Grandstream Networks, Inc. GXP User Manual Page 36 of 40
Firmware 1.1.6.16 Last Updated: 05/2008
Min-SE Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request Timer If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
Force Timer 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.
UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
Enable 100rel PRACK (Provisional Acknowledgment) method enables reliability to SIP
provisional responses (1xx series). This is required to support PSTN inter-
networking..
Account Ring Tone 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.
Send Anonymous
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.
Anonymous Method Whether to use “sip:anonymous@anonymous.invalid>” in the From Header or P-
Asserted-Identity header.
Anonymous Call
Rejection Default is NO. If set to YES, anonymous call will be rejected
Auto Answer 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.
Allow Auto Answer by
Call-Info If the Call-Info header contains answer-after=0, the call be answered
automatically (so called paging mode).
Turn off speaker on
remote disconnect When BYE is received, the phone will turn off its speaker automatically.
Check SIP User ID for
incoming INVITE Check the SIP User ID in Request URI. If they don’t match, the call will be
rejected.
Refer-To Use Target
Contact
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s Contact header information.