Asterisk no 11.04

[bfbicalho]

Boa tarde pessoal,

Estou precisando de uma ajuda de todos no asterisk.

eu tenho dois ramais internos:

1000 - admin
1001 - suporte

Eu consegui fazer o asterisk funcionar, porém estou tendo problemas para realizar chamadas entre ramais internos. eu fiz uns testes em épocas diferentes e tive 3 resultados diferentes:

As mensagens abaixo referem-se as tentativas de ligações aos ramais internos sem precisar acessar a telefonia comum.

leitura 1
-- Executing [1001@default:1] Macro("SIP/192.168.0.254-00000000", "stdexten,1001,SIP/1001&DAHDI/1") in new stack
[Aug  4 15:55:29] WARNING[26509]: app_macro.c:302 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'
    -- Auto fallthrough, channel 'SIP/192.168.0.254-00000000' status is 'UNKNOWN'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1000@default:1] Dial("SIP/192.168.0.254-00000001", "SIP/1000&DAHDI/1") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1000
[Aug  4 15:55:52] WARNING[26511]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
  == Spawn extension (default, 1000, 1) exited non-zero on 'SIP/192.168.0.254-00000001'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1001@default:1] Macro("SIP/192.168.0.254-00000003", "stdexten,1001,SIP/1001&DAHDI/1") in new stack
[Aug  4 15:57:41] WARNING[26513]: app_macro.c:302 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'
    -- Auto fallthrough, channel 'SIP/192.168.0.254-00000003' status is 'UNKNOWN'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1000@default:1] Dial("SIP/192.168.0.254-00000004", "SIP/1000&DAHDI/1") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1000
[Aug  4 16:00:40] WARNING[26612]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
  == Spawn extension (default, 1000, 1) exited non-zero on 'SIP/192.168.0.254-00000004'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
[Aug  4 16:02:59] NOTICE[2119]: chan_sip.c:20059 handle_request_invite: Call from '' to extension '192*168*0*202' rejected because extension not found.

leitura 2
-- Executing [1000@default:1] Dial("SIP/192.168.0.254-00000000", "SIP/1000&D                                                                             AHDI/1") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1000
[Aug  4 17:16:30] WARNING[1278]: channel.c:4035 ast_request: No channel type reg                                                                             istered for 'DAHDI'
[Aug  4 17:16:30] WARNING[1278]: app_dial.c:1745 dial_exec_full: Unable to creat                                                                             e channel of type 'DAHDI' (cause 66 - Channel not implemented)
  == Spawn extension (default, 1000, 1) exited non-zero on 'SIP/192.168.0.254-00000000'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1000@default:1] Dial("SIP/192.168.0.254-00000002", "SIP/1000&DAHDI/1") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1000
[Aug  4 17:17:30] WARNING[1283]: channel.c:4035 ast_request: No channel type registered for 'DAHDI'
[Aug  4 17:17:30] WARNING[1283]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)
  == Spawn extension (default, 1000, 1) exited non-zero on 'SIP/192.168.0.254-00000002'
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1001@default:1] Macro("SIP/192.168.0.254-00000004", "stdexten,1001,SIP/1001&DAHDI/1") in new stack
[Aug  4 17:18:19] WARNING[1285]: app_macro.c:302 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'
    -- Auto fallthrough, channel 'SIP/192.168.0.254-00000004' status is 'UNKNOWN'

leitura 3
root@asterisk:~# asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.6, Copyright (C) 1999 - 2010 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 1.6.2.6 currently running on asterisk (pid = 2134)
Verbosity was 0 and is now 15
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
[Aug 24 16:45:02] NOTICE[1792]: chan_sip.c:19978 handle_request_invite: Failed to authenticate device "bruno"<sip:admin@192.168.0.254>;tag=707afa12

andei configurando canal para o dadhi e aparece esta mensagem abaixo:
Eu sei que para funcionar de uma maineira mais completa eu preciso ter uma placa E1, pois assim o dadhi funcionará, porém que quero fazer ele funcionar somente no sip e assim poder até mesmo ter uma comunicação interna com alguém diretamente.

alguém sabe o que está ocorrendo?

Quero muito colocar o meu asterisk para funcionar, pois tenho interesse em trabalhar na montagem e configuração deste servidor.

Alguém pode me ajudar?

Obrigado pela ajuda.