[AsteriskBrasil] ENC: ATA GrandsStream HT-503 V1.4A nao atende ligações

Estefanio Brunhara estefanio em brunhara.com
Segunda Abril 6 17:26:37 BRT 2015


 

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-bounces at listas.asteriskbrasil.org
[mailto:asteriskbrasil-bounces at listas.asteriskbrasil.org] Em nome de Patrick
Enviada em: segunda-feira, 6 de abril de 2015 10:56
Para: asteriskbrasil at 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 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
_______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para
asteriskbrasil-unsubscribe at listas.asteriskbrasil.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20150406/60a635b3/attachment-0001.html>


Mais detalhes sobre a lista de discussão AsteriskBrasil