HandyTone -286 User Manual Grandstream Networks, Inc.
25
SIP Registration This parameter controls whether the IP phone needs to send REGISTER
messages to the proxy server. The default setting is “Yes”.
Unre
g
ister On
Reboot Default is No. If set to yes, the device will first send registration request to
remove previous bindings.
Re
g
istration
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.
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 Ke
y
Entr
y
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.