BudgeTone-200 SIP Phone User Manual
Contents
1. Figure 5 BU LO RN TABLE OF TABLES BT200 USER MANUAL Table 1 BT 200 Connectors Table 2 BT200 Key Features in a Table 3 Hardware Specifications Table 4 BT200 Technical Tape S ECOICONS EEN Table 6 200 Keypad Buttons Table 7 BT200 Call Grandstream Networks Inc BT200 User Manual Firmware 1 1 6 44 gt Arandstream Innovative IP Voice amp Video EE 4 5 m X 5 R 5 6 M 6 7 EM 10 H 9 10 12 14 mE 18 PRORA 19 PE 19 O 20 M9 30 30 SE 31 31 32 33 5 TREES 6 Eet 7 nor 7 10 5 7 EPEE IE E 8 T 8 10 13 Sou nase dn M 18 Page 2 of 33 Last Updated 12 2008 gt Aandstream Innovative IP Voice2. SEC EN Temperature 32 104 F 5 45C 00 0 0 Humidity 10 90 non condensing Compliance 0 0 FCC CE C Tick Table 4 BT200 Technical Specifications Protocol Support SIP 2 0 TCP UDP IP PPPoE RTP RTCP SRTP HTTP ARP RARP Support 0 Device NAT friendly remote software upgrade via TFIP HTTP for deployed devices Management including behind firewall NAT Auto manual provisioning system GUI Interface Support Layer 2 802 10 VLAN 802 1p and Layer QoS ToS DiffServ MPLS wm wm wm ww ww ww ww ww ww ww ww ww ww wm PPP Audio Features Full duplex hands free speakerphone headset enabled Advanced Digital Signal Processing DSP Dynamic negotiation of codec and voice payload length Support for PCMU GSM G 723 1 G 729A B G 726 32 G 722 wide band ILBC codecs In band and out of band DTMF in audio RFC2833 SIP INFO Silence Suppression VAD voice activity detection CNG comfort noise generation ANG automatic gain control Acoustic Echo Cancellation AEC with Acoustic Gain Control AGC for Speakerphone mode Support side tone Adaptive jitter buffer control patent pending and packet del
3. Innovative IP Voice amp Video BudgeTone 200 SIP Phone User Manual TABLE OF CONTENTS BT200 USER MANUAL WV EEG INS TAELA HON WHAT IS INCLUDED IN THE cce IIR neret rrr rein CONNECTING YOUR le OT SAFETY COMPLIANCES II eene emnes nem terrere VARRANT RR CREER PRODUCT e UE USING THE GXP SIP ENTERPRISE PHONE GETTING FAMILIAR WITH THE LCD GETTING FAMILIAR WITH KEYPAD HII memet rete rris MAKING PHONE e EE GALL dc ul cic CONFIGURATION GUIDE eegene dee CONFIGURATION MIARK PDAD CONFIGURATION VIA WEB 0 10 reete SAVING THE CONFIGURATION CHANGES 1 REBOOTING THE PHONE REMOTELY SOFTWARE UPGRADE amp FIRMWARE UPGRADE THROUGH TFTP HTTP eeeeennmm mmm CONFIGURATION FILE DOWNLOAD RESTORE FACTORY DEFAULT TABLE OF FIGURES BT200 USER MANUAL Figure 1 BT200 Back TE Figure 2 BT200 Internal Headset Wiring Gchema suus Figure 3 BT200 Front EEN Figure 4 BT200 Side View
4. Default is No If set to Yes BT200 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 supports up to 5 different Vocoder types including G 711 a u 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 Enable SRTP mode based on selection Default is No Default is Standard Choose the selection to meet special requirements from Soft Switch vendors BT200 User Manual Firmware 1 1 6 44 Page 29 of 33 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video SAVING THE CONFIGURATION CHANGES After the user makes a change to the configuration press the Update button in the Configura
5. ndstream Innovative IP Voice amp Video This parameter defines the local RTP RTCP port pair used to listen and transmit It is the base RTP port for channel 0 When configured channel will use this port _value for RTP and the port_value 1 for its RTCP channel 1 will use port_value 2 for RTP and port_value 3 for its RTCP The default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple BT200s are behind the same NAT Default is No This parameter specifies how often the BT200 sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open Default is 20 seconds NAT IP address used in SIP SDP message Default is blank IP address or Domain name of the STUN server STUN resolution result will display in the STATUS page of the Web Ul Default method is HTTP Firmware upgrade may take up to 10 minutes depending on network environment Do not interrupt the firmware upgrading process This is the IP address of the configured TFTP server If selected and it is non zero or not blank the BT200 will attempt to retrieve a new configuration file or new code image from the specified TFTP server at boot time It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory If a TFTP server is configured and a new code image is retrieved the new down
6. button to bring B C in the conference 4 f C does not answer the call can press FLASH back to talk to B NOTE During the conference if B or C drops the call the remaining two parties can still talk However if A the conference initiator hangs up all calls will be terminated Checking Message and Message Waiting Indication When BudgeTone 200 is on hook pressing the MESSAGE button will trigger the phone to call the VM server VMS configured for the Account The MWI Message Waiting Indicator LED will flash in red color in three quarters of a second when voicemail server sends message waiting information to BudgeTone 200 Mute and Delete When in conversation with an ACTIVE LINE pressing MUTE DEL will mute the conversation that is you can hear the other party but the other party cannot hear you Pressing the button again will resume the conversation When dialing a number press MUTE DEL will delete the last entered digit When receiving incoming call press MUTE DEL will Reject the call and forward to voice mail Grandstream Networks Inc BT200 User Manual Page 17 of 33 Firmware 1 1 6 44 Last Updated 12 2008 ndstream Innovative IP Voice amp Video CALL FEATURES BT200 series phone supports a list of call features Caller ID Block or Anonymous Call Disable Enable Call Waiting Call Forward on Busy Delay or Unconditional etc Following table shows the call features of BudgeTone 200 series
7. amp Video Real time Clock Synchronized to Internet time server Time zone configurable via web browser AM PM indicator Call Logs 01 10 for CALLED history dialed number 01 10 for CALLERS history Incoming caller ID Time Icon AM for the morning PM for the afternoon IP Address Separator Icons Numerical Numbers and Characters 0 9 xa TEN A b 6 9 H h 1 9 5 t Y Grandstream Networks Inc BT200 User Manual Page 11 of 33 Firmware 1 1 6 44 Last Updated 12 2008 GETTING FAMILIAR WITH KEYPAD Grandstream Networks Inc GRANDSTREAM Speaker Y Y Send Mute Delete BT200 User Manual Firmware 1 1 6 44 eam Innovative IP Voice amp Video LCD Up Down Keys Message Waiting Indicator Menu Button Called Message Callers Hold Transfer Conference Flash Standard Keypad Page 12 of 33 Last Updated 12 2008 ndstream Innovative IP Voice amp Video Table 6 BT200 Keypad Buttons Key Button Key Button Definitions 0 9 MENU CALLED CALLERS MESSAGE HOLD TRANSFER CONFERENCE FLASH MUTE DEL SEND RE DIAL SPEAKERPHONE Grandstream Networks Inc Digit star and pound keys are usually used to make phone calls 1 Reduce handset speakerphone headset volume after off hook the phone via handset or speaker 2 Reduce ring tone volume when phone in IDLE and off hook to confirm the changed ring tone volum
