I’m new to this forum as well as to asterisk and as beginner I would have a question before I’m starting with my project:
I have an openSuSE (Leap 15) server running which has a second soundcard which speaker and micorphone are connected to a ~20 years old analog intercom system of our house. Currently I’m sending and receiving event driven soundfiles by the server (e.g. send messages with advices to postman when ringing and nobody is at home).
I’m intending now to setup an asterisk server in a vmware instance (again openSuSE Leap 15) which is able to use soundcard - intercom connection from the host as SIP, so, if somebody is ringing I’m getting a connection to my mobile.
Now my question is if there is somebody with experince how to connect the asteriks VM with the hosts soundcard.
First possibility would be to register the soundcard also in the VM or
use a tool to transport sound via TCP from host to host (e.g. pulseaudio)
or give up and install asterisk directly on the host which I would like to prevent.
If somebody has a tip or link for me would be great.My concerns are realtime behavior and sound quality in both directions.