Asterisk 11.13.1~dfsg-2+b1, Copyright (C) 1999 - 2013 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.
=========================================================================
Connected to Asterisk 11.13.1~dfsg-2+b1 currently running on asterisk4 (pid = 499)
asterisk4*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [4311@phone:1] Goto("SIP/1401-00000043", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/1401-00000043", "") in new stack
       > 0x7ff3f801bcb0 -- Probation passed - setting RTP source address to 192.168.40.201:16460
    -- Executing [s@menu:2] BackGround("SIP/1401-00000043", "/var/tmp/asterisk/message106") in new stack
    -- <SIP/1401-00000043> Playing '/var/tmp/asterisk/message106.slin' (language 'ja')
    -- Executing [s@menu:3] WaitExten("SIP/1401-00000043", "2") in new stack
  == CDR updated on SIP/1401-00000043
    -- Executing [0@menu:1] Hangup("SIP/1401-00000043", "") in new stack
  == Spawn extension (menu, 0, 1) exited non-zero on 'SIP/1401-00000043'
asterisk4*CLI> quit
Asterisk cleanly ending (0).
Executing last minute cleanups