15 Sep
2016
15 Sep
'16
11:24 a.m.
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(a)gmail.com>om>:
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(a)fastway.com.br>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(a)gmail.com>om>:
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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(a)gmail.com>om>:
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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(a)fastway.com.br
> wrote:
> We won't need transcoding.
>
> Is b2b b2bua?
>
> Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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
>>>
>>>
>>>
>>>
>>> _______________________________________________
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>>> list
>>> sr-users(a)lists.sip-router.org
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>>>
>>>
>> _______________________________________________
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>> list
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>>
>>
> _______________________________________________
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