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@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@lists.kamailio.org> On Behalf Of Atux Atux
Sent: Tuesday, March 27, 2018 12:16 PM
To: Mack Hendricks <mack@dopensource.com>; Kamailio (SER) - Users Mailing List <sr-users@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@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@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

support: +888-907-2085

dSIPRouter - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services



On Mar 26, 2018, at 9:45 AM, Atux Atux <atuxnull@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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