Incoming SIP URI Calls That Work
Having done some experimentation, here's what works with incoming SIP URI calls to your dSIPRouter's FQDN...
First, if you're using Incredible PBX as your pass through Asterisk server, you still must WhiteList the IP address of the SIP URI caller even if the calls are directed to your Kamailio server's FQDN. Otherwise, incoming calls will ring with no answer ever. If you open UDP 5060 on your Incredible PBX server, this would change so that calls enumerated below would work for everybody, BUT your server would be exposed to the entire Internet for anonymous SIP calling and attempts to hack into your server's extensions and trunks. Not recommended! We had hoped that the dSIPRouter setup would allow anonymous SIP calls to the FQDN of your Kamailio server without having to open up UDP 5060 on your Asterisk server, but that is NOT the case.
Second, the from-trunk context setting in your Kamailio trunk setup on Incredible PBX means (1) SIP URI calls to all extensions will work and (2) SIP URI calls to DIDs enumerated in your inbound routes will work. All other calls will be processed by the Default Inbound Route which we recommend setting to Terminate Call: Play no service message. Since your firewall is blocking anonymous SIP URI calls, the other option is to change the from-trunk setting to from-internal which would mean that those whitelisted in IPtables could actually make outbound calls through your PBX using SIP URIs and your outbound routing rules, e.g. 18005551212@FQDN-of-your-Kamailio-server. Of course, if your firewall ever goes down, U R screwed. The whole world could make calls on your nickel. Hence, not recommended.