8. download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades A free Windows version TFTP server is available http support solarwinds net updates New customerFree cfm Grandstream Networks Inc BT200 User Manual Page 31 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Aandstream Innovative IP Voice amp Video Instructions for local TFTP Upgrade 1 Unzip the file and put all of them under the root directory of the TFTP server 2 The PC running the TFTP server and the BT200 should be in the same LAN segment 3 Go to File gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Startthe TFTP server in the phone s web configuration page 5 Configure the Firmware Server Path with the IP address of the PC 6 Update the change and reboot the unit User can also choose to download the free HTTP server from http httpd apache org or use Microsoft IIS web server NOTE e When BT200 phone boots up it will send TFTP or HTTP request to download configuration file cfgO00b82xxxxxx where 000 82 is the MAC address of the BT200 phone This file is for provisioning purpose For normal TFTP or HTTP firmware upgrades the following error messages in a TFTP or HTTP server log can be ignored TFTP Error from IP ADRESS requesting cfgO00b82023dd4 File does not exist Configuration File Download CONFIGURA
9. 711u 2 Press MENU to select new codec Press or 17 to browse list of available codecs 8 Display 8 codE rEL Press Menu to display the code releases Press or 177 to browse 9 Display Phy Addr Press MENU to display the physical MAC address 10 Display 10 ring 0 Grandstream Networks Inc BT200 User Manual Page 19 of 33 Firmware 1 1 6 44 Last Updated 12 2008 ndstream Innovative IP Voice amp Video Press MENU to hear the selected ring tone press or 1 to select the stored ring tones 11 Display 11 trANSPOrt Press MENU to select SIP Transport Factory Reset Display rESEt Please be very careful when using this function Two options factory reset or manual reboot Factory Reset 1 Key in the physical MAC address on back of the phone 2 Press MENU to reset phone to FACTORY DEFAULT settings All your setting will be erased Manual Reboot Press MENU without keying anything Others When phone is powered on and time is displayed Press or t Display ring 4 press or 1 again to hear and adjust the ring tone volume from off to 7 maximum off and on hook to set Press SPEAKERPHONE button or off hook and pick up handset press or 77 to adjust the speakerphone headset or handset volume CONFIGURATION VIA WEB BROWSER The BT200 embedded Web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow a us
10. LOG device MAC adaress error code error message For example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up This parameter defines the URI or IP address of the NTP Network Time Protocol serve It is used to display the current date time Caller ID must be configured Select a Distinctive Ring Tone 1 through 3 for a particular Caller ID The BT200 will ONLY use selected ring tones for particular Caller IDs For all other calls the BT200 will use System Ring Tone When selected and no Caller ID is configured the selected ring tone will be used for all incoming calls System Ring System ring tone Default is North American standard Tone Adjust system ring tone frequencies and cadences based on local telecom standard Grandstream Networks Inc BT200 User Manual Page 25 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Call Progress Tones Disable Call Waiting Disable Direct IP calls Use Quick IP Call Mode Disable Conference Lock keypad update Disable DND Send Flash Event Headset TX gain dB Headset RX gain dB ndstream Innovative IP Voice amp Video Using these settings users can configure ring or tone frequencies based on parameters from local telecom By default they are set to North American standard Frequencies should be configured with known values to avoid uncomfortable high pitch sounds Syntax f1 val f2 val c on1 off1 on2 off2 on3 off3 Fre
11. amp Video Table 8 Key Pad Configuration Men 19 Table 9 Device Configuration 21 Table 10 Device Configuration Basic Gettngs 21 Table 11 Advanced BE tre EE 22 Fable 12 SIP Account En dl e EE 26 GUI INTERFACE EXAMPLES BT200 USER MANUAL http Awww grandstream com user_manuals GUI GUI_BT200 rar SCREENSHOT OF CONFIGURATION LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE SCREENSHOT OF SIP ACCOUNT CONFIGURATION SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE E 22D gc Grandstream Networks Inc BT200 User Manual Page 3 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Aandstream Innovative IP Voice amp Video Welcome Thank you for purchasing Grandstream BudgeTone 200 IP Phone You made an excellent choice and we hope you will enjoy all its capabilities Grandstream s BT200 SIP IP phone is the innovative IP telephone that offers a rich set of functionality and superb sound quality They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market This document is subject to changes without notice The latest electronic version of this user manual is available for download from the following location http www grandstream com resources html Caution Changes or modifications to this product not expressl
12. from server and the phone will try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed e Busy tone keeps playing This means the phone has failed to receive the final response and decide to time out Be advised that this does not indicate the transfer has been successful nor does it indicate the transfer has failed ATTENDED TRANSFER User can transfer an active call to a third party with announcement User presses the FLASH button and hears a dial tone then dial the third party s phone number followed by pressing SEND button If the call is answered press TRANSFER to complete the Grandstream Networks Inc BT200 User Manual Page 16 of 33 Firmware 1 1 6 44 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video transfer operation and hand up if the call is not answered pressing FLASH button to resume the Original call NOTE e When Attended Transfer failed if A hangs up the BudgeTone phone will ring user A back again to remind A that B is still on the call A can pick up the phone to restore conversation with B Conference Call BT200 phone supports 3 way conference Assuming that call party A and B are in conversation A wants to bring C in a conference 1 A presses the CONFERENCE button to get a dial tone and put B on hold 2 Adials C s number then SEND key to make the call 3 answers the call then A presses CONFERENCE
