User Manual

BudgeTone-100 Series

IP Phone

For Firmware Version 1.0.8.32
Grandstream Networks, Inc.

www.grandstream.com

BudgeTone-100 User Manual

Table of Contents

1 WELCOME 2 INSTALLATION 2.1 INTERCONNECTION DIAGRAM 3 WHAT IS INCLUDED IN THE PACKAGE 3.1 SAFETY COMPLIANCES 3.2 WARRANTY 4 PRODUCT OVERVIEW 4.1 KEY FEATURES 4.2 HARDWARE SPECIFICATIONS 5 BASIC OPERATIONS 5.1 GET FAMILIAR WITH LCD/LED 5.2 GET FAMILIAR WITH KEYPAD 5.3 MAKE PHONE CALLS 5.3.1 Make Calls using Numbers 5.3.2 Make Calls using IP Address 5.3.3 Answer an Incoming Call 5.3.4 Handset Mode, Speakerphone/Headset Mode 5.3.5 Call Hold 5.3.6 Call Waiting and Call Flashing 5.3.7 Call Transfer 5.3.8 Conference Call 5.4 CALL FEATURES 6 CONFIGURATION GUIDE 6.1 CONFIGURATION WITH KEYPAD 6.2 CONFIGURATION WITH WEB BROWSER 6.2.1 Access the Web Configuration Menu 6.2.2 Configuration Menu 6.2.3 Saving the Configuration Changes 6.2.4 Rebooting the Phone from Remote 6.3 CONFIGURATION THROUGH A CENTRAL SERVER 7 SOFTWARE UPGRADE 7.1 UPGRADE THROUGH HTTP 7.2 UPGRADE THROUGH TFTP 7.3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX -2-4-5-6-7-7-7-8-8-9- 11 - 11 - 12 - 14 - 14 - 14 - 15 - 15 - 15 - 15 - 16 - 17 - 18 - 19 - 19 - 23 - 23 - 24 - 38 - 39 - 40 - 41 - 41 - 41 - 43 -
7.4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD 8 RESTORE FACTORY DEFAULT SETTING 9 HEADSET CONNECTION 10 GLOSSARY OF TERMS

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Welcome
Congratulations on becoming an owner of BudgeTone-100 IP telephone! You made an excellent choice and we hope you will enjoy all its capabilities. Grandstream's award-wining BudgeTone-100 series of SIP phones are innovative IP telephones that offer a rich set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market. Grandstream BudgeTone-100 IP telephone has been awarded the Best of Show product in 2003 Internet Telephony Conference and Expo. This document is subject to changes without notice. The latest electronic version of this user manual can be downloaded from Grandstream Networks official website: http://www.grandstream.com/user_manuals/budgetone100.pdf

Installation

BudgeTone-100 series IP phones are designed to look and feel like standard telephones. The following photo illustrates the appearance of a BudgeTone IP phone and the use of its key buttons.
LCD Message Light Menu Outgoing Call Log Message Access Hold Transfer Conference Flash
Volume & Menu Browser Key Incoming Call Log

Mute/Delete

Speakerphone

Send/(Re)Dial

Interconnection Diagram
There are several ways to connect the BT100 series IP telephone: 1. Connected directly behind Cable/DSL modem 2. Connected to LAN side of a (wireless) SOHO router (This is the most popular connection). 3. Connected to an Ethernet network Following diagram illustrates the interconnection of phones in above-mentioned networks:

What is Included in the Package
The BudgeTone-100 phone package contains: 1) One BudgeTone-100 phone 2) One universal power adaptor 3) One Ethernet cable

Safety Compliances

The BudgeTone-100 phone is compliant with various safety standards including FCC/CE. Its power adaptor is compliant with UL standard. The phone should only operate with the universal power adaptor provided with the package. Damages to the phone caused by using other unsupported power adaptors would not be covered by the manufacturers warranty.

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 Authorization) 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-100 IP telephone 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.

Product Overview

Grandstream IP Phone is a next generation IP network 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.

