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

Luciano Cavalcante Souza lucindio em gmail.com
Segunda Abril 6 11:13:26 BRT 2015


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 SouzaTecnólogo em Gestão da Tecnologia da Informação*



*Mobile: + 55798814.5895(vivo) e-mail: lucindio at gmail.com
<lucindio at gmail.com>Skype: lucindio**Concentre-se nos pontos FORTES,
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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@
> -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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> _______________________________________________
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> 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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