User Manual
Contents
1. Speakerphone Send Dial Mute Delete 14 Key Button 0 9 MENU CALLED CALLERS MESSAGE HOLD TRANSFER CONFERENCE FLASH MUTE DEL Key Button Definitions 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 volume 3 Next menu item browsing when phone is in IDLE mode after MENU key pressed off hook to interrupt and exit 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 Previous 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 between2. There are two ways to answer an incoming call l Pick up the handset to answer the call normally using handset or 2 Press the SPEAKERPHONE button to answer in speakerphone or headset mode 4 3 5 Handset Mode Speakerphone Headset Mode Handset mode and Speakerphone Headset mode cannot be enabled at the same time Pressing the hook switch or Speakerphone button would toggle the phone between these two modes 4 3 6 Call Hold While in conversation pressing the Hold button will put the remote end on hold Pressing the Hold button again will release the previously Hold state and resume the bi directional media 4 3 7 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 4 3 8 Call Transfer Two transfer operations are supported 18 4 3 8 1 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 1 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 ha
3. Info Turn off speaker on remote disconnect El El vas gt item ring tone custom ring tone I custom ring tone 2 custom ring tone 3 Ea El caller ID will be blocked if set to Yes Ei EJ ER El Eno EF 38 Preferred Vocoder choice 2 in listed order Special Feature Standard choice 1 choice 5 choice 6 T choice 3 choice 7 choice 4 choice 8 Update Individual Account Settings Account Active This field indicates whether the account is active or not The default Account Name SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Use DNS SRV value for the primary account Account 1 is Yes The default values for the other three accounts are No A name to identify an account which will be displayed in LCD SIP Server s IP address or Domain name provided by VoIP service provider IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by BudgeTone 200 for firewall or NAT penetration in different network environment If symmetric NAT is detected STUN will not work and ONLY outbound proxy can provide solution for it User account information provided by VoIP service provider ITSP usually has the form of digit similar to phone number or actually a phone number SIP service subscriber s Authenticate ID used for authentication Can be identical
4. Must recycle power to take effective Display 6 tFtP ai Press MENU to display the TFTP address Enter new TFTP server IP address Press MENU to save Press or 1 to exit Display 7 G 711u 2 Press MENU to select new codec Press or f to browse a list of available codecs line2 G 711A 2 SE G23 R A Gao ee Base at GS i Press 1 to 9 to indicate number of frames per TX packet Press MENU to save and exit Must recycle power to take effective Display 8 SIP SP 1 Reserve for future products Display 9 codE rEL Press Menu to display the code releases Press or f to browse line I b 2006 03 14 date boot code 2 1 1 0 1 version boot code 3 P 2006 04 28 date phone code 4 1 1 0 13 version phone code 5 lr 2004 05 12 date 1 ring tone 6 0 0 0 0 version ring tone 7 2r 2004 05 12 date 2 ring tone 8 0 0 0 0 version ring tone 9 3r 0000 00 00 date 3 ring tone 10 0 0 0 0 version ring tone all zeroes means unavailable or unsupported Press MENU to exit 23 Menu Item 10 11 Others Menu Functions Display 10 Phy Addr Press MENU to display the physical MAC address Press or 1 to exit Display 11 ring 0 Press MENU to hear the selected ring tone press or t to select the stored ring tones Now only 3 are availa
5. An FXO interface will accept calls from FXS or PSTN interfaces All countries and regions have their own standards FXO is complimentary to FXS and the PSTN Foreign eXchange Station An FXS device has hardware to generate the ring signal to the FXO extension usually an analog phone An FXS device will allow any FXO device to operate as if it were connected to the phone company This makes your PBX the POTS PSTN for the phone The FXS Interface connects to FXO devices by an FXO interface of course The Dynamic Host Configuration Protocol DHCP is an Internet protocol for automating the configuration of computers that use TCP IP DHCP can be used to automatically assign IP addresses to deliver TCP IP stack configuration parameters such as the subnet mask and default router and to provide other configuration information such as the addresses for printer time and news servers ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call In addition to improving quality this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network 50 There are two types of echo of relevance in telephony acoustic echo and hybrid echo Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks H 32
6. after clicking update and saved The maximum password length is 25 characters This controls the silence suppression VAD feature of G723 and G729 If set to Yes when a silence is detected small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first vocoder is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 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 BudgeTone 200 will use and save the maximum allowed value for the corresponding first vocoder choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729
7. Enable Call Features Disable Missed Call Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher When configured user will be able to dial voice mail server by pressing MSG button This parameter specifies the mechanism to transmit DTMF digit There are 3 modes supported 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 Default is No If set to Yes Call transfer Call Forwarding amp Do Not Disturb are supported locally Default is No If set to Yes missed calls will not be recorded for your review Grandstream implemented SIP Session Timer The 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 will be terminated Session Expiration is the time in seconds at which the session is considered timed out if 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
