<div dir="ltr">Qual o codec do seu tronco de entrada e o codec do ramal no freepbx como tamvem no ht503?<div>Se todos forem g711 blz ira passar normal.</div></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><font face="Tahoma"><div>
<i style="font-size:10pt"><b>Sds,</b></i><br><span style="font-size:10pt;color:rgb(0,0,255)"><b>Luciano Cavalcante Souza<br>Tecnólogo em Gestão da Tecnologia da Informação</b></span><br><span style="font-size:10pt;color:rgb(255,0,0)"><i>Mobile: + 55798814.5895(vivo) <br>e-mail: <b><span style="color:rgb(0,0,255)"><a href="mailto:lucindio@gmail.com" target="_blank">lucindio@gmail.com</a></span></b><br>Skype: <b><span style="color:rgb(0,0,255)">lucindio</span></b><br></i></span><font color="#ff0000"><span style="font-size:13.3333330154419px"><i>Concentre-se nos pontos FORTES, reconheça as FRAQUEZAS, agarre as OPORTUNIDADES e proteja-se contra as AMEAÇAS.</i></span></font><br><br><div style="font-size:10pt"><font color="#3366cc" face="Lucida Handwriting"><b><i>
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<br><div class="gmail_quote">2015-04-06 9:33 GMT-03:00 Estefanio Brunhara <span dir="ltr"><<a href="mailto:estefanio@brunhara.com" target="_blank">estefanio@brunhara.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="PT-BR" link="blue" vlink="purple"><div><p class="MsoNormal"><span lang="EN-US">Bom dia, lista!<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal">Configurei meu FreePbx bem básico, estou conseguindo fazer ligações, porém meu ATA não atende ligações originada na linha física.<u></u><u></u></p><p class="MsoNormal">Alguém poderia me dizer o que faltou?<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Pergunta, mesmo se o FreePbx estivesse configurado errado (rota de entrada) o ata teria que pelo menos atender a ligação?<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">#### A configuração da porta FXO <u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"> <span lang="EN-US">Number of Rings:1 (1-50. Default 4)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> PSTN Ring Thru FXS: (x) No Yes (Default Yes)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> PSTN Ring Thru Delay (sec): 1 (1-10 seconds. </span>Default 4 seconds)<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">######### A configuração completa do ATA<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Account Active: No Yes<u></u><u></u></p><p class="MsoNormal"><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)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Failover SIP Server: 192.168.77.169 (Optional, used when primary server no response)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Prefer Primary SIP Server: No (x) Yes ( yes - will register to Primary Server if Failover registration expires)<u></u><u></u></span></p><p class="MsoNormal"><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)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP Transport: (x)UDP TCP TLS (default is UDP)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">NAT Traversal: (x)No Keep-Alive STUN UPnP<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP User ID: 1111 (the user part of an SIP address)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Authenticate ID: 1111 (can be identical to or different from SIP User ID)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Authenticate Password: xxxx (purposely not displayed for security protection)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Name: (optional, e.g., John Doe)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US">DNS Mode: (x) A Record SRV NAPTR/SRV Use Configured IP<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Primary IP: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Backup IP1: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Backup IP2: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Tel URI: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP Registration: No (x) Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Unregister On Reboot: No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Outgoing Call without Registration: No (x) Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Register Expiration: 60 (in minutes. default 1 hour, max 45 days)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Reregister before Expiration: 0 (in seconds. Default 0 second)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP Registration Failure Retry Wait Time: 20 (in seconds. Between 1-3600, default is 20)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Local SIP port: 6062 (default 5062)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Local RTP port: 5012 (1024-65535, default 5012)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Use Random Port: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Remove OBP from Route Header: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Support SIP Instance ID: No (x) Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Validate Incoming SIP Message: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Check SIP User ID for incoming INVITE: (x) No Yes (no direct IP calling if Yes)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Authenticate incoming INVITE: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Allow Incoming SIP Messages<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">from SIP Proxy Only: (x) No Yes (no direct IP calling if Yes)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP T1 Timeout: 0.5 <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SIP T2 Interval: 4 <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">DTMF Payload Type: 101 <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Preferred DTMF method:<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">(in listed order) <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Priority 1: RFC2833<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Priority 2: SIP INFO<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Priority 3: In-audio<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US">Disable DTMF Negotiation: (x) No (default, negotiate with peer) Yes (use above DTMF order without negotiation)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Proxy-Require: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Use NAT IP: (used in SIP/SDP message if specified)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Use SIP User-Agent Header: <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Ring Timeout: 60 (10-300, default is 60 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Early Dial: (x) No Yes (use "Yes" only if proxy supports 484 response)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Dial Plan Prefix: (this prefix string is added to each dialed number)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Use # as Dial Key: No (x) Yes (if set to Yes, "#" will function as the "Dial" key)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Dial Plan: { x+ | *x+ | *xx*x+ } <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SUBSCRIBE for MWI: (x) No, do not send SUBSCRIBE for Message Waiting Indication<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Yes, send