On Mon, at 11:24 AM, Thomas Stein > wrote: I am the only one in the video chat room.Īm 10.10.16 um 11:43 schrieb Mirko Brankovic: Can I raise the log level somewhere?ĭo you see another participant in filmstrip div when you start the call ? In “Settings > Account Settings > Account”, “Codec Settings > SRTP Mode > Enabled And Forced” and under “Preferred Vocoder” for Wifi and Mobile networks I turned on Opus and G.722.Check the jitsi-videobridge log, should say something about attempt and response. I set the same settings for my extension and then went to set up my Grandstream Wave on my phone. Now Jitsi connects using SRTP and can use the opus codec. Now SRTP is alive it was then time to set some values in ps_endpoints for the extension: transport: null I had to modify the docker image again and include libsrtp2-dev in the operating system for Asterisk to be compiled with the necessary res_srtp module. The next problem I spotted in the log related to there being no res_srtp module loaded and therefore no SRTP available. I needed to install it into my docker image by downloading and extracting the. Looking at the Asterisk console and logs I noticed that it mentioned opus – I know opus is a codec and I then found it wasn’t installed – and isn’t included with Asterisk. I wasn’t sure what the problem was, but the usual candidate is a codec. The dreaded “488 Not Acceptable Here” has been the bane of my existence in the Asterisk setup. I faced a couple of niggling issues as I set things up. You’ll then see them hitting the Asterisk console as they are dialled to join in. You can invite your other callers or extensions in by entering their number in the dialog. When in a Jitsi conference you should see a + button toward the bottom left. With the Jigasi config entered if you have pjsip debug turned on for the Asterisk server you’ll see it getting connected. You’ll need to stop jigasi separately from the rest of the container set: $ docker stop $(docker container ps -f "name=jigasi" -q) It can get messy shutting things down when you use -d to detach. Pay attention to the way it uses two compose files. Then started up the container set: $ docker-compose -f docker-compose.yml -f jigasi.yml up -d Obviously you should use stronger passwords.# SIP server (use the SIP account domain if in doubt) # SIP URI for incoming / outgoing Password for the specified SIP account as a clear text # Basic Jigasi configuration options (needed for SIP gateway support) env, paying attention to the Jigasi section: # I followed the instructions, pulled it down and edited the. Jisti already has a docker-compose ready to rock – I added it into my ps_endpoints, ps_aors and ps_auths in exactly the same way as any other phone as extension 801. On the Asterisk side I treated Jitsi/Jigasi as just another SIP extension. So once my Asterisk server was alive and I could make calls to and from other mobile phone SIP clients hooking up Jitsi was next. I never had the need to be involved at the server end, configuring extensions and call routing, etc. Connecting SIP used to be – unbox a phone plug it into the LAN, let DHCP tell it where the VoIP server was and that’s all she wrote. The whole Asterisk thing has been an uphill challenge for me. To do this you need to use a SIP add-on called ‘Jigasi’. As I’m working on Asterisk right now the actual challenge is to get Jitsi configured so we can conference in audio users to our video chats.
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