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</o:shapelayout></xml><![endif]--></head><body lang=PT-BR link=blue vlink=purple><div class=WordSection1><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Estou usando o </span> PCMU que acredito ser o ULAW <o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><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>Luciano Cavalcante Souza<br><b>Enviada em:</b> segunda-feira, 6 de abril de 2015 11:13<br><b>Para:</b> asteriskbrasil@listas.asteriskbrasil.org<br><b>Assunto:</b> Re: [AsteriskBrasil] ATA GrandsStream HT-503 V1.4A nao atende ligações<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Qual o codec do seu tronco de entrada e o codec do ramal no freepbx como tamvem no ht503?<o:p></o:p></p><div><p class=MsoNormal>Se todos forem g711 blz ira passar normal.<o:p></o:p></p></div></div><div><p class=MsoNormal><br clear=all><o:p></o:p></p><div><div><div><div><div><div><div><div><div><div><div><div><div><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><b><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Sds,</span></i></b><span style='font-family:"Tahoma","sans-serif"'><br></span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>Luciano Cavalcante Souza<br>Tecnólogo em Gestão da Tecnologia da Informação</span></b><span style='font-family:"Tahoma","sans-serif"'><br></span><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:red'>Mobile: + 55798814.5895(vivo) <br>e-mail: </span></i><b><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'><a href="mailto:lucindio@gmail.com" target="_blank">lucindio@gmail.com</a></span></i></b><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:red'><br>Skype: </span></i><b><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>lucindio</span></i></b><i><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:red'><br>Concentre-se nos pontos FORTES, reconheça as FRAQUEZAS, agarre as OPORTUNIDADES e proteja-se contra as AMEAÇAS.</span></i><span style='font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>2015-04-06 9:33 GMT-03:00 Estefanio Brunhara <<a href="mailto:estefanio@brunhara.com" target="_blank">estefanio@brunhara.com</a>>:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Bom dia, lista!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Configurei meu FreePbx bem básico, estou conseguindo fazer ligações, porém meu ATA não atende ligações originada na linha física.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Alguém poderia me dizer o que faltou?<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Pergunta, mesmo se o FreePbx estivesse configurado errado (rota de entrada) o ata teria que pelo menos atender a ligação?<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>#### A configuração da porta FXO <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <span lang=EN-US>Number of Rings:1 (1-50. Default 4)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> PSTN Ring Thru FXS: (x) No Yes (Default Yes)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> PSTN Ring Thru Delay (sec): 1 (1-10 seconds. </span>Default 4 seconds)<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>######### A configuração completa do ATA<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Account Active: No Yes<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Primary SIP Server: 192.168.77.169 (e.g., <a href="http://sip.mycompany.com" target="_blank">sip.mycompany.com</a>, or IP address)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Failover SIP Server: 192.168.77.169 (Optional, used when primary server no response)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Prefer Primary SIP Server: No (x) Yes ( yes - will register to Primary Server if Failover registration expires)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Outbound Proxy: (e.g., <a href="http://proxy.myprovider.com" target="_blank">proxy.myprovider.com</a>, or IP address, if any)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP Transport: (x)UDP TCP TLS (default is UDP)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>NAT Traversal: (x)No Keep-Alive STUN UPnP</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP User ID: 1111 (the user part of an SIP address)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Authenticate ID: 1111 (can be identical to or different from SIP User ID)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Authenticate Password: xxxx (purposely not displayed for security protection)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Name: (optional, e.g., John Doe)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>DNS Mode: (x) A Record SRV NAPTR/SRV Use Configured IP</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Primary IP: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Backup IP1: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Backup IP2: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Tel URI: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP Registration: No (x) Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Unregister On Reboot: No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Outgoing Call without Registration: No (x) Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Register Expiration: 60 (in minutes. default 1 hour, max 45 days)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Reregister before Expiration: 0 (in seconds. Default 0 second)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP Registration Failure Retry Wait Time: 20 (in seconds. Between 1-3600, default is 20)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Local SIP port: 6062 (default 5062)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Local RTP port: 5012 (1024-65535, default 5012)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Use Random Port: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Remove OBP from Route Header: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Support SIP Instance ID: No (x) Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Validate Incoming SIP Message: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Check SIP User ID for incoming INVITE: (x) No Yes (no direct IP calling if Yes)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Authenticate incoming INVITE: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Allow Incoming SIP Messages</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>from SIP Proxy Only: (x) No Yes (no direct IP calling if Yes)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP T1 Timeout: 0.5 </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SIP T2 Interval: 4 </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>DTMF Payload Type: 101 </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Preferred DTMF method:</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>(in listed order) </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Priority 1: RFC2833</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Priority 2: SIP INFO</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Priority 3: In-audio</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Disable DTMF Negotiation: (x) No (default, negotiate with peer) Yes (use above DTMF order without negotiation)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Proxy-Require: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Use NAT IP: (used in SIP/SDP message if specified)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Use SIP User-Agent Header: </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Ring Timeout: 60 (10-300, default is 60 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Early Dial: (x) No Yes (use "Yes" only if proxy supports 484 response)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Dial Plan Prefix: (this prefix string is added to each dialed number)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Use # as Dial Key: No (x) Yes (if set to Yes, "#" will function as the "Dial" key)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Dial Plan: { x+ | *x+ | *xx*x+ } </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SUBSCRIBE for MWI: (x) No, do not send