Grandstream Networks, Inc.
GXP2124 User Manual
Page
45 of 50
Firmware version: 1.0.3.19 Last Updated: 03/2012
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:
System Ring Tone: when selected, all calls will ring with system ring tone.
3 Customer Ring Tones: when selected, incoming calls from designated
account will play selected ring tone.
Ring Timeout
Defines how long the phone will ring when receiving a call. Default is 60 seconds.
Line-seize Timeout
Defines how long before the line can be seized when Share Line is used. Default is
15 seconds.
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 Call
Rejection
Default is “No”. If set to “Yes”, anonymous call will be rejected.
Auto Answer
Default is “No”. If set to “Yes”, GXP2124 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.
This fields need to be set to Yes if users would like to have the phone to be
paged/intercom.
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.
Transfer on
Conference
Hangup
Defines whether or not the call is transferred to the other party if the initiator of the
conference hangs up. Default is “No”.
Check SIP User ID
for Incoming
Check the SIP User ID in Request URI. If they don’t match, the call will be rejected.
