[SR-Users] kamailio with media server

[Atux Atux]

Thanks a lot. My first steps are towards to make Kamailio have the
following:
-register to a provider, eg user:atux, passwd: null, domain:
sip.providerA.com. *Pending*.
-create users. *Done* already and have the users talk to each other by
using Jitsi.
-route DIDs to corresponding users. *Pending*

is there a guide for each one of them?

On Tue, Mar 27, 2018 at 9:03 PM, Michael Young <myoung at redmonsters.net>
wrote:

> Atux,
>
>
>
> Kamailio is not a PBX and will not replace your PBX. It can do some of the
> things you might expect from a PBX, but what you really want is Kamailio
> and a PBX integrated together. You may wish to review a presentation such
> as this one:
>
>
>
> https://www.slideshare.net/fredposner/using-asterisk-and-
> kamailio-for-reliable-scalable-and-secure-communication-solutions
>
>
>
> or a how-to such as this one:
>
>
>
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-
> asterisk-11.3.0-astdb
>
>
>
> Michael
>
>
>
> *From:* sr-users <sr-users-bounces at lists.kamailio.org> *On Behalf Of *Atux
> Atux
> *Sent:* Tuesday, March 27, 2018 12:16 PM
> *To:* Mack Hendricks <mack at dopensource.com>; Kamailio (SER) - Users
> Mailing List <sr-users at lists.kamailio.org>
> *Subject:* Re: [SR-Users] kamailio with media server
>
>
>
> Any suggestions, please? At least how do i register to a sip trunk and
> route the DIDs to extensions?
>
>
>
> On Tue, Mar 27, 2018 at 9:03 AM, Atux Atux <atuxnull at gmail.com> wrote:
>
> 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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