[asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Jared Smith]

On Mon, 2010-02-22 at 16:13 -0500, JT wrote:
> Is this something that is fixed in an update?  (Currently running 1.2)

Yes... modern versions of Asterisk support SIP session timers.  (If I
remember correctly, Asterisk 1.2 could tear down a call based on lack of
RTP data, but I never found it worked well enough in my tests to warrant
its use.)

--
Jared Smith
Digium, Inc.