BudgeTone-100 User Manual
Grandstream Networks, Inc.
NAT Traversal
This parameter defines whether the phone NAT traversal mechanism
will be activated or not. If activated (by choosing “Yes”) and a
STUN server is also specified, then the phone will behave according
to the STUN client specification. Under this mode, the embedded
STUN client inside the phone will attempt to detect if and what type
of firewall/NAT it is behind through communication with the
specified STUN server. If the detected NAT is a Full Cone,
Restricted Cone, or a Port-Restricted Cone, the phone will attempt to
use its mapped public IP address and port in all the SIP and SDP
messages it sends out.
If this field is set to “Yes” with no specified STUN server, then the
phone will periodically (every 10 seconds or so) send a blank UDP
packet (with no payload data) to the SIP server to keep the “hole” on
the NAT open.
TFTP Server
This is the IP address of the configured tftp server. If it is non-zero
or not blank, the IP phone will attempt to retrieve new configuration
file or new code image from the specified tftp server at boot time. It
will make up to 3 attempts before timeout and then it will start the
boot process using the existing code image in the Flash memory. If a
tftp server is configured and a new code image is retrieved, the new
downloaded image will be verified and then saved into the Flash
memory.
Voice Mail
User ID
This parameter defines the User ID (or extension number) of a 3
rd
party voice mail system where the user may have an account. By
defining this Voice Mail extension, when the user presses the
“Message” button on the phone, an INVITE message will be sent to
that extension to allow the user to retrieve messages.
Offhook
Auto-Dial
This parameter allows the user to configure a User ID or extension
number to be automatically dialed upon offhook. Please note that
only the user part of a SIP address needs to be entered here. The
phone will automatically append the “@” and the host portion of the
corresponding SIP address.
Send DTMF
This parameter controls the way DTMF events are transmitted.
There are 3 ways: in audio which means DTMF is combined in
audio signal (not very reliable with low-bit-rate codec), via RTP
(RFC2833), or via SIP INFO.
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