[SR-Users] Adding SIP Gateway Endpoint in Kamailio

David Villasmil david.villasmil.work at gmail.com
Fri Nov 15 14:30:27 CET 2019


If you're only using Asterisk as a media server, why use it? Why not just
use rtpproxy or mediaproxy? it'd be much simpler and you'd achieve the same
thing.
Unless you need to do something specific in Asterisk, there's really no
need.

https://dopensource.com/2017/05/31/installing-configuring-rtpproxy/

Should help you getting started.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Fri, Nov 15, 2019 at 1:21 PM Sujit Roy <sujitroydhk at gmail.com> wrote:

> Hello
>
> Here is my call flow scenario.
>
> SIP Gateway (A)
> Kamailio (K)
> Asterisk (AST)
> SIP Gateway (B)
>
> Now i want to send calls from A -> B by using Asterisk as media server.
> Kamailio shall be used to authenticate A and allow A to send calls to B.
>
> What are the configurations i need to make in Kamailio and Asterisk ?
>
> Thanks in advance.
>
> --
> Regards
> ===================
> Sujit Roy
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20191115/8ada4958/attachment.html>


More information about the sr-users mailing list