User Manual BudgeTone - 200 Series IP Phone For Firmware Version 1.1.0.16 Grandstream Networks, Inc. www.grandstream.
Table of Contents 1 WELCOME……………………………………………………………. 4 2 INSTALLATION……………………………………………………… 5 2.1 2.2 2.3 2.4 WHAT IS INCLUDED IN THE PACKAGE…………………………………5 CONNECTING YOUR PHONE…………………………………………...5 SAFETY COMPLIANCES………………………………………………..6 WARRANTY…………………………………………………………...6 3 PRODUCT OVERVIEW……………………………………………… 8 3.1 3.2 KEY FEATURES………………………………………………………. 9 HARDWARE SPECIFICATION………………………………………….10 4 USING BUDGETONE-200 IP PHONE……………………………... 12 4.1 4.2 4.3 4.3.1 4.3.2 4.3.3 4.3.4 4.3.5 4.3.6 4.3.7 4.3.8 4.
6.2 7 UPGRADE THROUGH TFTP………………………………………….. 45 RESTORE FACTORY DEFAULT SETTING……………………47 APPENDIX I GLOSSARY OF TERMS………………………………...
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.
2 Installation 2.1 What is Included in the Package The BudgeTone-200 phone package contains: 1. 2. 3. 4. 5. One BudgeTone -200 Main Case One Handset One Phone Cord One Universal Power Adapter One Ethernet Cable 2.
The table below describes the connectors on the BudgeTone-200 phone: LAN PC POWER HEADSET 10/100 Switch LAN port for connecting to Ethernet. 10/100 Switch port for connecting PC 5V power port 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.
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.
3 Product Overview The following photo illustrates the appearance of a BudgeTone-200 IP phone.
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: • • • • • • • • • • • • • • • • • • • Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP/SNTP, TFTP.
• allow user to specify different URL for configuration file and firmware files Hardware Features: • • • Support Headset which will auto switch to Headset when plugged in 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.
3.2 Hardware Specification The table below describes the hardware specification of BudgeTone-200: Model BudgeTone-200 LAN interface Headset Jack LED Phone Case 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: +5VDC, 1200mA, UL certified 18cm (W) 22cm (D) 6.5cm (H) 0.
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 when all segments illuminate: 010 上午 下午 AM PM 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.
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) AM PM Time Icon: AM for the morning PM for the afternoon IP Address Separator Icons: Numerical Numbers and Characters: 0-9 *=└ #=┘ A, b, C, c, d, E, F, G, g, H, h, I, L, n, O, o, P, q, r, S, t, U, u, Y 13
4.
Key Button 0 - 9, *, # ↓ ↑ MENU 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
SEND/(RE)DIAL SPEAKERPHONE 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 4.3 Making and Answering Phone Calls 4.3.
• • 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.
To make a direct IP to IP call, first off hook, then press “MENU” key, then enter a 12digit 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: • If the 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.
4.3.7.1 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 (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: • “Enable Call Feature” has to be configured to “Yes” in web configuration page in order to make the features to work.
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. If C does not answer the call, A can press FLASH back to talk to B. NOTE: • 4.3.9 During the conference, if B or C drops the call, the remaining two parties can still talk.
Key *30 *31 *67 *82 *70 *71 *72 *73 *90 *91 *92 *93 Call Features Block Caller ID (for all subsequent calls) Send Caller ID (for all subsequent calls) Block Caller ID (per call) Send Caller ID (per call) Disable Call Waiting. (Per Call) Enable Call Waiting (Per Call) 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.
5 Configuration 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.
Menu Item Menu Functions 5 Display “[5] dnS ” Press MENU to display the DNS address Enter new DNS address if DHCP is OFF Press ‘↓’ or ’↑’ to exit Press MENU to (save and) exit Must recycle power to take effective!!! 6 Display “[6] tFtP ” Press MENU to display the TFTP address Enter new TFTP server IP address Press MENU to save Press ‘↓’ or ’↑’ to exit 7 Display “[7] G-711u 2” Press MENU to select new codec Press ‘↓’ or ’↑’ to browse a list of available codecs line 2 “ - G-711A 2” 3 “ - G-723 1” 4 “ -
Menu Item Menu Functions 10 Display “[10] Phy Addr” Press MENU to display the physical / MAC address Press ‘↓’ or ’↑’ to exit 11 Display “[11] ring 0” Press MENU to hear the selected ring tone, press ‘↓’ or ’↑’ to select the stored ring tones. Now only 3 are available, ring 0 (default), ring 1 and ring 2. ring 3 is unavailable or unsupported.
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.
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.