Key Features
Support SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN, TFTP, etc. Powerful Digital Signal Processing (DSP) technology to ensure superior audio quality Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology Support various codecs including G.711 (PCM a-law and u-law), G.723.1 (5.3K/6.3K), G.729A, G.726 (32K) and iLBC Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold, Transfer, Forward, 3-Way Conference, in-band and out-of-band DTMF, off hook autodial, auto answer. Support syslog, full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator, downloadable ring tones. Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) Support standard encryption and authentication (DIGEST using MD5, MD5-sess) Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Support automated NAT traversal without manual manipulation of firewall/NAT Provide easy configuration through manual operation (phone keypad), Web interface or automated centralized configuration file via TFTP or HTTP. Support firmware upgrade via TFTP or HTTP.

Hardware Specifications

There are two models in the BudgeTone-100 family, namely: BudgeTone-101 BudgeTone-102 (As show in the below picture)
The ONLY difference between BT101 and BT102 is that the two RJ-45 ports of BT102 is actually a 10Base-T mini-Hub that allows user to share or sniffer the network using another data device like PC.
The table below describes the difference among these models.
Model LAN interface Phone Case Headset Plug Universal Switching Power Adaptor Dimension
BudgeTone-101 1xRJ45 10Base-T 25-button keypad 12-digit caller ID LCD 3.5 mm Input: 100-240VAC Output: +5VDC, 400mA, UL certified 18cm (W) 22cm (D) 6.5cm (H) 2 lbs (0.9 kg) 32 - 104oF 0 - 40oC 10% - 95% (non-condensing) FCC/CE/C-Tick
BudgeTone-102 2xRJ45 10Base-T 25-button keypad 12-digit caller ID LCD 3.5 mm Same as left

Same as left

Weight Operating Temperature Humidity Compliance
Same as left Same as left Same as left Same as left

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Basic Operations
Get Familiar with LCD/LED
BudgeTone-100 phone has a numeric LCD of 64mmx24mm size with backlight. The new model (which is shipping now) has a small red LED status reminder. Here is the display when all segments illuminate:
When the phone is in the normal idle state, the backlight is off. Whenever an event (call) occurs, the backlight and the red LED will turn on automatically to bring the users attention. In addition, if Voice Mail configured and there is a VM waiting, the backlight will be blinking and the red LED message light will light up to remind user there is a Voice Mail in the Voice Mail server.

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

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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)

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: 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 Both BudgeTone phone and other VoIP Device are on the same LAN using private or public IP addresses, or 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).
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.

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BudgeTone-100 User Manual Examples:
If the target IP address is 192.168.0.10, the dialing convention is MENU_key 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.

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
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.

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.
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.

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Call Transfer
Two transfer operations are supported. 5.3.7.1 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 partys 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. A can hold on to the phone and wait for one of the three following behaviors: A quick confirmation tone (temporarily using the call waiting indication tone) follows by a dial tone. This indicates the transfer has been successful. At this point, the user can either hang up or make another call. 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. 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.

5.3.7.2 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 partys 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: 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.

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Conference Call
BudgeTone 100 phone supports 3-way conference. Assuming that call party A and B are in conversation. A wants to bring C in a conference: 1. 2. 3. 4. A presses the CONFERENCE button to get a dial tone and put B on hold A dials Cs number then SEND key to make the call If C answers the call, then A presses CONFERENCE button to bring B, C in the conference. If C does not answer the call, A 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.

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Call Features
Following table shows the available call features of BudgeTone 100 by using keypad star(*) code, if the VoIP service provider supports these call features in the server side:
*30 *31 *67 *82 *50 *51 *70 *71 *72
Block CallerID (for all-config change) Send CallerID (for all-config change) Block CallerID (per call base) Send CallerID (per call base) Disable Call Waiting (for all-config change) Enable Call Waiting (for all-config change) Disable Call Waiting. (Per Call) Enable Call Waiting (Per Call) Unconditional Call Forward. To use this feature, dial *72 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone to confirm the cancel, then hang up. Busy Call Forward To use this feature, dial *90 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Busy Call Forward To cancel Unconditional Call Forward, dial *91 and get the dial tone to confirm the cancel, then hang up Delayed Call Forward To use this feature, dial *92 and get the dial tone. Then dial the forward number followed by pressing SEND button and hear dial tone again to confirm the forward, then hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone to confirm the cancel, then hang up When in conversation, this action will switch to the new incoming call if user hears the call waiting sound. When in conversation and no other incoming call, this action will switch to a new channel for a new call.

