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root@server2:~# asterisk -rvvvAsterisk 11.5.0, 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.
=========================================================================
Connected to Asterisk 11.5.0 currently running on server2 (pid = 6164)
    -- Registered SIP '100' at 192.168.1.20:57634
[Jul 26 22:23:48] NOTICE[6196]: chan_sip.c:27699 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 100
  == Using SIP RTP CoS mark 5
    -- Executing [4771390955@from-fly:1] Dial("SIP/46.19.210.14-00000000", "SIP/100") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/100
    -- SIP/100-00000001 is ringing
    -- SIP/100-00000001 answered SIP/46.19.210.14-00000000
    -- Remotely bridging SIP/46.19.210.14-00000000 and SIP/100-00000001
[Jul 26 22:24:45] WARNING[6196]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 03e94395215033ed061c35dd70910161@46.19.210.47-b2b_1 for seqno 200 (Critical Response) -- See https://w      iki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31998ms with no response
[Jul 26 22:24:45] WARNING[6196]: chan_sip.c:4203 retrans_pkt: Hanging up call 03e94395215033ed061c35dd70910161@46.19.210.47-b2b_1 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST      /SIP+Retransmissions).
  == Spawn extension (from-fly, 4771390955, 1) exited non-zero on 'SIP/46.19.210.14-00000000'
server2*CLI>