[SR-Users] Replacing Asterisk with Kamailio

anfecora anfecora at gmail.com
Tue Sep 13 18:07:35 CEST 2016

Valter i wouldnt take fully asterisk from the picture you can use it to
handle transcoding for example and still a b2b support.

Perhaps you can look for asterisk kamailio setup in the same server.

On Sep 13, 2016 8:42 AM, "Valter Nogueira" <valter at fastway.com.br> wrote:

> I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is
> not a SIP Proxy at all.
> Customer registers in a SIP account, sends the invite and thru de context
> Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy, since
> customer can't route directly to the SIP Trunk (altough it has a valida
> address, it don't have a public route allowed to it).
> I need limit customer concurrent calls, mangle some dial-in/dial-out
> numbers, keep track of ongoing call, control SIP dialog, retransmit correct
> hang-up causes and do media proxy (no transconding at all)
> After reading about Kamailio and Opensips, and due to the Kamailio Admin
> Book, I decided to go with Kamailio.
> Well, I understand that I have to use some kamailio modules, like auth,
> dialplan, rtpproxy and db_mysql.
> What make me stuck is how does everything fit together in kamailio.cfg and
> how do I get ongoing calls and CDR's?
> Can anyone point me a direction?
> Thanks
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