Grandstream Networks BudgeTone-200 User Manual page 29

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Force Timer
UAC Specify Refresher
UAS Specify Refresher
Force INVITE
Enable 100rel
Account Ring Tone
Ring Timeout
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
Disable Multiple Media
Attribute in SDP
Grandstream Networks, Inc.
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
N
ring tone.
Three (3) Customer Ring Tones: when selected, incoming calls from
N
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.
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", BT 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.
Default is No.
BT200/201 User Manual
Firmware 1.2.2.19
Page 29 of 33
Last Updated: 12/2009