User Manual
BudgeTone-100
Series
IP Phone
Version 1.0.5.6
Grandstream Networks, Inc.
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BudgeTone-100 User Manual
Grandstream Networks, Inc.
1 Welcome
Congratulations on becoming an owner of BudgeTone-100 IP telephone! You made an
excellent choice and we hope you will enjoy all its capabilities.
Grandstream's award-wining BudgeTone-100 series of SIP phones are innovative IP
telephones that offer a rich set of functionality and superb sound quality at ultra-
affordable price. They are fully compatible with SIP industry standard and can
interoperate with many other SIP compliant devices and software on the market.
This document is subject to changes without notice. The latest electronic version of this
user manual can be downloaded from the following location:
Grandstream Networks, Inc.
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2 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
Menu
Volume & Menu
Browser Key
Out-Call Log
Incoming Callers Log
Message
Hold
Transfer
Conference
Flash
MUTE/Delete
SEND/(Re)Dial
Speakerphone
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2.1 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
2.2 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
manufacturer’s warranty.
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3 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.
3.1 Key Features
• Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, HTTP, ARP/RARP, ICMP,
DNS, DHCP, NTP/SNTP, TFTP protocols
• Support NAT traversal using IETF STUN and Symmetric RTP
• Advanced Digital Signal Processing (DSP) technology to ensure superior hi-
fidelity audio quality, interoperable with various 3rd party SIP end user device,
Proxy/Registrar/Server and Gateway products
• Advanced and patent pending adaptive jitter buffer control, packet delay and loss
concealment technology
• Support popular codecs including G711 (a-law and u-law), G722, G.723.1
(5.3K/6.3K), G.726 (40K/32K/24K/16K), G.728, G.729A/B and iLBC. Dynamic
negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting,
Call Hold, Call Transfer, Do-Not-Disturb, Flash, 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.
• Full duplex hands-free speakerphone, redial, call log, volume control, voice mail
with indicator, downloadable ring tone, etc.
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort
Noise Generation), Line Echo Cancellation (G.168) and AGC (Automatic Gain
Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess)
• Provide easy configuration through manual operation (phone keypad), Web
interface or automated provisioning via central configuration through secure
TFTP file for mass deployment
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ,
MPLS)
• NAT friendly remote software upgrade capability (via TFTP) even from behind
firewall/NATs
• Support DNS SRV Look up and SIP Server Fail Over (pending)
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3.2 Hardware Specification
There are three models in the BudgeTone-100 family, namely:
BudgeTone-101
BudgeTone-102
The table below describes the difference among these models.
Model
BudgeTone-101
BudgeTone-102
LAN interface
Phone Case
1xRJ45 10Base-T
25-button keypad
2xRJ45 10Base-T
25-button keypad
12-digit caller ID LCD 12-digit caller ID
LCD
Universal
Switching
Power Adaptor
Input: 100-240VAC
Output: +5VDC, 1.2A,
UL certified
Same as left
Same as left
Dimension
18cm (W)
22cm (D)
6.5cm (H)
Weight
2 lbs (0.9 kg)
Same as left
Same as left
Operating
Temperature
32 - 104oF
0 - 40oC
Humidity
10% - 95%
(non-condensing)
Same as left
Same as left
Compliance
FCC/CE/C-Tick
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4 Basic Operations
4.1 Get Familiar with LCD
BudgeTone-100 phone has a numeric LCD of 64mmx24mm size and a backlight. Here
is the display when all segments illuminate:
上午
下午
AM
PM
010
The LCD is equipped with a backlight. When the phone is configured properly and in
the normal idle state, the backlight is off. Whenever an event occurs, the backlight turns
on automatically and brings the user’s attention.
