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 asterisk6 (pid = 6171)
asterisk6*CLI>
asterisk6*CLI>
  == Spawn extension (default, 2101, 1) exited non-zero on 'SIP/192.168.70.2-00000029'
  == Spawn extension (default, 1401, 1) exited non-zero on 'SIP/192.168.10.2-00000027'
  == Using SIP RTP CoS mark 5
    -- Executing [2401@default:1] Dial("SIP/192.168.10.2-0000002b", "SIP/kiev/2401") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/kiev/2401
    -- SIP/kiev-0000002c is ringing
    -- SIP/kiev-0000002c answered SIP/192.168.10.2-0000002b
    -- Locally bridging SIP/192.168.10.2-0000002b and SIP/kiev-0000002c
  == Spawn extension (default, 2401, 1) exited non-zero on 'SIP/192.168.10.2-0000002b'
  == Using SIP RTP CoS mark 5
    -- Executing [2301@default:1] Dial("SIP/192.168.10.2-0000002d", "SIP/kiev/2301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/kiev/2301
    -- SIP/kiev-0000002e is ringing
    -- SIP/kiev-0000002e answered SIP/192.168.10.2-0000002d
    -- Locally bridging SIP/192.168.10.2-0000002d and SIP/kiev-0000002e
  == Spawn extension (default, 2301, 1) exited non-zero on 'SIP/192.168.10.2-0000002d'
  == Using SIP RTP CoS mark 5
    -- Executing [2101@default:1] Dial("SIP/192.168.10.2-0000002f", "SIP/krakow/2101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/krakow/2101
    -- SIP/krakow-00000030 answered SIP/192.168.10.2-0000002f
    -- Locally bridging SIP/192.168.10.2-0000002f and SIP/krakow-00000030
  == Spawn extension (default, 2101, 1) exited non-zero on 'SIP/192.168.10.2-0000002f'
  == Using SIP RTP CoS mark 5
    -- Executing [2101@default:1] Dial("SIP/192.168.10.2-00000031", "SIP/krakow/2101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/krakow/2101
    -- SIP/krakow-00000032 answered SIP/192.168.10.2-00000031
    -- Locally bridging SIP/192.168.10.2-00000031 and SIP/krakow-00000032
  == Using SIP RTP CoS mark 5
    -- Executing [1401@default:1] Dial("SIP/2201-00000033", "SIP/minsk/1401") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/minsk/1401
    -- Got SIP response 486 "Busy Here" back from 192.168.60.1:1035
    -- SIP/minsk-00000034 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'SIP/2201-00000033' status is 'BUSY'
  == Spawn extension (default, 2101, 1) exited non-zero on 'SIP/192.168.10.2-00000031'
  == Using SIP RTP CoS mark 5
    -- Executing [2101@default:1] Dial("SIP/192.168.30.2-00000035", "SIP/krakow/2101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/krakow/2101
    -- SIP/krakow-00000036 answered SIP/192.168.30.2-00000035
    -- Locally bridging SIP/192.168.30.2-00000035 and SIP/krakow-00000036
  == Using SIP RTP CoS mark 5
    -- Executing [2101@default:1] Dial("SIP/192.168.70.2-00000037", "SIP/krakow/2101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/krakow/2101
    -- SIP/krakow-00000038 answered SIP/192.168.70.2-00000037
    -- Locally bridging SIP/192.168.70.2-00000037 and SIP/krakow-00000038
  == Spawn extension (default, 2101, 1) exited non-zero on 'SIP/192.168.30.2-00000035'
  == Using SIP RTP CoS mark 5
    -- Executing [1201@default:1] Dial("SIP/192.168.30.2-00000039", "SIP/rome/1201") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/rome/1201
    -- SIP/rome-0000003a is ringing
  == Spawn extension (default, 2101, 1) exited non-zero on 'SIP/192.168.70.2-00000037'
    -- SIP/rome-0000003a answered SIP/192.168.30.2-00000039
    -- Locally bridging SIP/192.168.30.2-00000039 and SIP/rome-0000003a
  == Spawn extension (default, 1201, 1) exited non-zero on 'SIP/192.168.30.2-00000039'
asterisk6*CLI> quit
Asterisk cleanly ending (0).
Executing last minute cleanups