----- "John Taylor" <j...@vetsurgeon.org.uk> wrote:
> I want to use an asterisk box to provide a voip service to a number
> of
> separate companies.
> I have a VOIP provider who I want to trunk with. As far as I can see
> it there are 2 options
> 1. Have 1 SIP trunk to one account at the provider who gives me
> multiple incoming numbers; this is less than optimal as the provider
> does not provide the DID number in the sip header; I only get the
> account number. I have the option to set "called line presentation"
> but this will stop CLID
> 2. Have multiple sip trunks to multiple accounts at the provider. Is
> this an advisable thing to do? I notice asterisk does not handle the
> incoming context correctly (all incoming calls go to the last
> incoming
> context defined in sip.conf), but I can extract the account called
> via
> the EXTEN variable.
> I would be looking at providing around 20 companies with accounts
> (all
> very small), and would prefer option (2) to enable failover to a
> number they specify.
> Thanks for any light shed
> John

Why not go with a real carrier that can send you proper DID and DNIS 
information for each call? Rather than trying to configure/code/etc around the 
problem with the ITSP, use an ITSP that does things correctly. There are many 
people here on asterisk-users that can recommend a proper ITSP. If you want 
pure business response, head over to asterisk-biz and ask there.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

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