[Asterisk-Users] determining cause of dropped calls?

[Bruce Komito]

I am trying to figure out why calls between SIP devices and the PSTN are
being regularly dropped after anywhere from 2-15 minutes.  I have turned
on everything I can think of, but I don't see any obvious reasons for the
drops.  All I can see from turning on debug and verbosity is two messages
advising of a destroyed call, followed by normal-looking SIP and ZAP
termination messages.

The first indication there is a problem, immediately before the call is
dropped is something like:

Destroying call '16118f2c0e1f992d7ee90db64dd58356 at 216.162.162.37'
Destroying call '0e5d498820a12c7d at 199.165.203.107'

Is there something I can turn on/enable that will tell me who is causing
the call to be destroyed and why?

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115