Grandstream Networks HT-286 User Manual
Page 20
Grandstream Networks, Inc.
HT–286 User Manual
Page 20 of 28
Firmware 1.1.0.31
Last Updated: 1/2009
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 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.
Allow outgoing call
without Registration
Default is set to “No”.
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
Timeout
Default is 4 seconds.
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.
SIP Registration
Failure Retry Wait
Time
Retry registration if the process failed. Default is 20 seconds.