[Asterisk-Users] Call Routing based on number dialed (using SIP)

[Geoff Manning]

Is it possible to route calls based on the number called when the inbound
call is SIP based?

Here is what we are trying to do:

1) Someone dials one of the companies 5 long standing, published phone
numbers which have been forwarded to ONE Voip telephone number by the telco.

2) The SER server where that Voip number terminates is passing it to our
Asterisk server

3) Is there a way to determine what the original number dialed was?

We want to avoid needing a Voip number for every forwarded number.


Thanks in advance.