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 <h...@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
> > 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?
>
> 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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