periodical SUBSCRIBE for Message Waiting Indication<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Anonymous Call Rejection: (x) No Yes <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Special Feature: Standard <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Session Expiration: 180 (in seconds. default 180 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Min-SE: 90 (in seconds. default and minimum 90 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Caller Request Timer: (x) No Yes (Request for timer when making outbound calls)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Callee Request Timer: (x)No Yes (When caller supports timer but did not request one)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Force Timer: (x) No Yes (Use timer even when remote party does not support)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">UAC Specify Refresher: UAC UAS (x) Omit (Recommended)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">UAS Specify Refresher: (x) UAC UAS (When UAC did not specify refresher tag)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Force INVITE: (x)No Yes (Always refresh with INVITE instead of UPDATE)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">INVITE Ring-No-Answer Timeout (sec): 40 (5-300 seconds. Default 40 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Enable 100rel: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US">Use First Matching Vocoder in 200OK SDP: (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Preferred Vocoder:<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">(in listed order) <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 1: PCMU<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 2: PCMA<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 3: G723<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 4: G729<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 5: G726-32<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 6: ILBC<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 7: G729E<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> choice 8: AAL2-G726-16<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Voice Frames per TX: 2 ( default 2, from 1 to 4 for G711/G726/G729)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">G723 Rate: (x) 6.3kbps encoding rate 5.3kbps encoding rate<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">iLBC Frame Size: (x) 20ms 30ms<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">iLBC Payload Type: 97 (between 96 and 127, default is 97)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">AAL2-G726-16 Payload Type: 100 (between 96 and 127, default is 100)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">AAL2-G726-24 Payload Type: 99 (between 96 and 127, default is 99)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">AAL2-G726-32 payload type: 104 (between 96 and 127, default is 104)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">AAL2-G726-40 Payload Type: 103 (between 96 and 127, default is 103)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">G729E Payload Type: 102 (between 96 and 127, default is 102)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US">VAD: (x)No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Symmetric RTP: (x)No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Fax Mode: (x) T.38 (Auto Detect) Pass-Through<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Fax Tone Detection Mode: Caller (x)Callee Caller or Callee<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Jitter Buffer Type: Fixed (x) Adaptive<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Jitter Buffer Length: Low (x) Medium High<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">SRTP Mode: (x) Disabled Enabled but not forced Enabled and forced<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US">Caller ID Scheme: Bellcore/Telcodia <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">FSK Caller ID Minimum RX Level (dB): -40 (-96 - 0dB. Default -40dB)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">FSK Caller ID Seizure Bits:70 (0 - 800 bits. Default 70)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">FSK Caller ID Mark Bits: 40 (1 - 800 bits. Default 40)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Caller ID Transport Type: Relay via SIP From <u></u><u></u></span></p><p class="MsoNormal"><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)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Hook Flash Duration (ms): 600 (200 - 1500 milliseconds. Default 600)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Gain:0 TX RX0<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US">Disable Line Echo Canceller (LEC): (x) No Yes<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US"> FXO Termination<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Enable Current Disconnect: No (x)Yes (Default Yes. If set to yes, enter threshold below)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Current Disconnect Threshold (ms):100 (50-800 milliseconds. Default 100 milliseconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Enable PSTN Disconnect Tone Detection: (x) No Yes (Default No)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (If set to yes, the following tone is used as the disconnect signal)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> PSTN Disconnect Tone: f1=425@-32,f2=0@-32,c=500/500 <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (Default: Busy Tone: f1=480@-32,f2=620@-32,c=500/500;)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US"> AC Termination Model Country-based (x) Impedance-based (Default Country-based )<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Country-based USA <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Impedance-based 900R 900ohms <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Number of Rings:1 (1-50. Default 4)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> PSTN Ring Thru FXS: (x) No Yes (Default Yes)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> PSTN Ring Thru Delay (sec): 1 (1-10 seconds. Default 4 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Channel Dialing<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> DTMF Digit Length (ms): 100 (40-127 milliseconds, Default 100 milliseconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> DTMF Dial Pause (ms): 100 (40-127 milliseconds, Default 100 milliseconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> First Digit Timeout (sec):10 (1-20 seconds. Default 10 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Inter-Digit Timeout (sec): 4 (1-15 seconds. Default 4 seconds)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Wait for Dial-Tone: (x) No Yes (Default Yes - dial upon dial-tone)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Stage Method (1/2): 1 (Default 2 - 2 stage dialing)<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-US"> Min Delay Before Dial PSTN Number: 500 (default 500ms, range 50 ~ 65000ms)<u></u><u></u></span></p></div></div><br>_______________________________________________<br>
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