[asterisk-users] Complex Dialplan Help Needed

[Jason Aarons (US)]

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Monday, July 12, 2010 6:10 PM
To: asterisk-users at 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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