Desculpa não entendi!
Hoje esta funcionando desta forma: A configuração de ramal esta OK, falo e recebe de qualquer ramal, consigo disca de qualquer rama para fora, usando a linha de par metálico da OI, porem as ligações que chegam na linha da OI, o ATA não atende Fiz as configurações abaixo no ATA. De: asteriskbrasil-boun...@listas.asteriskbrasil.org [mailto:asteriskbrasil-boun...@listas.asteriskbrasil.org] Em nome de Patrick Enviada em: segunda-feira, 6 de abril de 2015 10:56 Para: asteriskbrasil@listas.asteriskbrasil.org Assunto: Re: [AsteriskBrasil] ATA GrandsStream HT-503 V1.4A nao atende ligações Ta faltando configurei em Basic Setting a extensão e o IP de destino pra jogar pro PABX On 06-04-2015 09:33, Estefanio Brunhara wrote: Bom dia, lista! Configurei meu FreePbx bem básico, estou conseguindo fazer ligações, porém meu ATA não atende ligações originada na linha física. Alguém poderia me dizer o que faltou? Pergunta, mesmo se o FreePbx estivesse configurado errado (rota de entrada) o ata teria que pelo menos atender a ligação? #### A configuração da porta FXO Number of Rings:1 (1-50. Default 4) (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number) PSTN Ring Thru FXS: (x) No Yes (Default Yes) (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay) PSTN Ring Thru Delay (sec): 1 (1-10 seconds. Default 4 seconds) ######### A configuração completa do ATA Account Active: No Yes Primary SIP Server: 192.168.77.169 (e.g., sip.mycompany.com, or IP address) Failover SIP Server: 192.168.77.169 (Optional, used when primary server no response) Prefer Primary SIP Server: No (x) Yes ( yes - will register to Primary Server if Failover registration expires) Outbound Proxy: (e.g., proxy.myprovider.com, or IP address, if any) SIP Transport: (x)UDP TCP TLS (default is UDP) NAT Traversal: (x)No Keep-Alive STUN UPnP SIP User ID: 1111 (the user part of an SIP address) Authenticate ID: 1111 (can be identical to or different from SIP User ID) Authenticate Password: xxxx (purposely not displayed for security protection) Name: (optional, e.g., John Doe) DNS Mode: (x) A Record SRV NAPTR/SRV Use Configured IP Primary IP: Backup IP1: Backup IP2: Tel URI: SIP Registration: No (x) Yes Unregister On Reboot: No Yes Outgoing Call without Registration: No (x) Yes Register Expiration: 60 (in minutes. default 1 hour, max 45 days) Reregister before Expiration: 0 (in seconds. Default 0 second) SIP Registration Failure Retry Wait Time: 20 (in seconds. Between 1-3600, default is 20) Local SIP port: 6062 (default 5062) Local RTP port: 5012 (1024-65535, default 5012) Use Random Port: (x) No Yes Remove OBP from Route Header: (x) No Yes Support SIP Instance ID: No (x) Yes Validate Incoming SIP Message: (x) No Yes Check SIP User ID for incoming INVITE: (x) No Yes (no direct IP calling if Yes) Authenticate incoming INVITE: (x) No Yes Allow Incoming SIP Messages from SIP Proxy Only: (x) No Yes (no direct IP calling if Yes) SIP T1 Timeout: 0.5 SIP T2 Interval: 4 DTMF Payload Type: 101 Preferred DTMF method: (in listed order) Priority 1: RFC2833 Priority 2: SIP INFO Priority 3: In-audio Disable DTMF Negotiation: (x) No (default, negotiate with peer) Yes (use above DTMF order without negotiation) Proxy-Require: Use NAT IP: (used in SIP/SDP message if specified) Use SIP User-Agent Header: Ring Timeout: 60 (10-300, default is 60 seconds) Early Dial: (x) No Yes (use "Yes" only if proxy supports 484 response) Dial Plan Prefix: (this prefix string is added to each dialed number) Use # as Dial Key: No (x) Yes (if set to Yes, "#" will function as the "Dial" key) Dial Plan: { x+ | *x+ | *xx*x+ } SUBSCRIBE for MWI: (x) No, do not send SUBSCRIBE for Message Waiting Indication Yes, send periodical SUBSCRIBE for Message Waiting Indication Anonymous Call Rejection: (x) No Yes Special Feature: Standard Session Expiration: 180 (in seconds. default 180 seconds) Min-SE: 90 (in seconds. default and minimum 90 seconds) Caller Request Timer: (x) No Yes (Request for timer when making outbound calls) Callee Request Timer: (x)No Yes (When caller supports timer but did not request one) Force Timer: (x) No Yes (Use timer even when remote party does not support) UAC