Jigasi config SingleCallInProgressPolicy and CallWaitingDisabled not work

Hi Community,

I’m trying to disable ringing tone while dial-out, cause the ringing tone is too loud and may interfere the meeting.
Just like Option to disable ringing while in call? mentioned, I had tried set SingleCallInProgressPolicy=true and CallWaitingDisabled=true, neither of them not work.
Also, in source code I didn’t find any code about these two properties. Is there any other thing I can try for disabling ringing out notification

Are you sure that is bot coming from the sup side? In asterisk for example you can stop the ringing sip messages and sound by omitting r or R in the dial command.
Not sure there is an option to ignore ringing signalling in jigasi, need to check in code.

Thanks for reply! We had verify that the ringtone actually comes from sip server. But unfortunately, vendor of sip server said that they did not support mute during ring and unmute after phone connected. I was wondering that is there such feature or how to implement that:

  1. Mute the new phone participant by default while dialing out and the new phone participant not pick up the phone yet.
  2. Un-mute the new phone participant by default after the new phone participant pick up the phone.
    To implement this feature, what components should I modify? Web/New Prosody Module/Jicofo?Any hint?

Or is there a config that mute dial out participant by default? Then we could integrate with IVR and ask phone participant unmute themself with ‘*6’ or something like that.
I had set startAudioMuted=1 and startWithAudioMuted=true but seems these configs are for web participant. I’m wondering is there any alternative config for phone participant.

This one should work. The initial one is sent by the web, but then jigasi respects it. So the second participant which is jigasi, must be muted.
But you need to enable the option jigasi to support muting.

Thanks for your prompt reply sincerely! I try this option and it works!
So the left one is how to integrate with IVR? I just want the function that un-mute phone participant himself with press *6. Is this tutorial ‘Tutorial - Jitsi / Jigasi & FreePBX integration. Along with Asterisk IVR to use Jitsi conference mapper API’ good enough for that? I didn’t find any place about how to configure IVR to support unmute with ‘*6’ or something like that …

This one has mute examples: Documentation | Voximplant.com

Check the doc, I’m still confusing by what kind of protocol/message sent to jigasi from IVR.
I trace the muteIVR scenario, found the IVR script snippet, said that the mute/unmute command was send by json format with ‘call.sendInfo’. What does ‘call.sendInfo’ means? Does that mean IVR send message to jigasi via http something else? What’s the interface that jigasi provided to accept this message? It doesn’t look like via sip protocol at all …

function sendMuteRequestToCall(call, muted) {
    let request = {
        type: "muteRequest",
        id: uuidgen(),
        data: {
            audio: muted
        }
    };

    Logger.write(`DEBUG: Sending mute request: ${JSON.stringify(request)}`);
    call.sendInfo("application/json", JSON.stringify(request));
}

Yep, this is a sip info message with content that json.
https://voximplant.com/docs/references/voxengine/call#sendinfo

Finally I accomplish this scenario by modifying SIP server’s configuration’ which disable 183 signal and media port is disabled before connected.