Grandstream Networks GXP Series manual Firmware Last Updated 05/2008

SIP Registration Failure Retry Wait Time

SIP T1 Timeout

SIP T2 Interval

SIP Transport

Use RFC3581 Symmetric Routing

NAT Traversal (STUN)

Subscribe for MWI:

PUBLISH for Presence

Proxy-Require

Voice Mail UserID

Send DTMF

Early Dial

Dial Plan Prefix

Delayed Call Forward Wait Time

Enable Call Features

Call Log

Session Expiration

Retry registration if the process failed. Default is 20 seconds.

RFC 3261 SIP T1 timer. Default is 1 second.

RFC 3261 SIP T2 timer. Default is 0.5 seconds.

Choose SIP Transport between UDP and TCP. Default is UDP.

Default No. When selected the phone will follow the routing procedures specified in RFC3581.

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 GXP 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.

Enable Presence feature.

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 No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP support those features.

User can choose to disable Call Log and what kind of calls to 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.

Grandstream Networks, Inc.	GXP User Manual	Page 35 of 40
	Firmware 1.1.6.16	Last Updated: 05/2008