Session Expiration
Caller Request Timer
Callee Request Timer
Force Timer
UAC Specify Refresher
UAS Specify Refresher
Force INVITE
Enable 100rel
Account Ring Tone
Ring Timeout
Send Anonymous
Anonymous Call
Rejection
Transfer on Conference Hangup
The SIP Session Timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or
Session Expiration is the time (in seconds) at which the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds.
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
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.
Defines how long ring will ring when receiving a call. Default is 60 seconds.
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.
Default is “No”. If set to “Yes”, anonymous call will be rejected.
Default is “No”. If set to “Yes”, then for Attended Transfer, the
Defines whether or not the call is transferred to the other party if the initiator of the conference hangs up.
Default setting is set to “No”.
Grandstream Networks, Inc. | GXP1100/1105 User Manual | Page 29 of 34 |
| Firmware version: 1.0.1.110 | Last Updated: 01/2012 |