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 <[EMAIL PROTECTED]> 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 > _______________________________________________ --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