[Asterisk-Users] Asterisk SIP architecture question

[David Thomas]

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