> I may be wrong here, but I think you can only register once. The last > registration received will overwrite the first one. You will need to > specify a second entry and register that one separately. This is the > same reason you cannot register two devices to the same extension.
Yes, that's very likely what is happening. The provider is seeing two SIP registrations arrive, for the same provider account, from the same peer at the same IP address. It is very likely that the second registration is (by design) replacing the first. Then, whenever someone dials a DID associated with this provider account, the provider is routing the call based on the information in the most current registration... it's either going to the context and extension specified in that registration (if their is one) or to the "s" extension for the relevant context. (Some providers do allow multiple registration for a given account, and will INVITE all of them when an incoming call arrives, but (if I recall correctly) the registrations have to come from different IP addresses (and perhaps different peers) in order to be recognized as being distinct.) There are probably several ways around this: (1) Use two different provider accounts, and associate each DID with a different account. Use two "register" statements, one per account, and specify different routing extensions on these. (2) Use a provider which will let you register once, and will "pass through" the DID number which was dialed as the target extension. (3) Use a provider which will let you set up your DIDs for hardwired-IP-address routing (i.e. no "register" being required) and who passes through the DID as the extension to be called. I recently set up an account with Vitelity, and they support option (3). I simply entered the public IP address of my SIP server for the routing, and everything works correctly... the incoming INVITE requests say "sip:MYDID@MYIPADDRESS". Asterisk then uses "MYDID" as the desired extension in my dialplan, and routes the call appropriately. I'd suggest that the OP ask the current SIP provider whether they handle (2) i.e. whether it's possible for different DIDs associated with a single account to have different information in the INVITE requests sent to the registered client. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users