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(a)kamailio.org>rg>:
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(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just
accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com
<ACCOUNT1(a)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
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