Grandstream Networks 286 User ID is phone, Number, SIP Registration, Unregister On, Reboot

HandyTone -286 User Manual			Grandstream Networks, Inc.
	User ID is phone	If the HandyTone-286 has an assigned PSTN telephone number, then this field
	number	will be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone”
		parameter will be attached to the “From” header in SIP request.
	SIP Registration	This parameter controls whether the IP phone needs to send REGISTER
		messages to the proxy server. The default setting is “Yes”.
	Unregister On	Default is No. If set to yes, the device will first send registration request to
	Reboot	remove previous bindings.
	Registration	This parameter allows the user to specify the time frequency (in minutes) the
	Expiration	phone will refresh its registration with the specified registrar. The default
		interval is 3600 seconds (or 1 hour). The maximum interval is 45 days.
	Early Dial	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.
	Dial Plan Prefix	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.
	No Key Entry	Default is 4 seconds.
	Timeout
	Use # as	This parameter allows the user to configure the “#” key to be used as the
	Send Key	“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.
	Local SIP port	This parameter defines the local SIP port the IP phone will listen and transmit
		on. The default value is 5060.
	Local RTP port	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.
	Use Random Port	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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