I am trying to see if there is any interested in someone developing a method to link a Jigasi dial-in user to an existing web participant. I am looking to sponsor the work and the dev would be responsible for getting it merged into the Jitsi project. The reasons behind this are described here: Link JIGASI audio track to existing participant - #7 by DSchaef and is pretty common among conferencing solutions today. If you are interested feel free to PM me.
Requirements would include:
- Full functionality using Asterisk (PJSIP) or Freeswitch to include muting and unmuting participants.
- “Call me” functionality that will call a PSTN number and link the audio & mic feed to that number. Should be in the drop down under MIC selection and active speaker should reflect on the web participant.
- Dial code that will link a PSTN caller to a Web user. ie somewhere there should be a starcode that represents my user. After dialing in, I should be able to dial that starcode and then my phone number and my web session are then linked.
- Audio should not be transmitted over Web session when a PSTN number is linked to a web user.
- Mic control should control the PSTN Mic not PC when linked.
- There should be an indication of PSTN linked in the WEB UI. Maybe change MIC to phone icon?
- Would like a popup with dial-in options/directions if the web user’s mic is unavailable.