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:
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
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>>
>>
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