When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn.
Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? regards, David _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
