User manual GRANDSTREAM BT100

[. . . ] User Manual BudgeTone-100 Series IP Phone For Firmware Version 1. 0. 8. 32 Grandstream Networks, Inc. www. grandstream. com BudgeTone-100 User Manual Grandstream Networks, Inc. Table of Contents 1 WELCOME 2 INSTALLATION 2. 1 INTERCONNECTION DIAGRAM 3 WHAT IS INCLUDED IN THE PACKAGE 3. 1 SAFETY COMPLIANCES 3. 2 WARRANTY 4 PRODUCT OVERVIEW 4. 1 KEY FEATURES 4. 2 HARDWARE SPECIFICATIONS 5 BASIC OPERATIONS 5. 1 GET FAMILIAR WITH LCD/LED 5. 2 GET FAMILIAR WITH KEYPAD 5. 3 MAKE PHONE CALLS 5. 3. 1 Make Calls using Numbers 5. 3. 2 Make Calls using IP Address 5. 3. 3 Answer an Incoming Call 5. 3. 4 Handset Mode, Speakerphone/Headset Mode 5. 3. 5 Call Hold 5. 3. 6 Call Waiting and Call Flashing 5. 3. 7 Call Transfer 5. 3. 8 Conference Call 5. 4 CALL FEATURES 6 CONFIGURATION GUIDE 6. 1 CONFIGURATION WITH KEYPAD 6. 2 CONFIGURATION WITH WEB BROWSER 6. 2. 1 Access the Web Configuration Menu 6. 2. 2 Configuration Menu 6. 2. 3 Saving the Configuration Changes 6. 2. 4 Rebooting the Phone from Remote 6. 3 CONFIGURATION THROUGH A CENTRAL SERVER 7 SOFTWARE UPGRADE 7. 1 UPGRADE THROUGH HTTP 7. 2 UPGRADE THROUGH TFTP 7. 3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX -2-4-5-6-7-7-7-8-8-9- 11 - 11 - 12 - 14 - 14 - 14 - 15 - 15 - 15 - 15 - 16 - 17 - 18 - 19 - 19 - 23 - 23 - 24 - 38 - 39 - 40 - 41 - 41 - 41 - 43 - BudgeTone-100 User Manual Grandstream Networks, Inc. 7. 4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 8 RESTORE FACTORY DEFAULT SETTING 9 HEADSET CONNECTION 10 GLOSSARY OF TERMS - 43 - 44 - 45 - 48 - -3- BudgeTone-100 User Manual Grandstream Networks, Inc. 1 Welcome Congratulations on becoming an owner of BudgeTone-100 IP telephone!You made an excellent choice and we hope you will enjoy all its capabilities. Grandstream's award-wining BudgeTone-100 series of SIP phones are innovative IP telephones that offer a rich set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market. [. . . ] User can use * code to use this feature per call basis. Keep alive interval Use NAT IP Proxy-Require Voice Mail User ID Subscribe for MWI Auto Answer Offhook Auto-Dial Enable call features Disable Call Waiting - 34 - BudgeTone-100 User Manual Send DTMF Grandstream Networks, Inc. This parameter specifies the mechanism to transmit DTMF digits. There are 3 modes supported: in audio, which means DTMF is combined in inband audio signal (not very reliable with low-bit-rate codec); via RTP (RFC2833); or via SIP INFO. Make sure this setting is synchronized with SIP server otherwise will not work. This parameter sets the payload type for DTMF using RFC2833 Default is 101 DTMF Payload Type Send Flash Event Default is NO. If set to Yes, flash will be sent as DTMF event therefore the FLASH key to switch calls will NOT work. Onhook Threshold NTP server Default Ring Tone Time period when cradle pressed (Hook Flash) to simulate FLASH. Adjust this time value will prevent user on hook carelessly to activate Flash/Hold and cause the phone ring back. URI or IP address of the NTP (Network Time Protocol) server, which will be used by the phone to synchronize the date and time. · By default "System Ring Tone" is selected. All calls received that do not match any "incoming caller ID" number will generate the selected ring tone - system, 1, 2 or 3. · The user can setup up one number on each of the ring tones 1, 2, and 3. When a call is received from one of these numbers, the respective ring tone will be generated. · If the ring tone does not exist, the system default ring tone will be played. For BT-100, ring 3 is not supported. Send Anonymous If this parameter is set to "Yes", user ID will be sent as anonymous, essentially block the Caller ID from displaying. Anonymous Method Time to ring Special Feature This is decided by supported SIP server/proxy. Either "Use From Header" or "Use Privacy Header" depends on how SIP sever/proxy process the SIP header. Allow user to adjust the ring time of the phone. Default is 60 seconds Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Sonus, Broadsoft, etc. End user please do not touch this parameter. The IP address or URL of System log server, especially useful for ITSP (Internet Telephone Service Provider). End user don't encourage to touch this part Syslog Server - 35 - BudgeTone-100 User Manual Syslog level Grandstream Networks, Inc. Select the level for the phone to report the log messages. [. . . ] It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP. RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 SDP Session Description Protocol, is a format for describing streaming media initialization parameters. [. . . ]