[asterisk-users] alsa channel

[Jerry Geis]

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