Grandstream Networks, Inc | Grandstream Networks HandyTone-286

HandyTone -286 User Manual	Grandstream Networks, Inc.

SIP Registration

Unregister On Reboot

Registration

Expiration

Early Dial

Dial Plan Prefix

No Key Entry Timeout

Use # as

Send Key

Local SIP port

Local RTP port

Use Random Port

This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server. The default setting is “Yes”.

Default is No. If set to yes, the device will first send registration request to remove previous bindings.

This parameter allows the user to specify the time frequency (in minutes) the phone will refresh its registration with the specified registrar. The default interval is 3600 seconds (or 1 hour). The maximum interval is 45 days.

This parameter controls whether the phone will attempt to send an early INVITE each time a key is pressed when a user dials a number. If set to “Yes”, an INVITE is sent using the dial-number collected thus far; Otherwise, no INVITE is sent until the “(Re-)Dial” button is pressed or after about 5 seconds have elapsed if the user forgets to press the “(Re-)Dial” button.

The “Yes” option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response. Otherwise, the call will most likely be rejected by the proxy (with a 404 Not Found error).

Please note that this feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

This value contains the dial plan prefix string (typically an ASCII numeric string). If it is not blank, then this string will added to the dialed number.

Default is 4 seconds.

This parameter allows the user to configure the “#” key to be used as the “Send”(or “Dial”) key. Once set to “Yes”, pressing this key will immediately trigger the sending of dialed string collected so far. In this case, this key is essentially equivalent to the “(Re)Dial” key. If set to “No”, this # key will then be included as part of the dial string to be sent out.

This parameter defines the local SIP port the IP phone will listen and transmit on. The default value is 5060.

This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit on. It is the base RTP port for channel 0. When configured, channel 0 will use this port value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.

This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple IP phones are behind the same NAT.

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