13. phone Table 7 BT200 Call Features Key Call Features 90 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 50 Disable Call Waiting for all subsequent calls 51 Enable Call Waiting for all subsequent calls 70 Disable Call Waiting per Call en Enable Call Waiting per Call 72 Unconditional Call Forward To use this feature dial 72 and get the dial tone Dial the forward number and for a dial tone then hang up 73 Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up 90 Busy Call Forward To use this feature dial 90 and get the dial tone Dial the forward number and for a dial tone then hang up 91 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up 92 Delayed Call Forward To use this feature dial 92 and get the dial tone Dial the forward number and for a dial tone then hang up 93 Cancel Delayed Call Forward To cancel this Forward dial 03 and get the dial tone then hang up Grandstream Networks Inc BT200 User Manual Page 18 of 33 Firmware 1 1 6 44 Last Updated 12 2008 ndstream Innovative IP Voice amp Video Configuration Guide The BT200 can be configured in two ways Firstly using the Key Pad Configuratio
14. 00 Switch LAN port for connecting to Ethernet PC 10 100Mbps RJ 45 ports for PC downlink connection Power Jack 5V DC power port UL Certified HEADSET 2 5mm Headset port 1 Connect attached telephone cord to the handset and to the phone device Grandstream Networks Inc BT200 User Manual Page 5 of 33 Firmware 1 1 6 44 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video Put the handset on the base Connect attached LAN cable to the LAN port of the phone Connect attached power supply first to the 5V power port of the phone Connect attached power supply to the wall outlet pasce When power on the phone starts the initialization procedure Figure 2 BT200 Internal Headset Wiring Schema BT200 handset jack BI200 handset plug abcd headset Jack BT200 power jack Bottom SAFETY COMPLIANCES The BudgeTone 200 phone is compliant with various safety standards including FCC CE Its power adaptor is compliant with UL standard The phone should only be operated with the universal power adaptor provided with the package Damages to the phone caused by using other unsupported power adaptors are not covered by the manufacturer s warranty WARRANTY If you purchased your BT200 from a reseller please contact the company where you purchased your phone for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representa
15. TION FiLE DOWNLOAD The BT200 can be configured via Web Interface as well as via Configuration File through TFTP or HTTP Config Server Path is the TFTP or HTTP server path for the configuration file It needs to be set to a valid URL either in FQDN or IP address format A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 4 digit numeric numbers i e P2 is associated with Admin Password in the ADVANCED SETTINGS page For a detailed parameter list please refer to the corresponding configuration template of the firmware Once the BT200 boots up re booted it will request a configuration file named where is the MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes a Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check for upgrades at pre scheduled time intervals By defining different intervals in P193 for different devices a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time Grandstream Networks Inc BT200 User Manual Page 32 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Aandstream Inno
16. ay amp loss concealment Telephony Voice mail indicator downloadable custom ring tones call hold call transfer Features attended blind call forward call waiting caller ID mute redial call log caller ID display or block and volume control 3 way conference off hook auto dial auto answer early dial and speed dial e em em zm e em zm e em zm zm zm zm zm e e e o e e o e em e e e e o e e em e o e em e zm e e o e e e e e zm e e e e e e o e e e e e om e o zm em zm zm Network and Via keypad LCD Web browser or secure AES encrypted central configuration file Provisioning manual or dynamic host configuration protocol DHCP network setup Grandstream Networks Inc BT200 User Manual Page 8 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Innovative IP ndstream Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802 1p Q tagging VLAN Layer 3 TOS Firmware Support firmware upgrade via TFTP or HTTP Upgrades Support for Authenticating configuration file before accepting changes PESE User specific URL for configuration file and firmwarefiles n Advanced Message waiting indication support DNS SRV Look up and SIP Server Fail Over Server Features Security DIGEST authentication and encryption using MD5 an
17. d MD5 sess SIP over TLS pending Grandstream Networks Inc BT200 User Manual Page 9 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Innovative IP Voice Video Using the GXP SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD BT200 phone has a numeric LCD of 64mmx24mm size with backlight This model has a small red LED status reminder Here is the display when all segments illuminate Figure 5 BT200 LCD me O 225 When the phone is in the normal idle state the backlight is off Whenever an event call occurs the backlight will turn on automatically to bring the user s attention In addition if Voice Mail configured and there is a VM waiting the red LED will be blinking to remind user there is a Voice Mail in the Voice Mail server Table 5 LCD Icons LCD Icon Definitions Connectivity Status SIP Proxy Server Icon OFF IP address of Sip server is not found ON IP address of Sip server are located Blinking Ethernet link failure or the phone is not registered properly Phone Status Icon OFF when the handset is on hook ON when the handset is off hook Speaker Phone Status Icon FLASH when phone rings OFF when the speakerphone is off ON when the speakerphone is on Handset and Speakerphone Headset Volume Icons 0 7 scales to adjust handset speakerphone volume Grandstream Networks Inc BT200 User Manual Page 10 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Innovative IP Voice
18. d NOT touch these parameters Default is No Choose Yes to enable automatic HTTP upgrade and provisioning In Check for upgrade every field enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes When set to the phone will only perform HTTP upgrade and configuration check once at boot up To configure a User ID extension to dial automatically when the phone is taken offhook This parameter sets the payload type for DTMF using RFC2833 Default is 101 Time period when the cradle is pressed Hook Flash to simulate FLASH To prevent unwanted activation of the Flash Hold and automatic phone ring back adjust this time value The IP address or URL of System log server This feature is especially useful for ITSPs oelect the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e Sentor received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS
19. detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to this feature is disabled This field contains the number of voice frames to be transmitted in a single Ethernet packet be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte or 120kbps When setting this value be aware of the requested packet time ptime used in SDP message is a result of configuring this parameter This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first codec is configured as G 723 and the Voice Frames per TX is set to 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G 723 voice frame contains 30ms of audio Similarly if this field is set to 2 and the first codec is G 729 or G 711 or G 726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the IP phone will use and save the maximum allowed value for the corresponding first codec choice The maximum value for PCM is 10 x10ms frames for G 726 it is 20 x10ms frames for G 723 it is 32 x30ms frames for G 729 G 728 64 x10ms and 64 x2 5ms frames respectively Please be careful when editing these parameters Adjusting t
20. e 3 Next menu item browsing when phone is in IDLE mode after MENU key pressed off hook to interrupt and exit Enter keypad MENU mode when phone is in IDLE mode It is also the ENTER key once entering MENU After off hook press to display the dialed numbers When number displayed press the SEND key can make call using that displayed number After off hook press to display the incoming Caller IDs When number displayed press the SEND key can make call using that displayed number Enter to retrieve voice mails from Voice Mail Portal or Server Temporarily hold the active call Transfer the active call to another party Establish 3 way conferencing call Flash event to switch between two lines Mute an active call or Delete a key entry Also used to REJECT incoming call Dial a new number inputted or Redial the number last dialed After entering the phone number pressing this key would force a call to go out immediately before timeout Enter hands free mode 1 Increase handset speakerphone headset volume after off hook the phone via handset or speaker 2 Increase ring tone volume when phone in IDLE and off hook to confirm the changed ring tone volume 3 Next menu item browsing when phone is in IDLE mode after MENU key pressed off hook to interrupt and exit BT200 User Manual Firmware 1 1 6 44 Page 13 of 33 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video MAKING PHONE CALLS Handset Speake
21. e remote end on hold Pressing the Hold button again will release the previously Hold state and resume the bi directional media Call Waiting and Call Flashing If call waiting feature is enabled while the user is in a conversation he will hear a special stutter tone if there is another incoming call User then can press FLASH button to put the current call party on hold automatically and switch to the other call Pressing flash button toggles between two active calls Call Transfer Two transfer operations are supported BLIND TRANSFER User can transfer an active call to a third party without announcement User presses the TRANSFER button and if the other voice channel is available i e there is no other active conversation besides the current one user will hear a dial tone User can then dial the third party s phone number followed by pressing SEND button NOTE e Enable Call Feature has to be configured to Yes in web configuration page in order to make the features to work A can hold on to the phone and wait for one of the three following behaviors e quick confirmation tone temporarily using the call waiting indication tone follows by a dialtone This indicates the transfer has been successful At this point the user can either hang up or make another call e A quick busy tone followed by a restored call On supported platforms only This means the transfer has failed due to the failed response sent
22. each account displayed on LCD SIP Server s IP address or Domain name provided by VoIP service provider BT200 User Manual Firmware 1 1 6 44 Page 26 of 33 Last Updated 12 2008 Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Use DNS SRV User ID is Phone Number SIP Registration Un register on Reboot Register Expiration Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Use RFC3581 Symmetric Routing NAT Traversal STUN Grandstream Networks Inc ndstream Innovative IP Voice amp Video IP address or Domain name of Outbound Proxy Media Gateway or Session Border Controller Used for firewall or NAT penetration in different network environment If the system detects symmetric NAT STUN will not work ONLY outbound proxy can provide solution for symmetric NAT User account information provided by VoIP service provider ITSP either an actual phone number or formatted like one SIP service subscriber s Authenticate ID used for authentication It can be identical to or different from SIP User ID SIP service subscriber s account password for BT200 to register to SIP servers of ITSP SIP service subscriber s name that is used for Caller ID display Default is No If set to Yes the client will use DNS SRV to look up server If the phone has an assigned PSTN telephone number this field should be set to Yes O
23. er to configure the IP phone through a Web browser such as Microsoft s IE or Mozilla Firefox Access the Web Configuration Menu To access the 5 Web Configuration Menu e Connect the computer to the same network as the phone e sure the phone is turned on and shows its IP address e Start a Web browser on your computer e Enter the 5 IP address in the address bar of the browser e Enter the administrator s password to access the Web Configuration Menu The Web enabled computer has to be connected to the same sub network as the phone This can easily be done by connecting the computer to the same hub or switch as the phone is connected to In absence of a hub switch or free ports on the hub switch please connect the computer directly to the phone using the PC port on the phone If the phone is properly connected to a working Internet connection the phone will display its IP address This address has the format xxx xxx xxx xxx where xxx stands for a number from 0 255 You will need this number to access the Web Configuration Menu e g if the phone shows 192 168 0 60 please use hittp 192 168 0 60 in the address bar your browser wo The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing the button on the bottom of the page Reboot the phone to have the changes take effect If after having submit