Icon
LCD Icon Definitions
Network Status Icon:
FLASH in the case of Ethernet link failure
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
Speaker Phone Status Icon:
FLASH when phone rings or a call is pending
OFF when the speakerphone is off
ON when the speakerphone is on
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Alarm Clock Status Icon:
OFF when the alarm clock is not set
ON when the alarm clock is set
Lock Status Icon:
OFF when the lock is set
ON when the lock is not set
Encryption Status Icon:
010
OFF when the voice encryption is off
ON when the voice encryption is on
Handset and Speakerphone 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:
01-99 for CALLED history (dialed number)
01-99 for CALLERS history (caller ID)
Time Icon:
AM
PM
AM for the morning
PM for the afternoon
IP Address Separator Icons:
Three icons combine to indicate valid IP address
Numerical Numbers and Characters:
0 - 9
* = └
# = ┘
A, b, C, c, d, E, F, G, g, H, h, I, J, (k), L, (m), n, O, o, P,
q, r, S, t, U, u, (v, w, x), y, (z)
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4.2 Get Familiar with Keypad
BudgeTone-100 phone has a 25-button keypad. Underneath the keypad, there are 4
LEDs in red color.
Key Button
Key Button Definitions
Digit, star and pound keys are usually used to make phone
calls
0 - 9, *, #
Next menu item when phone is in IDLE mode
Or reduce handset/speakerphone volume
↓
↑
Previous menu item when phone is in IDLE mode
Or increase handset/speakerphone volume
Enter MENU mode when phone is in IDLE mode.
It is also the ENTER key once entering MENU
MENU
CALLED
CALLERS
MESSAGE
HOLD
Display the phone numbers called
Display the caller IDs
Enter to retrieve voice mails or other messages
Temporarily hold the active call
Transfer the active call to another number
Enter 3-way conferencing call
TRANSFER
CONFERENCE
FLASH
Flash event to switch between two lines
Mute an active call; or Delete a key entry, call log, voice
mail and etc
MUTE/DEL
Dial a new number or Redial the number last dialed. After
entering the phone number, pressing this key would force
a call to go out immediately before timeout
SEND/(RE)DIAL
SPEAKERPHONE
Enter hands-free mode
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4.3 Make Phone Calls
4.3.1 Make Calls Using Regular Phone or Extension Numbers
There are four ways to make phone calls:
1. Pick up handset or press SPEAKERPHONE button, and then enter the phone
numbers
2. Press the SEND/REDIAL button directly to redial the number just 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.
3. Browse the CALLED history and press the SEND/REDIAL button.
Pick up the handset or press the speakerphone button, and then press the “Called”
button to browse through the last 10 numbers dialed out. Once the desired number
is identified and displayed on the LCD screen, press the SEND/(Re)Dial button
and a new call to that displayed number will be sent out immediately.
4. Browse the CALLERS history and press the SEND/REDIAL button
Pick up the handset or press the speakerphone button, then press the “Callers”
button to browse thru the last 10 caller IDs received. Once the desired number is
identified and displayed on the LCD screen, press the SEND/(Re)Dial button and
a new call to that displayed number will be sent out immediately.
Examples:
If the phone is configured with a user part of “1000” with a SIP proxy, then to
dial the user extension “1008”, simply just dial 1008 and then press the
“SEND/(Re)Dial” button.
If the phone is configured with a regular PSTN number 16172223333 with a
service provider’s server, then to dial any other PSTN number (say, 16266667890) will
be simply to dial that number (16266667890) as if you were calling from a regular
analog phone, followed by pressing the “SEND/(Re)Dial” button.
If the “SEND/(Re)Dial” button is not pressed, the phone will wait for about 5
seconds before initiating the call.
4.3.2 Make Calls using IP Address
Direct IP calling allows 2 phones to talk to each other in an ad hoc fashion without a
SIP proxy. VoIP calls can be made between two phones if
• both phones have public IP addresses, or
• both phones are on a same LAN using private or public IP addresses, or
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• both phones can be connected through a router using public or private IP
addresses.
To make a direct IP calling, first pick up the phone or turn on the speakerphone, then
press “Menu” button followed by the 12-digit target IP address. If there is a user part,
press “Menu” button and then the encoded user part, followed by *3 (encoding for “@”)
and then followed by the 12-digit target IP address. Destination ports can also be
specified using *4 (encoding for “:”) followed by the encoded port number.
