Grandstream Networks GXP2130 Administration Manual page 38

Min-SE
Caller Request Timer
Callee Request Timer
Force Timer
UAC Specify Refresher
UAS Specify Refresher
Force INVITE
Account x  SIP Settings  Security Settings
Check Domain
Certificates
Validate Certificate Chain
Validate Incoming
Messages
Check SIP User ID for
incoming INVITE
Accept Incoming SIP
from Proxy Only
Authenticate Incoming
INVITE
Account x  Audio Settings
GXP2130/GXP2140/GXP2160
FIRMWARE VERSION 1.0.5.18
terminated once the session interval expires. Session Expiration is the time (in
seconds) where the session is considered timed out, provided no successful
session refresh transaction occurs beforehand. The default setting is 180. The
valid range is from 90 to 64800.
The minimum session expiration (in seconds). The default value is 90
seconds. The valid range is from 90 to 64800.
If set to "Yes" and the remote party supports session timers, the phone will use
a session timer when it makes outbound calls. The default setting is "No".
If set to "Yes" and the remote party supports session timers, the phone will use
a session timer when it receives inbound calls. The default setting is "No".
If Force Timer is set to "Yes", the phone will use the session timer even if the
remote party does not support this feature. If Force Timer is set to "No", the
phone will enable the session timer only when the remote party supports this
feature. To turn off the session timer, select "No". The default setting is "No".
As a Caller, select UAC to use the phone as the refresher; or select UAS to
use the Callee or proxy server as the refresher. The default setting is "Omit".
As a Callee, select UAC to use caller or proxy server as the refresher; or select
UAS to use the phone as the refresher. The default setting is "UAC".
The Session Timer can be refreshed using the INVITE method or the UPDATE
method. Select "Yes" to use the INVITE method to refresh the session timer.
The default setting is "No".
Choose whether the domain certificates will be checked or not when TLS/TCP
is used for SIP Transport. The default setting is "No".
Validate certification chain when TCP/TLS is configured. The default setting is
"No".
Choose whether the incoming messages will be validated or not. The default
setting is "No".
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
If set to "Yes", the phone will challenge the incoming INVITE for authentication
with SIP 401 Unauthorized response. The default setting is "No".
GXP2130/GXP2140/GXP2160
ADMINISTRATION GUIDE
Page 36 of 87