I have had a case where after a hangup on the Alsa channel
asterisk still thinks the line or call is active.
I have:
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
in my sip.conf file to help with this but it had no effect.
How can I ensure a session HANGS up and is not stale????
Is there a way for the next incoming call to VERIFY that console/ALSA
channel is still valid.
I dont want to hangup a real connection - I want to give a busy tone for
sure.
But if the session is not valid I need it gone.
How can I do that. I am using 1.4.43
Jerry
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