----- "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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
