I was dialed into a conference room on Talk earlier and was disconnected after one minute of being on mute.
If your system stops sending RTP packets when you are muted, Asterisk will disconnect you after one minute (rtptimeout=60 in /etc/asterisk/sip.conf).
I've not been able to re-create this with my soft phone nor my linksys phone. I suspect the issue is settings on the client.
Closing.
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