I think you need to ask your SIP provider about Redirecting Header, ask what 
they support and how-to.

I work more with Cisco CallManager and SIP Rediversion Header is new in 
CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco 
Mobility/Single Number Reach, providers usually won't accept a Calling Party 
Number that isn't in your range, some will.

http://www.voip-info.org/wiki/view/RDNIS


-----Original Message-----
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Monday, July 12, 2010 6:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Complex Dialplan Help Needed

Hello all,

I have a project which requires me to rout calls from ten blocks of sequential 
numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in 
from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box 
through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant 
for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers 
from the blocks of numbers from the SS7 gateway, not the CLI of the originating 
caller.

The Asterisk is behind a firewall with NAT setup. The traffic is one way only. 
Calls going to the switch goes to Asterisk, Asterisk accepts the call, looks at 
the CLI from the line (not the caller), routs the call to its assigned outside 
number through the primary trunk. If primary trunk is unavailable, trixbox will 
then rout the call to the spare trunks on the list.

Hope anyone who has setup this before could give me some good tips on how to 
set this up.

Geoffrey




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