[asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?

Kirill Marchuk 62mkv at mail.ru
Thu Jan 29 02:43:02 CST 2015

Hi all

  Have recently watched Matt Jordan's session on Kamailio World 2014

On slides 26-29 of his presentation 
he speaks about a (completely new, for me at least) approach to build 
scalable telephony systems, using N instances of Kamailio and N 
instances of Asterisk

Are there any whitepapers, howtos, "implementation experience reports", 
whatever, available, that would describe such an approach in details and 
help some not-so-advanced admins to at least understand "if is it what 
they need, or not exactly, or not at all" ?

We are planning to look closer at Kamailio (or any other proxy, like 
OpenSip) as a way to do both load-balancing and failover solutions, so 
that refusal of any Asterisk instance should have minimal possible 
effect on the overall system availability.

A lot of questions howevere arise, like: what if one SIP user got 
REGISTERed at Server 1, and the other on Server 3, so how can they call 
one another ?

Also, outbound registrations can be done from one instance at a time, 
say it's done from Server1 for Trunk1, so how can users, that got 
authenticated at Server2, call thru that registration (Trunk1) ?

Also, Kamailio itself has to be protected from failing, and probably 
even from overload...

Would be great to read something in-depth about that


Kirill Marchuk

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