8. border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems IVR IVR is a software application that accepts a combination of voice telephone input and touch tone keypad selection and provides appropriate responses in the form of voice fax callback e mail and perhaps other media MTU A Maximum Transmission Unit MTU is the largest size packet or frame specified in octets eight bit bytes that can be sent in a packet or frame based network such as the Internet The maximum for Ethernet is 1500 byte 51 NAT Network Address Translation NTP Network Time Protocol a protocol to exchange and synchronize time over networks The port used is UDP 123 Grandstream products using NTP to get time from Internet OBP SBC Outbound Proxy or another name Session Border Controller A device used in VoIP networks OBP SBCs are put into the signaling and media path between calling and called party The OBP SBC acts as if it was the called VoIP phone and places a second call to the called party The effect of this behaviour is that not only the signaling traffic but also the media traffic voice video etc crosses the OBP SBC Without an OBP SBC the media traffic travels directly between the VoIP phones Private OBP SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network Public VoIP service providers use OBP SBCs to allow the use of VoIP protocols from pri
9. 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 41 Force INVITE Enable 100rel Account Ring Tone Send Anonymous Auto Answer Allow Auto Answer by Call Info Turn off speaker on remote disconnect Preferred Vocoder Special Feature 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 signalling messages There are 4 different ring tone that are de
10. will be sent out immediately Examples To 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 is assigned with a PSTN like 16 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 4 3 3 Make Calls using IP Address Direct IP calling allows two parties that is a BudgeTone phone 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 BudgeTone phone and other VoIP Device i e another IP Phone or BudgeTone SIP phone or other VoIP unit have public IP addresses or e Both BudgeTone phone and other VoIP Device are on the same LAN using private or public IP addresses or e Both BudgeTone 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 m
11. 2 1 Value is set in units of hours Default value is 120hr 5 Days The time IP address is assigned to the LAN clients Forward all WAN IP traffic to a specific IP address if no matching port is used by HandyTone 486 itself or in the defined port forwarding Allow the user to forward a matching TCP UDP port to a specific LAN IP address with a specific TCP UDP port In addition to the Basic Settings configuration page end user also has access to the device Status page The following is a screen shot of the device Status page Details are explained next MAC Address IP Address Product Model Software Version System Up Time Registered PPPoE Link Up _ ADVANCED SETTINGS ACCOUNT 00 0B 82 08 3D 6E 192 168 1 113 BT200 Program 1 1 0 13 Bootloader 1 1 0 1 0 day s 7 hour s 7 minute s Account Yes disabled detected NAT type is full cone 29 MAC Address IP Address Product Model Software Version System Up Time Registered PPPoE Link Up Detected NAT Type The device ID in HEX format This is a very important ID for ISP troubleshooting This field shows LAN IP address of BudgeTone 200 This field contains the product model info e Program This is the main software release its number is always used for firmware upgrade e Bootloader This is normally not changed This field shows system up time since the last reboot This field indicates whether the device is reg
12. 3 A suite of standards for multimedia conferences on traditional packet switched networks HTTP Hyper Text Transfer Protocol the World Wide Web protocol that performs the request and retrieve functions of a server IP Internet Protocol A packet based protocol for delivering data across networks IP PBX IP based Private Branch Exchange IP Telephony Internet Protocol telephony also known as Voice over IP Telephony A general term for the technologies that use the Internet Protocol s packet switched connections to exchange voice fax and other forms of information that have traditionally been carried over the dedicated circuit switched connections of the public switched telephone network PSTN The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression translation of the signal into Internet protocol IP packets for transmission over the Internet or other packet switched networks the process is reversed at the receiving end The terms IP Telephony and Internet Telephony are often used to mean the same however they are not 100 per cent interchangeable since Internet is only a subcase of packet switched networks For users who have free or fixed price Internet access IP Telephony software essentially provides free telephone calls anywhere in the world However the challenge of IP Telephony is maintaining the quality of service expected by subscribers Session
13. D 53 Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing wherein the presence or absence of human speech is detected from the audio samples VLAN A virtual LAN known as a VLAN is a logically independent network Several VLANs can co exist on a single physical switch It is usually refer to the IEEE 802 1Q tagging protocol VoIP Voice over IP VoIP encompasses many protocols All the protocols do some form of signalling of call capabilities and transport of voice data from one point to another e g SIP H 323 etc 54
14. G728 64 x10ms and 64 x2 5ms frames respectively This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv or MPLS Default value is 48 This contains the value used for layer 2 VLAN tag Default setting is blank Default is 4 seconds This parameter allows users to configure the key to be used as the Send or Dial key If set to Yes pressing this key will immediately trigger the sending of dialed string collected so far In this case this key is essentially equivalent to the Re Dial key If set to No this key will then be included as part of the dial string to be sent out 33 Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server Firmware Upgrade and provisioning Via TFTP Server Via HTTP Server Allow DHCP Option 66 to override server This parameter defines the local RTP RTCP port pair the BudgeTone 200 will listen and transmit It is the base RTP port for channel 0 When configured channel 0 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 BudgeTone 200s are behind the same NAT This parameter specifies how often the BudgeTone 200 send
15. GUIDE esevoesnvvenvevnnnvsnnnennnennnnnnnnnnenenee 22 5 1 CONFIGURATION WITH KEYPAD 2 2 i0 scecsceeeiocs aiid buses aeedtdeeeweres 22 5 2 CONFIGURATION WITH WEB BROWSER seceeeee ence ence ees 25 5 2 1 Access the Web Configuration Menu ronnrnanonnrnanernrnennrr 25 5 2 2 End User Conhguratnonsanseas aassaasnorrenmnonei 4 25 5 2 3 Advanced User Configuration rornnnrnonnrnnnonnnnrnnrnrnenernen 30 5 2 4 Saving the Configuration Changes 43 5 2 5 Rebooting the Phone from Remote 43 5 3 CONFIGURATION THROUGH CENTRAL PROVISIONING SERVER 44 6 FIRMWARE UPGRADE Giss 45 6 1 UPGRADE THROUGH HTTP 0 ccc cece cece cece ence ceeee cee ceeaeeecs 45 6 2 UPGRADE THROUGH TFTP kann 45 7 RESTORE FACTORY DEFAULT SETTING 00 47 APPENDIX I GLOSSARY OF TERMG cccccccsccccccccccccees 48 1 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 BudgeTone 200 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 ww