The follow is a table of the encoding scheme for the most commonly used characters:
00
01
02
03
04
05
06
07
08
09
*0
*1
*2
*3
*4
21
22
23
31
32
33
41
42
43
51
52
53
0
1
2
3
4
5
6
7
8
9
. (dot character)
_ (underscore character)
- (hyphen character)
@
: (column character)
a
b
c
d
e
f
g
h
i
j
k
l
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61
62
63
71
72
73
74
81
82
83
91
92
93
94
m
n
o
p
q
r
s
t
u
v
w
x
y
z
The rule of thumb to remember these encoding is: “a” is the first letter on button “1” so
its encoding is “11”. “b” is the 2nd letter on button “1” and its encoding is “12”. “c” is
the 3rd letter on button “1” and its encoding is “13”. Likewise, “d” is the first letter on
button “2” and its encoding is “21”. This pattern and rule applies to all other alphabetic
encoding.
Examples:
If the target IP address is 192.168.0.160, the dialing convention is
Menu_key 192168000160
followed by pressing the “(Re)Dial” button or the “#” key is it is configured as a send
key. In this case, the default destination port 5060 is used if no port is specified.
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 “(Re)Dial” button or the “#” key is it is configured as a send
key.
Menu_key 51634262*3192168001100*45062
followed by pressing the “SEND/(Re)Dial” button or the “#” key is it is configured as a
send key.
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4.3.3 Answer an Incoming Call
There are two ways to answer an incoming call:
1. Pick up the handset to answer the call normally
2. Press the SPEAKERPHONE button to answer in speakerphone mode
4.3.4 Handset Mode and Speakerphone Mode
Handset mode and Speakerphone mode cannot be enabled at the same time. Pressing the
hook-switch or Speakerphone button would toggle the phone between these two modes.
4.3.5 Call Hold
While in conversation, pressing the “Hold” button will put the remote end on hold. This
is achieved by sending a Re-INVITE with “a=sendonly” attribute and a zero IP address
for media in the SDP message. Pressing the “Hold” button again will release the
previously Hold state and the bi-directional media will resume. This is triggered by
sending another Re-INVITE with “a=sendrecv” attribute and a non-zero IP address for
media in the SDP message.
4.3.6 Flash
This button is equivalent to putting an active call on Hold and then switching to the
other voice channel. If the other channel has an active conversation going on, this is
essentially a switching of the “talking” channel and the other channel will be activated.
If the other channel is idle with no active conversation going on, then the user will hear
a dial tone.
4.3.7 Call Transfer
The user can transfer an active call to a third phone by using the “Transfer” button. The
sequence is like this:
The 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),
he/she will hear a dial tone. He/She can then dial the 3rd phone and then
hangs up his own phone.
Two kinds of blind call transfers are supported: using REFER and using BYE/Also.
The SIP message flow based on SIP REGER method looks something like this:
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Call Flow Diagram For Blind Call Transfer
Transferee
Transferor
Recipient
INVITE
100/180/200
ACK
RTP Media
REFER
202
NOTIFY
200
BYE
200
INVITE
100/180/200
ACK
RTP Media
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The SIP message flow based on BYE/Also: method looks something like this:
Transferee
Transferor
Recipient
INVITE
100/180/200
ACK
RTP Media
REFER
501 Not Implemented
BYE with “Also:”
200
INVITE
100/180/200
ACK
RTP Media
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4.4 Call Features
Following table shows the call features of BudgeTone series phone.
Dial Key
*70
Call Features
Do not disturb, it will give caller busy tone when called. This
setting will be released when user hangs up the phone.
Unconditional Call Forward.
To use this feature, dial “*72” and get the dial tone. Then dial the
forward number and hang up.
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get the
dial tone, then hang up.
Busy Call Forward
To use this feature, dial “*90” and get the dial tone. Then dial the
forward number and hang up.
Cancel Busy Call Forward
To cancel “Unconditional Call Forward”, dial “*91” and get the
dial tone, then hang up
*72
*73
*90
*91
*92
*93
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Then dial the
forward number and hang up.
