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 asterisk2 (pid = 691)
asterisk2*CLI>
asterisk2*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [4211@gr1:1] Goto("SIP/1201-00000029", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/1201-00000029", "") in new stack
    -- Executing [s@menu:2] BackGround("SIP/1201-00000029", "/var/tmp/asterisk/message06") in new stack
    -- <SIP/1201-00000029> Playing '/var/tmp/asterisk/message06.gsm' (language 'ru')
    -- Executing [s@menu:3] WaitExten("SIP/1201-00000029", "2") in new stack
    -- Timeout on SIP/1201-00000029, continuing...
    -- Executing [s@menu:4] BackGround("SIP/1201-00000029", "/var/tmp/asterisk/message07") in new stack
    -- <SIP/1201-00000029> Playing '/var/tmp/asterisk/message07.gsm' (language 'ru')
    -- Executing [9@menu:1] Goto("SIP/1201-00000029", "s,start") in new stack
    -- Goto (menu,s,4)
    -- Executing [s@menu:4] BackGround("SIP/1201-00000029", "/var/tmp/asterisk/message07") in new stack
    -- <SIP/1201-00000029> Playing '/var/tmp/asterisk/message07.gsm' (language 'ru')
    -- Executing [0@menu:1] Playback("SIP/1201-00000029", "/var/tmp/asterisk/message09") in new stack
    -- <SIP/1201-00000029> Playing '/var/tmp/asterisk/message09.gsm' (language 'ru')
    -- Executing [0@menu:2] Hangup("SIP/1201-00000029", "") in new stack
  == Spawn extension (menu, 0, 2) exited non-zero on 'SIP/1201-00000029'
  == Using SIP RTP CoS mark 5
    -- Executing [4211@gr1:1] Goto("SIP/1201-0000002a", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/1201-0000002a", "") in new stack
    -- Executing [s@menu:2] BackGround("SIP/1201-0000002a", "/var/tmp/asterisk/message06") in new stack
    -- <SIP/1201-0000002a> Playing '/var/tmp/asterisk/message06.gsm' (language 'ru')
    -- Executing [s@menu:3] WaitExten("SIP/1201-0000002a", "2") in new stack
[Mar 31 14:11:19] WARNING[4788][C-0000002b]: pbx.c:6696 __ast_pbx_run: Invalid extension '2', but no rule 'i' or 'e' in context 'menu'
  == Using SIP RTP CoS mark 5
    -- Executing [4211@gr1:1] Goto("SIP/1201-0000002b", "menu,s,1") in new stack
    -- Goto (menu,s,1)
    -- Executing [s@menu:1] Answer("SIP/1201-0000002b", "") in new stack
    -- Executing [s@menu:2] BackGround("SIP/1201-0000002b", "/var/tmp/asterisk/message06") in new stack
    -- <SIP/1201-0000002b> Playing '/var/tmp/asterisk/message06.gsm' (language 'ru')
[Mar 31 14:11:33] WARNING[4789][C-0000002c]: pbx.c:6696 __ast_pbx_run: Invalid extension '17', but no rule 'i' or 'e' in context 'menu'
asterisk2*CLI> quit
Asterisk cleanly ending (0).
Executing last minute cleanups