> My dialplan is for the outgoing SIP call is:
>
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
>I can't help with presenting busy to the SIP devices, but if you have the above on any sort of PSTN gateway you are going to annoy the PSTN users - as if the number selected is busy or otherwise unavailable you will still 'Answer' the PSTN call, causing the person calling to pay whatever call establishment charges/minimum charges appropriate to their tariff. Linus _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
