[SR-Users] Replacing Asterisk with Kamailio

Thu Sep 15 08:21:18 CEST 2016

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
OpenSER
Flavio E. Goncalves
April 2008

which 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 at fastway.com.br>
wrote:

> Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
>
> I prefer installing it from sources.
>
> What I get until now, is that kamailio.cfg is more a program than a
> configuration file at all.
>
> I really appreciate the links and I will try to understand them.
>
> Thank you
>
> Valter
>
>
> 2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko <ovoshlook at 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-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.sabery at 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 at fastway.com.br>
>>> wrote:
>>>
>>>> We won't need transcoding.
>>>>
>>>> Is b2b b2bua?
>>>>
>>>> Em 13 de set de 2016 13:07, "anfecora" <anfecora at 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 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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>>> list
>>>>>> sr-users at lists.sip-router.org
>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>>>
>>>>> _______________________________________________
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>>>>> sr-users at lists.sip-router.org
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>>>>>
>>>>>
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>>>>
>>>>
>>>
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>>>
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>>
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>> sr-users at lists.sip-router.org
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>>
>>
>
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