*73 *90

*91 *92

*93 Flash or Hook Flash

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Configuration Guide

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] dhcP On or [1] dhcP oFF for the current selection Press MENU key to enter edit mode Press or 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 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 to exit Press MENU to (save and) exit Must recycle power to take effective!!! Display [4] routEr Press MENU to display the Router/Gateway address Enter new Router/Gateway address if DHCP is OFF Press or to exit Press MENU to (save and) exit Must recycle power to take effective!!!

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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!!! Display [6] tFtP Press MENU to display the TFTP address Enter new TFTP server IP address Press MENU to save Press or to exit Display [7] G-711u 2 Press MENU to select new codec Press or to browse a list of available codecs line 1 - G-711A - G-3 - G-4 - G-5 - G-6 - G-7 - iLBC 1 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.

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Display [9] codE rEL Press Menu to display the code releases Press or to browse line 1 b 2006-03-06 date: boot code 2 1. 0. 8. 11 version: boot code 3 P 2006-09-14 date: phone code 4 1. 0. 8. 32 version: phone code 5 c 2005-03-05 date: codec 6 1. 0. 1. 0 version: codec 7 h 2006-09-14 date: web server 8 1. 0. 8. 32 version: web server 9 1r 2004-05-12 date: 1st ring tone 10 1. 0. 0. 0 version: ring tone 11 2r 2005-07-21 date: 2nd ring tone 12 1. 0. 0. 0 version: ring tone 13 3r 0000-00-00 date: 3rd ring tone 14 0. 0. 0. 0 version: ring tone (all zeroes means unavailable or unsupported) Press MENU to exit Display [10] Phy Addr Press MENU to display the physical / MAC address Press or to exit 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. Press MENU to select and exit Display -- rESEt --, please be very CAREFUL here. Only shown when key to pressed: 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. Press MENU key without key in anything, phone will function the same as power cycle or reboot. This is called soft reboot.

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When phone is powered on and time is displayed Press or , Display ring [4] , press or again to hear and adjust the ring tone volume, from 0 (off) to 7 (maximum), off and on hook to set Press SPEAKERPHONE button, or off hook and pick up handset, press or to adjust the speakerphone/headset or handset volume

Others

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Configuration with Web Browser
BudgeTone-100 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 Microsofts IE or Mozillas Firefox.
Access the Web Configuration Menu
The BudgeTone-100 IP Phone Web Configuration Pages can be accessed by input the phones IP into browsers URL address field like: http://Phone-IP-Address where the Phone-IP-Address is the IP address of the phone. There are two ways to retrieve this IP address from the phone: 1) When the phone is off-hook or in speakerphone mode, simply press MENU button. (This is most common way to get the IP address of the phone) 2) When the phone is on-hook, press MENU button and then the browsing arrow keys to [2] IP Addr, press MENU again. NOTE: To type IP address into browser to bring up the configuration pages, please strip out the leading 0 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.
Once the correct IP address of the phone is input into browser and Enter key pressed, the web log in page will come up like following:
Grandstream Device Configuration

Password

All Rights Reserved Grandstream Networks, Inc. 2005

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The password is case sensitive with maximum length of 25 characters. The factory default password for End User is 123, for Administrator is admin respectively. Only administrator has the privilege to get access to ADVANCED SETTINGS configuration page. NOTE: If you cannot log into the configuration pages by using default password, please check with the VoIP service provider. Most likely, the service provider has already provisioned and automatically configured the device for you and has changed the default password.

Configuration Menu

After input the correct password into the login screen, the embedded Web server of the IP phone will respond with the Configuration Pages screen, which is explained in details below.

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BudgeTone-100 User Manual End user Password
This contains the password for end user to access the Web Configuration Menu. This field is case sensitive and maximum length is 25 characters. The default end user password is 123. End user only has privilege to see the Status page and change parameters in Basic page. End user can NOT access to Advanced Settings pages and will get error message if try to.