16. T between the client and server transactions If the network latency is high select bigger value for reliable usage This element sets the value of the SIP protocol T2 timer in seconds Timer T2 defines the retransmit interval for INVITE responses and non INVITE requests The SIP protocol default value is 4 seconds This parameter defines whether the BudgeTone 200 NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the BudgeTone 200 will behave according to the STUN client specification Under this mode the embedded STUN client inside the BudgeTone 200 will attempt to detect if and what type of firewall NAT it is sitting behind through communication with the specified STUN server If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the BudgeTone 200 will attempt to 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 BudgeTone 200 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 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 40 Voice Mail User ID Send DIMF Early Dial Dial Plan Prefix
17. User Manual BudgeTone 200 Series IP Phone For Firmware Version 1 1 0 13 Grandstream Networks Inc WWW grandstream com CA DET DRES essen Table of Contents L WELCOME 4 2 INSTALLATION wecsvesvesieesncads pep verve tedvaetesiiidveresskeucaed 5 2 1 WHAT IS INCLUDED IN THE PACKAGE ooouenanonnnnonnnonnnnennanenenenen 5 2 2 CONNECTING YOUR PHONE ssrnsesran barnevern sa bang 5 20 SAFETY COMPLIANCES ep a E une 6 2A MRS 6 3 PRODUCT OVERVIEW Luse 8 Sol KREV FEATURES sc 205i e ea o aa Mhiaeadoae REER 9 3 2 HARDWARE SPECIFICATION ci csecavawseuenysnusnse biden daesuabnawnrsskseans 10 4 USING BUDGETONE 200 IP PHONE cccseccscccessccesess 12 4 1 GETTING FAMILIAR WITH UCD is ciiccsscgstetns elagicoieeresinoeasestile 12 4 2 GETTING FAMILIAR WITH KEYPAD ec cee eee eee e eee cnet eeeeees 14 4 3 MAKING AND ANSWERING PHONE CALLS 0ceeceeee ence eee ees 16 4 3 1 Handset Speakerphone and Headset Mode 0 000 00000 16 4 3 2 Multiple SIP Accounts and Lines 16 4 3 3 Making CANS pasna a Vane cian aba nag thes Saat amas 17 4 3 4 Making Calls using IP Address 18 4 3 5 Receiving CA Busasskseomer dekke a aaaea 18 4 3 6 El Hoa 18 4 3 7 Call Waiting and Switch between Calls rornrnnnnnnrnnnonnnnrr 18 4 3 8 GE ER OSE EEE EE ON 18 4 3 9 Ce edE 19 4 3 10 Checking Message and Message Waiting Indication 19 4 3 11 Mute and Dl us Aae 20 AA CPA eee 20 5 CONFIGURATION
18. When BudgeTone 200 boots up it will send TFTP or HTTP request to download configuration files there are two configuration files one is cfg txt and the other is cfg000b82xxxxxx where 000b82xxxxxx is the MAC address of the BudgeTone 200 These two files are for initial automatically provisioning purpose only for normal TFTP or HTTP firmware upgrade the following error messages in a TFTP or HTTP server log can be ignored FTP Error from IP ADRESS requesting cfg000b82023dd4 File does not exist FTP Error from IP ADRESS requesting cfg txt File does not exist 46 7 Restore Factory Default Setting Warning Restore the Factory Default Setting will DELETE all configuration information of the device Please backup or print out all the settings before you approach to following steps Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your service provider Step 1 Find the MAC Address of the device The MAC address of the device is located on the bottom of the device It is a 12 digit number User can also use Menu option 10 to find out the phone s MAC address Step 2 Encode the MAC address Please use the following mapping 0 9 0 9 A 22 B 222 C 2222 D 33 E 333 F 3333 For example if the MAC address is 000b8200e395 it should be encoded as 0002228200333395 Step 3 Access the phone screen menu select reset wi
19. 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 lt 255 phone will make direct IP call to aaa bbb ccc X XX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask XX or X are also valid so leading 0 is not required 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 If this parameter is set to Yes the configuration updates via keypad for Menu Item 7 9 12 are disabled 36 Following is the screenshot of the Account Configuration Page Account Active ER FR Account Name Mycompany te g MyCompany SIP Server sip mycompany com co g Sip mycompany com or IP address Er e g proxy myprovider com or IP address if any SIP User ID E user part of an SIP address Authenticate ID FE be identical to or different from SIP User ID PG purposely not displayed for security protection Name TTT sona e g John Doe Use DNS SRV ER EN User ID is p
20. ayer 3 QoS ToS DiffServ MPLS e Support firmware upgrade via TFTP or HTTP e Support DNS SRV Look up and SIP Server Fail Over e Acoustic Echo Cancellation AEC with Acoustic Gain Control AGC for speakerphone mode e Support for Authenticating configuration file before accepting changes e allow user to specify different URL for configuration file and firmware files Hardware Features e Support Headset which will auto switch to Headset when plugged in e Support 10 100 Full Half Duplex Ethernet Switch with LAN and PC port Ethernet polarity can be auto detected thus either straight through or twist cable can be used e Support Message Waiting Indication LED 10 3 2 Hardware Specification The table below describes the hardware specification of BudgeTone 200 Model LAN interface Headset Jack LED Phone Case Universal Switching Power Adaptor Dimension Weight Temperature Humidity Compliance BudgeTone 200 2xRJ45 10 100Base T 2 5mm Headset port 1 LED in RED color 25 button keypad 12 digit caller ID LCD Input 100 240VAC 50 60 Hz Output SVDC 1200mA UL certified 18cm W 22cm D 6 5cm H 0 9kg 21bs 40 130 F 5 45 C 10 90 non condensing FCC CE C Tick 11 4 Using BudgeTone 200 IP Phone 4 1 Getting Familiar with LCD LED BudgeTone 200 phone has a numeric LCD of 64mmx24mm size with backlight This model has a small red LED status reminder Here is the display whe