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then
hang up
Flash/Hook When in conversation, this action will switch to the new incoming
call if user heard the call waiting sound.
When in conversation and no incoming call heard, this action will
switch to a new channel for a new call.
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5 Configuration Guide
5.1 Configuration with Keypad
When the phone is on-hook, press the MENU button to enter MENU state. When the
phone goes off-hook or a call comes in, the phone automatically exits the MENU state
and prepare for the call. It also exits the MENU state if left idle for 20 seconds.
Here are the Menu options supported:
Menu Item
Menu Functions
Display “[1] dhcP On ”
or “[1] dhcP oFF” for the current selection
Press Menu to enter edit mode
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu to save and exit
1
Must recycle power to take effect
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 effect
2
3
4
Display “[3] SubNet ”
Press Menu to display the Subnet address
Enter new Subnet address if DHCP is OFF
Press ‘↓’ or ’↑’ to exit
Press Menu to (save and) exit
Must recycle power to take effect
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 effect
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Menu Item
Menu Functions
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 effect
5
6
Display “[6] tFtP
”
Press Menu to display the TFTP address
Enter new TFTP server address
Press ‘↓’ or ’↑’ to exit
Press Menu to save and 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 2”
2 “ - G-722 2”
3 “ - G-723 1”
7
4 “ - G-726 2”
5 “ - G-728 8”
6 “ - G-729 2”
7 “ - iLBC 1”
Press 1 to 9 to indicate number of frames per TX packet
Press Menu to save and exit. The codec selected takes effect
immediately. No power cycle needed
Display “[8] SIP SP-1”
Press Menu to display the SIP Server/Service Provider
Press ‘↓’ or ’↑’ to browse the valid SIP Server (1-9)
Press Menu to save and exit
8
SIP Server(s) must be configured via Web browser
Only configured SIP server(s) are displayed
Take effect immediately
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Menu Item
Menu Functions
Display “[9] codE rEL”
Press Menu to display the code releases
Press ‘↓’ or ’↑’ to browse
line 1 “b 2004-06-12” – date: boot code
2 “ 1. 0. 0.18” – version: boot code
3 “P 2004-06-21” – date: phone code
4 “ 1. 0. 5. 3” – version: phone code
5 “c 2004-05-06” – date: vocoder
6 “ 1. 0. 0. 6” – version: vocoder
7 “h 2004-06-17” – date: web server
8 “ 1. 0. 0. 36” – version: web server
9 “1r 2004-05-12” – date: 1st ring tone
10 “ 1. 0. 0. 0” – version: ring tone
11 “2r 2004-05-12” – 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, take effect immediately
9
Display “[10] Phy Addr”
10
11
Press Menu to display the physical / MAC address
Press Menu, ‘↓’ 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, take effect immediately
Display “ -- rESEt --”, please be very CAREFUL here
A> Key in the physical / MAC address on back of the
phone, Press Menu, phone will be reset back to
FACTORY DEFAULT setting, all your setting
will be erased and gone
B> Press Menu without key in anything, phone will
function the same as power cycle / reboot
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Menu Item
Menu Functions
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), to designed one, then off and on the
handset to set the ring tone volume
Others
• Press “SPEAKERPHONE” button, or off hook and pick
up handset, press ‘↓’ or ’↑’ to adjust the speakerphone or
phone volume
5.2 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 Microsoft’s IE.
5.2.1 Access the Web Configuration Menu
The IP Phone Web Configuration Menu can be accessed by the following URI:
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 in on-hook state, press Menu button and then the browsing
arrow keys to check “[2] IP Addr ”
2) When the phone is in off-hook or speakerphone state, simply press Menu button
Once this request is entered and sent from a Web browser, the IP phone will respond
with the following login screen:
Welcome to Grandstream IP Phone
Password
Login
The password is case sensitive and the factory default password is lower case ‘admin’.
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5.2.2 Configuration Menu
After key in the correct password in the login screen, the embedded web server inside
the IP phone will respond with the Configuration Menu screen. It will be displayed like
following. You will know the MAC address of the phone, product model,
software/firmware version and the available custom ring tones from the showed page.