24. hese parameters will also change the dynamic jitter buffer The BT200 has a patent dynamic jitter buffer handling algorithm The jitter buffer range is 20 200 ms Grandstream recommends using the default settings provided Grandstream does not recommend adjusting these parameters if you are an average user Incorrect settings will affect the voice quality Please refer to the Codec FAQ at http www grandstream com fagscodec html for more technical detail Layer 3 QoS This field defines the layer 3 QoS parameter It is the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Layer 2 QoS This contains the value used for layer 2 VLAN tag Default setting is blank No Key Entry Default is 4 seconds Timeout Use as This parameter allows users to configure the key as the Send or Dial key If set Dial Key to Yes the key will immediately send the call In this case this key is essentially equivalent to the Dial key If set to No the key is included as part of the dial string Grandstream Networks Inc BT200 User Manual Page 23 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server Firmware Upgrade and Provisioning Via TFTP Server Via HTTP Server Config Server Path Firmware File Prefix Postfix Config File Prefix Postfix Allow DHCP Option 66 to override server Authenticate Conf File
25. igit IP address Then press MENU button to save Must reboot the phone to take effect Web Configuration Interface To configure the Upgrade Server via the Web configuration interface open the web browser Enter the BI200 IP address Enter the admin password to access the web configuration interface In the ADVANCED SETTINGS page enter the Upgrade Server s IP address or FQDN in the Firmware Server Path field Select TFTP or HTTP upgrade method Update the change by clicking the Update button Reboot or power cycle the phone to update the new firmware During this stage the LCD will display the firmware file downloading process If a firmware upgrade fails for any reason e g TFTP HTTP server is not responding there are no code image files available for upgrade or checksum test fails etc the phone will stop the upgrading process and re boot using the existing firmware software Firmware upgrades take around 60 seconds in a controlled LAN or 5 10 minutes over the Internet Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible No Local TFTP Server For users who do not have a local TFTP server Grandstream provides a NAT friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically Please check the Support Download section of our website to obtain this TFTP server IP address http www grandstream com firmware html Alternatively
26. ll is forward to a number or VM Default is 20 seconds Default is Yes If set to Yes all star code call features will be supported locally User can choose to disable Call Log 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 Defines the minimum session expiration in seconds Default is 90 seconds If set to Yes the phone will use session timer when it makes outbound calls if remote party supports session timer Defines how long ring will ring when receiving a call Default is 60 seconds If selecting Yes the phone will use session timer when it receives inbound calls with session timer request 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 t
27. loaded image will be verified and then saved into the Flash memory Note Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device The HTTP server URL used for firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream 1 1 6 44 Here 6688 is the specific TCP port that the HTTP server is using omit if using default port 80 Note If Auto Upgrade is set to No BT200 will only perform HTTP download once at boot up IP address or domain name of firmware server Default is blank If configured BT200 will request the firmware file with the prefix postfix This setting is useful for ITSPs End user should keep it blank Default is blank End user should keep it blank Default is Yes This allows device gets provisioned automatically Default is If set to Yes configuration file would be authenticated before acceptance End user should use default setting Grandstream Networks Inc BT200 User Manual Page 24 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Automatic Upgrade Offhook Auto Dial DTMF Payload Type Onhook Threshold Syslog Server Syslog Level NTP server Distinctive Ring Tone ndstream Innovative IP Voice amp Video This function is used by ITSP End user shoul
28. n Menu on the phone secondly through embedded web configuration menu CONFIGURATION VIA KEYPAD The BT200 keypad is the same as a traditional phone with additional feature functionality including speaker menu keys LCD navigation keys and advanced feature keys which include hold transfer conference and flash 1 sure the phone is idle 2 Press the MENU button to enter the keypad MENU to configure the phone Using the Keypad 1 To enter the MENU press MENU button 2 Navigate the menu by using the UP DOWN 3 Press the MENU button to confirm a menu selection 4 delete an entry by pressing the MUTE DEL button Press the MENU button to enter the key the Key Pad Menu The menu options available are listed in table 8 Table 8 Key Pad Configuration Menu Manual Option Call Features 1 Display 1 dhcP On or 1 dhcP ott for the current selection Press MENU key to enter edit mode 2 Display 2 IP Addr Press MENU to display the current IP address Enter new IP address if DHCP is OFF 3 Display 3 SubNet Press MENU to display the Subnet mask Enter new Subnet mask if DHCP is OFF 4 Display 4 routEr Press MENU to display the Router Gateway address Enter new Router Gateway address if DHCP is OFF 5 Display 5 dnS Press MENU to display the DNS address Enter new DNS address if DHCP is OFF 6 Display 6 tFtP Press MENU to display the TFTP address Enter new TFTP server IP address 7 Display 7 G
29. o 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 BT200 User Manual Firmware 1 1 6 44 Page 28 of 33 Last Updated 12 2008 Enable 100rel Account Ring Tone 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 Preferred Vocoder SRTP Mode Special Feature Grandstream Networks Inc ndstream Innovative IP Voice amp Video 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 e One 1 System Ring Tone when selected all calls will ring with system ring tone e Three 3 Customer Ring Tones when selected incoming calls from designated account will play selected ring tone 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 Qanonymous invalid2 in the From Header or P Asserted Identity header Default is NO If set to YES anonymous call will be rejected
