Hi Henning,
thanks for your tip.
I just checked it and I am sure it will be valuable.
Atenciosamente,
2018-11-13 19:04 GMT-02:00 Henning Westerholt hw@kamailio.org:
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