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" <val...@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 > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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