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</o:shapelayout></xml><![endif]--></head><body lang=PT-BR link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US>Bom dia, lista!<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal>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>Alguém poderia me dizer o que faltou?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>#### A configuração da porta FXO <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> <span lang=EN-US>Number of Rings:1 (1-50. Default 4)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> PSTN Ring Thru FXS: (x) No Yes (Default Yes)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><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><o:p> </o:p></p><p class=MsoNormal>######### A configuração completa do ATA<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Account Active: No Yes<o:p></o:p></p><p class=MsoNormal><span lang=EN-US>Primary SIP Server: 192.168.77.169 (e.g., sip.mycompany.com, or IP address)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Failover SIP Server: 192.168.77.169 (Optional, used when primary server no response)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Outbound Proxy: (e.g., proxy.myprovider.com, or IP address, if any)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP Transport: (x)UDP TCP TLS (default is UDP)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>NAT Traversal: (x)No Keep-Alive STUN UPnP<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP User ID: 1111 (the user part of an SIP address)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Authenticate ID: 1111 (can be identical to or different from SIP User ID)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Authenticate Password: xxxx (purposely not displayed for security protection)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Name: (optional, e.g., John Doe)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>DNS Mode: (x) A Record SRV NAPTR/SRV Use Configured IP<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Primary IP: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Backup IP1: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Backup IP2: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Tel URI: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP Registration: No (x) Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Unregister On Reboot: No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Outgoing Call without Registration: No (x) Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Register Expiration: 60 (in minutes. default 1 hour, max 45 days)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Reregister before Expiration: 0 (in seconds. Default 0 second)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP Registration Failure Retry Wait Time: 20 (in seconds. Between 1-3600, default is 20)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Local SIP port: 6062 (default 5062)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Local RTP port: 5012 (1024-65535, default 5012)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Use Random Port: (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Remove OBP from Route Header: (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Support SIP Instance ID: No (x) Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Validate Incoming SIP Message: (x) No Yes<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Authenticate incoming INVITE: (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Allow Incoming SIP Messages<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>from SIP Proxy Only: (x) No Yes (no direct IP calling if Yes)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP T1 Timeout: 0.5 <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SIP T2 Interval: 4 <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>DTMF Payload Type: 101 <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Preferred DTMF method:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>(in listed order) <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Priority 1: RFC2833<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Priority 2: SIP INFO<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Priority 3: In-audio<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Proxy-Require: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Use NAT IP: (used in SIP/SDP message if specified)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Use SIP User-Agent Header: <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Ring Timeout: 60 (10-300, default is 60 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Early Dial: (x) No Yes (use "Yes" only if proxy supports 484 response)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Dial Plan Prefix: (this prefix string is added to each dialed number)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Dial Plan: { x+ | *x+ | *xx*x+ } <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SUBSCRIBE for MWI: (x) No, do not send SUBSCRIBE for Message Waiting Indication<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Yes, send periodical SUBSCRIBE for Message Waiting Indication<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Anonymous Call Rejection: (x) No Yes <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Special Feature: Standard <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Session Expiration: 180 (in seconds. default 180 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Min-SE: 90 (in seconds. default and minimum 90 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Caller Request Timer: (x) No Yes (Request for timer when making outbound calls)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Callee Request Timer: (x)No Yes (When caller supports timer but did not request one)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Force Timer: (x) No Yes (Use timer even when remote party does not support)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>UAC Specify Refresher: UAC UAS (x) Omit (Recommended)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>UAS Specify Refresher: (x) UAC UAS (When UAC did not specify refresher tag)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Force INVITE: (x)No Yes (Always refresh with INVITE instead of UPDATE)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>INVITE Ring-No-Answer Timeout (sec): 40 (5-300 seconds. Default 40 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Enable 100rel: (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Use First Matching Vocoder in 200OK SDP: (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Preferred Vocoder:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>(in listed order) <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 1: PCMU<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 2: PCMA<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 3: G723<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 4: G729<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 5: G726-32<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 6: ILBC<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 7: G729E<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> choice 8: AAL2-G726-16<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Voice Frames per TX: 2 ( default 2, from 1 to 4 for G711/G726/G729)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>G723 Rate: (x) 6.3kbps encoding rate 5.3kbps encoding rate<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>iLBC Frame Size: (x) 20ms 30ms<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>iLBC Payload Type: 97 (between 96 and 127, default is 97)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>AAL2-G726-16 Payload Type: 100 (between 96 and 127, default is 100)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>AAL2-G726-24 Payload Type: 99 (between 96 and 127, default is 99)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>AAL2-G726-32 payload type: 104 (between 96 and 127, default is 104)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>AAL2-G726-40 Payload Type: 103 (between 96 and 127, default is 103)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>G729E Payload Type: 102 (between 96 and 127, default is 102)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>VAD: (x)No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Symmetric RTP: (x)No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Fax Mode: (x) T.38 (Auto Detect) Pass-Through<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Fax Tone Detection Mode: Caller (x)Callee Caller or Callee<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Jitter Buffer Type: Fixed (x) Adaptive<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Jitter Buffer Length: Low (x) Medium High<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>SRTP Mode: (x) Disabled Enabled but not forced Enabled and forced<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Caller ID Scheme: Bellcore/Telcodia <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>FSK Caller ID Minimum RX Level (dB): -40 (-96 - 0dB. Default -40dB)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>FSK Caller ID Seizure Bits:70 (0 - 800 bits. Default 70)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>FSK Caller ID Mark Bits: 40 (1 - 800 bits. Default 40)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Caller ID Transport Type: Relay via SIP From <o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Hook Flash Duration (ms): 600 (200 - 1500 milliseconds. Default 600)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Gain:0 TX RX0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Disable Line Echo Canceller (LEC): (x) No Yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US> FXO Termination<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Enable Current Disconnect: No (x)Yes (Default Yes. If set to yes, enter threshold below)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Current Disconnect Threshold (ms):100 (50-800 milliseconds. Default 100 milliseconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Enable PSTN Disconnect Tone Detection: (x) No Yes (Default No)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> (If set to yes, the following tone is used as the disconnect signal)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> PSTN Disconnect Tone: f1=425@-32,f2=0@-32,c=500/500 <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> (Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> (Default: Busy Tone: f1=480@-32,f2=620@-32,c=500/500;)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US> AC Termination Model Country-based (x) Impedance-based (Default Country-based )<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Country-based USA <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Impedance-based 900R 900ohms <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Number of Rings:1 (1-50. Default 4)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> PSTN Ring Thru FXS: (x) No Yes (Default Yes)<o:p></o:p></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)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> PSTN Ring Thru Delay (sec): 1 (1-10 seconds. Default 4 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US> Channel Dialing<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> DTMF Digit Length (ms): 100 (40-127 milliseconds, Default 100 milliseconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> DTMF Dial Pause (ms): 100 (40-127 milliseconds, Default 100 milliseconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> First Digit Timeout (sec):10 (1-20 seconds. Default 10 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Inter-Digit Timeout (sec): 4 (1-15 seconds. Default 4 seconds)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Wait for Dial-Tone: (x) No Yes (Default Yes - dial upon dial-tone)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Stage Method (1/2): 1 (Default 2 - 2 stage dialing)<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> Min Delay Before Dial PSTN Number: 500 (default 500ms, range 50 ~ 65000ms)<o:p></o:p></span></p></div></body></html>