Thanks Max, with pleasure :-)
Ali
On 12/18/20 6:44 PM, Maxim Solodovnik wrote:
The configuration is described here
https://openmeetings.apache.org/AsteriskIntegration.html
If you'll find any missing part - please contribute :)))
On Fri, 18 Dec 2020 at 22:03, Ali Alhaidary
<[email protected] <mailto:[email protected]>> wrote:
which should be configured?
sip.hostname=
sip.manager.port=5038
sip.manager.user=openmeetings
sip.manager.password=12345
sip.manager.timeout=10000
sip.ws.local.port.min=6666
sip.ws.local.port.max=7666
## 127.0.0.1 is NOT working here
sip.ws.local.host=
sip.ws.remote.port=8088
sip.ws.remote.user=omsip_user
sip.ws.remote.password=12345
On 12/18/20 4:32 PM, Maxim Solodovnik wrote:
As you can see from commit messages:
So far only audio from room to SIP seems to work
(I'm using Linphone in my tests and for whatever reason video is
not working :((( )
I'll try to implement other direction (SIP to room)
would appreciate any advice on Asterisk config
and early tests result
the thing I was unable to test so far: is there any echo (audio
stream duplication)?
On Fri, 18 Dec 2020 at 18:07, Ali Alhaidary
<[email protected]
<mailto:[email protected]>> wrote:
Hi.
is SIP fully implemented in 6.0.0 snapshots?
Ali
--
Best regards,
Maxim
--
Best regards,
Maxim
