Am Freitag, 9. November 2018, 21:25:15 CET schrieb 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@customer_ip_address*
>
> In Kamailio, I have to define the account's domain like *kamctl add
> ACCOUNT1@mydomain.com <ACCOUNT1@mydomain.com> password. *Kamailio just
> accepts a REGISTER/INVITE from *ACCOUNT1@mydomain.com
> <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?

Hello Valter,

did you already looked into this tutorials? They are for a bit older version
of Kamailio and asterisk, but should give you ideas about the direction.

https://kb.asipto.com/asterisk:index

Best regards,

Henning

--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services
Kamailio security assessment - https://skalatan.de/de/assessment