One reinvite problem

hello,every body.
I has one problem, so i ask for help here.

I use asterisk to transfer webrtc to my sip gateway. but my sip gateway don’t support webrtc. so I use asterisk.

I find this work well, when i call one video or two video .

but when a call has set up. sip gateway add a new video (change one video to two video). the other one don’t receive invite.

Can i change asterisk configure to support this?

I find some community topic say asterisk is b2b model don’t tranfer reivite.

some body can tell me answer。 thinks。

Asterisk 16 may support this but it hasn’t really been tested for that use case. What version are you using?

I use asterisk 16.0.1.

If it doesn’t work out of the box, then it would likely require someone to specifically go through the scenario, identify what has to happen, and code it.

thank you.
because I don’t has ability to code asterisk. i change my plan.

may be i can add some funny function to asterisk.