Today, I use Asterisk as a SIP/RTP PROXY

I proxy from customers Asterisks to a VOIP provider, in a multi-homed server.

Now, I want to move to Kamailio without any rupture in customer's configuration.

As anyone can imagine I am kind of lost.

USER ACCOUNTS

In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive in FROM HEADER sip:ACCOUNT1@customer_ip_address

In Kamailio, I have to define the account's domain like kamctl add ACCOUNT1@mydomain.com password. Kamailio just accepts a REGISTER/INVITE from ACCOUNT1@mydomain.com 


SIP/RTP PROXY

In Asterisk, I just dialout to the VOIP PROVIDER like dial (SIP/VOIP_ACCOUNT/${EXTENSION})

Asterisk does all the magic (it is a B2BUA). It bridges the new call and media to the original call. Moreover, user don't know anything about how call are completed, nor how credentials are setup and soon.

In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and maybe uac. I am not sure how to setup it.


Can someone send me a clue?


Thank you,

Valter