BudgeTone-200 SIP Phone
User Manual
Updated – 01/2008
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Table 8: Key Pad Configuration Menu.......................................................................................... 19
Table 9: Device Configuration - Status ......................................................................................... 22
Table 10: Device Configuration – Basic Settings.......................................................................... 22
Table 11: Advanced Settings........................................................................................................ 23
Table 12: SIP Account Settings .................................................................................................... 27
GUI INTERFACE EXAMPLES
BT200 USER MANUAL
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE
2. SCREENSHOT OF STATUS PAGE
3. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
4. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE
5. SCREENSHOT OF SIP ACCOUNT CONFIGURATION
6. SCREENSHOT OF SAVED CONFIGURATION CHANGES
7. SCREENSHOT OF REBOOT PAGE
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Welcome
Thank you for purchasing Grandstream BudgeTone-200 IP Phone. You made an excellent choice and
we hope you will enjoy all its capabilities.
Grandstream's BT200 SIP IP phone is the innovative IP telephone that offers a rich set of functionality
and superb sound quality. They are fully compatible with SIP industry standard and can interoperate with
many other SIP compliant devices and software on the market.
This document is subject to changes without notice. The latest electronic version of this user manual is
available for download from the following location:
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.
Warning: Please do not use a different power adaptor with the BT200 as it may cause damage to the
products and void the manufacturer warranty.
•
•
•
This document is contains links to Grandstream GUI Interfaces. Please download these examples
http://www.grandstream.com/user_manuals/GUI/GUI_BT200.rar for your reference.
This document is subject to change without notice. The latest electronic version of this user manual
is available for download @: http://www.grandstream.com/user_manuals/BT200_User_Manual.pdf
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.
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Installation
WHAT IS INCLUDED IN THE PACKAGE
The BudgeTone-200 phone package contains:
1. One BudgeTone -200 Main Case
2. One Handset
3. One Phone Cord
4. One Universal Power Adapter
5. One Ethernet Cable
CONNECTING YOUR PHONE
Following is a backside picture of BT200, each connection port is labeled with the name in the following
table:
Figure 1: BT200 Back Panel
PC
LAN
POWER HEADSET
Table 1: BT200 Connectors
LAN
10/100 Switch LAN port for connecting to Ethernet.
10/100Mbps RJ-45 ports for PC (downlink) connection.
5V DC power port; UL Certified
PC
Power Jack
HEADSET
2.5mm Headset port
1. Connect attached telephone cord to the handset and to the phone device.
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2. Put the handset on the base.
3. Connect attached LAN cable to the LAN port of the phone.
4. Connect attached power supply first to the 5V power port of the phone.
5. Connect attached power supply to the wall outlet.
When power on, the phone starts the initialization procedure.
Figure 2: BT200 Internal Headset Wiring Schema
SAFETY COMPLIANCES
The BudgeTone-200 phone is compliant with various safety standards including FCC/CE. Its power
adaptor is compliant with UL standard. The phone should only be operated with the universal power
adaptor provided with the package. Damages to the phone caused by using other unsupported power
adaptors are not covered by the manufacturer’s warranty.
WARRANTY
If you purchased your BT200 from a reseller, please contact the company where you purchased your
phone 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
before you return the product. Grandstream reserves the right to remedy warranty policy without prior
notification.
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Product Overview
The following photo illustrates the appearance of a BudgeTone-200 IP phone.
Figure 3: BT200 Front View
Figure 4: BT200 Side View
Table 2: BT200 Key Features in a Glance
Features
Benefits
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Table 3: Hardware Specifications
LAN Interface (Ethernet ports)
Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and
PC port with auto detection
Table 4: BT200 Technical Specifications
Protocol
Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP/SRTP, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP/SNTP, TFTP.
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723,1 (5.3/6.3K), G.729A/B, G.711 a/µ-law, G.726-32, G.722 (wide-
band), GSM and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay & loss concealment
3-way-conference, off-hook auto dial, auto answer, early dial and speed dial
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Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS
User specific URL for configuration file and firmware files
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Using the GXP SIP Enterprise Phone
GETTING FAMILIAR WITH THE LCD
BT200 phone has a numeric LCD of 64mmx24mm size with backlight. This model has a small red LED
status reminder. Here is the display when all segments illuminate:
Figure 5: BT200 LCD
When the phone is in the normal idle state, the backlight is off. Whenever an event (call) occurs, the
backlight will turn on automatically to bring the user’s attention. In addition, if Voice Mail configured and
there is a VM waiting, the red LED will be blinking to remind user there is a Voice Mail in the Voice Mail
server.