21. ble ring 0 default ring I and ring 2 ring 3 is unavailable or unsupported Press MENU to select and exit Display rESEt please be very CAREFUL here e Key in the physical MAC address on back of the phone Press MENU phone will be reset to FACTORY DEFAULT setting and all your setting will be erased e Press MENU key without key in anything phone will function the same as power cycle or reboot When phone is powered on and time is displayed e Press or t Display ring 4 press or t again to hear and adjust the ring tone volume from 0 off to 7 maximum off and on hook to set e Press SPEAKERPHONE button or off hook and pick up handset press or T to adjust the speakerphone headset or handset volume 24 5 2 Configuration with Web Browser BudgeTone 200 series IP phone has an embedded Web server that will respond to HTTP GET POST requests It also has embedded HTML pages that allow a user to configure the IP phone through a Web browser such as Microsoft s IE 5 2 1 Access the Web Configuration Menu The IP Phone Web Configuration Menu can be accessed by the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone When the phone is on hook press Menu button and then select the Status item to see IP IP Address NOTE e To type IP address into browser to get into the configuration page please strip out the
22. e used in SIP SDP message STUN server Mn or w por Firmware Upgrade and Upgrade Via El Frp Elurre Provisioning Firmware Server Path Config Server Path fmoranastream convas Firmware File Prefix ER Firmware File Postfix Config File Prefix ER Config File Postfix 31 Allow DHCP Option 66 to override server o es Automatic Upgrade EAN El check for upgrade every FE nutes default 7 days Bi say Check for New Firmware Fi check New Firmware only when F W pre suffix changes ER EF cfg file would be authenticated before acceptance if set to Yes DTMF Payload Type 10 Syslog Server SSS Syslog Level NONE DEES Ne finest gov Ry or IP address Allow DHCP Option 42 to override NTP server Eno Efyss Custom ring tone 1 used if incoming caller ID is Authenticate Conf File T Custom ring tone 2 used if incoming caller ID is Distinctive Ring Tone Custom ring tone 3 used if incoming caller ID is Disable Call Waiting FR EH Use Quick IP call mode ER EN Lock keypad update ER EF configuration update via keypad is disabled if set to Yes 32 Admin Password Silence Suppression Voice Frames per TX Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use as Send Key Administrator password Only administrator can configure the Advanced Settings page Password field is purposely left blank for security reason
23. efault Router IP address DNS Server I primary DNS Server 2 secondary fields will need to be configured These fields are set to zero by default This parameter controls how the date time is displayed according to the specified time zone This parameter controls whether the time will be displayed in daylight savings time or not If set to Yes then the displayed time will be 1 hour ahead of normal time Allow user to choose among the following three formats Year Month Day Month Day Y ear Day Month Y ear This parameter controls whether the device is working in NAT router mode or Bridge mode Need save the setting and reboot the device before the setting start to work If set to Yes user can access the configuration page through the WAN port instead of connecting PC and GXP2000 through the PC port to do the configuration On the other hand it exposes the GXP2000 to others and may cause some security issues for users Default is No If set to Yes The GXP2000 will respond to the PING command from other computers for testing but it also is vulnerable to the DOS attack Default is No Cloned WAN MAC Addr LAN Subnet Mask LAN DHCP Base IP DHCP IP Lease Time DMZ IP Port Forwarding Allow the user to set a specific MAC address Set in Hex format Sets the LAN subnet mask Default value is 255 255 255 0 Base IP for the LAN port which function as a Gateway for the subnet Default value is 192 168
24. er when the sname field in the DHCP header has been used for DHCP options If you choose yes GXP2000 will use the TFTP server resolved from DHCP instead of the one you specified in the TFTP Server option above 34 Automatic Upgrade Authenticate Conf File DTMF Payload Type Syslog Server Syslog Level NTP server Choose Yes to enable automatic upgrade and provisioning In Check for new firmware every field enter the number of days to enable BudgeTone 200 to check the server for firmware upgrade or configuration in the defined period of days When set to No BudgeTone 200 will only do upgrade once at boot up Always check for New Firmware Check New Firmware only when F W pre suffix changes if set to Yes cfg file would be authenticated before acceptance This mechanism is useful for the protection of configuration on the device from unauthorized change This parameter sets the payload type for DTMF using RFC2833 The IP address or URL of System log server This feature is especially useful for ITSP Internet Telephone Service Provider Select 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 product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level regis
25. fined e System Ring Tone when selected all calls will ring with system ring tone e Customer Ring Tone 1 to 3 when selected BudgeTone 200 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 BudgeTone 200 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 The BudgeTone 200 supports up to 5 different Vocoder types including G 711 A U law GSM G 723 1 G 729A B User can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last Vocoder in this list can be entered by choosing the appropriate option in Choice 8 Default is Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc 42 5 2 4 Saving the Configuration Changes Once a change is made the user should press the Update button in the Configu
26. gital Signal Processing DSP technology to ensure superior hi fidelity audio quality interoperable with various 3 party SIP end user device Proxy Registrar Server and Gateway products e Advanced and patent pending adaptive jitter buffer control packet delay and loss concealment technology e Support popular codecs including G711 a law and u law G 723 1 6 3K G 729A B and GSM Dynamic negotiation of codec and voice payload length e Support standard voice features such as Caller ID Display or Block Call Waiting Call Waiting Caller ID Call Hold Call Transfer attended blind Do Not Disturb Call Forwarding in band and out of band DTMF RFC2833 SIP INFO Dial Plans Off Hook Auto Dial Auto Answer Early Dial and Speed Dial etc e Full duplex hands free speakerphone redial call log volume control voice mail with indicator downloadable ring tone etc e Support Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control e Support Acoustic Echo Cancellation AEC with Acoustic Gain Control AGC for speakerphone mode e Support sidetone e Support DIGEST authentication and encryption using MD5 and MDS sess e Provide easy configuration through manual operation phone keypad Web interface or automated provisioning by downloading encrypted configuration file via HTTP TFTP for mass deployment e Support for Layer 2 802 1Q VLAN 802 1p and L
