User Manual
BudgeTone-100 Series
IP Phone
Version 1.00
OvisLink (Canada) Inc.
http://www.ovislink.ca/news/bt100.htm
BudgeTone-100 User Manual OvisLink(Canada) Inc.
Table of Contents
1 WELCOME ................................................................................................................3
2 INSTALLATION .......................................................................................................4
2.1 WHAT IS INCLUDED IN THE PACKAGE ......................................................................4
2.2 SAFETY COMPLIANCES ............................................................................................5
3 PRODUCT OVERVIEW...........................................................................................6
3.1 KEY FEATURES ........................................................................................................6
3.2 HARDWARE SPECIFICATION .....................................................................................7
4 BASIC OPERATIONS ..............................................................................................8
4.1 GET FAMILIAR WITH LCD .......................................................................................8
4.2 GET FAMILIAR WITH KEYPAD..................................................................................9
4.3 MAKE PHONE CALLS .............................................................................................10
4.3.1 Make Calls Using Regular Phone or Extension Numbers...........................10
4.3.2 Make Calls using IP Address.......................................................................11
4.3.3 Answer an Incoming Call.............................................................................13
4.3.4 Handset Mode and Speakerphone Mode .....................................................14
4.3.5 Call Hold......................................................................................................14
4.3.6 Flash.............................................................................................................14
4.3.7 Call Transfer ................................................................................................14
5 CONFIGURATION GUIDE...................................................................................17
5.1 CONFIGURATION WITH KEYPAD.............................................................................17
5.2 CONFIGURATION WITH WEB BROWSER..................................................................19
5.2.1 Access the Web Configuration Menu...........................................................19
5.2.2 Configuration Menu.....................................................................................20
5.2.3 Saving the Configuration Changes ..............................................................27
5.3 CONFIGURATION THROUGH A CENTRAL SERVER ...................................................28
6 SOFTWARE UPGRADE ........................................................................................29
6.1 UPGRADE WITH TFTP............................................................................................29
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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.
Budge Tone award-wining BudgeTone-100 series of SIP phones are innovative IP
telephones that offer a rich set of functionality and superb sound quality at ultraaffordable 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:
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(Re)
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
Speakerphone
Message
Hold
Transfer
Conference
Flash
Delete
Dial
2.1 What is Included in the Package
The BudgeTone-100 phone package contains:
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1) One BudgeTone-100 phone
2) One universal power adaptor
3) One Ethernet cable
4) User Manual
2.2 Safety Compliances
The BudgeTone-100 phone is compliant with various safety standards including
FCC/CE/UL. 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
Budge Tone IP Phone is a next generation IP network telephone based on industry open
standard SIP (Session Initiation Protocol). Built on innovative technology, Budge Tone
IP Phone features market leading superb sound quality and rich functionalities at massaffordable price.
3.1 Key Features
• Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP, DNS, DHCP,
NTP/SNTP, TFTP protocols
• Support IETF STUN and SIMPLE standard extension (Model 102D)
• Interoperable with various 3rd party SIP end user device, Proxy / Registrar /
Server, and gateway products.
• Advanced 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 popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D)
• Support standard voice features such as Caller ID Display or Block, Call Waiting,
Hold, Transfer, Forward, in-band and out-of-band DTMF (RFC2833), Dial Plans
• Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone,
redial, call log, volume control, voice mail with indicator, downloadable ring tone
(Model 102D)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort
Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain
Control)
• Support BASIC and DIGEST authentication (MD5, MD5-sess)
• Provide easy configuration thru manual operation (phone keypad and Web
interface) or automated centralized configuration file.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ,
MPLS)
• Remote software upgrade capability via TFTP
• Built-in alarm-clock with downloadable music ringing tone (Model 102D)
• Optional voice encryption (secure RTP, Model 102D)
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3.2 Hardware Specification
There are three models in the BudgeTone-100 family, namely:
BudgeTone-101
BudgeTone-102
BudgeTone-102D
The table below describes the difference among these models.
