<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'>
<div>Quanto ao Log de autenticaçao do Trunk, tem algum comando especifico?</div><div><br></div><div><b>Status Ramais</b></div><div><br></div><div><div>Name/username Host Dyn Nat ACL Port Status</div><div>RAMAL IP RAMAL D 5060 OK (19 ms)</div><div>RAMAL IP RAMAL D 5060 OK (13 ms)</div><div>RAMAL IP RAMAL D 5060 OK (14 ms)</div><div>RAMAL IP RAMAL D 5060 OK (15 ms)</div><div>RAMAL IP RAMAL D 5060 OK (22 ms)</div><div>RAMAL IP RAMAL D 5060 OK (13 ms)</div><div>RAMAL IP RAMAL D 5060 OK (15 ms)</div><div>RAMAL IP RAMAL D 5060 OK (14 ms)</div><div>RAMAL IP RAMAL D 5060 OK (11 ms)</div><div>RAMAL IP RAMAL D 5060 OK (14 ms)</div><div>RAMAL IP RAMAL D 5060 OK (14 ms)</div><div>RAMAL IP RAMAL D 5060 OK (14 ms)</div><div>RAMAL IP RAMAL D 5060 OK (12 ms)</div><div>RAMAL IP RAMAL D 5060 OK (24 ms)</div><div>RAMAL IP RAMAL D 5060 OK (10 ms)</div><div>RAMAL IP RAMAL D 5060 OK (15 ms)</div><div>RAMAL IP RAMAL D 5060 OK (11 ms)</div><div>RAMAL IP RAMAL D 5060 OK (13 ms)</div><div>RAMAL (Unspecified) D N 5060 UNKNOWN</div><div>RAMAL (Unspecified) D 5060 UNKNOWN</div><div>RAMAL (Unspecified) D N 5060 UNKNOWN</div><div>RAMAL (Unspecified) D N 5060 UNKNOWN</div><div>RAMAL IP RAMAL D N 62653 OK (58 ms)</div><div>RAMAL (Unspecified) D N 5060 UNKNOWN</div><div>trunk-G1/XXXX IP OPERADORA N 5060 OK (8 ms)</div><div>trunk-sps-G1 IP OPERADORA N 5060 OK (8 ms)</div></div><div><br></div><div><br></div><b>Log da Chamada</b><br><br><div><div> Now forwarding SIP/XXX-00000054 to 'Local/XXXXXXXXXXX@from-trunk' (thanks to SIP/trunk-G1-00000055)</div><div>[Dec 16 11:42:30] NOTICE[3206]: chan_local.c:550 local_call: No such extension/context XXXXXXXXXXX@from-trunk while calling Local channel</div><div>[Dec 16 11:42:30] NOTICE[3206]: app_dial.c:787 do_forward: Failed to dial on local channel for call forward to 'XXXXXXXXXXX@from-trunk'</div><div> == Everyone is busy/congested at this time (1:0/0/1)</div><div> -- Executing [1-dial@macro-trunkdial-failover-0.3:5] Goto("SIP/XXX-00000054", "1-CHANUNAVAIL,1") in new stack</div><div> -- Goto (macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1)</div><div> -- Executing [1-CHANUNAVAIL@macro-trunkdial-failover-0.3:1] Goto("SIP/XXX-00000054", "2-dial,1)") in new stack</div><div> -- Goto (macro-trunkdial-failover-0.3,2-dial,1)</div><div> -- Executing [2-dial@macro-trunkdial-failover-0.3:1] Set("SIP/XXX-00000054", "TCOUNT=5") in new stack</div><div> -- Executing [2-dial@macro-trunkdial-failover-0.3:2] Goto("SIP/XXX-00000054", "1-dial,1") in new stack</div><div> -- Goto (macro-trunkdial-failover-0.3,1-dial,1)</div><div> -- Executing [1-dial@macro-trunkdial-failover-0.3:1] GotoIf("SIP/XXX-00000054", "1?1-out,1") in new stack</div><div> -- Goto (macro-trunkdial-failover-0.3,1-out,1)</div><div> -- Executing [1-out@macro-trunkdial-failover-0.3:1] Playback("SIP/XXX-00000054", "all-busy-now-try-call-later") in new stack</div><div> -- lintog729_new</div><div> -- use count: 1</div><div> -- <SIP/XXX-00000054> Playing 'all-busy-now-try-call-later.gsm' (language 'en')</div><div> == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/XXX-00000054' in macro 'trunkdial-failover-0.3'</div><div> == Spawn extension (DLPN_DialPlanXXX, XXXXXXXXXXX, 1) exited non-zero on 