27. ground noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection DATAGRAM A data packet carrying its own address information so it can be independently routed from its source to the destination computer DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or compressed Lossy compression algorithms ordinarily decimate while subsampling DECT Digital Enhanced Cordless Telecommunications A standard developed by the European Telecommunication Standard Institute from 1988 governing pan 48 DNS DID DSP DTMF FQDN European digital mobile telephony DECT covers wireless PBXs telepoint residential cordless telephones wireless access to the public switched telephone network Closed User Groups CUGs Local Area Networks and wireless local loop The DECT Common Interface radio standard is a multicarrier time division multiple access time division duplex MC TDMA TDD radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz each divided into 24 time slots of 10ms and twelve full duplex accesses per carrier for a total of 120 possible combinations A DECT base station an RFP Radio Fixed Part can transmit all 12 possible accesses time slots simultaneously by using different frequencies or using only one frequency All signaling information is transmitted from the RFP within a m
28. hone number No Yes SIP Registration No Yes Unregister On Reboot No Yes Register Expiration EM minutes default I hour max 45 days local SIP port KEM default 5060 SIP TI Timeout isee ov SIP T2 Interval 4sec ov NAT Traversal STUN Eno EY no but send keep alive Ei SUBSCRIBE for MWI EM Ely Proxy Require MR vr re User ID extension for 3rd party voice mail system Outbound Proxy Authenticate Password 37 Send DTMF Early Dial Dial Plan Prefix Enable Call Features Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE gt indio RTP RFC2833 EN SIP INFO EJ F ves use Yes only if proxy supports 484 response HE i prefix string is added to each dialed number Elin EJ if Yes Call Forwarding amp Call Waiting Disable are supported locally FM seconds default 180 seconds EN seconds default and minimum 90 seconds El calls gt request one No support Elluac Evas gt Recommended gt uac Elvas When UAC did not specify refresher tag El Yes Request for timer when making outbound Yes When caller supports timer but did not Yes Use timer even when remote party does not Yes Always refresh with INVITE instead of UPDATE Enable 100rel Account Ring Tone Send Anonymous Auto Answer Allow Auto Answer by Call
29. horization number Grandstream reserves the right to remedy warranty policy without prior notification Warning Please do not attempt to use a different power adaptor Using other power adaptor may damage the BudgeTone 200 and will void the manufacturer warranty Caution Changes or modifications to this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Information in this document is subject to change without notice No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of Grandstream Networks Inc 3 Product Overview The following photo illustrates the appearance of a BudgeTone 200 IP phone Front View Side View 3 1 Key Features Grandstream BudgeTone 200 IP Phone is a next generation IP telephone based on industry open standard SIP Session Initiation Protocol Built on innovative technology Grandstream IP Phone features market leading superb sound quality and rich functionalities at mass affordable price Software Features e Support SIP 2 0 TCP UDP IP PPPoE RTP RTCP HTTP ARP RARP ICMP DNS DHCP NTP SNTP TFTP e Support multiparty conferencing e Supports Quick IP Call Mode e Support NAT traversal using IETF STUN and Symmetric RTP e Advanced Di
30. ion Guide 5 1 Configuration with Keypad When the phone is IDLE or On Hook press the MENU button to enter key pad menu state When the phone goes off hook or a call comes in the phone automatically exits the key pad menu state and prepare for the call It also exits the key pad menu state if left idle for 20 seconds Here are the key pad menu options supported Menu Item Menu Functions Display 1 dheP On or 1 dheP oFF for the current selection Press MENU key to enter edit mode Press or 1 to toggle the selection Press MENU to save and exit Must recycle power to take effective Display 2 IP Addr Press MENU to display the current IP address Enter new IP address if DHCP is OFF Press or T to exit Press MENU to save and exit Must recycle power to take effective Display 3 SubNet Press MENU to display the Subnet mask Enter new Subnet mask if DHCP is OFF Press or T to exit Press MENU to save and exit Must recycle power to take effective 59 Display 4 routEr Press MENU to display the Router Gateway address Enter new Router Gateway address if DHCP is OFF Press or 1 to exit Press MENU to save and exit Must recycle power to take effective 22 Menu Item Menu Functions Display 5 dns i Press MENU to display the DNS address Enter new DNS address if DHCP is OFF Press or 1 to exit Press MENU to save and exit
31. istered to the SIP server s This field shows whether the PPPoE connection is up if connected to DSL modem This field shows what kind NAT the BudgeTone 200 is connected to via its LAN port It is based on STUN protocol 5 2 3 Advanced User Configuration To login to the Advanced User Configuration page please follow the instructions in section 5 2 1 to get to the following login page The password is case sensitive with a maximum length of 25 characters and the factory default password for Advanced User is admin Password ii Login 30 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 Following is a screen shot of the advanced configuration page ee E not displayed for security protection Silence Suppression ER EH Voice Frames per TX Em to 10 20 32 64 for G711 G726 G723 other codecs respectively Layer 3 QoS FP wite serv or Precedence value Layer 2 QoS 802 1Q VLAN Tag 802 1p priority value PO 0 7 No Key Entry Timeout nm seconds default is 4 seconds Admin Password Use as Dial Key Eno Elve if set to Yes will function as the Re Dial key local RTP port 1504 1024 65535 default 5004 Use random port FR FR keep alive interval PM seconds default 20 seconds Use NAT IP HEEN specified this will b
