23.04.2019 0:27, Joshua C. Colp wrote:
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:
Tried already.
"line" is good, but not perfect.
Every time I restart asterisk, it will generate new random string for ";line=".
So, every time I restart asterisk, registrar (Server1) will save one
more
Hi,
Thank for your answer.
22.04.2019 23:47, Joshua C. Colp пишет:
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
Eric,
As I mentioned before that does not seem to work with PHP
== Using SIP RTP CoS mark 5
-- Executing [my_test_test@from-external:1]
NoOp("SIP/fpp--09ca", "") in new stack
-- Executing [my_test_test@from-external:2]
AGI("SIP/fpp--09ca",
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:
> Tried already.
>
> "line" is good, but not perfect.
>
> Every time I restart asterisk, it will generate new random string for
> ";line=".
>
> So, every time I restart asterisk, registrar (Server1) will save one
> more contact in it's
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
Registration is OK.
Server2 outgoing calls are OK.
INVITE, unauthorized, INVITE with password, OK, RINGING,...
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
> Hi,
>
> Got problems with incoming SIP calls.
>
> Scenario:
>
> Server1: 3cx or any other server
>
> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
>
> Server2 registers on Server1 with SIP ID 1121.
>
> Registration is OK.
>
> Server2 outgoing
