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]> 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]> > wrote: > >> Hi. >> >> is SIP fully implemented in 6.0.0 snapshots? >> >> >> Ali >> >> > > -- > Best regards, > Maxim > > -- Best regards, Maxim
