Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux5 (pid = 6351)
Verbosity was 0 and is now 3
== Using SIP RTP CoS mark 5
-- Executing [2101@default:1] Dial("SIP/2102-00000000", "SIP/2101") in new stack
== Using SIP RTP CoS mark 5
[Jun 20 17:16:46] WARNING[6399]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2102-00000000' status is 'CHANUNAVAIL'
== Using SIP RTP CoS mark 5
-- Executing [2102@default:1] Dial("SIP/2102-00000001", "SIP/2102") in new stack
== Using SIP RTP CoS mark 5
-- Called 2102
-- SIP/2102-00000002 is ringing
-- SIP/2102-00000002 answered SIP/2102-00000001
-- Native bridging SIP/2102-00000001 and SIP/2102-00000002
-- Music class default requested but no musiconhold loaded.
== Spawn extension (default, 2102, 1) exited non-zero on 'SIP/2102-00000001'
-- Remote UNIX connection
Executing last minute cleanups
== Destroying musiconhold processes
linux5*CLI>
Disconnected from Asterisk server
Executing last minute cleanups