'SIP/XXX-00000054'</div><div><br></div><br><div><br></div><div><br></div><div>Att;</div><div><br></div><div><b><font class="ecxApple-style-span" face="Tahoma"><span class="ecxApple-style-span" style="font-size:12pt">Gleidison C. Sampaio</span></font></b></div><div><br></div><div><font class="Apple-style-span" color="#17365D"><br></font></div><hr id="stopSpelling">From: sidnei_rp@ig.com.br<br>To: asteriskbrasil@listas.asteriskbrasil.org<br>Date: Thu, 16 Dec 2010 11:37:12 -0200<br>Subject: [AsteriskBrasil] RES: Problema de Registro Trunk VoIP Asterisk 1.6<br><br>
<meta http-equiv="Content-Type" content="text/html; charset=unicode">
<meta name="Generator" content="Microsoft SafeHTML"><style>
.ExternalClass p.ecxMsoNormal, .ExternalClass li.ecxMsoNormal, .ExternalClass div.ecxMsoNormal
{margin-bottom:.0001pt;font-size:12.0pt;font-family:'Times New Roman','serif';}
.ExternalClass a:link, .ExternalClass span.ecxMsoHyperlink
{color:blue;text-decoration:underline;}
.ExternalClass a:visited, .ExternalClass span.ecxMsoHyperlinkFollowed
{color:purple;text-decoration:underline;}
.ExternalClass p
{margin-right:0cm;margin-left:0cm;font-size:12.0pt;font-family:'Times New Roman','serif';}
.ExternalClass span.ecxecxapple-style-span
{;}
.ExternalClass span.ecxEstiloDeEmail19
{font-family:'Calibri','sans-serif';color:#1F497D;}
.ExternalClass .ecxMsoChpDefault
{font-size:10.0pt;}
@page WordSection1
{size:612.0pt 792.0pt;}
.ExternalClass div.ecxWordSection1
{page:WordSection1;}
</style><div class="ecxWordSection1"><p class="ecxMsoNormal"><span style="font-size:11.0pt;font-family:'Calibri','sans-serif';color:#1F497D">Posta os logs ai de chamada, de registro, de ramal...</span></p><p class="ecxMsoNormal"><span style="font-size:11.0pt;font-family:'Calibri','sans-serif';color:#1F497D"> </span></p><div><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="ecxMsoNormal"><b><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'">De:</span></b><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> asteriskbrasil-bounces@listas.asteriskbrasil.org [mailto:asteriskbrasil-bounces@listas.asteriskbrasil.org] <b>Em nome de </b>Gleidison Sampaio<br><b>Enviada em:</b> quinta-feira, 16 de dezembro de 2010 11:21<br><b>Para:</b> Asterisk Lista<br><b>Assunto:</b> [AsteriskBrasil] Problema de Registro Trunk VoIP Asterisk 1.6</span></p></div></div><p class="ecxMsoNormal"> </p><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'">Galera,</span></p><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'">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.</span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'">alguem ja passou por essa situaçao?</span></p><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'">Att;</span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span class="ecxapple-style-span"><b><span style="font-family:'Tahoma','sans-serif'">Gleidison C. Sampaio</span></b></span><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"></span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><div><p class="ecxMsoNormal"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div><p class="ecxMsoNormal" style="margin-bottom:12.0pt"><span style="font-size:10.0pt;font-family:'Tahoma','sans-serif'"> </span></p></div></div></div> </body>
</html>