Oct 10 02:35:29 jicofo2477110: cont-finish. I am trying to configure Jitsi to operate as a softphone for my Telekom Germany VoIP-connection. In the conference I made a kicked sip-participant and ended the conference. The call was successful, the sound was in 2 directions. packetlogging. I made a call from a conference to a sip number.
AudioMediaStream.DISABLE_DTMF_HANDLING=true Enabling UDP port in server resolved this issue for me, in your server make sure below ports are enabled, 80/tcp for Web UI HTTP (really just to redirect, after uncommenting ENABLEHTTPREDIRECT1 in.
# allow dtmf pass through to the sip side The issue may be because of External ports. Everything works well even with invite people (phone number).
#Jitsi call pause number install#
Hi, I was able to install and configure Jitsi.
#Jitsi call pause number how to#
Jigasi: sip-communicator.properties .=false How to call to person (number) by Jitsi Meet API (JavaScript in web browser) Developers. Here are my logs and configuration files: The other versions are listed here: $ apt list -installed | grep ji I use the latest version from the master branch of Jigasi. Also no received data is seen in the docker logs of the vosk container. The breakpoints are not triggered in the methods to send the audio to the vosk container. The problem lies within receiving the audio data from the call. I can trigger the breakpoints in the initialzation of the transcription service. RTCAGENTNAME var agentSession Number() var agentSessionName. This is the sip user for the sip (vox) side. I run the code directly from my IDE (Visual Studio Code) in debug mode and set some breakpoints in all methods of the VoskTranscriptionService to understand the behavior of this code.Īs a result I can create and join Jitsi calls, active the subtitle option, the transcriber joins and I can converse with myself. I have changed sip-communicator.properties to have the user 2. which should be the host user. I followed the setup for the transcription part as stated in the README and disabled SIP. Jigasi is checked out from the github repository. To keep it simple i just used 127.0.0.1 as host. There is no history of the video calls/conference (no history of video. I installed all components except jigasi from the package repository as described in the quickstart guide. Open video from UC Assistant, CC Agent, Switchboard Call a user with video.
In preparation for this I tried to set up a working JitsiJigasiVosk environment without altering the code. I want to develop a new connector from Jigasi to our own transcription service.
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