asterisk users - May 2006 - Dropped SIP connections never being closed?

Hi all

I have noticed an interresting problem with Wireless SIP Connections.

When a Phone gets out of reach during a call (for example into a MeetMe 
Conference) of course the connection gets lost. The Phone hangs up.

But 'show channels' still shows that call, and the user is still in the 
Conference.
I did retry that a few times and ended up with a couple of zombie callers in 
Conferences. (Or even to foreign SIP test destinations without timeout like 
echo tests etc.)

Why doesn't Asterisk notice when a call is uncleanly dropped?

-Benoit-

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,