IP Address

There are 2 modes under which the IP phone 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) and the IP phone will acquire its IP address from the first DHCP server it discovers on the LAN it attaches to. PPPoE account settings are configured here if the user is connecting the IP phone directly to the DSL/ADSL modem. Users can specify DNS server manually by entering DNS servers IP address. The IP Phone will attempt to establish a PPPoE session if PPPoE account is set. For most users, just leave them blank - If Static IP mode is selected, the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (mandatory), DNS Server 2 (optional) fields need to be configured. These fields are set to zero by default.
Time zone Day light savings time Date display format
Displayed date/time will be adjusted according to the specified time zone. Default NO. If set to Yes, then the displayed time will be 1 hour ahead of normal time. This parameter controls the date display format.

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Advanced Settings:
Grandstream Device Configuration STATUS BASIC SETTINGS ADVANCED SETTINGS Admin Password: SIP Server: Outbound Proxy: SIP User ID: Authenticate ID: Authenticate Password: Name:
Tom Smith abc.sipprovider.com
(purposely not displayed for security protection) (e.g., sip.mycompany.com, or IP address) (e.g., proxy.myprovider.com, or IP address, if any) (the user part of an SIP address) (can be identical to or different from SIP User ID) (purposely not displayed for security protection) (optional, e.g., John Doe)
Advanced Options: Preferred Vocoder: (in listed order) choice 1: choice 2: choice 3: choice 4: choice 5: choice 6: choice 7: choice 8: G723 rate: iLBC frame size: iLBC payload type: Silence Suppression: Voice Frames per TX: Layer 3 QoS:

97 G.729A/B iLBC G.723.1 PCMU PCMA G.726-32 G.722 (w ide band) PCMA
6.3kbps encoding rate 20ms 30ms

5.3kbps encoding rate

(between 96 and 127, default is 97) No Yes (up to 10/20/32/64 for G711/G726/G723/other codecs respectively) (Diff-Serv or Precedence value)
Layer 2 QoS: 802.1Q/VLAN Tag Allow incoming SIP messages from SIP proxy only: No Yes

802.1p priority value

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Use DNS SRV: User ID is phone number: SIP Registration: Unregister On Reboot: Register Expiration: Early Dial: Allow outgoing call without Registration: Dial Plan Prefix: No Key Entry Timeout: Use # as Dial Key: local SIP port: local RTP port: Use random port: NAT Traversal:

No No Yes Yes

Yes Yes No No (in seconds. default 1 hour, max 45 days) Yes (use "Yes" only if proxy supports 484 response) Yes (this prefix string is added to each dialed number) (in seconds, default is 4 seconds)

5060 5004

Yes (if set to Yes, "#" will function as the Dial key) (default 5060) (1024-65535, default 5004) Yes
Yes, STUN server is: keep-alive interval: Use NAT IP

stun.fw dnet.net:3478

(URI or IP:port)
(in seconds, default 20 seconds) (if specified, this IP address is used in SIP/SDP
message) (if specified, the content will appear in ProxyRequire header)
Proxy-Require: Voice Mail UserID: SUBSCRIBE for MWI:
(User ID/extension for 3rd party voice mail system)
No, do not send SUBSCRIBE for Message Waiting Indication Yes, send periodical SUBSCRIBE for Message Waiting Indication
Auto Answer: Offhook Auto-Dial: Enable Call Features:

No offhook)

Yes (User ID/extension to dial automatically when
No Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported locally) No Yes

Disable Call-Waiting:

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Send DTMF: DTMF Payload Type: Send Flash Event: Onhook Threshold: NTP Server:

in-audio

via RTP (RFC2833)

via SIP INFO

800 ms time.nist.gov
Yes (Flash will be sent as a DTMF event if set to Yes)

(URI or IP address)

system ring tone custom ring tone 1, used if incoming caller ID is Default Ring Tone:
custom ring tone 2, used if incoming caller ID is
custom ring tone 3, used if incoming caller ID is
Send Anonymous: Anonymous Method: Time to ring: Special Feature: Syslog Server: Syslog Level:
Yes (caller ID will be blocked if set to Yes) Use Privacy Header

