My configuration of SIP profile are: param name="nat-options-ping" value="true" param name="all-reg-options-ping" value="true" param name="unregister-on-options-fail" value="true" param name="enable-timer" value="false" and this is because, i was experiencing video session starting up within audio call, SDP session update once session timers completes, That was totally unexplained, and i eventually had to disable timer, which ultimately resolved that issue. What left is , RTP timers, and i have param name="rtp-timer-name" value="soft" param name="rtp-timeout-sec" value="15" but it is not helpful. Channel hangs up once only a client becomes unreachable. But if both becomes unreachable, Freeswitch is helpless!! which logs are required to dig it in details and what approach shall be considered? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6... Sent from the freeswitch-users mailing list archive at Nabble.com.