[AsteriskBrasil] RES: ATA GrandsStream HT-503 V1.4A nao atende ligações
Estefanio Brunhara
estefanio em brunhara.com
Segunda Abril 6 17:22:27 BRT 2015
Estou usando o PCMU que acredito ser o ULAW
De: asteriskbrasil-bounces at listas.asteriskbrasil.org [mailto:asteriskbrasil-bounces at listas.asteriskbrasil.org] Em nome de Luciano Cavalcante Souza
Enviada em: segunda-feira, 6 de abril de 2015 11:13
Para: asteriskbrasil at listas.asteriskbrasil.org
Assunto: Re: [AsteriskBrasil] ATA GrandsStream HT-503 V1.4A nao atende ligações
Qual o codec do seu tronco de entrada e o codec do ramal no freepbx como tamvem no ht503?
Se todos forem g711 blz ira passar normal.
Sds,
Luciano Cavalcante Souza
Tecnólogo em Gestão da Tecnologia da Informação
Mobile: + 55798814.5895(vivo)
e-mail: lucindio at gmail.com
Skype: lucindio
Concentre-se nos pontos FORTES, reconheça as FRAQUEZAS, agarre as OPORTUNIDADES e proteja-se contra as AMEAÇAS.
2015-04-06 9:33 GMT-03:00 Estefanio Brunhara <estefanio at brunhara.com>:
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)
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