Grandstream Networks BT200 user manual Saving The Configuration Changes, Special Feature

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Check SIP User ID for Incoming Invite

Check SIP User ID for Incoming Invite

Preferred Vocoder

SRTP Mode

Special Feature

When turned on, it will validate the To URI for incoming INVITE before accepting it. If it differs from locally configured URI, it should reject the call with 404 Not Found message.

The BT200 supports up to 5 different Vocoder types including G.711(a/µ) (also known as PCMU/PCMA), GSM, G.723.1, G.729A/B.

Configure Vocoders in a preference list that is included with the same preference order in SDP message. Enter the first Vocoder in this list by choosing the appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by choosing the appropriate option in “Choice 8”.

Disabled: BT200 will use RTP for both inbound and outbound calls. (Default setting)

Enabled but not forced: BT200 provides crypto suites to others SIP phones. If they support SRTP, then use SRTP, otherwise, use RTP.

Enabled and forced: All calls using this account must use SRTP. If the other party doesn’t support SRTP, BT200 will decline the call. If we invite others, and receive a “200 Ok” message without crypto suite, we will send a “BYE” message followed by the “ACK” message to terminate the call. If someone invites us without crypto suite, we will response a “488 Not Acceptable Here” message.

Default is Standard. Choose the selection to meet some special requirements from SoftSwitch vendors.

SAVING THE CONFIGURATION CHANGES

Once a change is made, click the “Update” button in the Configuration Menu. The IP phone will display a confirmation page. Reboot or power cycle the BT200 after all changes to ensure the changes take effect.

FIGURE 5: SAVE CONFIGURATION SCREENSHOT

REBOOTING THE PHONE FROM REMOTE

The administrator can remotely reboot the phone by clicking the “Reboot” button at the bottom of the configuration menu. Once done, the web server will display a confirmation page to indicate that rebooting is underway. When finished, login again after waiting for about 30 seconds.

Grandstream Networks, Inc.

BT200 User Manual

Page 29 of 37

 

Firmware 1.1.1.14

Last Updated: 12/2006

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Contents Grandstream Networks, Inc BT200 User ManualBT-Series BT200 Dual Ethernet Port SIP Phone Firmware VersionBT200 USER MANUAL TABLE OF CONTENTSAccess the Web Configuration Menu Placing a Quick IP CallGUI INTERFACES TABLE OF FIGURESTABLE OF TABLES WELCOME SAFETY COMPLIANCESWARRANTY CONNECT YOUR PHONE CONNECTING YOUR PHONEEQUIPMENT PACKAGING CONNECT YOUR PHONE TO THE VOIP NETWORK PRODUCT OVERVIEW KEY FEATURESHARDWARE FEATURES Device Management Audio Features Call Handling Features Lines Protocol Support1 Display Feature Keys Ethernet PortsNetwork and Provisioning Advanced Server Features Security TABLE 3 TECHNICAL SOFTWARE FEATURESGETTING FAMILIAR WITH THE LCD / LED USING THE BT200 IP PHONEAlarm Clock Icon IconMenu and Call Keys Advanced Feature Buttons Speaker, Send Mute/Del GETTING FAMILIAR WITH KEYPADAM PM Real-time ClockKey Button Definitions Key ButtonUP ↑ DOWN ↓MAKING AND ANSWERING PHONE CALLS PLACING CALLS USING AN IP ADDRESSHANDSET, SPEAKERPHONE AND HEADSET MODE PLACING A TRADITIONAL PHONE CALLANSWER AN INCOMING CALL Placing a Quick IP CallCall Hold Call Waiting and Call FlashingExpected outcomes Blind and Attended Call Transfer3-way Conference Retrieving MessagesCall Features CALL FEATURESDisable Call Waiting per Call Enable Call Waiting per CallCONFIGURATION WITH KEYPAD CONFIGURATION GUIDEMenu Functions Menu ItemAccess the Web Configuration Menu CONFIGURATION WITH WEB BROWSEREND USER CONFIGURATION Examples End User Password IP Address Time Zone Daylight Savings TimeDate Display Format Display Account Name instead of Date Device Mode WAN side HTTP access Reply to ICMP on WAN port Cloned WAN MAC AddrLAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port ForwardingMAC Address ADVANCED USER CONFIGURATIONIP Address Product ModelLayer 2 QoS No Key Entry Timeout Use # as Send Key Local RTP port Admin Password Silence Suppression Voice Frames per TX Layer 3 QoSAllow DHCP Option 66 to override server Automatic Upgrade Firmware Upgrade and provisioning Via TFTP Server Via HTTP ServerAuthenticate Conf File Off-hook Auto Dial Use Random Port Keep-alive interval Use NAT IP STUN ServerAllow DHCP Option 42 to override NTP server Distinctive Ring Tone Disable Call WaitingDTMF Payload Type Syslog Server Syslog Level NTP server Authenticate ID Lock keypad updateAuthenticate PasswordLocal SIP port SIP T1 Timeout SIP T2 Interval NAT Traversal SIP Registration Unregister on RebootEarly Dial Dial Plan Prefix Enable Call Features Disable Missed-Call Register ExpirationForce Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Session Expiration Min-SE Caller Request Timer Callee Request TimerEnable 100rel Account Ring Tone Send Anonymous Auto Answer Allow Auto Answer by Call-Info Turn off speaker on remote disconnectSpecial Feature SAVING THE CONFIGURATION CHANGESREBOOTING THE PHONE FROM REMOTE Check SIP User ID for Incoming Invite Preferred Vocoder SRTP ModeCONFIGURATION THROUGH CENTRAL PROVISIONING SERVER UPGRADE THROUGH HTTP FIRMWARE UPGRADEUPGRADE THROUGH TFTP Keypad Configuration Prompt MenuNo Local TFTP Server NOTE for TFTP/HTTP Firmware UpgradeDirections for Restoration RESTORE FACTORY DEFAULT SETTINGGLOSSARY 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