We recently installed VitalPBX as our phonesys. We are having the issue where, while on a conference with multiple people, our extension connects to our Conference while muted(the phone itself has a mic mute option that isn't digital). After an uncertain time interval (I've tried it up to 6 minutes without replicating the issue,) we can no longer communicate with the rest of the users in the conference after unmuting the phone. Is there anything (log, live feed, etc) I could use to determine what is happening on the backend? I'd like to determine if it's my phone or the system doing this.
for debug, you may check the asterisk log from the Asterisk CLI, also, you may get the call log from Asterisk CLI and share it here, in case that you want we analyze it.
Could you please instruct me how to get the Asterisk CLI log? I'd like to post it on here if possible.
You must go to the Linux console, and execute the following command
then, make the call and copy the console output