[Asterisk-Users] outgoing Sip-call problem URI and Phone-number

Roland.Knoerl at student.fh-nuernberg.de Roland.Knoerl at student.fh-nuernberg.de
Mon Jul 5 13:14:55 MST 2004


Hi there,
my * is running very well in standard mode, but
I would like to implement some more features.
that´s why I hope that someone could help me !

For example:

I have Asterisk working behind a Cisco Gateway.
This is working well, but I would like to
have the ID translated, so that the right telephone-number is transfered.

e.g.:  SIP-user roland.knoerl should have the outgoing number 772 .
I´m sure that it would be easier to change the username into 772, but 
I have to implement it that way.


Maybe it has something to do with the line:
exten => _0.,1,Dial(SIP/${EXTEN:0}@mycisco,30,r)
Put something in front of the ${EXTEN:0}  ???

For incoming calls, I already did a few lines that makes the user roland.knoerl
available by dialing 772.

Thanks in advance !


Roland Knoerl  , Nuremberg , Germany



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