Model
BudgeTone-101
BudgeTone-102
BudgeTone-102D
LAN interface 1xRJ45 10Base-T 2xRJ45 10Base-T 2xRJ45 10Base-T
Phone Case 25-button keypad
12-digit caller ID LCD
25-button keypad
12-digit caller ID
25-button keypad
16x2 character LCD
LCD
Universal
Switching
Power Adaptor
Dimension 18cm (W)
Input: 100-240VAC
Output: +5VDC,
400mA
Same as left Same as left
Same as left Same as left
22cm (D)
6.5cm (H)
Weight 2 lbs (0.9 kg) Same as left Same as left
Operating
Temperature
Humidity 10% - 95%
32 - 104oF
0 - 40oC
Same as left Same as left
Same as left Same as left
(non-condensing)
Compliance UL/FCC/CE Same as left Same as left
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A
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:
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
上午
下午
M
PM
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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A
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:
M
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)
4.2 Get Familiar with Keypad
BudgeTone-100 phone has a 25-button keypad. Underneath the keypad, there are 4
LEDs in red color.
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Key Button Key Button Definitions
0 - 9, *, #
Digit, star and pound keys are usually used to make phone
calls
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.
MENU
CALLED
CALLERS
It is also the ENTER key once entering MENU
Display the phone numbers called
Display the caller IDs
MESSAGE Enter to retrieve voice mails or other messages
HOLD Temporarily hold the active call
TRANSFER Transfer the active call to another number
CONFERENCE Enter 3-way conferencing call
FLASH Flash event to switch between two lines
DEL Delete a key entry, call log, voice mail and etc
Redial the number dialed last time. After entering the
(RE)DIAL
phone number, pressing this key would force a call to go
out immediately before timeout
SPEAKERPHONE Enter hands-free mode
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
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2. Press the 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 REDIAL button.
Pick up the handset or press the speakerphone button, then press the “Called”
button to browse thru the last 10 numbers dialed out. Once the desired number is
identified and displayed on the LCD screen, press the (Re)Dial button and a new
call to that displayed number will be sent out immediately.
4. Browse the CALLERS history and press the 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 (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 “(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 “(Re)Dial” button.
If the “(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
• 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.
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The follow is a table of the encoding scheme for the most commonly used characters:
00 0
01 1
02 2
03 3
04 4
05 5
06 6
07 7
08 8
09 9
*0 . (dot character)
*1 _ (underscore character)
*2 - (hyphen character)
*3 @
*4 : (column character)
21 a
22 b
23 c
31 d
32 e
33 f
41 g
42 h
43 i
51 j
52 k
53 l
61 m
62 n
63 o
71 p
72 q
73 r
74 s
81 t
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82 u
83 v
91 w
92 x
93 y
94 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 2
rd
the 3
letter on button “1” and its encoding is “13”. Likewise, “d” is the first letter on
nd
letter on button “1” and its encoding is “12”. “c” is
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.
If the target address is john@192.168.1.100:5062
, then the dialing convention would be:
Menu_key 51634262*3192168001100*45062
followed by pressing the “(Re)Dial” button or the “#” key is it is configured as a send
key.
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
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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 again. 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 basically 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 3
rd
phone and then
hangs up his own phone.
2 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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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
1
2
3
Press Menu 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 address
Enter new Subnet address if DHCP is OFF
Press ‘↓’ or ’↑’ to exit
Press Menu to (save and) exit
Must recycle power to take effective!!!
4
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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Menu Item Menu Functions
Display “[5] dnS ”
Press Menu to display the DNS address
5
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
6
Enter new TFTP server address
Press ‘↓’ or ’↑’ to exit
Press Menu to save and exit
Display “[7] G-723 1”
Press Menu to select new vocoder
Press ‘↓’ or ’↑’ to browse a list of available vocoders
line 1 “ - G-711u 2”
7
8
2 “ - G-711A 2”
3 “ - G-723 1”
4 “ - G-726 1”
5 “ - G-728 4”
6 “ - G-729 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”
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
SIP Server(s) must be configured via Web browser
Only configured SIP server(s) are displayed
Take effective 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 2003-02-15” – date: boot code
2 “ 1. 0. 0. 2” – version: boot code
3 “P 2003-02-16” – date: phone code
9
5 “c 2003-02-16” – date: vocoder
6 “ 1. 0. 0. 5” – version: vocoder
7 “h 2003-02-16” – date: web server
8 “ 1. 0. 0. 5” – version: web server
9 “r 2003-02-16” – date: ring tone
10 “ 1. 0. 0. 5” – version: ring tone
Press Menu to exit
Display “10] Phy Addr”
4 “ 1. 0. 0. 5” – version: phone code
10
Press Menu to display the physical / MAC address
Press Menu, ‘↓’ or ’↑’ to exit
5.2 Configuration with Web Browser
BudgeTone-100 series IP phone has an embedded Web server that will respond to
HTTP GET/POST requests from a Web browser. 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. 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 ”
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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 Budge Tone IP Phone
Password
Login
The password is case sensitive and the factory default password is ‘admin’.