Grandstream IP Phone Configuration
MAC Address: 00.0B.82.00.A1.25
Product Model: BT100
Software Version: Program--1.0.5.3 Bootloader--1.0.0.18 HTML--1.0.0.36 VOC--1.0.0.6
Custom Ring ring1--1.0.0.0 ring2--1.0.0.0 ring3--0.0.0.0
Tone: (all zeroes means unavailable or unsupported)
Admin Password:
IP Address:
(password to configure this IP phone)
dynamically assigned via DHCP (default) or PPPoE
(will attempt PPPoE if DHCP fails and following is non-blank)
PPPoE account ID:
PPPoE password:
0
0
0
0
Use this DNS server (if specified):
statically configured as:
.
.
.
192
168
0
0
160
0
IP Address:
.
.
.
0
0
0
0
0
Subnet Mask:
.
.
.
.
.
.
.
.
.
0
0
0
Default Router:
.
0
0
0
DNS Server 1:
.
0
0
0
DNS Server 2:
.
SIP Server:
Outbound Proxy:
SIP User ID:
(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)
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate
Password:
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Name:
(optional, e.g., John Doe)
Advanced Options:
current setting is "PCMU"
Preferred Vocoder:
choice 1:
(in listed order)
current setting is "PCMA"
current setting is "G723"
current setting is "G729"
current setting is "G726-32"
current setting is "G728"
current setting is "iLBC"
current setting is "G722"
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
choice 7:
choice 8:
G723 rate:
6.3kbps encoding rate
5.3kbps encoding rate
iLBC frame size:
20ms
30ms
99
iLBC payload type:
(between 96 and 127, default is 98)
Yes
Silence
Suppression:
No
Voice Frames per
TX:
2
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
(Diff-Serv or Precedence value)
48
Layer 3 QoS:
Layer 2 QoS:
Use DNS SRV:
0
0
802.1Q/VLAN Tag
802.1p priority value
(0-7)
No
Yes
Yes
No
User ID is phone
number:
No
SIP Registration:
Yes
Yes
Unregister On
Reboot:
No
Register
Expiration:
60
(in minutes. default 1 hour, max 45 days)
Early Dial:
No
Yes (use "Yes" only if proxy supports 484 response)
(this prefix string is added to each dialed number)
Dial Plan Prefix:
No Key Entry
Timeout:
4
(in seconds, default is 4 seconds)
Use # as Dial Key:
local SIP port:
No
5060
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
(default 5060)
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5004
local RTP port:
(1024-65535, default 5004)
Yes
Use random port:
No
NAT Traversal:
No
Yes, STUN server is:
(URI or IP:port)
20
keep-alive interval:
Use NAT IP
(in seconds, default 20 seconds)
(if specified, this IP address is used in SIP/SDP
message)
67
153
142
69
TFTP Server:
.
.
.
(for remote software upgrade and configuration)
(User ID/extension for 3rd party voice mail system)
Voice Mail
UserID:
SUBSCRIBE for
MWI:
No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Auto Answer:
No
Yes
(User ID/extension to dial automatically when
Offhook Auto-Dial:
offhook)
Enable Call
Features:
No
locally)
Yes (if Yes, Call Forwarding & Do-Not-Disturb are supported
Disable Call-
Waiting:
No
Yes
Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
DTMF Payload
Type:
101
Send Flash Event:
No
time.nist.gov
Yes (Flash will be sent as a DTMF event if set to Yes)
(URI or IP address)
NTP Server:
Time Zone:
current setting is "GMT-5:00 (US Eastern Time, New York)"
Year-Month-Day
Month-Day-Year
Day-Month-Year
Date Display
Format:
Daylight Savings
Time:
No
Yes (if set to Yes, display time will be 1 hour ahead of normal
time)
system ring tone
Default Ring Tone:
custom ring tone 1, used if incoming caller ID is
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custom ring tone 2, used if incoming caller ID is
custom ring tone 3, used if incoming caller ID is
Send Anonymous:
No
No
Yes (caller ID will be blocked if set to Yes)
Yes (configuration update via keypad is disabled if set to Yes)
Lock keypad
update:
Update
Cancel
Reboot
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The definitions for all the configuration parameters in the Configuration Menu are:
This contains the password to access the Web Configuration Page.
