Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on aster-3 (pid = 2628)
Verbosity is at least 16
aster-3*CLI> sip reload
Reloading SIP
[Apr 23 12:20:49] WARNING[2717]: chan_sip.c:20720 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'berlin' to '192.168.12.2'
aster-3*CLI> sip reload
Reloading SIP
[Apr 23 12:21:04] WARNING[2717]: chan_sip.c:20720 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'berlin' to '192.168.12.2'
[Apr 23 12:21:08] NOTICE[2717]: chan_sip.c:25030 handle_request_register: Registration from '<sip:vienna@192.168.12.3>' failed for '192.168.12.5:5060' - Wrong password
[Apr 23 12:21:12] NOTICE[2717]: chan_sip.c:25030 handle_request_register: Registration from '<sip:vienna@192.168.12.3>' failed for '192.168.12.5:5060' - Wrong password
[Apr 23 12:21:15] NOTICE[2717]: chan_sip.c:25030 handle_request_register: Registration from '<sip:vienna@192.168.12.3>' failed for '192.168.12.5:5060' - Wrong password
[Apr 23 12:21:29] NOTICE[2717]: chan_sip.c:25030 handle_request_register: Registration from '<sip:vienna@192.168.12.3>' failed for '192.168.12.5:5060' - Wrong password
-- Remote UNIX connection
-- Remote UNIX connection disconnected
[Apr 23 12:22:29] NOTICE[2717]: chan_sip.c:25030 handle_request_register: Registration from '<sip:vienna@192.168.12.3>' failed for '192.168.12.5:5060' - Wrong password
== Using SIP RTP CoS mark 5
-- Executing [1301@default:1] Dial("SIP/192.168.12.5-0000000e", "SIP/1301") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1301
aster-3*CLI> sip reload
Reloading SIP
-- SIP/1301-0000000f is ringing
aster-3*CLI> sip reload
Reloading SIP
== Spawn extension (default, 1301, 1) exited non-zero on 'SIP/192.168.12.5-0000000e'
aster-3*CLI> sip reload
Reloading SIP
== Using SIP RTP CoS mark 5
-- Executing [1302@default:1] Dial("SIP/192.168.12.5-00000010", "SIP/1302") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1302
-- SIP/1302-00000011 is ringing
-- SIP/1302-00000011 answered SIP/192.168.12.5-00000010
-- Locally bridging SIP/192.168.12.5-00000010 and SIP/1302-00000011
== Spawn extension (default, 1302, 1) exited non-zero on 'SIP/192.168.12.5-00000010'
aster-3*CLI> exit
Executing last minute cleanups
Asterisk cleanly ending (0).