SUBSCRIBE for Message Waiting Indication</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Yes, send periodical SUBSCRIBE for Message Waiting Indication</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Anonymous Call Rejection: (x) No Yes </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Special Feature: Standard </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Session Expiration: 180 (in seconds. default 180 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Min-SE: 90 (in seconds. default and minimum 90 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Caller Request Timer: (x) No Yes (Request for timer when making outbound calls)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Callee Request Timer: (x)No Yes (When caller supports timer but did not request one)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Force Timer: (x) No Yes (Use timer even when remote party does not support)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>UAC Specify Refresher: UAC UAS (x) Omit (Recommended)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>UAS Specify Refresher: (x) UAC UAS (When UAC did not specify refresher tag)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Force INVITE: (x)No Yes (Always refresh with INVITE instead of UPDATE)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>INVITE Ring-No-Answer Timeout (sec): 40 (5-300 seconds. Default 40 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Enable 100rel: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Use First Matching Vocoder in 200OK SDP: (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Preferred Vocoder:</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>(in listed order) </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 1: PCMU</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 2: PCMA</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 3: G723</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 4: G729</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 5: G726-32</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 6: ILBC</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 7: G729E</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> choice 8: AAL2-G726-16</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Voice Frames per TX: 2 ( default 2, from 1 to 4 for G711/G726/G729)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>G723 Rate: (x) 6.3kbps encoding rate 5.3kbps encoding rate</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>iLBC Frame Size: (x) 20ms 30ms</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>iLBC Payload Type: 97 (between 96 and 127, default is 97)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>AAL2-G726-16 Payload Type: 100 (between 96 and 127, default is 100)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>AAL2-G726-24 Payload Type: 99 (between 96 and 127, default is 99)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>AAL2-G726-32 payload type: 104 (between 96 and 127, default is 104)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>AAL2-G726-40 Payload Type: 103 (between 96 and 127, default is 103)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>G729E Payload Type: 102 (between 96 and 127, default is 102)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>VAD: (x)No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Symmetric RTP: (x)No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Fax Mode: (x) T.38 (Auto Detect) Pass-Through</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Fax Tone Detection Mode: Caller (x)Callee Caller or Callee</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Jitter Buffer Type: Fixed (x) Adaptive</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Jitter Buffer Length: Low (x) Medium High</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>SRTP Mode: (x) Disabled Enabled but not forced Enabled and forced</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Caller ID Scheme: Bellcore/Telcodia </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>FSK Caller ID Minimum RX Level (dB): -40 (-96 - 0dB. Default -40dB)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>FSK Caller ID Seizure Bits:70 (0 - 800 bits. Default 70)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>FSK Caller ID Mark Bits: 40 (1 - 800 bits. Default 40)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Caller ID Transport Type: Relay via SIP From </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Send Hook Flash To PSTN: (x) No Yes (If Yes, hook flash will be sent to PSTN upon receiving flash event from RFC2833 or SIP INFO)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Hook Flash Duration (ms): 600 (200 - 1500 milliseconds. Default 600)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Gain:0 TX RX0</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Disable Line Echo Canceller (LEC): (x) No Yes</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> FXO Termination</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Enable Current Disconnect: No (x)Yes (Default Yes. If set to yes, enter threshold below)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Current Disconnect Threshold (ms):100 (50-800 milliseconds. Default 100 milliseconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Enable PSTN Disconnect Tone Detection: (x) No Yes (Default No)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (If set to yes, the following tone is used as the disconnect signal)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> PSTN Disconnect Tone: f1=425@-32,f2=0@-32,c=500/500 </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (Default: Busy Tone: f1=480@-32,f2=620@-32,c=500/500;)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> AC Termination Model Country-based (x) Impedance-based (Default Country-based )</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Country-based USA </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Impedance-based 900R 900ohms </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Number of Rings:1 (1-50. Default 4)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> PSTN Ring Thru FXS: (x) No Yes (Default Yes)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> PSTN Ring Thru Delay (sec): 1 (1-10 seconds. Default 4 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Channel Dialing</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> DTMF Digit Length (ms): 100 (40-127 milliseconds, Default 100 milliseconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> DTMF Dial Pause (ms): 100 (40-127 milliseconds, Default 100 milliseconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> First Digit Timeout (sec):10 (1-20 seconds. Default 10 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Inter-Digit Timeout (sec): 4 (1-15 seconds. Default 4 seconds)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Wait for Dial-Tone: (x) No Yes (Default Yes - dial upon dial-tone)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Stage Method (1/2): 1 (Default 2 - 2 stage dialing)</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> Min Delay Before Dial PSTN Number: 500 (default 500ms, range 50 ~ 65000ms)</span><o:p></o:p></p></div></div><p class=MsoNormal><br>_______________________________________________<br>WORKOFFEE KHOMP: A Khomp renovou sua agenda de workshops<br>gratuitos em 2015. Participe da próxima edição no Rio de<br>Janeiro, dia 10 de abril, e conheça o lançamento UMG 100.<br>Garanta a sua vaga e saiba mais em: <a href="http://www.workoffee.com.br" target="_blank">www.workoffee.com.br</a><br>_______________________________________________<br>DIGIVOICE: Fabricante pioneiro em Banco de Canais e Placas E1, GSM, FXO e FXS para Asterisk e Elastix. Temos Cursos de Telefonia IP e Asterisk.<br>Construa soluções de PABX IP com produtos DigiVoice - visite <a href="http://www.digivoice.com.br" target="_blank">www.digivoice.com.br</a><br>_______________________________________________<br>Para remover seu email desta lista, basta enviar um email em branco para <a href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>