30. quencies are in Hz and cadence on and off are in 10ms ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Up to three cadences are supported Default is No If set to Yes the call waiting will be disabled Default is No If set to Yes direct IP calls will be disabled Dial an IP address under the same LAN VPN segment by entering the last octet in the IP address In the Advanced Settings page there is an option Use Quick IP call mode Default setting is No When set to YES and XXX is dialed where X is 0 9 and XXX 7255 phone will make direct IP call to aaa bbb ccc XXX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask 4 XX or X are also valid so leading 0 is not required but OK See Quick IP Call Mode for details Default is No If set to Yes conference calls will be disabled If set to Yes the configuration changes via keypad are disabled Default is No If set to Yes DND key on keypad will be disabled Default is No If set to yes flash will be sent as DTMF event Transmission gain Its a headset setting to control the voice intensity Receive gain It s a headset setting to control the voice intensity Table 12 SIP Account Settings Account Name SIP Server Grandstream Networks Inc The name associated with
31. quired but OK eg 192 168 0 2 calling 192 168 0 3 just dial 3 follow by SEND or 192 168 0 2 calling 192 168 0 23 just dial 23 follow by SEND or 192 168 0 2 calling 192 168 0 123 just dial 123 follow by SEND or 192 168 0 2 dial 3 and 03 and 003 has same effect gt call 192 168 0 3 Note If you have a SIP Server configured Direct IP IP call will still work However if you are using STUN Direct IP IP call will also use STUN OR To make a direct IP to IP call first off hook then press MENU key then enter a 12 digit target IP address to make the call If port is not default 5060 destination ports can be specified by using 4 encoding for followed by the port number Examples Ifthe target IP address is 192 168 0 10 the dialing convention is MENU key 192 168 000 010 followed by pressing the SEND key or wait for seconds in the No Key Entry Timeout If the target IP address port is 192 168 1 20 5062 then the dialing convention would be MENU key 192168001020 45062 followed by pressing the SEND key wait for seconds in the No Key Entry Timeout Quick IP Call Mode The BT200 also supports Quick IP call mode This enables the phone to make direct IP calls using only the last few digits last octet of the target phone s IP number This is possible only if both phones are in under the same LAN VPN This simulates a PBX function using the CMSA CD without a SIP server Controlled
32. rphone and Headset Mode The regular Handset mode can be switched with either the Speaker mode Hand free or the Headset mode however whenever the Headset is plugged in Speaker mode will be switched to the Headset mode automatically To Switch between Handset and Speaker Headset simply press the Hook Flash in the Handset cradle or the Speaker button Make Calls using Numbers There are FIVE ways to make phone calls e Pick up handset or press SPEAKERPHONE button and then enter the phone numbers e Press the SEND button directly to redial the number last called e Once pressed the last dialed number will be displayed on the LCD as the corresponding tones are played out and an outgoing call is sent e Browse the CALLED CALLER history and press the SEND REDIAL button e Pick up the handset or press the speakerphone button then press the CALLED CALLERS button to browse thru the last 10 numbers dialed out Once the desired number is identified and displayed on the LCD screen press the SEND button and a new call to that displayed number will be sent out immediately Examples e dial another extension on the same proxy such as 1008 simply pick up handset or press speakerphone dial 1008 and then press the SEND button To dial a PSTN number such as 6266667890 you might need to enter in some prefix number followed by the phone number Please check with your VoIP service provider to get the information If you phone i
33. s assigned with a PSTN like number such as 6265556789 most likely you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone followed by pressing the SEND button Make Calls using IP Address Direct IP calling allows two parties that is a BT200 and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy This kind of VoIP calls can be made between two parties if e Both BT200 phone and other VoIP Device i e another IP Phone or BT200 SIP phone or other VoIP unit have public IP addresses or e Both BT200 phone and other VoIP Device are on the same LAN using private or public IP addresses or e Both BT200 phone and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ This model has the ability to dial an IP address under the same LAN segment by simply pressing the last octet in the IP address In the Advanced Settings page there is an option Use Quick IP call mode by default it is set to No When this option is set to YES and XXX is dialed where X is 0 9 and XXX 7 255 phone will make direct IP call to aaa bbb ccc XXX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask Grandstream Networks Inc BT200 User Manual Page 14 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Aandstream Innovative IP Voice amp Video XX or X are also valid so leading is not re
34. s from the second Sunday of March at 2AM and ends the first Sunday of November at 2AM The saving is 60 minutes Choose one of the following formats e Year Month Day Month Day Year e Day Month Year Choose to display Account Name or date on LCD Default is No This field lets user to choose whether to ring the phone Speaker when headset is connected Advanced User configuration includes not only the end user configuration but also advanced configuration such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration Table 11 Advanced Settings Admin Administrator password Only the administrator can access the Advanced Settings and Password Account Settings page Password field is purposely blank for security reasons after clicking update and saved The maximum password length is 25 characters Grandstream Networks Inc BT200 User Manual Page 22 of 33 Firmware 1 1 6 44 Last Updated 12 2008 G723 rate iLBC frame size ILBC payload type Silence Suppression Voice Frames per TX ndstream Innovative IP Voice amp Video Encoding rate for G723 codec By default 6 3kbps rate is set iLBC packet frame size Default is 20ms For Asterisk PBX 30ms might be required Payload type for iLBC Default value is 97 The valid range is between 96 and 127 This controls the silence suppression VAD feature of the audio codec 723 and G 729 If set to Yes when silence is
