[SR-Users] Replacing Asterisk with Kamailio

[Yuriy Gorlichenko]

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.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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