Grandstream Networks HandyTone-486 User Manual page 21

HandyTone-486 User Manual
NAT Traversal
Use NAT IP:
TFTP Server
Subscribe for
MWI:
Offhook
Auto-Dial
Enable Call
Features
Disable Call
Waiting
Send DTMF
DTMF Payload
Type
This parameter defines whether the HandyTone ATA NAT traversal
mechanism will be activated or not. If activated (by choosing "Yes") and
a STUN server is also specified, then the HandyTone ATA will behave
according to the STUN client specification. Under this mode, the
embedded STUN client inside the HandyTone ATA 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 HandyTone ATA 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
HandyTone ATA 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.
NAT IP address used in SIP/SDP message. Default is blank.
This is the IP address of the configured tftp server. If it is non-zero or not
blank, the HandyTone ATA 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.
Default is NO. When set to Yes a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
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
HandyTone ATA will automatically append the "@" and the host portion
of the corresponding SIP address.
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are
supported locally
Default is Not.
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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