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 = 6034)
Verbosity was 0 and is now 3
== Using SIP RTP CoS mark 5
-- Executing [8000@default:1] Playback("SIP/2102-00000000", "demo-congrats") in new stack
-- <SIP/2102-00000000> Playing 'demo-congrats.gsm' (language 'en')
== Spawn extension (default, 8000, 1) exited non-zero on 'SIP/2102-00000000'
linux5*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2101 (Unspecified) D 5060 Unmonitored
2102/2102 192.168.105.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
== Using SIP RTP CoS mark 5
[Jun 20 17:09:38] NOTICE[6059]: chan_sip.c:20281 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
-- Remote UNIX connection
Executing last minute cleanups
== Destroying musiconhold processes
linux5*CLI>
Disconnected from Asterisk server
Executing last minute cleanups