32. leading O as the browser will parse in octet e g if the IP address is 192 168 001 014 please type in 192 168 1 14 5 2 2 End User Configuration Once this HTTP request is entered and sent from a Web browser the BudgeTone 200 will respond with the following login screen Password ia Login 25 The password is case sensitive with maximum length of 25 characters and the factory default password for End User is 123 After a correct password is entered in the login screen the embedded Web server inside the BudgeTone 200 will respond with the Configuration page which is explained in details below Exd User gees purposely not displayed for security protection Password IP Address P namically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID en PPPoE password Sa Option 12 ae E 3 Option 15 Vendor Class ID Option 60 Preferred DNS server H ee i Ef taticany configured as IP Address Subnet Mask Default Router DNS Server 1 DNS Server 2 Time Zone Allow DHCP Eo 2 to override Time Zone setting o es Daylight EF EF if set to Yes display time will be 1 hour ahead of 26 Savings normal time Time Date EBV ear Month Day Display EB Month Day Year Format Eb ay Month Year System Device Mode Device Mode Eb switch default EAT Router NAT Router Configura
33. ll A can pick up the phone to restore conversation with B 4 3 9 Conference Call BudgeTone 200 phone supports 3 way conference 19 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 A dials C s number then SEND key to make the call 3 If C answers the call then A presses CONFERENCE button to bring B C in the conference 4 IfC does not answer the call A can press FLASH back to talk to B NOTE e 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 4 3 10 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 4 3 11 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 wi
34. ll Reject the call and forward to voice mail 4 4 Call Features BudgeTone 200 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 phone 20 Key Call Features 30 Block Caller ID for all subsequent calls wel Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 70 Disable Call Waiting Per Call 71 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 79 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 01 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 93 and get the dial tone then hang up 21 5 Configurat
35. m provides a licensed provisioning system called GAPS that can be used to support automated configuration of BudgeTone 200 GAPS Grandstream Automated Provisioning System uses enhanced NAT friendly TFTP or HTTP thus no NAT issues and other communication protocols to communicate with each individual BudgeTone 200 for firmware upgrade remote reboot etc Grandstream provide GAPS Grandstream Automated Provisioning System service to VoIP service providers It could be either simple redirection or with certain special provisioning settings Initially upon booting up Grandstream devices by default point to Grandstream provisioning server GAPS based on the unique MAC address of each device GAPS provision the devices with redirection settings so that they will be redirected to customer s TFTP or http server for further provisioning Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files The GAPSLite configuration tool is now free to end users The tool and configuration templates can be downloaded from http www grandstream com DOWNLOAD Configuration_Tool For details on how GAPS works please refer to the documentation of GAPS product 44 6 Firmware Upgrade 6 1 Upgrade through HTTP To upgrade software BudgeTone 200 can be configured with an HTTP server where the new code image file is located For e
36. n all segments illuminate E S do O 9 ull BIH G88008888 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 Icon LCD Icon Definitions Network Status Icon FLASH in the case of Ethernet link failure or the phone is not registered properly OFF if IP address or SIP server is not found ON if IP address and SIP server are located Phone Status Icon OFF when the handset is on hook ON when the handset is off hook Speakerphone Headset Status Icon FLASH when phone rings OFF when the speakerphone headset is off ON when the speakerphone headset is on 12 Handset and Speakerphone Headset Volume Icons 0 7 scales to adjust handset speakerphone volume Real time Clock Synchronized to Internet time server Time zone configurable via web browser 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 Ax byCrcevd EEG oe Hoh lek nO 0 Pq m Sst U u Y 13 4 2 Getting Familiar with Keypad Menu Called Buton Numbers GRANDSTREA Up Down Keys Message
37. nd is considerably less complex than H 323 All Grandstream products are SIP based STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT or multiple NATs to find out its public address the type of NAT it is behind and the internet side port associated by the NAT with a particular local port This information is used to set up UDP communication between two hosts that are both behind NAT routers The protocol is defined in RFC 3489 STUN will usually work good with non symmetric NAT routers TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite Using TCP applications on networked hosts can create connections to one another over which they can exchange data or packets The protocol guarantees reliable and in order delivery of sender to receiver data TFTP Trivial File Transfer Protocol is a very simple file transfer protocol with the functionality of a very basic form of FTP It uses UDP port 69 as its transport protocol 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 VA
38. ode 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 lt 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 XX or X are also valid so leading 0 is not required 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 66 99 be specified by using 4 encoding for followed by the port number Examples 17 e 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 e Ifthe 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 4 3 4 Answer an Incoming Call
39. ration Menu The IP phone will then display the following screen to confirm that the changes have been saved Your configuration changes have been saved They will take effect on next reboot User is recommended to power cycle the IP phone after seeing the above message 5 25 Rebooting the Phone from Remote The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the bottom of the configuration menu Once done the following screen will be displayed to indicate that rebooting is underway The device is rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin At this point user can relogin to the phone after waiting for about 30 seconds 43 5 3 Configuration through Central Provisioning Server Grandstream BudgeTone 200 can be automatically configured from a central provisioning system When BudgeTone 200 boots up it will send TFTP or HTTP request to download configuration files there are two configuration files one is cfg txt and the other is cfg000b82xxxxxx where O00b82xxxxxx is the MAC address of the BudgeTone 200 The configuration files can be downloaded via TFTP or HTTP from the central server A service provider or an enterprise with large deployment of BudgeTone 200 can easily manage the configuration and service provisioning of individual devices remotely from a central server Grandstrea
