In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip.
You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Chris _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users