Grandstream Networks BT200 Session Expiration Min-SE Caller Request Timer Callee Request Timer

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Session Expiration

Session Expiration

Min-SE

Caller Request Timer

Callee Request Timer

Force Timer

UAC Specify

Refresher

UAS Specify

Refresher

Force INVITE

Enable 100rel

Account Ring Tone

Send Anonymous

Auto Answer

Allow Auto Answer by Call-Info

Turn off speaker on remote disconnect

The SIP Session Timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session is terminated.

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.

The minimum session expiration (in seconds). The default value is 90 seconds.

If selecting “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 selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support 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.

The use of the PRACK (Provisional Acknowledgment) method enables reliability to be offered to SIP provisional responses (1xx series). This is very important if PSTN internetworking is to be supported. A user’s wish to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signaling messages.

There are 4 different ring tone that are defined:

System Ring Tone: when selected, all calls will ring with system ring tone.

Customer Ring Tone 1 to 3: when selected, BT200 will ONLY play this ring tone for all the incoming calls for this account.

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.

When set to “Yes”, BT200 will automatically switch to speaker when there is an incoming call.

Default is No. If set to Yes, auto answer depends on the Call-Info in the SIP message. This feature needs the support of IP-PBX.

Default is No. If set to Yes, the speaker will turn off, and the phone will go back to idle status, after the other party of the call hands up.

Grandstream Networks, Inc.

BT200 User Manual

Page 28 of 37

 

Firmware 1.1.1.14

Last Updated: 12/2006

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Contents BT200 User Manual Grandstream Networks, IncBT-Series BT200 Dual Ethernet Port SIP Phone Firmware VersionTABLE OF CONTENTS BT200 USER MANUALAccess the Web Configuration Menu Placing a Quick IP CallTABLE OF TABLES TABLE OF FIGURESGUI INTERFACES WARRANTY SAFETY COMPLIANCESWELCOME EQUIPMENT PACKAGING CONNECTING YOUR PHONECONNECT YOUR PHONE CONNECT YOUR PHONE TO THE VOIP NETWORK HARDWARE FEATURES KEY FEATURESPRODUCT OVERVIEW Lines Protocol Support1 Display Feature Keys Ethernet Ports Device Management Audio Features Call Handling FeaturesNetwork and Provisioning Advanced Server Features Security TABLE 3 TECHNICAL SOFTWARE FEATURESUSING THE BT200 IP PHONE GETTING FAMILIAR WITH THE LCD / LEDAlarm Clock Icon IconGETTING FAMILIAR WITH KEYPAD Menu and Call Keys Advanced Feature Buttons Speaker, Send Mute/DelAM PM Real-time ClockKey Button Key Button DefinitionsUP ↑ DOWN ↓PLACING CALLS USING AN IP ADDRESS MAKING AND ANSWERING PHONE CALLSHANDSET, SPEAKERPHONE AND HEADSET MODE PLACING A TRADITIONAL PHONE CALLPlacing a Quick IP Call ANSWER AN INCOMING CALLCall Hold Call Waiting and Call FlashingBlind and Attended Call Transfer Expected outcomes3-way Conference Retrieving MessagesCALL FEATURES Call FeaturesDisable Call Waiting per Call Enable Call Waiting per CallCONFIGURATION GUIDE CONFIGURATION WITH KEYPADMenu Item Menu FunctionsCONFIGURATION WITH WEB BROWSER Access the Web Configuration MenuEND USER CONFIGURATION End User Password IP Address Time Zone Daylight Savings Time ExamplesWAN side HTTP access Reply to ICMP on WAN port Cloned WAN MAC Addr Date Display Format Display Account Name instead of Date Device ModeLAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port ForwardingADVANCED USER CONFIGURATION MAC AddressIP Address Product ModelAdmin Password Silence Suppression Voice Frames per TX Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use # as Send Key Local RTP portFirmware Upgrade and provisioning Via TFTP Server Via HTTP Server Allow DHCP Option 66 to override server Automatic UpgradeAuthenticate Conf File Off-hook Auto Dial Use Random Port Keep-alive interval Use NAT IP STUN ServerDTMF Payload Type Syslog Server Syslog Level NTP server Disable Call WaitingAllow DHCP Option 42 to override NTP server Distinctive Ring Tone Lock keypad update Authenticate IDAuthenticate PasswordSIP Registration Unregister on Reboot Local SIP port SIP T1 Timeout SIP T2 Interval NAT TraversalEarly Dial Dial Plan Prefix Enable Call Features Disable Missed-Call Register ExpirationSession Expiration Min-SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITEEnable 100rel Account Ring Tone Send Anonymous Auto Answer Allow Auto Answer by Call-Info Turn off speaker on remote disconnectSAVING THE CONFIGURATION CHANGES Special FeatureREBOOTING THE PHONE FROM REMOTE Check SIP User ID for Incoming Invite Preferred Vocoder SRTP ModeCONFIGURATION THROUGH CENTRAL PROVISIONING SERVER FIRMWARE UPGRADE UPGRADE THROUGH HTTPUPGRADE THROUGH TFTP Keypad Configuration Prompt MenuNOTE for TFTP/HTTP Firmware Upgrade No Local TFTP ServerRESTORE FACTORY DEFAULT SETTING Directions for RestorationGLOSSARY OF TERMS FXS is complimentary to FXS and the PSTN NAT Network Address Translation UDP User Datagram Protocol UDP is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages known as datagrams to one another. UDP does not provide the reliability and ordering guarantees that TCP does datagrams may arrive out of order or go missing without notice. However, as a result, UDP is faster and more efficient for many lightweight or time-sensitive purposes