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Assessments - 6, GPA: 2.8 ( )

Instruções de Operação Grandstream Networks, Modelo HT704

Fabricante : Grandstream Networks
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Língua de Ensino: en
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Check SIP User ID for Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the incoming INVITE call will be rejected. If this option is enabled, the device will not be able to make direct IP calls. Allow Incoming SIP Messages from SIP Proxy Only Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls. SIP T1 Timeout T1 is an estimate of the round-trip time between the client and server transactions. If the network latency is high, select larger value for more reliable usage. Default is 0.5 Sec. SIP T2 Interval Maximum retransmission interval for non-INVITE requests and INVITE responses. Default is 4 Sec. DTMF Payload Type Sets the payload type for DTMF using RFC2833. Default is 101. Preferred DTMF method The HT70X supports up to 3 different DTMF methods including in-audio, via RTP (RFC2833) and via Sip Info using SIP INFO messages. The user can configure DTMF method in a priority list. Disable DTMF Negotiation Default is No. If set to yes, use above DTMF order without negotiation Send Flash Event Default is No. If set to yes, flash will be sent as DTMF event. Enable Call Features Default is Yes. (If Yes, call features using star codes will be supported locally) Offhook Auto-Dial This parameter allows users to configure a User ID or extension number that is automatically dialed when off-hook. Only the user part of a SIP address needs to be entered here. The HT70X will automatically append the “@” and the host portion of the corresponding SIP address. HT701 and HT702 only Offhook Auto-Dial Delay The auto-dial delay after off hook. Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Use NAT IP NAT IP address used in SIP/SDP message. Default is blank. Use SIP User-Agent Header Configurable SIP User-Agent Header. Distinctive Ring Tone Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is configured, then the device will ONLY uses this ring tone when the incoming call is from the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls using the FXS port or Profile. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol * (star) will be used. For example: if configured as *617, Ring Tone 1 will be used in case of call arrived from the area code 617. Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page. Note: If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be used. Bellcore rings and tones are independent from custom ring tones. The custom ring tones can also be specified by alert-info header, for example Alert-Info: Este manual também é adequado para os modelos :
fones de ouvido sem fio e outros acessórios - HT701 (891.15 kb)
fones de ouvido sem fio e outros acessórios - HT702 (891.15 kb)


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