35. static IP usage is recommended Setting up the phone to make Quick IP calls To enable Quick IP calls the phone has to be setup first This is done through the web setup function In the Advanced Settings page set the Use Quick IP call mode to YES When xxx is dialed where x is 0 9 and xxx lt 255 a direct IP call to aaa bbb ccc XXX is completed aaa bbb ccc is from the local IP address regardless of subnet mask The numbers xx or x are also valid The leading is not required but OK For example 192 168 0 2 calling 192 168 0 3 dial 3 follow by SEND or 192 168 0 2 calling 192 168 0 23 dial 23 follow by SEND or 192 168 0 2 calling 192 168 0 123 dial 123 follow by SEND or 192 168 0 2 dial 3 and 03 and 003 results in the same call call 192 168 0 3 NOTE f you have a SIP Server configured a Direct IP IP still works If you are using STUN the Direct IP IP call will also use STUN Configure the Use Random to NO when completing Direct IP calls Grandstream Networks Inc BT200 User Manual Page 15 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Aandstream Innovative IP Voice amp Video Answer an Incoming Call There are two ways to answer an incoming call 1 Pick up the handset to answer the call normally using handset or 2 Press the SPEAKERPHONE button to answer in speakerphone or headset mode Call Hold While in conversation pressing the Hold button will put th
36. t Restricted Cone the phone will use its mapped public IP address and port in all of its SIP and SDP messages If the NAT Traversal field is set to Yes with no specified STUN server the BT200 will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open BT200 User Manual Page 27 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Subscribe for MWI Proxy Require Voice Mail UserlD Send DTMF Early Dial Dial Plan Prefix Delayed Call Forward Wait Time Enable Call Features Disable Call Log Session Expiration Min SE Caller Request Timer Ring Timeout Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Grandstream Networks Inc ndstream Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically SIP Extension to notify SIP server that the unit is behind the NAT Firewall When configured user can access messages by pressing MSG button This ID is usually the VM portal access number This parameter specifies the mechanism to transmit DTMF digit There are 3 supported modes in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO Default is No Use only if proxy supports 484 response Sets the prefix added to each dialed number Time waited before the ca
37. ted some changes more settings have to be changed press the menu option needed Definitions Grandstream Networks Inc BT200 User Manual Page 20 of 33 Firmware 1 1 6 44 Last Updated 12 2008 ndstream Innovative IP Voice amp Video This section will describe the options in the Web configuration user interface As mentioned a used can log in as an administrator or end user Functions available for the end user are e Status Displays the network status account statuses software version and MAC address of the phone e Basic Basic preferences such as date and time settings multi purpose keys and LCD settings can be set here Additional functions available to administrators are e Advanced Settings To set advanced network settings codec settings e Account To configure each of the SIP accounts Table 9 Device Configuration Status MAC Address The device ID in HEXADECIMAL format IP Address This field shows IP address of BT200 Product Model This field contains the product model information Part Number This field contains the product part number Software Version e Program This is the main software firmware release number always used to identify the software firmware system of the phone e Boot Booting code version number System Up Time This field shows system up time since the last reboot System Time This field shows the current time in the phone Registered Indicates whether accounts are registered to
38. the related SIP server s BT200 can support four unique SIP profiles PPPoE Link Up Indicates whether the PPPoE connection is enabled connected to a modem Table 10 Device Configuration Basic Settings End User This contains the password to access the Web Configuration Menu This field is Password case sensitive with a maximum length of 25 characters IP Address There BT200 operates in two modes 1 DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The BT200 acquires its IP address from the first DHCP server it discovers on its LAN The DHCP option is reserved for NAT router mode To use the PPPoE feature set the PPPoE account settings The BT200 establishes a PPPoE session if any of the PPPoE fields are set 2 Static IP mode configure all of the following fields IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary These fields are set to zero by default Grandstream Networks Inc BT200 User Manual Page 21 of 33 Firmware 1 1 6 44 Last Updated 12 2008 Time Zone Daylight Savings Time Date Display Format Display Account Name instead of Date Mute Speaker Ringer ndstream Innovative IP Voice amp Video This parameter controls the date time display according to the specified time zone This parameter controls time displayed in daylight savings time If set to Yes then the displayed time
39. therwise set it to No If Yes is set a user phone parameter will be attached to the From header in SIP request This parameter controls sending REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s registration information will be cleared on reboot This parameter allows user to specify the time frequency in minutes that BT200 refreshes its registration with the specified registrar The default interval is 60 minutes The maximum interval is 65 535 minutes about 45 days This parameter defines the local SIP port used to listen and transmit The default value for Account 1 is 5060 It is 5062 5064 5066 for Account 2 Account 3 and Account 4 respectively Retry registration if the process failed Default is 20 seconds RFC 3261 SIP T1 timer Default is 1 second RFC 3261 SIP T2 timer Default is 0 5 seconds Choose SIP Transport between UDP and TCP Default is UDP This option allows a SIP client to request that the server send the response back to the source IP address and port from which the request originated This parameter activates the NAT traversal mechanism If activated by choosing Yes and a STUN server is also specified the phone performs according to the STUN client specification Using this mode the embedded STUN client detects if and what type of NAT Firewall configuration is used If the detected NAT is a Full Cone Restricted Cone or a Por
