[SR-Users] Replacing Asterisk with Kamailio

Valter Nogueira valter at fastway.com.br
Tue Sep 13 18:27:17 CEST 2016


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
>>
>>
>>
>>
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>>
>>
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