Benoit Panizzon wrote: > Why doesn't Asterisk notice when a call is uncleanly dropped?
Because it can't. There is no continuous signaling in a SIP call, so there's no way to know that the peer is gone. You can use 'rtptimeout' to make Asterisk notice when the RTP stream has stopped and then drop the call, _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
