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
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