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> dialplan re
reload  remove
aster-3*CLI> dialplan reload
Dialplan reloaded.
  == Parsing '/etc/asterisk/extensions.conf':   == Found
    -- Registered extension context 'default'; registrar: pbx_config
    -- Added extension '_13XX' priority 1 to default
    -- Added extension '_23XX' priority 1 to default
  == Parsing '/etc/asterisk/users.conf':   == Found
    -- Registered extension context 'app_dial_gosub_virtual_context'; registrar: app_dial
    -- merging incls/swits/igpats from old(app_dial_gosub_virtual_context) to new(app_dial_gosub_virtual_context) context, registrar = pbx_config
    -- Added extension 's' priority 1 to app_dial_gosub_virtual_context
    -- Registered extension context 'parkedcalls'; registrar: features
    -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config
    -- Added extension '700' priority 1 to parkedcalls
    -- Registered extension context 'app_queue_gosub_virtual_context'; registrar: app_queue
    -- merging incls/swits/igpats from old(app_queue_gosub_virtual_context) to new(app_queue_gosub_virtual_context) context, registrar = pbx_config
    -- Added extension 's' priority 1 to app_queue_gosub_virtual_context
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000062 sec
    -- Time to restore hints and swap in new dialplan: 0.000002 sec
    -- Time to delete the old dialplan: 0.000008 sec
    -- Total time merge_contexts_delete: 0.000072 sec
  == Using SIP RTP CoS mark 5
    -- Executing [2301@default:1] Dial("SIP/1302-00000002", "SIP/frankfurt/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/frankfurt/2301
    -- SIP/frankfurt-00000003 is ringing
  == Spawn extension (default, 2301, 1) exited non-zero on 'SIP/1302-00000002'
  == Using SIP RTP CoS mark 5
    -- Executing [1301@default:1] Dial("SIP/frankfurt-00000004", "SIP/1301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1301
    -- SIP/1301-00000005 is ringing
    -- SIP/1301-00000005 answered SIP/frankfurt-00000004
    -- Locally bridging SIP/frankfurt-00000004 and SIP/1301-00000005
  == Spawn extension (default, 1301, 1) exited non-zero on 'SIP/frankfurt-00000004'
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Executing [2301@default:1] Dial("SIP/1302-00000006", "SIP/frankfurt/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/frankfurt/2301
    -- SIP/frankfurt-00000007 is ringing
    -- SIP/frankfurt-00000007 answered SIP/1302-00000006
    -- Locally bridging SIP/1302-00000006 and SIP/frankfurt-00000007
  == Spawn extension (default, 2301, 1) exited non-zero on 'SIP/1302-00000006'
  == Using SIP RTP CoS mark 5
    -- Executing [1301@default:1] Dial("SIP/frankfurt-00000008", "SIP/1301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1301
    -- SIP/1301-00000009 is ringing
  == Spawn extension (default, 1301, 1) exited non-zero on 'SIP/frankfurt-00000008'
  == Using SIP RTP CoS mark 5
    -- Executing [1301@default:1] Dial("SIP/frankfurt-0000000a", "SIP/1301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1301
    -- SIP/1301-0000000b is ringing
  == Spawn extension (default, 1301, 1) exited non-zero on 'SIP/frankfurt-0000000a'
  == Using SIP RTP CoS mark 5
    -- Executing [1302@default:1] Dial("SIP/frankfurt-0000000c", "SIP/1302") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1302
    -- SIP/1302-0000000d is ringing
    -- SIP/1302-0000000d answered SIP/frankfurt-0000000c
    -- Locally bridging SIP/frankfurt-0000000c and SIP/1302-0000000d
  == Spawn extension (default, 1302, 1) exited non-zero on 'SIP/frankfurt-0000000c'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
aster-3*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).