Savings Time: Date Display Format: normal time) Year-Month-Day Month-Day-Year Day-Month-Year System Device Mode Device Switch (default) Mode: NAT/Router NAT/Router Configuration WAN side No Yes (WAN side access to http server will be rejected if set http access: to No) Reply to No Yes (Unit will not respond to PING from WAN side if set to ICMP on No) WAN port: Cloned WAN MAC (in hex format) Addr: LAN Subnet 255.255.255.0 (default is 255.255.255.0) Mask: LAN DHCP 192.168.2.
Update All Rights Reserved Grandstream Networks, Inc. 2004, 2005 End User Password This contains the password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters. IP Address There are two modes under which the BudgeTone 200 can operate: • 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.
Cloned WAN MAC Addr 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 LAN Subnet Mask LAN DHCP Base IP Base IP for the LAN port, which function as a Gateway for the subnet. Default value is 192.168.2.1. DHCP IP Lease Time Value is set in units of hours. Default value is 120hr (5 Days.) The time IP address is assigned to the LAN clients.
MAC Address IP Address 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. Product Model Software Version • • Program: This is the main software release, its number is always used for firmware upgrade. Bootloader: This is normally not changed. System Up Time This field shows system up time since the last reboot.
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.
Allow DHCP Option 66 to override server: No Yes Automatic Upgrade: No 7 days) Yes, check for upgrade every 10080 minutes (default Always Check for New Firmware Check New Firmware only when F/W pre/suffix changes Authenticate Conf File: DTMF Payload Type: No Yes (cfg file would be authenticated before acceptance if set to Yes) 101 Syslog Server: Syslog Level: NTP Server: NONE time.nist.
Admin Password Administrator password. Only administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reason after clicking update and saved. The maximum password length is 25 characters. Silence Suppression 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.
Local RTP port 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. Use Random Port This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports.
Automatic Upgrade 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” Authenticate Conf File if set to Yes, cfg file would be authenticated before acceptance.
Allow DHCP Option 42 to override NTP server DHCP Option 42 specifies a list of IP addresses for Network Time Protocol (NTP) servers available to the client. If you choose yes, BT200 will use the NTP servers resolved from DHCP, instead of the one you specified in the "NTP Server" option above.
Following is the screenshot of the Account Configuration Page:- Grandstream Device Configuration STATUS BASIC SETTINGS ADVANCED SETTINGS ACCOUNT Account Active: Account Name: SIP Server: Outbound Proxy: SIP User ID: Authenticate ID: Authenticate Password: Name: No Yes MyCompany (e.g., MyCompany) sip.mycompany.com (e.g., sip.mycompany.com, or IP address) (e.g., proxy.myprovider.
mail system) Send DTMF: Early Dial: in-audio No response) Dial Plan Prefix: Enable Call Features: Session Expiration: Min-SE: Caller Request Timer: Callee Request Timer: Force Timer: via RTP (RFC2833) via SIP INFO Yes (use "Yes" only if proxy supports 484 (this prefix string is added to each dialed number) No Yes (if Yes, Call Forwarding & Call-WaitingDisable are supported locally) 180 (in seconds. default 180 seconds) 90 (in seconds.
Turn off speaker on remote disconnect: Preferred Vocoder: (in listed order) No choice 1: PCMU choice 5: GSM choice 2: PCMA choice 6: PCMU choice 3: G.729A/B choice 7: PCMA choice 4: Special Feature: Yes G.723.1 choice 8: G.729A/B Standard Update All Rights Reserved Grandstream Networks, Inc. 2004, 2005 Individual Account Settings Account Active This field indicates whether the account is active or not. Account Name A name to identify an account which will be displayed in LCD.
Use DNS SRV: Default is No. If set to Yes the client will use DNS SRV to look up server. User ID is Phone Number 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 SIP Registration This parameter controls whether the BudgeTone 200 needs to send REGISTER messages to the proxy server. The default setting is “Yes”.
Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Voice Mail User ID When configured, user will be able to dial voice mail server by pressing “MSG” button. Send DTMF 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. Early Dial Default is No. Use only if proxy supports 484 response.
UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer. Enable 100rel 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.
5.2.4 Saving the Configuration Changes Once a change is made, the user should press the “Update” button in the Configuration Menu. The IP phone will then display the following screen to confirm that the changes have been saved: Grandstream Device Configuration STATUS BASIC SETTINGS ADVANCED SETTINGS ACCOUNT Your configuration changes have been saved. They will take effect on next reboot. All Rights Reserved Grandstream Networks, Inc.
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 “000b82xxxxxx” is the MAC address of the BudgeTone 200. The configuration files can be downloaded via TFTP or HTTP from the central server.
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 example, 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.
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.
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. Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default.
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.
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.
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. FXO 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&T. They should also operate the same way when connected to an FXS interface.
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.323 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.
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.
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).
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.
