Hello!
we need some help with our dialplan. we installed one asterisk (in a servercenter) connected to a carrier SIP trunk, no sip phones on the system, no special features. we connected some different SIP telephone systems (such as Cisco, other asterisk PBXen, alcatel)
asterisk should only be used as a gateway, should not be play “invalid number” or busy tones …
how can i do that?
example:
Cisco telephones system make a call through SIP to the asterisk. asterisk passes the call to the SIP carrier (already works fine).
now the destination is busy or the number is invalid (…),
asterisk should pass the original hangup code to the cisco machine.
is it all? can’t believe it 
exten => _0.,1,Dial(SIP/${EXTEN},,tToR)
exten => _0.,n,Hangup(${HANGUPCAUSE})
thanks for any help!
regards
philip