Thanks a lot. My first steps are towards to make Kamailio have the
following:
-register to a provider, eg user:atux, passwd: null, domain:
. *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(a)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(a)lists.kamailio.org> *On Behalf Of *Atux
Atux
*Sent:* Tuesday, March 27, 2018 12:16 PM
*To:* Mack Hendricks <mack(a)dopensource.com>om>; Kamailio (SER) - Users
Mailing List <sr-users(a)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(a)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(a)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(a)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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