Grandstream Networks 100 Series user manual Grandstream Networks, Inc

BudgeTone-100 User Manual Grandstream Networks, Inc.

Unregister On Reboot

Register

Expiration

Early Dial

Dial Plan Prefix

No key Entry Timeout

Use # as

Dial Key

Local SIP port

Local RTP port

Use Random port

NAT Traversal

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

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.

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.

Sets the prefix added to each dialed number. If configured, the prefix will be added to EVERY number input

Default is 4 seconds.

User can short or extend that depends on digits dialed habit

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.

This parameter defines the local SIP port the IP phone will listen and transmit on. The default value is 5060.

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.

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.

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.

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