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HandyTone 286 User Manual Grandstream Networks, Inc.
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Registration
Expiration
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 60 minutes (or 1 hour). The maximum
interval is 65535 minutes (about 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 be used as a
prefix to the target URI string in the “To” header field of an INVITE
message.
Use # as
Send Key
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.
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.