Den 2015-09-15 kl. 16:52, skrev asandoval...@gmail.com:
Hello Marek! I’ve been running on an issue with my Asterisk 12
configuration for using WebRTC on a LAN environment for about a month! I
really need some help …
My calls from the browser are done fine. I get ringing, they can be
answered
Hi,
I'd like to use queues in Asterisk and I have a few basic questions
(I'm a newbie) about Queues:
1.- What are the differences between Agents and Members (if any)?
2.- I want to implemment a small call center and I think the best way it is by
using dynamic members
Sebastian,
If I have understood you correctly, the SIP communication is now via NAT
instead forwarded ports. For safety, it is much better.
I think it is not because of a UDP timeout, but rather because of a NAT
timeout. For this is "qualify" exactly the right thing to let the NAT port
Greetings All, Regarding this archived post.
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
Did anyone ever find an solution to this? I've got a new box running
13.3.0 with the exact same issue.
For those that don't read the link.
I've got SIP Peers in
