Yes, I've done something similar. What you really need is a conference service. That way you can give out dialing instructions to allow people to connect by telephone.

Your audio output from vMix would be sent to the conference bridge as if it was a telephone. I've done this using a SIP soft phone, so the connection is actually done via IP...not an analog phone line.

I wrote it up on my blog. If the conference bridge offered a WebRTC client, you may only need Chrome and some virtual audio cables. Back then I was using Wirecast. Also, vMix had fewer audio busses. All the new audio busses would make it easier now.

In the conference bridge, you would set the call so that your vMix connection was the host/presenter and everyone one else was automatically muted.

While there are some free conference services, you may find these not suitable. You may end up paying something to have the telephone access.

For a dozen years I co-produced a podcast about telecom topics. Along the way I met a gentleman was had just started a new specialty conference service with some amazing features.

We used his company ZipDX in the production of the show for years. It allowed us to have dial-in access lines in many countries.

As his company grew he eventually hired me.

Incidentally, some very large outfits, like webcasts.com use ZipDX in this very manner. We provide telephone access so people have the flexibility to connect by phone or web.