> 
> It seems whatever attempts i made at adding audio in the special 'h' 
> hangup extension would fail.  As you can see the PlayTones("busy") is 
> happening before the SIP BYE message but it still doesn't make it to the 
> audio stream.  Does the Dial cmd tear down the rtp session before 
> returning?  Did you mean for me to check the DIALSTATUS in the special 
> hangup extension or somewhere else?
> 
>  > >
>  > > 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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