Table 5: LCD Icons
Icon
LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon:
OFF – IP address of Sip server is not found
ON – IP address of Sip server are located
Blinking –Ethernet link failure or the phone is not registered properly
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
OFF when the speakerphone is off
ON when the speakerphone is on
Handset and Speakerphone/Headset Volume Icons:
0-7 scales to adjust handset / speakerphone volume
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Real–time Clock:
Synchronized to Internet time server
AM
PM
Time zone configurable via web browser
AM/PM indicator
Call Logs:
01-10 for CALLED history (dialed number)
01-10 for CALLERS history (Incoming caller ID)
Time Icon:
AM for the morning
PM for the afternoon
AM
PM
IP Address Separator Icons
Numerical Numbers and Characters:
0 - 9
* = └
# = ┘
A, b, C, c, d, E, F, G, g, H, h, I, L, n, O, o, P, q, r, S, t, U, u, Y
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GETTING FAMILIAR WITH KEYPAD
LCD
Up/Down Keys
Message Waiting Indicator
Menu Button
Called
Message
Callers
Hold
Transfer
Conference
Flash
Standard Keypad
Speaker
Send
Mute/Delete
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Table 6: BT200 Keypad Buttons
Key Button Key Button Definitions
0 - 9, *, #
Digit, star and pound keys are usually used to make phone calls
1) Reduce handset, speakerphone/headset volume after off hook the phone
via handset or speaker
2) Reduce ring tone volume when phone in IDLE and off hook to confirm the
changed ring tone volume
3) Next menu item browsing when phone is in IDLE mode after MENU key
pressed, off hook to interrupt and exit
↓
Enter keypad MENU mode when phone is in IDLE mode.
It is also the ENTER key once entering MENU
MENU
CALLED
CALLERS
After off hook, press to display the dialed numbers. When number displayed,
press the SEND key can make call using that displayed number
After off hook, press to display the incoming Caller IDs. When number
displayed, press the SEND key can make call using that displayed number
Enter to retrieve voice mails from Voice Mail Portal or Server
Temporarily hold the active call
MESSAGE
HOLD
Transfer the active call to another party
Establish 3-way conferencing call
TRANSFER
CONFERENCE
FLASH
Flash event to switch between two lines
Mute an active call; or Delete a key entry
Also used to ‘REJECT’ incoming call.
MUTE/DEL
Dial a new number inputted or Redial the number last dialed. After entering
the phone number, pressing this key would force a call to go out immediately
before timeout
SEND/(RE)DIAL
Enter hands-free mode
SPEAKERPHONE
MENU
Enter keypad MENU mode when phone is in IDLE mode.
It is also the ENTER key once entering MENU
After off hook, press to display the dialed numbers. When number displayed,
press the SEND key can make call using that displayed number
CALLED
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MAKING PHONE CALLS
Handset, Speakerphone and Headset Mode
The regular Handset mode can be switched with either the Speaker mode (Hand free) or the Headset
mode, however, whenever the Headset is plugged in, Speaker mode will be switched to the Headset
mode automatically.
To Switch between Handset and Speaker/Headset, simply press the Hook Flash in the Handset cradle or
the Speaker button.
Make Calls using Numbers
There are FIVE ways to make phone calls:
•
•
•
Pick up handset or press SPEAKERPHONE button, and then enter the phone numbers
Press the SEND button directly to redial the number last called.
Once pressed, the last dialed number will be displayed on the LCD as the corresponding DTMF
tones are played out and an outgoing call is sent.
•
•
Browse the CALLED/CALLER history and press the SEND/REDIAL button.
Pick up the handset or press the speakerphone button, then press the “CALLED/CALLERS”
button to browse thru the last 10 numbers dialed out. Once the desired number is identified and
displayed on the LCD screen, press the SEND button and a new call to that displayed number will
be sent out immediately.
Examples:
•
•
To dial another extension on the same proxy, such as 1008, simply pick up handset or press
speakerphone, dial 1008 and then press the “SEND” button.
To dial a PSTN number such as 6266667890, you might need to enter in some prefix number
followed by the phone number. Please check with your VoIP service provider to get the
information. If you phone is assigned with a PSTN-like number such as 6265556789, most likely
you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone,
followed by pressing the “SEND” button.
