We won't need transcoding. Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" <anfec...@gmail.com> escreveu: > 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 >> >> > _______________________________________________ > 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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