This field is case sensitive and maximum length is 16 characters.
Admin Password
There are 2 modes under which the IP phone can operate:
1. DHCP (default) or PPPoE Mode
If DHCP mode is enabled (default), 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. In addition,
the phone will attempt PPPoE if DHCP fails.
Be careful if the PPPoE account ID or password field is not empty,
the phone will try PPPoE from then on and never use DHCP. So
leave these two fields BLANK if using DHCP.
With this feature, the phone can be directly connected to ISP or
broadband service providers’ Cable or DSL modem.
IP Address
Use this DNS server (if specified), any IP put here will become the
primary DNS server, if also the other field “Use DNS SVR” is set to
“Yes”. This helps in some cases if the DNS server get from DHCP
or PPPoE not working properly, client can put own DNS server here.
Default is blank and “Use DNS SVR” is “No”.
2. Static IP Mode
If Static IP mode is selected, then the IP address, Subnet Mask,
Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary) fields will need to be configured. These fields are reset
to zero by default.
This field contains the URI string or the IP address (and port, if
different from 5060) of the SIP proxy server. e.g., the following are
SIP Server
some valid examples:
sip.my-voip-provider.com, or sip:my-
company-sip-server.com, or 192.168.1.200:5066
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This field contains the URI string or the IP address (and port, if
different from 5060) of the outbound proxy. If there is no outbound
proxy, this field SHOULD be left blank. If not blank, all outgoing
requests will be sent to this outbound proxy.
Outbound Proxy
SIP User ID
This field contains the user part of the SIP address for this phone.
e.g., if the SIP address is: sip:my_user_id@my_provider.com, then
the SIP User ID is: my_user_id. Please do NOT include the
preceding “sip:” scheme or the host portion of the SIP address in this
field.
This field contains the login ID used for SIP authentication.
Typically, this is the account number on an SIP server for this IP
phone. It can be the same as or different from the above SIP User
ID, depending on whether a separate account ID is used for
authentication.
Authenticate ID
This field contains the password used for SIP authentication. It is
used together with the above SIP User ID.
Authenticate
Password
This field contains user’s real name, it’s optional
Name
The BudgeTone IP phone supports up to 8 different vocoder types
including G711-ulaw, G711-alaw, G722, G723, G729A/B, G726-32
(ADPCM), G728 and iLBC. Depending on the product model, some
of these codecs may not be provided in standard release.
Preferred
Vocoder
A user can configure codecs 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”.
This defines the encoding rate for G723 codec. By default, 6.3kbps
rate is chosen.
G723 Rate
This defines the size of the iLBC codec frame. The default setting is
20ms.
iLBC
frame size
This defines payload time for iLBC. Default value is 98. The valid
range is between 96 and 127.
iLBC
payload type
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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.
Silence
Suppression
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.
Voice Frames per
TX
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 phone 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/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.
Layer 3 QoS
Layer 2 QoS
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
Default is No. If set to Yes, the phone will use the previous specified
DNS server and put DHCP/PPPoE DNS server as 2nd/backup choice
Use DNS SRV:
If the BudgeTone phone has an assigned PSTN telephone number,
then this field will 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
User ID is Phone
Number
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This parameter controls whether the BudgeTone phone needs to send
REGISTER messages to the proxy server. The default setting is
“Yes”.
SIP Registration
Default is No. If set to yes, the SIP user will be unregistered on
reboot.
Unregister on
Reboot
This parameter allows the user to specify the time frequency (in
minutes) the BudgeTone phone will refresh its registration with the
specified registrar. The default interval is 60 minutes (or 1 hour).
The maximum interval is 65535 minutes (about 45 days).
Register
Expiration
Default is No. Use only if proxy supports 484 response
Sets the prefix added to each dialed number
Default is 4 seconds.
