Grandstream Networks GXV-3000 user manual Grandstream Networks, Inc

GXV-3000 User Manual Grandstream Networks, Inc.

NAT Traversal

Subscribe for MWI:

Proxy-Require

Voice Mail User ID

Send DTMF

Early Dial

Dial Plan Prefix

Enable Call Features

Session Expiration

Min-SE

Caller Request Timer

This parameter defines whether the GXV-3000 NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the GXV-3000 will behave according to the STUN client speci- fication. Under this mode, the embedded STUN client inside the GXV-3000 will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXV-3000 will attempt to 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 GXV-3000 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 Indica- tion will be sent periodically.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

When configured, user will be able to dial voice mail server 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 modes supported: 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.

Default is No. If set to Yes, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP’s SIP server supporting those features.

Grandstream implemented SIP Session Timer. The session timer extension en- ables 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 will be terminated.

Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

The minimum session expiration (in seconds). The default value is 90 sec- onds.

If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer.

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