Use From Header

60 seconds Standard
Firmware Upgrade and Provisioning: Upgrade Via

fm.grandstream.com/gs

Firmware Server Path: Config Server Path: Firmware File Prefix: Config File Prefix: Automatic Upgrade: No

SIP Server

Outbound Proxy
SIP User ID Authentication ID Authenticate Password Preferred Vocoder

G723 Rate:

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BudgeTone-100 User Manual iLBC frame size iLBC payload type Silence Suppression
iLBC packet frame size. Default is 20ms. For Asterisk IP-PBX, 30ms might need to be configured for compatibility Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. 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.
Voice Frames per This field contains the number of voice frames to be transmitted in a single Ethernet packet (be advised the max. size of Ethernet packet is 1500 byte (or TX 120k bit) so user should be aware that there IS a limit there). When setting this value, the user should be aware of the requested packet time (ptime, used in SDP message) as 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 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 codec 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 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 G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively. Please be very careful when massage those parameters. By adjust this, user also get jitter buffer changed accordingly. BT-100 phone has patent dynamic jitter buffer handling algorithm. The jitter buffer range from 20 ~ 200 ms. Incorrect setting will affect voice quality so do not touch the parameter if not understand and most of the case the default value will work in GS products. Please refer to the Codec FAQ in our website for more technical details: http://www.grandstream.com/FAQ-Codec.pdf Layer 3 QoS Layer 2 QoS This field defines the layer 3 QoS parameter, which can be used for IP Precedence or Diff-Serv or MPLS. Default value is 48. Layer 2 QoS settings. Default setting is blank or 0 Other VLAN supported equipments like VLAN switch/router required if user wants to configure these settings. If set to Yes, the phone will ignore any SIP message that does not come from the IP address (Source IP in the IP header, the SIP server) that it is registered to. Default is No.

Allow incoming SIP messages from SIP proxy only

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BudgeTone-100 User Manual Use DNS SRV Use ID is phone number SIP registration
Default is No. If set to Yes, the phone will use DNS SRV configured to lookup for the server If Yes is set, a user=phone parameter will be attached to the From header in SIP request, which will be processed by supported SIP proxy. Default is No. This parameter controls whether the BudgeTone phone needs to send REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the phone will send remove all register request to the server (* in the contact header) to remove all previous bindings. If server does not support this it will cause some problems This parameter allows the user to specify the time frequency (in seconds) the IP phone will refresh its registration with the specified registrar (SIP Server). The default interval is 3600 seconds (or 1 hour). The maximum interval is 45 days. Default setting is No. The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response (like Asterisk). Otherwise, the call will most likely be rejected by the proxy (with a 404 Not Found error). Please note that this feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling. Default is No. If set to Yes, if ITSP permit, phone can make outgoing call even not registered. But can not get incoming calls. This feature is highly depended on ITSPs. Sets the prefix added to each dialed number. If configured, the prefix will be added to EVERY number input Default is 4 seconds. User can short or extend that depends on digits dialed habit This parameter allows the user to configure the # key to be used as the SEND key. Once 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 SEND key. If set to No, this # key will then be included as part of the dial string to be sent out. This parameter defines the local SIP port the IP phone will listen and transmit on. The default value is 5060.
Unregister On Reboot Register Expiration Early Dial
Allow outgoing call without Registration Dial Plan Prefix No key Entry Timeout Use # as Dial Key

