[AsteriskBrasil] RES: Problema de Registro Trunk VoIP Asterisk 1.6
Gleidison Sampaio
gleidison.sampaio em hotmail.com
Quinta Dezembro 16 11:51:44 BRST 2010
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/XXXXXXXXXXX em 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 XXXXXXXXXXX em 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 'XXXXXXXXXXX em from-trunk' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1-dial em 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-CHANUNAVAIL em 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-dial em macro-trunkdial-failover-0.3:1] Set("SIP/XXX-00000054", "TCOUNT=5") in new stack -- Executing [2-dial em 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-dial em 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-out em 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_rp em ig.com.br
To: asteriskbrasil em 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-bounces em listas.asteriskbrasil.org [mailto:asteriskbrasil-bounces em 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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