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BudgeTone-100 User Manual Grandstream Networks, Inc.
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Unregister On
Reboot
Default is No. If set to Yes, the phone will send “remove all register” request to
the server (“*” in the contact header) to remove all previous bindings. If server
does not support this it will cause some problems
Register
Expiration
This parameter allows the user to specify the time frequency (in seconds) the IP
phone will refresh its registration with the specified registrar (SIP Server). The
default interval is 3600 seconds (or 1 hour). The maximum interval is 45 days.
Early Dial
Default setting is No. The “Yes” option should be used ONLY if there is a SIP
proxy configured and the proxy server supports 484 Incomplete Address
response (like Asterisk). 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
Sets the prefix added to each dialed number. If configured, the prefix will be
added to EVERY number input
No key Entry
Timeout
Default is 4 seconds.
User can short or extend that depends on digits dialed habit
Use # as
Dial Key
This parameter allows the user to configure the “#” key to be used as the
“SEND” 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 “SEND” 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. 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
Default No. If set to Yes, the device will pick randomly generated SIP and RTP
ports. This is usually necessary and useful when multiple IP Phones are behind
the same full cone NAT router.
NAT Traversal
Defines whether the NAT traversal mechanism is activated.
It should be set to YES if the device is behind NAT router.
If Outbound Proxy is NOT configured, STUN server needs to be set to activate
STUN detection mechanism. Usually ITSP will provide these settings for device
to work properly behind NAT/Firewall
If this field is set to “Yes” without STUN server, then the device will
periodically (every Keep-alive interval) send a dummy UDP packet to the SIP
server to pinhole the NAT in the router side.