Valter, you can buy the the draft of Kamailio Admin Book. It will really help you understand Kamilio how it works , how it porcess calls and what each part of the cfg means.yes it is true that you need to know about sip and how request correlate with responses etc because it does correlate the way it is handled.kamailio.cfg is a very simple configuration file, you define the modules that you want to use and then how to process the calls, it is not as explicit as asterisk because asterisk is a different tool, kamailio is a full proxy/sip router while asterisk is only a registrar.you can also try to download/ buy the book Building Telephony Systems with OpenSERFlavio E. GoncalvesApril 2008which is a lso very explicit on how is the cfg divided.also try to understand the pseudo variables which will help you pin point each header within the configuration for example.hope that helps, it is not an easy dry cut learning kamailio neither is to understand the rfc3261, but you must understand both to master the technology.
On Wed, Sep 14, 2016 at 10:59 PM, Valter Nogueira <valter@fastway.com.br> wrote:Thank youI really appreciate the links and I will try to understand them.What I get until now, is that kamailio.cfg is more a program than a configuration file at all.Yes, you are absolutely right: I don't understand (yet) how Kamailio works!I prefer installing it from sources.Valter2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko <ovoshlook@gmail.com>: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-asteri sk-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-tuto rial/ 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.sabery@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 <valter@fastway.com.br> wrote:We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" <anfecora@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" <valter@fastway.com.br> wrote:ThanksCan anyone point me a direction?What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.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)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).
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