[AsteriskBrasil] ATA GrandsStream HT-503 V1.4A nao atende ligações
Patrick
wushumasters em gmail.com
Segunda Abril 6 10:55:45 BRT 2015
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 at -32,f2=0 at -32,c=500/500
>
> (Syntax: f1=freq at vol, f2=freq at vol, c=on1/off1-on2/off2-on3/off3;)
>
> (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm)
>
> (Default: Busy Tone: f1=480 at -32,f2=620 at -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
> _______________________________________________
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