Log de registro
[Dec 16 11:52:40] NOTICE[366]: chan_sip.c:21742 handle_request_register:
Registration from '<sip:@XXX.XXX.XXX.XXX:5060>' failed for 'XXX.XXX.XXX.XXX' -
No matching peer found[Dec 16 11:52:40] WARNING[366]: chan_sip.c:18276
handle_response_register: Got 404 Not found on SIP register to service
x...@xxx.xxx.xxx.xxx@XXX.XXX.XXX.XXX, giving up[Dec 16 11:52:40] NOTICE[366]:
chan_sip.c:21742 handle_request_register: Registration from
'<sip:x...@xxx.xxx.xxx.xxx>' failed for 'XXX.XXX.XXX.XXX' - No matching peer
found[Dec 16 11:52:40] WARNING[366]: chan_sip.c:18276 handle_response_register:
Got 404 Not found on SIP register to service
x...@xxx.xxx.xxx.xxx@aut.sipsrv.com.br, giving up[Dec 16 11:52:40] NOTICE[366]:
chan_sip.c:21742 handle_request_register: Registration from
'<sip:x...@xxx.xxx.xxx.xxx:5060>' failed for 'XXX.XXX.XXX.XXX' - No matching
peer found[Dec 16 11:52:40] WARNING[366]: chan_sip.c:18276
handle_response_register: Got 404 Not found on SIP register to
servicex...@xxx.xxx.xxx.xxx@XXX.XXX.XXX.XXX, giving up[Dec 16 11:52:41]
NOTICE[366]: chan_sip.c:18430 handle_response_peerpoke: Peer 'trunk-CARRIER' is
now Reachable. (8ms / 2000ms)
Att;
Gleidison C. Sampaio
From: gleidison.samp...@hotmail.com
To: sidnei...@ig.com.br; asteriskbrasil@listas.asteriskbrasil.org
Date: Thu, 16 Dec 2010 09:51:44 -0400
Subject: Re: [AsteriskBrasil] RES: Problema de Registro Trunk VoIP Asterisk 1.6
Quanto ao Log de autenticaçao do Trunk, tem algum comando especifico?
Status Ramais
Name/username Host Dyn Nat ACL Port StatusRAMAL
IP RAMAL D 5060 OK (19 ms)RAMAL
IP RAMAL D 5060 OK (13 ms)RAMAL IP
RAMAL D 5060 OK (14 ms)RAMAL IP RAMAL
D 5060 OK (15 ms)RAMAL IP RAMAL D
5060 OK (22 ms)RAMAL IP RAMAL D 5060
OK (13 ms)RAMAL IP RAMAL D 5060 OK (15
ms)RAMAL IP RAMAL D 5060 OK (14 ms)RAMAL
IP RAMAL D 5060 OK (11 ms)RAMAL
IP RAMAL D 5060 OK (14 ms)RAMAL IP
RAMAL D 5060 OK (14 ms)RAMAL IP RAMAL
D 5060 OK (14 ms)RAMAL IP RAMAL D
5060 OK (12 ms)RAMAL IP RAMAL D 5060
OK (24 ms)RAMAL IP RAMAL D 5060 OK (10
ms)RAMAL IP RAMAL D 5060 OK (15 ms)RAMAL
IP RAMAL D 5060 OK (11 ms)RAMAL
IP RAMAL D 5060 OK (13 ms)RAMAL
(Unspecified) D N 5060 UNKNOWNRAMAL
(Unspecified) D 5060 UNKNOWNRAMAL
(Unspecified) D N 5060 UNKNOWNRAMAL
(Unspecified) D N 5060 UNKNOWNRAMAL IP RAMAL
D N 62653 OK (58 ms)RAMAL (Unspecified) D
N 5060 UNKNOWNtrunk-G1/XXXX IP OPERADORA N 5060
OK (8 ms)trunk-sps-G1 IP OPERADORA N 5060 OK (8 ms)
Log da Chamada
Now forwarding SIP/XXX-00000054 to 'Local/xxxxxxxx...@from-trunk' (thanks to
SIP/trunk-G1-00000055)[Dec 16 11:42:30] NOTICE[3206]: chan_local.c:550
local_call: No such extension/context xxxxxxxx...@from-trunk while calling
Local channel[Dec 16 11:42:30] NOTICE[3206]: app_dial.c:787 do_forward: Failed
to dial on local channel for call forward to 'xxxxxxxx...@from-trunk' ==
Everyone is busy/congested at this time (1:0/0/1) -- Executing
[1-d...@macro-trunkdial-failover-0.3:5] Goto("SIP/XXX-00000054",
"1-CHANUNAVAIL,1") in new stack -- Goto
(macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1) -- Executing
[1-chanunav...@macro-trunkdial-failover-0.3:1] Goto("SIP/XXX-00000054",
"2-dial,1)") in new stack -- Goto (macro-trunkdial-failover-0.3,2-dial,1)
-- Executing [2-d...@macro-trunkdial-failover-0.3:1] Set("SIP/XXX-00000054",
"TCOUNT=5") in new stack -- Executing
[2-d...@macro-trunkdial-failover-0.3:2] Goto("SIP/XXX-00000054", "1-dial,1") in
new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing
[1-d...@macro-trunkdial-failover-0.3:1] GotoIf("SIP/XXX-00000054", "1?1-out,1")
in new stack -- Goto (macro-trunkdial-failover-0.3,1-out,1) -- Executing
[1-...@macro-trunkdial-failover-0.3:1] Playback("SIP/XXX-00000054",
"all-busy-now-try-call-later") in new stack -- lintog729_new -- use
count: 1 -- <SIP/XXX-00000054> Playing 'all-busy-now-try-call-later.gsm'
(language 'en') == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1)
exited non-zero on 'SIP/XXX-00000054' in macro 'trunkdial-failover-0.3' ==
Spawn extension (DLPN_DialPlanXXX, XXXXXXXXXXX, 1) exited non-zero on
'SIP/XXX-00000054'
Att;
Gleidison C. Sampaio
From: sidnei...@ig.com.br
To: asteriskbrasil@listas.asteriskbrasil.org
Date: Thu, 16 Dec 2010 11:37:12 -0200
Subject: [AsteriskBrasil] RES: Problema de Registro Trunk VoIP Asterisk 1.6
Posta os logs ai de chamada, de registro, de ramal... De:
asteriskbrasil-boun...@listas.asteriskbrasil.org
[mailto:asteriskbrasil-boun...@listas.asteriskbrasil.org] Em nome de Gleidison
Sampaio
Enviada em: quinta-feira, 16 de dezembro de 2010 11:21
Para: Asterisk Lista
Assunto: [AsteriskBrasil] Problema de Registro Trunk VoIP Asterisk 1.6 Galera,
estou com o seguinte problema, tenho um tronco Voip no Asterisk com interface
freePBX, na interface grafica sinaliza que o tronco nao esta logado, com a
mensagem de status "REJECTED" ja quando eu rodo o comando "sip show peers" por
ssh ele aparece que o tronco esta logado, e realmnte verifiquei na operadora
nao esta logado mesmo, ja conferi todas as senhas, estao ok, coloquei o DID da
operadora em um soft fone loga normalmente, ja no asterisk nao loga. alguem ja
passou por essa situaçao? Att; Gleidison C. Sampaio
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