[asterisk-users] PJ SIP realtime with Kamailio / opensips

[Chirag Desai]

Hi all,

I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.

All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.

In order to do what Matt suggested would I be correct in assuming I would
have to use the asterisk database rather than the Kamailio database? I've
compared the two and the table structures are different.

If I use the asterisk database I guess asterisk still needs to be
responsible for handling authentication, registration and writing the
contacts to the database. If I use the Kamailio database how would I dial a
local extension from asterisk if I'm using multiple domains?

For example 100 at domainA.com -> 200 at domainA.com

Or even

100 at domainA.com -> 300 at domainB.com

How would pjsip find the contact to dial?

As far as I can tell asterisk will have no idea who is registered or how to
find them (contact details). Maybe I'm over thinking something simple, or
maybe I'm not. Either way I would love your thoughts on how this could be
done.

My Kamailio is public facing, but talks to asterisk over an internal
network. Asterisk can face the internet but I'd rather not.

Thanks in advance for your help,

C
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