Geoff Manning wrote:
Is this even possible or am I better off getting a voip number for each of
the existing numbers I want to forward.
Thanks!
-----Original Message-----
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 4:53 PM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Call Routing based on number dialed (using
SIP)
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.
_______________________________________________
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
_______________________________________________
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
You can use the sipgetheader() application. If you pass a call to
asterisk, the field "To" in the SIP header stay as originally dialed.
So, with
sipgetheader(or_To=To)
Cut(or_To,or_To,:,2)
Cut(or_To,or_To,@,1)
in your dialplan, you can get the original dialed number.
with the cut function you can cut the "sip:" and the "@domain.asd"
substrings.
Mirko
_______________________________________________
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
