You don’t need your users registering on asterisk, then. You’d setup 2
active-active dns-balanced kamailio’s dispatching outbound calls to the
asterisks, since user-to-user calls don’t need to leave kamailio. You’d
also generate CDRs in kamailio.
No sure what service you intent to provide with the Asterisks, I assume
VoiceMail, conferencing, etc. For outbound termination you can also do it
from the Kamailio itself.
Hope that helps.
David
On Sat, 4 May 2019 at 20:19, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
I will use Asterisk PjSip because it provide more
functionality than
"clear" SIP like for example login multiple endpoints on that same
credentials without logout previous endpoint.
What I want? Proof of concept HA clouded PBX for multi-tenant clients
sob., 4 maj 2019 o 20:59 David Villasmil <david.villasmil.work(a)gmail.com>
napisał(a):
You’re probably better off doing registration on
kamailio only. Depending
on what exactly you’re trying to achieve.
A trace would help a lot. Check the IP addresses in the SDP actually
belong to the user and the asterisk.
On Sat, 4 May 2019 at 17:58, przeqpiciel <przeqpiciel(a)gmail.com> wrote:
Hi,
There is any working examples how to configure dispatcher module of
Kamailio? I have working registrations but working in half dialing
function.Signalling are come to asterisk and rtp start sending to my IP but
i cant hear anything.
Asterisk are doing registrar job - i dont know if it is good idea, maybe
not.
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David Villasmil
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phone: +34669448337