Grandstream Networks BudgeTone-200 User Manual page 28

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NAT Traversal (STUN)
Subscribe for MWI
SUBSCRIBE for
Registration Event
Proxy-Require
Voice Mail UserID
Send DTMF
Early Dial
Dial Plan Prefix
Delayed Call Forward
Wait Time
Enable Call Features
Disable Call Log
Session Expiration
Min-SE
Caller Request Timer
Callee Request Timer
Grandstream Networks, Inc.
This parameter activates the NAT traversal mechanism. If activated (by choosing
"Yes") and a STUN server is also specified, the phone performs according to the
STUN client specification. Using this mode, the embedded STUN client detects if
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped
public IP address and port in all of its SIP and SDP messages. If the NAT
Traversal field is set to "Yes" with no specified STUN server, the BT will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the "hole" on the NAT open.
Default is No. When set to "Yes" a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Default is No. This is mainly used for IMS purposes. When enabled, the terminals
should store the Service-Route header values after successfully registered, and
thereafter add a route header with the values stored in the Service-Route when
initiating a request excluding REGISTER.
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
When configured, user can access messages by pressing "MSG" button. This ID
is usually the VM portal access number.
This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: 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.
Default is No. Use only if proxy supports 484 response.
Sets the prefix added to each dialed number.
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Default is Yes. If set to "Yes", all star code call features will be supported locally.
User can choose to disable Call Log.
The SIP Session Timer extension enables SIP sessions to be periodically
"refreshed" via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVITE message, the session is
terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Defines the minimum session expiration (in seconds). Default is 90 seconds.
If set to "Yes", the phone will use session timer when it makes outbound calls if
remote party supports session timer.
If selecting "Yes", the phone will use session timer when it receives inbound calls
with session timer request.
BT200/201 User Manual
Firmware 1.2.2.19
Page 28 of 33
Last Updated: 12/2009