[SR-Users] kamailio with media server

Atux Atux atuxnull at gmail.com
Tue Mar 27 08:03:00 CEST 2018


Hi. At the moment i am trying to learn Kamailio and it is in a test lab
only. My intention is to move my PBX to Kamailio if possible and have:
-a connection with the carriers (SIP)
-Registration of the extensions (users)
-Route DIDs between the carriers and the extensions
-Offers PBX services (voicemails, announcements to the extensions)
-have as less hardware implication as possible. If possible have everything
in a single machine/vm


On Mon, Mar 26, 2018 at 11:40 PM, Mack Hendricks <ap at goflyball.com> wrote:

> Hey Atux,
>
> Can you give a little more detail on your use case?  Are you looking for
> Kamailio to:
>
> - route requests to a media server for playing announcements
> - proxy requests between your endpoints and your media server(s)
> - distribute calls to your carriers based on some logic
>
> The answer may be all three - this will help us point you in the right
> direction.
>
> *Mack Hendricks / Head of Support / dOpenSource*
> web: http://dopensource.com
> support: +888-907-2085
> dSIPRouter <http://dsiprouter.org> - GUI focused on implementing Kamailio
> to provide SIP Trunking and PBX Hosting Services
>
> On Mar 26, 2018, at 9:45 AM, Atux Atux <atuxnull at gmail.com> wrote:
>
> Hi. New to the area of Kamailio.
> i am did install in debian kamailio with rtpproxy and i have created 3
> users 1000-10002 (one for each jitsi user) they talk nice between them.
> I have followed this tutorial http://kb.asipto.com/kamailio:
> skype-like-service-in-less-than-one-hour and in less than 5 minutes i had
> my accounts registered.
> i would like to have a media server so the users could hears announcements
> and stuff. At the end i would like to have kamailio as a test lab PBX where
> i could connect my SIP trunk providers and my users to route calls.
>
>
> Is there any guide on how to setup a media server and the services, please?
>
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>
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>
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