Hello,
Not a typical issue, usually it’s the opposite. I’d like to set up the extension (I use grandstream ATA) so that parties calling from/to this extension would NOT hear each other. So in other words, all the SIP signalling will go thru, but the RTP stream would get lost somewhere. Calls would be going from/to this extension only (two ports registered under the same ID).
Reason is, this is a test equipment, I need to set up a modem transmission between two ports and since this doesn’t work well with VoIP, I’m using electrical coupling (capacitors) between the lines. But still it needs to accept the number dialed, ring the other line etc.
I asked ChatGPT and I think it’s bs’ing me 
Tried it, I don’t think it’s intended for muting the voice:
To set up an extension in Asterisk PBX such that calling parties cannot hear each other, you can utilize the “Privacy” feature.
and then in extensions:
exten => 100,1,NoOp()
same => n,Set(PRIVACY()=full)
Any hint appreciated, it doesn’t have to be asterisk side. I tried breaking some config in the ATA settings, like reporting a bogus NAT IP and it worked, but in 30 seconds goes “busy”, so any transmission longer than that would break.