Local SIP port

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BudgeTone-100 User Manual Local RTP port
This parameter defines the local RTP-RTCP port pair the IP phone 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 Default No. If set to Yes, the device will pick randomly generated SIP and RTP ports. This is usually necessary and useful when multiple IP Phones are behind the same full cone NAT router. NAT Traversal Defines whether the NAT traversal mechanism is activated. It should be set to YES if the device is behind NAT router. If Outbound Proxy is NOT configured, STUN server needs to be set to activate STUN detection mechanism. Usually ITSP will provide these settings for device to work properly behind NAT/Firewall If this field is set to Yes without STUN server, then the device will periodically (every Keep-alive interval) send a dummy UDP packet to the SIP server to pinhole the NAT in the router side. Default is 20 seconds. The interval of sending dummy UDP packet to keep NAT pin hole open in the router side. Min. value is 10 seconds. NAT IP address (WAN side) used in SIP/SDP message. Default is blank. SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Required by some soft switch vendor like Nortel MCS. User ID (extension or access number) of a 3rd party VoiceMail system where the user may have an account. By defining it, user presses the MESSAGE button on the phone, an INVITE message will send to that ID/number to allow the user to retrieve VM. Default is No. When set to Yes, a SUBSCRIBE for Message Waiting Indication will be sent periodically to server. BT-100 support both synchronize and nonsynchronized SUBSCRIBE SIP message. Default is No. When set to Yes, the phone will automatically pick up the call after a short beep and turn on the speaker. This parameter allows the user to configure a User ID or extension number to be automatically dialed upon off hook (like hot line). Please note that only the user part of a SIP address needs to be entered here. The phone will automatically append the @ and the host portion of the corresponding SIP address. Default is Yes. Advance call features or feature codes functions (Star code, see Section 5.4 of this manual) are supported locally Default is No. User can use * code to use this feature per call basis.

32 digits in Hexadecimal. Once configured, the firmware will ONLY be changed if the key is matched. This will lock the unit and firmware by ITSP. Useful for ITSP to encrypt firmware. End user should keep it blank. Default No. Useful by ITSPs. End user should use default setting. Once configured, only authenticated configuration file will be used. Default is No. The configuration update via keypad is disabled if set to Yes. User can not change settings via key pad and only WebUI and configuration file can be used to change setting if set to Yes. Be very careful for this setting Default NO. If set to Yes, User ID, Authentication ID and Password can be configured in basic settings. This parameter is decided by related ITSP. Override the MTU size to meet some special network settings. Default is 0, means no override.
Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Override MTU Size
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.

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User is recommended to reboot or power cycle the IP phone after all the changes are made so that those changes can take effect.
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, the user can relogin to the phone after waiting for about 30 seconds.

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Configuration through a Central Server
Grandstream IP phone can be automatically configured from a central provisioning system. When BudgeTone phone boots up, it will send TFTP or HTTP request to download configuration file. The name of the configuration file is cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the BudgeTone. 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 phone can easily manage the configuration and service provisioning of individual devices remotely from a central server. Grandstream provides a licensed provisioning system called GAPS that can be used to support automated configuration of BudgeTone. 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 phone for firmware upgrade, configuration parameters change or 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 customers 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.

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WARNING !!!
Restore Factory Default Setting
Restore the Factory Default Setting will DELETE all configuration information of the phone. 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 VoIP service provider. Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default. The steps are as follows: Step 1: Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of the unit. 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 (when pressed 2 twice, the A letter will show on the LCD)
For example, if the MAC address is 000b8200e395, it should be encoded as 0002228200333395. Step 3: To perform factory reset: a. b. c. d. e. Press the MENU button for Key Pad Menu options. Press the Up or Down button to see reset. Enter the encoded MAC address. Press the MENU button again Wait for phone reboot and the LCD backlight finish flashing.

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Headset Connection
The BT-100 phone has a headset socket allowing user to plug the headset into the phone. The picture below shows the handset and headset connectors wiring schema.
As show in the schema, the left side is pin assignment for a RJ11 interface headset; while the right side is showing a normal 3.5mm headset plug. A 3.5mm to 2.5mm plug converter is required if user want to user normal 2.5mm cell phone headset. The plug converter can be purchased from any electronics component store.

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Cross Over Cable For Plantronic Headset
Some users want to use headset products with RJ22 plug, like the Plantronic headset. In this case, a special cable is required. The handset twisted cable is a roll-over cable standard for ALL handset, using RJ22 plug. Since the default handset plug lay out is Asia standard which is just reversed to North American or Europe standard, therefore US and Europe customer can not use popular Plantronic Handset without tweaking the connection wiring. The quick and easy solution will be a special cable: a cross-over cable. Please ask for help from electrician if user can not understand this part. Here is the example and instruction to hand made such an adapter or cable and confirmed to work with Plantronic M12 headset with amplifier, which is most popular headset used in call centers. Here are the schemas of the two cables; the plug is viewed with Pin facing user, with PINs as specified above: d: SP + a: SP c: Mic + b: Mic -