[AsteriskBrasil] Grandstream HT503

[Henrique Oliveira]

André,

Podes testar isso:

Basic Settings

* IP Address: (*) “dynamically assigned via DHCP” ou (*) “statically
configured as” de acordo com a rede.

* Device Mode: (*) Bridge

* Reply to ICMP on WAN port: (Yes)

* WAN side HTTP/Telnet access: (Yes)

* Enable LAN DHCP: (No)
Advances Settings* Call Progress Tones:

Dial Tone: f1=425 at -10,f2=425 at -10,c=0/0;
Ringback Tone: f1=425 at -10,f2=425 at -10,c=1000/4000;
Busy Tone: f1=425 at -10,f2=425 at -10,c=250/250;
Reorder Tone: f1=425 at -10,f2=425 at -10,c=250/250;
Confirmation Tone: f1=350 at -11,f2=440 at -11,c=100/100-100/100-100/100;
Call Waiting Tone: f1=440 at -13,c=300/10000-300/10000-0/0;
Prompt Tone: f1=350 at -13,f2=440 at -13,c=0/0;

FXO Port* Primary SIP Server: (IP_DO_SERVIDOR)

* Failover SIP Server: (IP_DO_SERVIDOR)

* Prefer Primary SIP Server: (Yes)

* SIP Transport: (UDP)

* NAT Traversal: (No)

* SIP User ID: (NUMERO_DO_RAMAL)

* Authenticate ID: (NUMERO_DO_RAMAL)

* Authenticate Password: (SENHA_DO_RAMAL)

* Name: (NUMERO_DO_RAMAL)

* SIP Registration: (Yes)

* Unregister On Reboot: (Yes)

* Support SIP Instance ID: (Yes)

* Preferred DTMF method: (RFC2833) em todos

* Use # as Dial Key: (Yes)

* UAC Specify Refresher: (Omit)

* 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

* Fax Mode: (*) Pass-Through

* Fax Tone Detection Mode: (*) Caller or Callee

* Jitter Buffer Type: (Adaptive)

* Jitter Buffer Length: (Medium)

* Enable Current Disconnect: (Yes)

* Enable PSTN Disconnect Tone Detection: (Yes)

* PSTN Disconnect Tone:

f1=425 at -10,f2=425 at -10,c=250/250;

* AC Termination Model: (Impedance-based)

* Country-based (USA)

* Impedance-based (900R – 900 ohms)

* PSTN Ring Thru FXS: (NO)

*Henrique Antonio de Oliveira*

*Tel: 47-84022327*
*E-Mail: haooliveira at gmail.com <haooliveira at gmail.com>*
*Skype: henrique-o*


Em 26 de abril de 2016 13:55, André Luís Barbosa <andre.lbarbosa75 at gmail.com
> escreveu:

> Estou com este ATA e não estou conseguindo configurar o  mesmo como um
> gateway da PSTN para o asterisk.
> Gostaria que qualquer ligação que chegar na FXO(Line), seja encaminhada
> para o Asterisk.
> Procurei na net alguns procedimentos, mas não consegui.
>
> --
> Atenciosamente,
>
>
>
> André Luís Barbosa
> 81-997278169
> andre.lbarbosa75 at gmail.com
> https://br.linkedin.com/in/andreluisbarbosa
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
> Intercomunicador e acesso remoto via rede IP e telefones IP
> Conheça todo o portfólio em www.Khomp.com
> _______________________________________________
> 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/20160426/6c952cf9/attachment.html>