40. tion Menu The web browser will then display a message window to confirm saved changes Grandstream recommends reboot or power cycle the IP phone after saving changes REBOOTING THE PHONE REMOTELY Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again Grandstream Networks Inc BT200 User Manual Page 30 of 33 Firmware 1 1 6 44 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video Software Upgrade amp Customization Software or firmware upgrades are completed via either TFTP or HTTP The corresponding configuration settings are in the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP To upgrade via TFTP or HTTP select TFTP or HTTP upgrade method Upgrade Server needs to be set to a valid URL of a HTTP server Server name can be in either FQDN or IP address format Here are examples of some valid URLs e firmware mycompany com 6688 Grandstream 1 1 6 44 e 168 75 215 189 There are two ways to set up the Upgrade Server to upgrade firmware via Key Pad Menu and Web Configuration Interface Key Pad Menu Only TFTP would be used when configure the phone using key pad To configure the Upgrade Server via Key Pad Menu options select 6 tFtP from the Main Menu then press MENU button to enter this option Enter the 12 d
41. tive for a RMA Return Materials Authorization number before you return the product Grandstream reserves the right to remedy warranty policy without prior notification Grandstream Networks Inc BT200 User Manual Page 6 of 33 Firmware 1 1 6 44 Last Updated 12 2008 ream Innovative IP Voice amp Video Product Overview The following photo illustrates the appearance of a BudgeTone 200 IP phone Figure 3 BT200 Front View Figure 4 BT200 Side View Table 2 BT200 Key Features in a Glance Features Benefits vm wm wm e wm wm e wm wm wm wm wm A wm wm em em wm em wm wm wm em wm wm wm Open Standards Compatible Support SIP 2 0 TCP UDP IP PPPoE RTP RTCP SRTP HTTP ARP RARP ICMP DNS DHCP NTP TFTP Superb Audio Quality Advanced Digital Signal Processing DSP Silence suppression O EES VAD CNG AGC Network Interfaces Dual 10 100mbps Ethernet ports Feature Rich Traditional voice features including caller ID call waiting hold transfer forward block autodial off hook dial and click
42. to dial Advanced Features Dedicated buttons for hold send speakerphone headset transfer 3 way conference mute message Grandstream Networks Inc BT200 User Manual Page 7 of 33 Firmware 1 1 6 44 Last Updated 12 2008 gt Aandstream Innovative IP Voice amp Video Advanced Functionality Custom down loadable ring tones SRTP SIP over TLS pending multi language support and adjustable positioning angles wall mountable AES encryption Table 3 Hardware Specifications LAN Interface Ethernet ports Two 2 10 100 Mbps Full Half Duplex Ethernet Switch with LAN and PC port with auto detection e mm gt wm mm wm wm wm wm ww wm wm www ww www ww ww Headset Jack 2 5mm Headset port LED A LEDINRED olor Phone Case 25 button keypad 12 digit caller ID LCD Universal Switching 100 240 50 60Hz Power Adaptor amp Output SvDC 1200mA UL certified 1 Dimension 220mm x 180mm w x Domm
43. vative IP Voice amp Video Restore Factory Default Setting WARNING Restoring the Factory Default Setting will delete all configuration information of the phone Please backup or print all the settings before you restoring factory default settings Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider INSTRUCTIONS FOR RESTORATION Step 1 Press the MENU button for Key Pad Menu options and press the Up button to see reset Step 2 Enter the MAC address printed on the bottom of the sticker Please use the following mapping 0 9 0 9 A 22 press the 2 key twice A will show on the LCD B 222 C 2222 D 33 press the 3 key twice D will show on the LCD E 333 F 3333 Example if the MAC address is 000582006395 it should be key in as 0002228200333395 NOTE f there are digits like 22 in the MAC you need to type 2 then press gt right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2 Step 3 Press the MENU button again If the MAC address is correct the phone will reboot Wait for phone reboot and the LCD backlight finish flashing Grandstream Networks Inc BT200 User Manual Page 33 of 33 Firmware 1 1 6 44 Last Updated 12 2008
44. will be 1 hour ahead of normal time The Optional Rule is configured to automatically adjust the Daylight Savings Time DST based on the rule set in this field Rule Syntax e Start time end time saving e Both start time and end time have the same syntax month day weekday hour minute o month 1 2 3 12 for Jan Feb Dec o day 1 2 3 31 o weekday 1 2 3 7 for Tue Sun or 0 which means the daylight saving rule is not based on week days but based on the day of the month o hour hour 0 23 minute minute 0 59 If weekday is 0 it means the date to start or end daylight saving is at exactly the given date In that case the day value must not be negative If weekday is not zero and day is positive then the daylight saving starts on the first day the iteration of the weekday e g 1st Sunday 3rd Tuesday etc If weekday is not zero and day is negative then the daylight saving starts on the last day the iteration of the weekday e g last Sunday 3rd last Tuesday etc The saving is in the unit of minutes The saving time may also be preceded by a negative sign if subtraction is desired instead of addition The default value is set for US the Automatic Daylight Saving Time Rule shall be set to 43 2 7 2 0 11 1 7 2 0 60 Examples US Canada where daylight saving time is applicable 03 02 7 02 00 11 1 7 02 00 60 This means the daylight saving time start
45. y approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Warning Please do not use a different power adaptor with the BT200 as it may cause damage to the products and void the manufacturer warranty e This document is contains links to Grandstream GUI Interfaces Please download these examples http www grandstream com user manuals GUI GUI BT200 rar for your reference e This document is subject to change without notice The latest electronic version of this user manual is available for download http www grandstream com user_manuals BT200 User Manual pdf e Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted Grandstream Networks Inc BT200 User Manual Page 4 of 33 Firmware 1 1 6 44 Last Updated 12 2008 fandstream Innovative IP Voice amp Video Installation WHAT IS INCLUDED IN THE PACKAGE The BudgeTone 200 phone package contains One BudgeTone 200 Main Case One Handset One Phone Cord One Universal Power Adapter One Ethernet Cable pua oh gt CONNECTING YOUR PHONE Following is a backside picture of BT200 each connection port is labeled with the name in the following table Figure 1 BT200 Back Panel PC LAN POWER HEADSET Table 1 BT200 Connectors LAN 10 1
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