Make Calls using IP Address
Direct IP calling allows two parties, that is, a BT200 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 BT200 phone and other VoIP Device(i.e., another IP Phone or BT200 SIP phone or other
VoIP unit) have public IP addresses, or
Both BT200 phone and other VoIP Device are on the same LAN using private or public IP
addresses, or
Both BT200 phone and other VoIP Device can be connected through a router using public or
private IP addresses (with necessary port forwarding or DMZ).
This model has the ability to dial an IP address under the same LAN segment by simply pressing the last
octet in the IP address.
In the Advanced Settings page there is an option "Use Quick IP-call mode", by default it is set to No.
When this option is set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255, phone will make
direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of
subnet mask.
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#XX or #X are also valid so leading 0 is not required (but OK).
eg.
192.168.0.2 calling 192.168.0.3 just dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 just dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or #
192.168.0.2 dial #3 and #03 and #003 has same effect --> call 192.168.0.3
Note:- If you have a SIP Server configured, Direct IP-IP call will still work. However, if you are using
STUN, Direct IP-IP call will also use STUN.
OR
To make a direct IP to IP call, first off hook, then press “MENU” key, then enter a 12-digit target IP
address to make the call. If port is not default 5060, destination ports can be specified by using “*4”
(encoding for “:”) followed by the port number.
Examples:
•
If the target IP address is 192.168.0.10, the dialing convention is
MENU_key 192 168 000 010
followed by pressing the “SEND” key or wait for seconds in the No Key Entry Timeout.
•
If the target IP address/port is 192.168.1.20:5062, 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.
Quick IP Call Mode
The BT200 also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only
the last few digits (last octet) of the target phone’s IP-number.
This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using
the CMSA/CD without a SIP server. Controlled static IP usage is recommended.
Setting up the phone to make Quick IP calls
To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In
the “Advanced Settings” page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is
0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the local IP
address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required
(but OK).
For example:
192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or #
192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3
NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct
IP-IP call will also use STUN. Configure the “Use Random Port” to “NO” when completing Direct IP calls.
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Answer an Incoming Call
There are two ways to answer an incoming call:
1.
2.
Pick up the handset to answer the call normally using handset, or
Press the SPEAKERPHONE button to answer in speakerphone or headset mode
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.
Call Transfer
Two transfer operations are supported.
BLIND TRANSFER
User can transfer an active call to a third party without announcement.
User presses the “TRANSFER” button and if the other voice channel is available (i.e., there is no
other active conversation besides the current one), user will hear a dial tone. User can then dial the
third party’s phone number followed by pressing SEND button.
NOTE:
•
“Enable Call Feature” has to be configured to “Yes” in web configuration page in order to make
the features to work.
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.
ATTENDED TRANSFER
User can transfer an active call to a third party with announcement.
User presses the “FLASH” button and hears a dial tone, then dial the third party’s phone number
followed by pressing “SEND” button. If the call is answered, press “TRANSFER” to complete the
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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.
Conference Call
BT200 phone supports 3-way conference.
Assuming that call party A and B are in conversation. A wants to bring C in a conference:
1. A presses the “CONFERENCE” button to get a dial tone and put B on hold
2. A dials C’s number then “SEND” key to make the call
3. If C answers the call, then A presses “CONFERENCE” button to bring B, C in the conference.
4. If C does not answer the call, A can press FLASH back to talk to B.
NOTE:
•
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.
Checking Message and Message Waiting Indication
When BudgeTone-200 is on-hook, pressing the MESSAGE button will trigger the phone to call the VM
Server (VMS) configured for the Account.
The MWI (Message Waiting Indicator) LED will flash in red color in three quarters of a second when
voicemail server sends message waiting information to BudgeTone-200.
Mute and Delete
When in conversation with an ACTIVE LINE, pressing “MUTE/DEL” will mute the conversation, that is,
you can hear the other party but the other party cannot hear you. Pressing the button again will resume
the conversation.
When dialing a number, press “MUTE/DEL” will delete the last entered digit.
When receiving incoming call, press “MUTE/DEL” will Reject the call and forward to voice mail.
CALL FEATURES
BT200 series phone supports a list of call features: Caller ID Block (or Anonymous Call), Disable/Enable
Call Waiting, Call Forward on Busy, Delay, or Unconditional, etc.
Following table shows the call features of BudgeTone-200 series phone.
Table 7: BT200 Call Features
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Key
*30
*31
*67
*82
*70
*71
*72
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
Disable Call Waiting (per Call)
Enable Call Waiting (per Call)
Unconditional Call Forward
To use this feature, dial “*72” and get the dial tone. Dial the forward number and “#”
for a dial tone, then hang up.
*73
*90
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. Dial the forward number and “#”
for a dial tone, then hang up.