5.2.2 Configuration Menu
After the correct password is entered in the login screen, the embedded Web server
inside the IP phone will respond with the Configuration Menu screen which is explained
in details below.
The definitions for all the configuration parameters in the Configuration Menu are:
Password
This contains the password to access the Web Configuration Menu.
This field is case sensitive.
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.
- 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.
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OvisLink(Canada) Inc.
SIP Server
Outbound Proxy
SIP User ID
IP User ID is
Phone Number
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
some valid examples: sip.my-voip-provider.com, or sip:mycompany-sip-server.com, or 192.168.1.200:5066
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.
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.
If the IP 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.
SIP Login ID
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.
SIP Password
This field contains the password used for SIP authentication. It is
used together with the above SIP Login ID
MAC Address:
Software Version:
Admin Password:
IP Address:
00.0B.82.00.10.0A
Program--1.0.3.49 Bootloader--1.0.0.3 HTML--1.0.0.8
(password to configure this IP phone)
Budge Tone IP Phone Configuration
dynamically assigned via DHCP, or
statically configured as:
IP Address:
192 168 0 160
. . .
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p
D
R
SIP Server:
Outbound Proxy:
SIP User ID:
Authenticate ID:
Authenticate Password:
Name:
Advanced Options:
Preferred Vocoder:
(in listed order)
Subnet Mask:
Default Router:
DNS Server 1:
DNS Server 2:
sipprov ider.com
(e.g., sip.mycompany.com, or IP address)
255. 255. 255
0 0 0 0
. . .
0 0 0 0
. . .
0 0 0 0
. . .
0
.
(e.g., proxy.myprovider.com, or IP address, if any)
8001
8001
(can be identical to or different from SIP User ID)
John Doe
(optional, e.g., John Doe)
Choice1:
choice2:
choice3:
choice4:
choice5:
choice6:
current setting is "PCMU"
current setting is "G729"
G.723.1
PCMA
c setting is "G726-32"urrent
current setting is "G728"
G723 rate:
Silence Suppression:
Voice Frames per TX:
IP QoS:
VLAN Tag:
6.3kbps encoding rate 5.3kbps encoding rate
No Yes
2
(up to 10/20/32/64 frames for G711/G726/G723/othercodecs respectively)
48
(IP Diff-Serv or Precedence value for RTP)
0
(VLAN classification for RTP)
SIP User ID is
hone number:
ial Plan:
SIP Registration:
egister Expiration:
No Yes
(dial plan prefix, used only when SIP User ID is phone number)
Yes No
1
(in minutes. default 1 hour, max 45 days)
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N
N
S
Use # as Dial Key:
local SIP port:
local RTP port:
Voice Mail UserID:
Offhook Auto-Dial:
Send Anonymous:
Early Dial:
AT Traversal:
TFTP Server:
TP Server:
Time Zone:
No Yes (use "Yes" only if proxy supports 484 response)
No Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
5060
No Yes, STUN server is: . . .
. . . (for remote software upgrade and configuration)
(User ID/extension for 3rd party voice mail system)
(User ID/extension to dial automatically when offhook)
current setting is "GMT-5:00 (US Eastern Time, New Y ork)"
No Yes (caller ID will be blocked if set to Yes)
(default 5060)
5004
(1024-65535, default 5004)
time.nist.gov
(URI or IP address)
Update
0000
Preferred
Vocoder
G723 Rate:
ilence
Suppression
The BudgeTone IP phone supports up to 6 different vocoder types
including G711-ulaw, G711-alaw, G723, G729A/B, G726-32
(ADPCM), and G728. Depending on the product model, some of
these vocoders may not be provided in standard release.
