[SR-Users] Replacing Asterisk with Kamailio

Thu Sep 15 17:24:04 CEST 2016

When a call cames from my customer, it will be autheticated by auth module.

After authetication, I need to proxy it to the sip provider, but it expects
my credentials, not my customer's one.

It not seems to me, correct if I am wrong, a proxy operation, is it?

Since I am a programmer, I expected to have little problem to undertand
kamailio.cfg, but when I read some code, I wonder how could someone think
in this sip sequence and how could do I know that I am doing right?

Atenciosamente,

2016-09-15 3:04 GMT-03:00 Yuriy Gorlichenko <ovoshlook at gmail.com>:

> Yes. Thats correct
> Kamailio.cfg is a script. It is a sublnguage. You must think as programmer
> for using it
>
> Input data is sip method. Thist script using it for handling making
> changes if it need and make him to know where it must be proxyed
> That is a main idea.
>
> Actually in asteirks for example it is a same btut asteirsk works at the
> extensions.conf/ael/lua with invite method and
> Taking Leg A and creates Leg B
>
> kamailio is different. It is just proxies same method through itself. and
> working with every method like invite and his replies, and other methods.
>
> You must think not about dialplan there but about method handling. And it
> need to very good know SIP RFC for understanding what is going on and why.
>
> I suppose everyone who uses kamailio thought before thant knows SIP. But
> it was wrong.
>
> 2016-09-15 5:59 GMT+03:00 Valter Nogueira <valter at fastway.com.br>:
>
>> 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
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>> 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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