[Asterisk-Users] PRI problem

[C F]

Looks like you got a configuration issue, you should test for the
${DIALSTATUS} variable and set the signalling to the phones based on
that.

You can do:
exten => _X.,1,Dial(Zap/g1/${EXTEN})
exten => _X.,2,Goto,s-${DIALSTATUS},1)
exten => s-CANCEL,1,Playtones(congestion)
exten => s-CANCEL,2,Congestion
exten => s-NOANSWER,1,Goto(s-CANCEL,1)
exten => s-BUSY,1,Playtones(busy)
exten => s-BUSY,2,Busy
exten => s-CONGESTION,1,Goto(s-CANCEL,1)
exten => s-CHANUNAVAIL,1,Goto(s-CANCEL,1)

Check this:
http://www.voip-info.org/wiki-asterisk+cmd+dial
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS

On 1/6/06, Joseph Rothstein <jrothstein at comcentrixs.com> wrote:
> We have an Asterisk server with a single Digium E1. Everzthign works as it
> should except for one minor issue.
>
> When we place a call to a number that is busy, Asterisk does not seem to
> properly send the busy signal back to the SIP phones. There is no indication
> on the phone of anything at all, just silence, like the call did not go
> through. As you might imagine, this can be quite frustrating. The only
> indication is that we see a 403 Forbidden SIP message on softphones.
>
> I would appreciate any ideas of how to solve this issue. I have yet to do
> extensive PRI debugging to see what the Telecom provider sends back, so I am
> assuming that it correct signaling.
>
> Regards,
> Joe
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>