A user can configure vocoders in a preference list that will be
included with the same preference order in SDP message. The first
vocoder in this list can be entered by choosing the appropriate option
in “Choice 1”. Similarly, the last vocoder in this list can be entered
by choosing the appropriate option in “Choice 6”.
This defines the encoding rate for G723 vocoder. By default, 6.3kbps
rate is chosen.
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.
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b
Voice Frames
per TX
This field contains the number of voice frames to be transmitted in a
single packet. When setting this value, the user should be aware of
the requested packet time (used in SDP message) as a result of
configuring this parameter. This parameter is associated with the
first vocoder in the above vocoder Preference List or the actual used
payload type negotiated between the 2 conversation parties at run
time.
e.g., if the first vocoder is configured as G723 and the “Voice
Frames per TX” is set to be 2, then the “ptime” value in the SDP
message of an INVITE request will be 60ms because each G723
voice frame contains 30ms of audio. Similarly, if this field is set to
e 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 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.
IP Qos
VLAN Tag
Dial Plan
This field defines the layer 3 QoS parameter which can be the value
used for IP Precedence or Diff-Serv or MPLS. Default value is 48.
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
This value contains the dial plan prefix string (typically an ASCII
numeric string). If it is not blank, then this string will be used as a
prefix to the target URI string in the “To” header field of an INVITE
message.
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Reg
Early Dial
Use # as
Send Key
This parameter controls whether the phone will attempt to send an
early INVITE each time a key is pressed when a user dials a number.
If set to “Yes”, an INVITE is sent using the dial-number collected
thus far; Otherwise, no INVITE is sent until the “(Re-)Dial” button
is pressed or after about 5 seconds have elapsed if the user forgets to
press the “(Re-)Dial” button.
The “Yes” option should be used ONLY if there is a SIP proxy
configured and the proxy server supports 484 Incomplete Address
response. 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.
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.
IP
Registration
istration
Interval
Local SIP port
Local RTP port
This parameter controls whether the IP phone needs to send
REGISTER messages to the proxy server. The default setting is
“Yes”.
This parameter allows the user to specify the time frequency (in
minutes) the 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).
This parameter defines the local SIP port the IP phone will listen and
transmit on. The default value is 5060.
This parameter defines the local RTP-RTCP port pair the IP phone
will listen 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.
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NAT Traversal
TFTP Server
This parameter defines whether the phone NAT traversal mechanism
will be activated or not. If activated (by choosing “Yes”) and a
STUN server is also specified, then the phone will behave according
to the STUN client specification. Under this mode, the embedded
STUN client inside the 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 phone will attempt to
use its mapped public IP address and port in all the SIP and SDP
messages it sends out.
If this field is set to “Yes” with no specified STUN server, then the
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.
This is the IP address of the configured tftp server. If it is non-zero
or not blank, the IP 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.
Voice Mail
User ID
Offhook
Auto-Dial
NTP server
Time Zone
This parameter defines the User ID (or extension number) of a 3
rd
party voice mail system where the user may have an account. By
defining this Voice Mail extension, when the user presses the
“Message” button on the phone, an INVITE message will be sent to
that extension to allow the user to retrieve messages.
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
phone will automatically append the “@” and the host portion of the
corresponding SIP address.
This parameter defines the URI or IP address of the NTP server
which the IP phone will use to display the current date/time.
This parameter controls how the displayed date/time will be adjusted
according to the specified time zone.
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If this parameter is set to “Yes”, the “From” header in outgoing
end
Anonymous
INVITE message will be set to anonymous, essentially blocking the
Caller ID from displaying.
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.
Budge Tone IP Phone Configuration Update Status
Your configuration changes have been saved.
They will take effect on next reboot.
Back to Home Page
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.
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Budge Tone 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.
5.3 Configuration through a Central Server
The content of this section will be provided when this feature is implemented in the near
future.
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6 Software Upgrade
6.1 Upgrade with TFTP
To upgrade software, BudgeTone-100 phone can be configured with a tftp server where
the new code image is stored. The phone can be configured in either static IP or DHCP
mode using private or public IP address. TFTP server must have either public IP address
or be on the same LAN with the phone. Once this is done, 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
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 4 minutes over Internet, or just a few seconds if it is
performed on a LAN. It is generally recommended to use conduct tftp upgrade in a
controlled LAN environment.
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