Indeed, this is a very good suggestion.
I can propose a little plan for the first step:
1. Learn how sip registration works
2. Configure kamailio as a sip registrar server.
3. Try to route calls between registered phones.
4. Read about in-dialog requests routing (record_route(), loose_route())
and configure calls between connected endpoint in a "right" way.
And if you want to learn mode:
5. Try to handle NAT issues at kamailio (symmetric sip / rtp relay)
Asterisk supports symmetric rtp pretty well, so if you want to route all
calls through asterisk boxes you can skip rtp relay configuration step.
We use Kamailio for registration of users and loadbalancing.
I think you can start with next steps:
1. Single installation of kamailio with registration of users, fixing
NAT problems etc.
2. After you get some experience with Kamailio - you can add Asterisk
servers.
In asterisk servers - I accept all calls which comes from Kamailio -
registration of users is purely Kamailio "job".
In Kamailio - again I accept all calls which come from any of Asterisk
servers.
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