Hello,
I've started playing with an idea to add multiple asterisk servers and
using dispatcher to balance the sip load between them. I added the code
according to dispatcher module documentation (
http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I
think there's something
Well, this
*if (from_uri!=myself uri!=myself)*
Means neither source nor destination is our user. Which implies that if our
domain is A, then call from domain B to C is not possible. However, calls
from B or C to A and A to B or C are possible. That is way an
unauthorized user gets passed and
Hi guys,
I'm designing a new service for a client and I was wondering what your
opinion is of the 2 options I'm considering to separate users on the
server. Basically I want to create closed groups where each user can only
call and receive calls from members of the same group (single domain
Hi,
On my server, I have the option of using either Rtpengine for NAT traversal
or pure TURN without rtpengine.
Rtpengine has the obvious plus that it only needs 1 public IP, while TURN
(with STUN) will need 2 public IPs, although that's not a problem in my
case.
Having said that, I'd like to
