[SR-Users] from Asterisk to Kamailio

[Valter Nogueira]

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 at customer_ip_address*

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