Early Dial
Dial Plan Prefix
No Key Entry
Timeout
This parameter allows the user to configure the “#” key to be used as
the “Send”(or “Dial”) 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 “(Re)Dial” key. If
set to “No”, this # key will then be included as part of the dial string
to be sent out.
Use # as
Dial Key
This parameter defines the local SIP port the BudgeTone phone will
listen to and transmit on. The default value is 5060.
Local SIP port
Local RTP port
This parameter defines the local RTP-RTCP port pair the
BudgeTone phone will listen to and transmit on. 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 phones are behind the same NAT.
Use Random Port
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This parameter defines whether the BudgeTone phone NAT
traversal mechanism will be activated or not. If activated (by
choosing “Yes”) and a STUN server is also specified, then the
BudgeTone phone will behave according to the STUN client
specification. Under this mode, the embedded STUN client inside
the BudgeTone phone will attempt to detect if and what type of
firewall/NAT it is 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 phone will attempt to use
its mapped public IP address and port in all the SIP and SDP
messages it sends out.
NAT
Traversal
If this field is set to “Yes” with no specified STUN server, then the
BudgeTone phone will periodically (every 10 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 20 seconds. This field specify the time period to send out
packets to keep router’s NAT mapping port open
keep-alive
interval
NAT IP address used in SIP/SDP message. Default is blank.
Use NAT IP
TFTP Server
This is the IP address of the configured tftp server. If it is non-zero
or not blank, the BudgeTone phone 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.
If the ID number put here, when “MESSAGE” button pressed, the
phone will automatically send out a SIP request. For example, if
8005 put here, when “MESSAGE” button pressed, a SIP message
“sip:8005@my_provider.com” will be sent to SIP server.
Voice Mail
UserID
Default is NO. When set to Yes a SUBSCRIBE for Message
Waiting Indication will be sent periodically.
Subscribe for
MWI
Default is NO. When set to Yes, any incoming call will be
automatically answered via speakerphone after a Beep. This is
somewhat similar to Intercom but still different.
Auto Answer
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This parameter allows the user to configure a User ID or extension
number to be automatically dialed upon offhook. Please note that
only the user part of a SIP address needs to be entered here. The
BudgeTone phone will automatically append the “@” and the host
portion of the corresponding SIP address.
Offhook
Auto-Dial
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are
supported locally
Enable Call
Features
Default is No.
Disable Call
Waiting
This parameter controls the way DTMF events are transmitted.
There are 3 ways: 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.
Send DTMF
Default is 101. This is the inband DTMF payload
DTMF Payload
Type
This parameter allows the user to control whether to send an SIP
NOTIFY message indicating the Flash event, or just to switch to the
voice channel when the user presses the Flash key.
Send
Flash Event
This parameter defines the URI or IP address of the NTP server that
the BudgeTone phone will use to display the current date/time.
Default is “time.nist.gov”
NTP server
Time Zone
This parameter controls how the displayed date/time will be adjusted
according to the specified time zone.
Date display format for different countries
Date Display
Format
This parameter controls whether the displayed time will be daylight
savings time or not. If set to Yes, then the displayed time will be 1
hour ahead of normal time.
Daylight Savings
Time
Different system ring tones (now have 3 ) can be selected. Also can
use different ring tone to identify different incoming call by caller ID
Default
Ring Tone
If this parameter is set to “Yes”, the “From” header in outgoing
INVITE message will be set to anonymous, essentially blocking the
Caller ID from displaying.
Send Anonymous
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If this parameter is set to “Yes”, the configuration update via keypad
is disabled.
Lock keypad
update
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5.2.3 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 IP Phone Configuration Update Status
Your configuration changes have been saved.
They will take effect on next reboot.
Click to view your changes
The user is recommended to power cycle the IP phone after seeing the above
message.
5.2.4 Rebooting the phone from remotely
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.
Grandstream IP Phone Rebooting Status
The IP phone 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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5.3 Configuration through a Central Server
Grandstream IP phones can be automatically configured via a central provisioning
system called Grandstream Automated Provisioning System (GAPS).