Specify Refresher: UAC UAS (x) Omit (Recommended) UAS Specify Refresher: (x) UAC UAS (When UAC did not specify refresher tag) Force INVITE: (x)No Yes (Always refresh with INVITE instead of UPDATE) INVITE Ring-No-Answer Timeout (sec): 40 (5-300 seconds. Default 40 seconds) Enable 100rel: (x) No Yes Use First Matching Vocoder in 200OK SDP: (x) No Yes Preferred Vocoder: (in listed order) choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: ILBC choice 7: G729E choice 8: AAL2-G726-16 Voice Frames per TX: 2 ( default 2, from 1 to 4 for G711/G726/G729) G723 Rate: (x) 6.3kbps encoding rate 5.3kbps encoding rate iLBC Frame Size: (x) 20ms 30ms iLBC Payload Type: 97 (between 96 and 127, default is 97) AAL2-G726-16 Payload Type: 100 (between 96 and 127, default is 100) AAL2-G726-24 Payload Type: 99 (between 96 and 127, default is 99) AAL2-G726-32 payload type: 104 (between 96 and 127, default is 104) AAL2-G726-40 Payload Type: 103 (between 96 and 127, default is 103) G729E Payload Type: 102 (between 96 and 127, default is 102) VAD: (x)No Yes Symmetric RTP: (x)No Yes Fax Mode: (x) T.38 (Auto Detect) Pass-Through Fax Tone Detection Mode: Caller (x)Callee Caller or Callee Jitter Buffer Type: Fixed (x) Adaptive Jitter Buffer Length: Low (x) Medium High SRTP Mode: (x) Disabled Enabled but not forced Enabled and forced Caller ID Scheme: Bellcore/Telcodia FSK Caller ID Minimum RX Level (dB): -40 (-96 - 0dB. Default -40dB) FSK Caller ID Seizure Bits:70 (0 - 800 bits. Default 70) FSK Caller ID Mark Bits: 40 (1 - 800 bits. Default 40) Caller ID Transport Type: Relay via SIP From 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) Hook Flash Duration (ms): 600 (200 - 1500 milliseconds. Default 600) Gain:0 TX RX0 Disable Line Echo Canceller (LEC): (x) No Yes FXO Termination Enable Current Disconnect: No (x)Yes (Default Yes. If set to yes, enter threshold below) Current Disconnect Threshold (ms):100 (50-800 milliseconds. Default 100 milliseconds) Enable PSTN Disconnect Tone Detection: (x) No Yes (Default No) (If set to yes, the following tone is used as the disconnect signal) PSTN Disconnect Tone: f1=425@-32,f2=0@-32,c=500/500 (Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;) (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm) (Default: Busy Tone: f1=480@-32,f2=620@-32,c=500/500;) AC Termination Model Country-based (x) Impedance-based (Default Country-based ) Country-based USA Impedance-based 900R 900ohms Number of Rings:1 (1-50. Default 4) (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number) PSTN Ring Thru FXS: (x) No Yes (Default Yes) (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay) PSTN Ring Thru Delay (sec): 1 (1-10 seconds. Default 4 seconds) Channel Dialing DTMF Digit Length (ms): 100 (40-127 milliseconds, Default 100 milliseconds) DTMF Dial Pause (ms): 100 (40-127 milliseconds, Default 100 milliseconds) First Digit Timeout (sec):10 (1-20 seconds. Default 10 seconds) Inter-Digit Timeout (sec): 4 (1-15 seconds. Default 4 seconds) Wait for Dial-Tone: (x) No Yes (Default Yes - dial upon dial-tone) Stage Method (1/2): 1 (Default 2 - 2 stage dialing) Min Delay Before Dial PSTN Number: 500 (default 500ms, range 50 ~ 65000ms) _______________________________________________ WORKOFFEE KHOMP: A Khomp renovou sua agenda de workshops gratuitos em 2015. Participe da próxima edição no Rio de Janeiro, dia 10 de abril, e conheça o lançamento UMG 100. Garanta a sua vaga e saiba mais em: www.workoffee.com.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. Construa soluções de PABX IP com produtos DigiVoice - visite www.digivoice.com.br _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscr...@listas.asteriskbrasil.org
_______________________________________________ WORKOFFEE KHOMP: A Khomp renovou sua agenda de workshops gratuitos em 2015. Participe da próxima edição no Rio de Janeiro, dia 10 de abril, e conheça o lançamento UMG 100. Garanta a sua vaga e saiba mais em: www.workoffee.com.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. Construa soluções de PABX IP com produtos DigiVoice - visite www.digivoice.com.br _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscr...@listas.asteriskbrasil.org