[AsteriskBrasil] Grandstream HT503
Henrique Oliveira
haooliveira em gmail.com
Terça Abril 26 14:17:18 BRT 2016
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
>
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