BudgeTone-100 User Manual
Grandstream Networks, Inc.
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.
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.
SIP
Registration
This parameter controls whether the IP phone needs to send
REGISTER messages to the proxy server. The default setting is
“Yes”.
Registration
Interval
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).
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
por1 for its RTCP; channel 1 will use por2 for RTP
and por3 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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