Trying to make SIP to SIP calls through Asterisk networks with anonymous connection

I am trying to setup SIP to SIP calling between Asterisk managed networks. I want to make it so that people can call IP:[email protected] and they connect to my Asterisk and get my external IVR then they can dial my extension or navigate extensions just like they would if they had called using a PSTN line. I also want to call other people using my Asterisk and dialing an external SIP like so SIP:[email protected] I want out going SIP calls to me managed by my Asterisk so I can transfer them to other people in my office or conference or use any of the other great features that Asterisk provides. I do not want to go SIP direct to SIP, I want to go SIP to Asterisk to Asterisk to SIP and connect to the far end Asterisk without requiring me to register my Asterisk server with the far end Asterisk server.

For testing I have setup two servers running Asterisk. Both are on the Internet with static IP addresses and behind firewalls. The firewalls are configured to allow TCP & UDP ports 5060 to 5082 and 10001 to 20000 to connect directly to the Asterisk servers. This allows SIP and RTP connections from the outside. I have tested with Twinkle (a Linux softphone) and can connect to a registered account with NAT from external IPs. I have also set the Asterisk servers to allow incoming anonymous SIP calls to connect to the from-external. When I try to dial SIP:[email protected]_url Asterisk tries to dial the some_extension on my local network not the other network. I reconnect to the running asterisk using -r and watch when I dial and it does not report the @other_url only the some_extension.

I am not having much luck finding the documentation I need. Can someone point me to a How-To on doing this?

I am using Asterisk 1.2.18 with FreePBX 2.2.0.