I have strange problem some time I can see call is continue even the caller and called party hangup the phone. I am using mySQL realtime configuration for our SIP users. I added the all necessary fields for user configuration. And I already given 60 seconds value for rtptimeout and rtpholdtimeout but it does not seems to disconnect the call if there is not RTP from 60 seconds.
When I am restarting asterisk then call is going to terminate. So Please give me some idea to stop such kind of problem.