40. s 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 This radio button will enable BudgeTone 200 to download firmware or configuration file through either TFTP or HTTP This is the IP address of the configured TFTP server If selected and it is non zero or not blank the BudgeTone 200 will attempt to retrieve 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 downloaded image will be verified and then saved into the Flash memory Note Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the network environment this process can take up to 15 or 20 minutes The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream 1 0 5 16 Here 6688 is the specific TCP port that the HTTP server is listening at it can be omitted if using default port 80 Note If Auto Upgrade is set to No BudgeTone 200 will only do HTTP download once at boot up DHCP Option 66 is used to identify a TFTP serv
41. s 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 A 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 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 4 3 8 2 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 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 ca
42. th the up or down arrows keys Step 4 Enter the encoded MAC address Once the correct MAC address is displayed in the LCD screen press MENU button the device will reboot automatically and restore to factory default setting 47 8 Appendix I Glossary of Terms ADSL Asymmetric Digital Subscriber Line Modems attached to twisted pair copper wiring that transmit from 1 5 Mbps to 9 Mbps downstream to the subscriber and from 16 kbps to 800 kbps upstream depending on line distance AGC Automatic Gain Control is an electronic system found in many types of devices Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions ARP Address Resolution Protocol is a protocol used by the Internet Protocol IP RFC826 pecifically IPv4 to map IP network addresses to the hardware addresses used by a data link protocol The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer It is used when IPv4 is used over Ethernet ATA Analogue Telephone Adapter Covert analogue telephone to be used in data network for VoIP like Grandstream HT series products CODEC Abbreviation for Coder Decoder It s an analog to digital A D and digital to analog D A converter for translating the signals from the outside world to digital and back again CNG Comfort Noise Generator geneate artificial back
43. there enter the TFTP server address in the designated field towards the bottom of the configuration screen Once the TFTP server is set user needs to update the change by clicking the Update button Then Reboot or power cycle the phone the firmware files will be fetched upon booting up TFTP checking is only performed during the initial power up If the configured TFTP server is found and a new code image is available the BudgeTone 200 will attempt to 45 retrieve the new image files by downloading them into the BudgeTone 200 s SRAM During this stage the BudgeTone 200 s LEDs will blink until the checking downloading process is completed Upon verification of checksum the new code image will then be saved into the Flash If TFTP fails for any reason e g TFTP server is not responding there are no code image files available for upgrade or checksum test fails etc the BudgeTone 200 will stop the TFTP process and simply boot using the existing code image in the flash TFTP process may take as long as 1 to 2 minutes over the Internet or just 20 seconds if it is performed on a LAN Users are recommended to conduct TFTP upgrade in a controlled LAN environment if possible For those who do not have a local TFTP server Grandstream provides a NAT friendly TFTP server on the public Internet for firmware upgrade Please check the Services section of Grandstream s Web site to obtain this TFTP server s IP address NOTE e
44. tion WAN side EAN EF WAN side access to http server will be rejected if set to http access No Reply to OMe an EA El Unit will not respond to PING from WAN side if set to WAN port No Cloned WAN MAC E E o m E hex format Addr LAN Subnet e Mask default is 255 255 255 0 LAN DEC Base IP base IP for the LAN port default is 192 168 2 1 DHCP IP 120 ay Lease Time in units of hours default is 120 hours or 5 days pmzre NE AN port LANIP L Z gt A port Protocol Port Forwarding N 7 Update End User Password IP Address Time Zone Daylight Savings Time Date Display Format Device Mode WAN side http access Reply to ICMP on WAN port This contains the password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters There are two modes under which the BudgeTone 200 can operate e If DHCP mode is enabled then all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The BudgeTone 200 will acquire its IP address from the first DHCP server it discovers from the LAN it is connected e To use the PPPoE feature the PPPoE account settings need to be set The BudgeTone 200 will attempt to establish a PPPoE session if any of the PPPoE fields is set e If Static IP mode is enabled then the IP address Subnet Mask D
45. to or different from SIP User ID SIP service subscriber s account password for BudgeTone 200 to register to SIP servers of ITSP SIP service subscriber s name which will be used for Caller ID display Default is No If set to Yes the client will use DNS SRV to look up server User ID is Phone Number SIP Registration Unregister on Reboot Register Expiration Local SIP port SIP T1 Timeout SIP T2 Interval NAT Traversal Subscribe for MWI Proxy Require If the BudgeTone 200 has an assigned PSTN telephone number this field should be set to Yes Otherwise 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 whether the BudgeTone 200 needs to send 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 BudgeTone 200 refreshes its registration with the specified registrar The default interval is 60 minutes or I hour The maximum interval is 65535 minutes about 45 days This parameter defines the local SIP port the BudgeTone 200 will 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 T1 is an estimate of the round trip time RT
