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 vilen (pid = 23059)
Verbosity was 1 and is now 4
  == Using SIP RTP CoS mark 5
    -- Executing [3301@gr_relay:1] Dial("SIP/vienna-000000fd", "SIP/3301,2") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3301
    -- SIP/3301-000000fe is ringing
    -- Nobody picked up in 2000 ms
    -- Executing [3301@gr_relay:2] NoOp("SIP/vienna-000000fd", "Dial Status: NOANSWER") in new stack
    -- Executing [3301@gr_relay:3] Dial("SIP/vienna-000000fd", "SIP/3302") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3302
    -- SIP/3302-000000ff is ringing
  == Spawn extension (gr_relay, 3301, 3) exited non-zero on 'SIP/vienna-000000fd'
    -- Executing [h@gr_relay:1] Playback("SIP/vienna-000000fd", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr_relay, h, 1) exited non-zero on 'SIP/vienna-000000fd'
  == Using SIP RTP CoS mark 5
    -- Executing [3301@gr_relay:1] Dial("SIP/vienna-00000100", "SIP/3301,2") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3301
    -- SIP/3301-00000101 is ringing
    -- Nobody picked up in 2000 ms
    -- Executing [3301@gr_relay:2] NoOp("SIP/vienna-00000100", "Dial Status: NOANSWER") in new stack
    -- Executing [3301@gr_relay:3] Dial("SIP/vienna-00000100", "SIP/3302") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3302
    -- SIP/3302-00000102 is ringing
  == Spawn extension (gr_relay, 3301, 3) exited non-zero on 'SIP/vienna-00000100'
    -- Executing [h@gr_relay:1] Playback("SIP/vienna-00000100", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr_relay, h, 1) exited non-zero on 'SIP/vienna-00000100'
  == Using SIP RTP CoS mark 5
    -- Executing [3309@gr1:1] Playback("SIP/3301-00000103", "/etc/asterisk/sound/message") in new stack
    -- <SIP/3301-00000103> Playing '/etc/asterisk/sound/message.gsm' (language 'en')
  == Spawn extension (gr1, 3309, 1) exited non-zero on 'SIP/3301-00000103'
    -- Executing [h@gr1:1] Playback("SIP/3301-00000103", "/etc/asterisk/sound/message") in new stack
[Apr 23 16:41:13] WARNING[24498]: file.c:766 ast_readaudio_callback: Failed to write frame
    -- <SIP/3301-00000103> Playing '/etc/asterisk/sound/message.gsm' (language 'en')
[Apr 23 16:41:13] WARNING[24498]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/3301-00000103 for /etc/asterisk/sound/message
  == Using SIP RTP CoS mark 5
    -- Executing [8001@gr1:1] Answer("SIP/3301-00000104", "") in new stack
    -- Executing [8001@gr1:2] Record("SIP/3301-00000104", "/etc/asterisk/sound/message:gsm,5") in new stack
    -- <SIP/3301-00000104> Playing 'beep.gsm' (language 'en')
    -- Executing [8001@gr1:3] Playback("SIP/3301-00000104", "/etc/asterisk/sound/messege") in new stack
[Apr 23 16:41:24] WARNING[24500]: file.c:663 ast_openstream_full: File /etc/asterisk/sound/messege does not exist in any format
[Apr 23 16:41:24] WARNING[24500]: file.c:958 ast_streamfile: Unable to open /etc/asterisk/sound/messege (format 0x4 (ulaw)): No such file or directory
[Apr 23 16:41:24] WARNING[24500]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/3301-00000104 for /etc/asterisk/sound/messege
    -- Executing [8001@gr1:4] Hangup("SIP/3301-00000104", "") in new stack
  == Spawn extension (gr1, 8001, 4) exited non-zero on 'SIP/3301-00000104'
    -- Executing [h@gr1:1] Playback("SIP/3301-00000104", "/etc/asterisk/sound/message") in new stack
[Apr 23 16:41:24] WARNING[24500]: file.c:766 ast_readaudio_callback: Failed to write frame
    -- <SIP/3301-00000104> Playing '/etc/asterisk/sound/message.gsm' (language 'en')
[Apr 23 16:41:24] WARNING[24500]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/3301-00000104 for /etc/asterisk/sound/message
  == Using SIP RTP CoS mark 5
    -- Executing [2301@gr1:1] Dial("SIP/3301-00000105", "SIP/vienna/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/vienna/2301
    -- SIP/vienna-00000106 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.12.5:5060
    -- SIP/vienna-00000106 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/3301-00000105' status is 'BUSY'
    -- Executing [h@gr1:1] Playback("SIP/3301-00000105", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr1, h, 1) exited non-zero on 'SIP/3301-00000105'
  == Using SIP RTP CoS mark 5
    -- Executing [2301@gr1:1] Dial("SIP/3301-00000107", "SIP/vienna/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/vienna/2301
    -- SIP/vienna-00000108 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.12.5:5060
    -- SIP/vienna-00000108 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/3301-00000107' status is 'BUSY'
    -- Executing [h@gr1:1] Playback("SIP/3301-00000107", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr1, h, 1) exited non-zero on 'SIP/3301-00000107'
  == Using SIP RTP CoS mark 5
    -- Executing [2301@gr1:1] Dial("SIP/3301-00000109", "SIP/vienna/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/vienna/2301
    -- SIP/vienna-0000010a is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.12.5:5060
    -- SIP/vienna-0000010a is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/3301-00000109' status is 'BUSY'
    -- Executing [h@gr1:1] Playback("SIP/3301-00000109", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr1, h, 1) exited non-zero on 'SIP/3301-00000109'
  == Using SIP RTP CoS mark 5
    -- Executing [2301@gr1:1] Dial("SIP/3301-0000010b", "SIP/vienna/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/vienna/2301
    -- SIP/vienna-0000010c is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.12.5:5060
    -- SIP/vienna-0000010c is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/3301-0000010b' status is 'BUSY'
    -- Executing [h@gr1:1] Playback("SIP/3301-0000010b", "/etc/asterisk/sound/message") in new stack
  == Spawn extension (gr1, h, 1) exited non-zero on 'SIP/3301-0000010b'
vilen*CLI> quit
Executing last minute cleanups