With GAPS, a service provider or an enterprise with large deployment of IP phones can
easily manage the configuration and service provisioning of individual devices remotely
and automatically via a central server. GAPS uses enhanced (NAT friendly) tftp and
other communication protocols to communicate with each individual IP phone even if
the phone is behind a NAT.
GAPS must be used to support automated configuration of an IP phone. To enable this
feature on the phone, the user just needs to enter the IP address of GAPS server in the
tftp server field of the configuration screen. Then power cycle the phone.
For details on how GAPS works, please refer to the documentation of GAPS product.
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6 Software Upgrade
6.1 Upgrade with TFTP
Warning !!!
The device WILL get damaged if there is a power outage during firmware upgrade.
Grandstream STRONGLY recommend customer maintain UNINTERRUPTED
POWER SUPPLY during firmware upgrade. This damage is NOT covered by the
manufacture warranty. Grandstream will NOT take any responsibility for this kind of
damage. Please be very CAREFUL when doing firmware upgrade.
To upgrade software, BudgeTone-100 phone can be configured with a TFTP server
where the new code image is located. The TFTP upgrade can work in either static IP or
DHCP mode using private or public IP address. It is recommended that the TFTP server
must have either public IP address or be on the same LAN segment with the phone.
There are 2 ways to set up the TFTP server to upgrade the firmware, namely via keypad
and via the phone’s Web configuration interface. To configure the TFTP server via
keypad, press “Menu” key and then press the down arrow key 5 times until you see the
“tftp” option. Press the “Menu” again and the current TFTP server IP address will be
displayed on the LCD. Enter the new TFTP server’s full 12-digit IP address and wait
about 2 seconds for the 3 separating “.” to show up on the LCD. E.g., if the new TFTP
server IP is 192.168.1.90, enter 192168001090. After about 2 seconds, the LCD will
display 192.168.001.090. At that time, press the “Menu” key to confirm and quit the
TFTP IP editing mode.
To configure the TFTP server via the Web configuration interface, open up your
browser to point at the IP address of the phone. Input the admin password to enter the
configuration screen. From there, enter the TFTP server address in the designated field
towards the bottom of the configuration screen.
Once the TFTP server is configured, power cycle the IP phone.
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 phone will attempt to retrieve the
new image files by downloading them into the phone’s SRAM. During this stage, the
Keypad LEDs will blink with 0.25 second ON and 0.25 second OFF pattern until the
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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 phone will stop the TFTP process and simply boot using
the existing code image in the flash.
TFTP may take as long as 1—10 minutes over Internet, or just 20+ seconds if it is
performed on a LAN. It is generally recommended to conduct TFTP upgrade in a
controlled LAN environment if possible because the network environment like firewall,
NAT could affect TFTP function properly. For users who do not have local TFTP
server, Grandstream provides a NAT-friendly TFTP server on the public Internet for
users to download the latest firmware upgrade automatically. Please check the Service
or Support section of Grandstream’s Web site to obtain this TFTP server IP address.
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7 Connection Diagram
Network connection diagrams for Grandstream series products.
Internet
SIP SVR / IP PBX
BT 101/102
HT 286/486
Cable/xDSL Modem
Cable/xDSL Modem
Switch/Hub
Computer
Switch/Hub
Home Router
HT 286/486
HT 286/486
BT 101/102
HT 486
Analog Phone
Computer
Laptop
BT 101/102
Cordless
Phone
Computer
Computer
BT 102
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8 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.
Step 2:
Encode the Mac address. The encode rule is:
“2” is the first letter on the button “2” so its encoding is “2”.
“A” is the second letter on button “2” so its encoding is “22”.
“B” is the third letter on button “2” and its encoding is “222”.
“C” is the fourth letter on button “2” and its encoding is “2222”.
For example, the Mac address is 000b8200e395,
User should encode it as “0002228200333395”.
Step 3:
Access the phone screen menu, then select the “-- reset --" with the up or down
arrows keys.
Step 4:
Dial in the encode of the Mac address. Once the correct Mac address is displayed
in the LCD screen, press “MENU” button, the device will reboot automatically
and restore to the factory default setting.
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