Grandstream Networks GXP Series manual Firmware Last Updated 05/2008

Min-SE

Caller Request Timer

Callee Request Timer

Force Timer

UAC Specify Refresher

UAS Specify Refresher

Force INVITE

Enable 100rel

Account Ring Tone

Send Anonymous

Anonymous Method

Anonymous Call

Rejection

Auto Answer

Allow Auto Answer by Call-Info

Turn off speaker on remote disconnect

Check SIP User ID for incoming INVITE

Refer-To Use Target Contact

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.

If set to “Yes”, the phone will use session timer even if the remote party does not support this feature. If set to “No”, the session timer is enabled only when the remote party supports this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher.

Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer.

PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is required to support PSTN inter- networking..

There are 4 uniquely defined ring tones:

•One (1) System Ring Tone: when selected, all calls will ring with system ring tone.

•Three (3) Customer Ring Tones: when selected, incoming calls from designated account will play selected ring tone.

If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will be set to anonymous, essentially blocking the Caller ID from displaying.

Whether to use “sip:anonymous@anonymous.invalid>” in the From Header or P- Asserted-Identity header.

Default is NO. If set to YES, anonymous call will be rejected

Default is No. If set to “Yes”, GXP will automatically switch on speaker to answer the incoming call. Set to Intercom/Paging mode, it will answer the call based on the SIP info header from the server.

If the Call-Info header contains answer-after=0, the call be answered automatically (so called paging mode).

When BYE is received, the phone will turn off its speaker automatically.

Check the SIP User ID in Request URI. If they don’t match, the call will be rejected.

Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information.

Grandstream Networks, Inc.	GXP User Manual	Page 36 of 40
	Firmware 1.1.6.16	Last Updated: 05/2008