46. tration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG device MAC address error code error message Here is an example May 19 02 40 38 192 168 1 14 GS_LOG 00 0b 82 00 a1 be 000 Ethernet link is up URI or IP address of the NTP Network Time Protocol server which will be used by the phone to synchronize the date and time 35 Allow DHCP Option 42 to override NTP server Distinctive Ring Tone Disable Call Waiting Quick IP Call Mode Lock keypad update DHCP Option 42 specifies a list of IP addresses for Network Time Protocol NTP servers available to the client If you choose yes GXP2000 will use the NTP servers resolved from DHCP instead of the one you specified in the NTP Server option above Customer Ring Tone I to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY sound this ring tone when the incoming call is from the Caller ID device will use System Ring Tone for all other calls When selected but no Caller ID is configured the selected ring tone will be used for all incoming calls Default is No This model has the ability to dial an IP address under the same LAN segment by simply pressing the last octet in the IP
47. two lines Mute an active call or Delete a key entry call log etc Also used to REJECT incoming call 15 SEND RE DIAL 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 SPEAKERPHONE Enter hands free mode 4 3 Making and Answering Phone Calls 4 3 1 Handset Speakerphone and Headset Mode The regular Handset mode can be switched with either the Speaker mode Hand fee 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 4 3 2 Make Calls using Numbers There are FIVE ways to make phone calls Pick up handset or press SPEAKERPHONE button and then enter the phone numbers Press the SEND button directly to redial the number last called Once pressed the last dialed number will be displayed on the LCD as the corresponding DTMF tones are played out and an outgoing call is sent Browse the CALLED CALLER history and press the SEND REDIAL button 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
48. ultiframe 16 frames Voice signals are digitally encoded into a 32 kbit s signal using Adaptive Differential Pulse Code Modulation Short for Domain Name System or Service or Server an Internet service that translates domain names into IP addresses Direct Inward Dialing Direct Inward Dialing The ability for an outside caller to dial to a PBX extension without going through an attendant or auto attendant Digital Signal Processing Using computers to process signals such as sound video and other analog signals which have been converted to digital form Digital Signal Processor A specialized CPU used for digital signal processing Grandstream products all have DSP chips built inside Dual Tone Multi Frequency The standard tone pairs used on telephone terminals for dialing using in band signaling The standards define 16 tone pairs 0 9 and A F although most terminals support only 12 of them 0 9 and Fully Qualified Domain Name 49 FXO FXS DHCP A FQDN consists of a host and domain name including top level domain For example www grandstream com is a fully qualified domain name www is the host grandstream is the second level domain and com is the top level domain Foreign eXchange Office An FXO device can be an analog phone answering machine fax or anything that handles a call from the telephone company like AT amp T They should also operate the same way when connected to an FXS interface
49. vate networks with internet connections using NAT PPPoE Point to Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames It is used mainly with cable modem and DSL services PSTN Public Switched Telephone Network i e the phone service we use for every ordinary phone call or called POT Plain Old Telephone or circuit switched network RTCP Real time Transport Control Protocol defined in RFC 3550 a sister protocol of the Real time Transport Protocol RTP It partners RTP in the delivery and packaging of multimedia data but does not transport any data itself 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 32 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 It has been published by the IETF as RFC 2327 SIP Session Initiation Protocol An IP telephony signaling protocol developed by the IETF RFC3261 SIP is a text based protocol suitable for integrated voice data applications SIP is designed for voice transmission and uses fewer resources a
50. w grandstream com user_manuals BudgeTone200 pdf 2 Installation 2 1 What is Included in the Package The BudgeTone 200 phone package contains 1 One BudgeTone 200 Main Case 2 One Handset 3 4 5 One Phone Cord One Universal Power Adapter One Ethernet Cable 2 2 Connecting Your Phone Following is a backside picture of BudgeTone 200 each connection port is labeled with the name in the following table Grandstream Networks BudgeTone 200 Series BudgeTone 200 7 8145 LO 100M6ps parte PC and LAN HEADSET The table below describes the connectors on the BudgeTone 200 phone LAN 10 100 Switch LAN port for connecting to Ethernet PC 10 100 Switch port for connecting PC POWER 5V power port HEADSET 2 5mm Headset port 2 3 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 2 4 Warranty Grandstream has a reseller agreement with our reseller customer End user should contact the company from whom you purchased the product for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for a RMA Return Materials Aut
51. xample following URL in the HTTP Upgrade Server http firmware mycompany com 6688 Grandstream 1 0 1 12 Where firmware mycompany com is the FQDN of the HTTP server 6688 is the TCP port the HTTP server listening to Grandstream 1 0 0 4 is the RELATIVE directory to the root dir in HTTP server Thus you can put different firmware into different directory as well NOTE e If Auto Upgrade field is set to No HTTP upgrade will be performed only once during boot up If it is set to Yes the device will check the HTTP server in the number of days that is defined in Check for new firmware every field 6 2 Upgrade through TFTP To upgrade software BudgeTone 200 can be configured with a TFTP server where the new code image is located It is recommended to set the TFTP server address in either a public IP address or on the same LAN with the BudgeTone 200 There are two ways to set up the TFTP server to upgrade the firmware namely through voice menu prompt or via the BudgeTone 200 s Web configuration interface To configure the TFTP server via voice prompt please refer to section 5 1 with option 06 once set up the TFTP IP address power cycle the device the firmware will be fetched once the device boots up To configure the TFTP server via the Web configuration interface open up your browser to point at the IP address of the BudgeTone 200 Input the admin password to enter the configuration screen From
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