Hey all. I have webrtc up and running with asterisk 11. All is going well
with TLS now working.
At least I hope it is using TLS and wss. Based on what I am seeing I have
UDP, WSS listed in the Allowed transports, but every time I connect the
Primary transport shows WS.. Why is this? Am I
Hi Andre,
Some routers just simply won't support this double-nat scenario you
describe. Othera will... And without any special forwarding.
Is it possible to put the first router into "bridge" mode, and use the
second router as the actual NAT router?
This may be the quickest solution to your
Hi all,
I have a setup which is not working right now:
Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) -
Asterisk (10.17.46.99)
My issue: Everything works, but RTP is only going from my Asterisk
towards the provider. Asterisk is configured to use SIP-ports 55060 and
Hi all,
I'm needing to upgrade Asterisk from 10.x to whatever the recommended version
is that will allow me to continue to use Fax For Asterisk.
I don't have many upgrade windows, I'd like to get the most bang for my buck,
but I can't afford to be a beta tester on this server.
The FFA site
Hey guys, any thoughts on that? Probably a bug or is it a default behavior?
On Thu, Mar 9, 2017, 2:05 PM Kseniya Blashchuk wrote:
> Hi all!
> I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP
> addresses from the same subnet set on one interface, and
Hi!
Apparently this is possible; my asterisk server is doing this when my
SIP phone redirects the call with a SIP REFER message. The phone is
excluded from the call after it transfers the call.
I'll contact my ITSP if their trunk can also do this.
Regards,
Sree
On 03/09/2017 11:03 PM, Sree