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
accou...@mydomain.com <accou...@mydomain.com> password. *Kamailio just
accepts a REGISTER/INVITE from *accou...@mydomain.com
<accou...@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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