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@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@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@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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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