*91
*92
Cancel Busy Call Forward
To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then hang up.
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Dial the forward number and “#”
for a dial tone, then hang up.
*93
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then hang up.
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Configuration Guide
The BT200 can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone;
secondly, through embedded web-configuration menu.
CONFIGURATION VIA KEYPAD
The BT200 keypad is the same as a traditional phone, with additional feature functionality including
speaker, menu keys, LCD navigation keys, and advanced feature keys which include hold, transfer,
conference and flash.
1. Be sure the phone is idle.
2. Press the MENU button to enter the keypad MENU to configure the phone.
Using the Keypad
1. To enter the MENU, press MENU button.
2. Navigate the menu by using the UP/DOWN
3. Press the MENU button to confirm a menu selection.
4. To delete an entry by pressing the MUTE/DEL button.
Press the MENU button to enter the key the Key Pad Menu. The menu options available are listed in table 8.
Table 8: Key Pad Configuration Menu
Manual Option Call Features
1
Display “[1] dhcP On ” or “[1] dhcP oFF” for the current selection
Press MENU key to enter edit mode
2
Display “[2] IP Addr ”
Press MENU to display the current IP address
Enter new IP address if DHCP is OFF
3
4
5
6
7
8
Display “[3] SubNet ”
Press MENU to display the Subnet mask
Enter new Subnet mask if DHCP is OFF
Display “[4] routEr ”
Press MENU to display the Router/Gateway address
Enter new Router/Gateway address if DHCP is OFF
Display “[5] dnS ”
Press MENU to display the DNS address
Enter new DNS address if DHCP is OFF
Display “[6] tFtP ”
Press MENU to display the TFTP address
Enter new TFTP server IP address
Display “[7] G-711u 2”
Press MENU to select new codec
Press ‘↓’ or ’↑’ to browse a list of available codecs
Display “[8] codE rEL”
Press Menu to display the code releases
Press ‘↓’ or ’↑’ to browse
9
Display “[9] Phy Addr”
Press MENU to display the physical / MAC address.
10
Display “[10] ring 0”
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Press MENU to hear the selected ring tone, press ‘↓’ or ’↑’ to select the stored ring tones.
11
Display “[11]trANSPOrt”
Press MENU to select SIP Transport.
Factory Reset
Display “ -- rESEt --”. Please be very careful when using this function.
Two options: factory reset or manual reboot.
Factory Reset:
1. Key in the physical / MAC address on back of the phone.
2. Press MENU to reset phone to FACTORY DEFAULT settings. All your setting will be
erased.
Manual Reboot: Press MENU without keying anything.
Others
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
CONFIGURATION VIA WEB BROWSER
The BT200 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE or Mozilla
Firefox.
Access the Web Configuration Menu
To access the phone’s Web Configuration Menu
•
•
•
•
•
Connect the computer to the same network as the phone1
Make sure the phone is turned on and shows its IP-address
Start a Web-browser on your computer
Enter the phone’s IP-address in the address bar of the browser2
Enter the administrator’s password to access the Web Configuration Menu3
1
2
3
The Web-enabled computer has to be connected to the same sub-network as the phone. This can
easily be done by connecting the computer to the same hub or switch as the phone is connected to. In
absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the
phone using the PC-port on the phone.
If the phone is properly connected to a working Internet connection, the phone will display its IP
address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. You
will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60,
The default administrator password is “admin”; the default end-user password is “123”.
NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of the
page. Reboot the phone to have the changes take effect. If, after having submitted some changes, more
settings have to be changed, press the menu option needed.
Definitions
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This section will describe the options in the Web configuration user interface. As mentioned, a used can
log in as an administrator or end-user.
Functions available for the end-user are:
•
Status: Displays the network status, account statuses, software version and MAC-address of the
phone
•
Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings
can be set here.
Additional functions available to administrators are:
•
•
Advanced Settings: To set advanced network settings, codec settings.
Account: To configure each of the SIP accounts.
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Table 9: Device Configuration - Status
MAC Address
IP Address
The device ID, in HEXADECIMAL format.
This field shows IP address of BT200
Product Model
Software Version
This field contains the product model information.
•
•
Program: This is the main software (firmware) release number, always used to
identify the software (firmware) system of the phone.
Boot: Booting code version number
System Up Time
Registered
This field shows system up time since the last reboot.
Indicates whether accounts are registered to the related SIP server(s). BT200 can
support four unique SIP profiles.
PPPoE Link Up
Indicates whether the PPPoE connection is enabled (connected to a modem).
