it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working. Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help Read this http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime-integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration and this (dont see that it is old.Logis is the same) https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio-3-1-realtime-integration-tutorial/ All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself. 2016-09-13 21:11 GMT+03:00 Gholamreza Sabery <gr.sab...@gmail.com>: > For testing purpose you can use example config file it is a very good > place to start. Also if you want automatic installation and deployment you > can use this project: > > https://github.com/ghrst/Kamailio-HA > > > On Tue, Sep 13, 2016 at 8:57 PM, Valter Nogueira <val...@fastway.com.br> > wrote: > >> 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 >>> >>> >> _______________________________________________ >> 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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