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 <[email protected]> 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 <[email protected]> *On Behalf Of *Atux > Atux > *Sent:* Tuesday, March 27, 2018 12:16 PM > *To:* Mack Hendricks <[email protected]>; Kamailio (SER) - Users > Mailing List <[email protected]> > *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 <[email protected]> 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 <[email protected]> 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 <[email protected]> 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? > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
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