Thanks Maxim.
Greetings
El mié., 15 may. 2019 a las 6:16, Maxim Solodovnik ()
escribió:
> Hello Siovel,
>
> sorry for the long response we had long public holidays :)))
>
> If I do remember correctly to connect you need to:
> 1) run Asterisk
> 2) run OM with Asterisk support
> 3) run red5-sip co
... and the port defined in the base-url.
Am 15.05.2019 um 17:53 schrieb Scholz, Rene:
Hello Maxim,
it ever uses the base-url from the configuration for generating the link.
Best regards,
René
Am Mittwoch, den 15.05.2019 um 12:20 schrieb Maxim Solodovnik:
Hello Rene,
If I do r
Hello Maxim,
it ever uses the base-url from the configuration for generating the
link.
Best regards,
René
Am Mittwoch, den 15.05.2019 um 12:20 schrieb Maxim Solodovnik:
Hello Rene,
If I do remember correctly current base URL (retrieved via JS
callback) is used while invitations gen
Hi,
I had a little problem during the last update (from 3.0.7 to 4.0.8). I
didn't realise, that ffmpeg didn't work after server update. The
following om update went throug, despite the fact, that
screen-recordings were not converted.
That why I needed to figure out (with a little bit help of Maxi
Hello Rene,
If I do remember correctly current base URL (retrieved via JS
callback) is used while invitations generation
So this should work as follows:
1) user A access OM at port 5443
2) generates the link (send invitation)
Result: base URL has port 5443
1) user B access OM at port 5444
2) gen
Hello Siovel,
sorry for the long response we had long public holidays :)))
If I do remember correctly to connect you need to:
1) run Asterisk
2) run OM with Asterisk support
3) run red5-sip connected with OM
4) open Admin->Rooms
5) go to particular room settings and check "Enable SIP transport in
