| Delayed Call Forward | Time waited before the call is forward to a number or VM. | ||
| Wait Time | Default is 20 seconds. |
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| Enable Call Features | Default is Yes. If set to “Yes”, all star code call features will be supported locally. | ||
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| Disable Call Log | User can choose to disable Call Log. |
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| Session Expiration | The SIP Session Timer extension enables SIP sessions to be periodically | ||
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| “refreshed” via a SIP request (UPDATE, or | ||
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| expires, if there is no refresh via a UPDATE or | ||
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| terminated. |
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| Session Expiration is the time (in seconds) at which the session is considered | ||
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| timed out, provided no successful session refresh transaction occurs beforehand. | ||
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| The default value is 180 seconds. |
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| Defines the minimum session expiration (in seconds). Default is 90 seconds. | ||
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| Caller Request Timer | If set to “Yes”, the phone will use session timer when it makes outbound calls if | ||
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| remote party supports session timer. |
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| Callee Request Timer | If selecting “Yes”, the phone will use session timer when it receives inbound calls | ||
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| with session timer request. |
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| Force Timer | If set to “Yes”, the phone will use session timer even if the remote party does not | ||
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| support this feature. If set to “No”, the session timer is enabled only when the | ||
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| remote party supports this feature. To turn off Session Timer, select “No” for | ||
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| Caller Request Timer, Callee Request Timer, and Force Timer. | ||
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| UAC Specify Refresher | As a Caller, select UAC to use the phone as the refresher, or UAS to use the | ||
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| Callee or proxy server as the refresher. |
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| UAS Specify Refresher | As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to | ||
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| use the phone as the refresher. |
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| Force INVITE | Session Timer can be refreshed using INVITE method or UPDATE method. | ||
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| Select “Yes” to use INVITE method to refresh the session timer. | ||
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| Enable 100rel | PRACK (Provisional Acknowledgment) method enables reliability to SIP | ||
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| provisional responses (1xx series). This is required to support PSTN inter- | ||
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| networking.. |
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| There are 4 uniquely defined ring tones: |
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| Account Ring Tone |
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| • One (1) System Ring Tone: when selected, all calls will ring with system | ||
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| ring tone. |
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| • Three (3) Customer Ring Tones: when selected, incoming calls from | ||
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| designated account will play selected ring tone. |
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| Send Anonymous | If this parameter is set to “Yes”, the “From” header in outgoing INVITE message | ||
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| will be set to anonymous, essentially blocking the Caller ID from displaying. | ||
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| Anonymous Method | Whether to use “sip:anonymous@anonymous.invalid>” in the From Header or P- | ||
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| Anonymous Call | Default is NO. If set to YES, anonymous call will be rejected | ||
| Rejection |
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Grandstream Networks, Inc. | BT200 User Manual | Page 29 of 33 | ||
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| Firmware 1.1.5.15 | Last Updated: 01/2008 |