Table 10: Device Configuration – Basic Settings
End User
Password
This contains the password to access the Web Configuration Menu. This field is
case sensitive with a maximum length of 25 characters.
IP Address
There BT200 operates in two modes:
1. DHCP mode: all the field values for the Static IP mode are not used (even
though they are still saved in the Flash memory.) The BT200 acquires its IP
address from the first DHCP server it discovers on its LAN. The DHCP
option is reserved for NAT router mode. To use the PPPoE feature, set the
PPPoE account settings. The BT200 establishes a PPPoE session if any of
the PPPoE fields are set.
2. Static IP mode: configure all of the following fields: IP address, Subnet
Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary). These fields are set to zero by default.
Time Zone
This parameter controls the date/time display according to the specified time zone.
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This parameter controls time displayed in daylight savings time. If set to “Yes”, then the
displayed time will be 1 hour ahead of normal time.
Daylight Savings Time
The “Optional Rule” is configured to automatically adjust the Daylight Savings Time (DST)
based on the rule set in this field.
Rule Syntax:
•
•
start-time; end-time; saving
Both start-time and end-time have the same syntax: month,day,weekday,hour,minute
o
o
o
month: 1,2,3,..,12 (for Jan, Feb, .., Dec)
day: [+|-]1,2,3,..,31
weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight
saving rule is not based on week days but based on the day of the month.
hour: hour (0-23), minute: minute (0-59)
o
If “weekday” is 0, it means the date to start or end daylight saving is at exactly the given date.
In that case, the “day” value must not be negative. If “weekday” is not zero and “day” is
positive, then the daylight saving starts on the first “day” the iteration of the weekday (e.g.:
1st Sunday, 3rd Tuesday etc). If “weekday” is not zero and “day” is negative, then the
daylight saving starts on the last “day” the iteration of the weekday (e.g.: last Sunday, 3rd last
Tuesday etc).
The saving is in the unit of minutes. The saving time may also be preceded by a negative (-)
sign if subtraction is desired instead of addition.
The default value is set for US, the “Automatic Daylight Saving Time Rule” shall be set to
“3,2,7,2,0;11,1,7,2,0;60”
Examples
US/Canada where daylight saving time is applicable:
03,02,7,02,00;11,1,7,02,00;60
This means the daylight saving time starts from the second Sunday of March at 2AM and
ends the first Sunday of November at 2AM. The saving is 60 minutes.
Date Display Format
Choose one of the following formats:
•
•
•
Year-Month-Day
Month-Day-Year
Day-Month-Year
Display Account Name
instead of Date
Choose to display Account Name or date on LCD.
Mute Speaker Ringer
in Headset Mode
Default is No. This field lets user to choose whether to ring the phone Speaker
when headset is connected.
Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous
configuration.
Table 11: Advanced Settings
Admin
Password
Administrator password. Only the administrator can access the “Advanced Settings” and
“Account Settings” page. Password field is purposely blank for security reasons after
clicking update and saved. The maximum password length is 25 characters.
G723 rate
Encoding rate for G723 codec. By default, 6.3kbps rate is set.
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iLBC frame size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required.
iLBC payload
type
Payload type for iLBC. Default value is 97. The valid range is between 96 and 127.
Silence
Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and G.729.
If set to “Yes”, when silence is detected, a 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 TX
This field contains the number of voice frames to be transmitted in a single Ethernet
packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500
byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is 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 G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in
the SDP message of an INVITE request will be 60ms because each G.723 voice frame
contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or
G.711 or G.726, 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 G.726, it is 20 (x10ms) frames;
for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames
respectively.
Please be careful when editing these parameters. Adjusting these parameters will also
change the dynamic jitter buffer. The BT200 has a patent dynamic jitter buffer handling
algorithm. The jitter buffer range is 20 ~ 200 ms.
Grandstream recommends using the default settings provided. Grandstream does not
recommend adjusting these parameters if you are an average user. Incorrect settings
will affect the voice quality.
Please refer to the Codec FAQ at
http://www.grandstream.com/faqscodec.html for more technical detail.
Layer 3 QoS
Layer 2 QoS
This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or
Diff-Serv or MPLS. Default value is 48.
This contains the value used for layer 2 VLAN tag. Default setting is blank.
Default is 4 seconds.
No Key Entry
Timeout
Use # as
Dial Key
This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If set
to “Yes”, the “#” key will immediately send the call. In this case, this key is essentially
equivalent to the “Dial” key. If set to “No”, the “#” key is included as part of the dial string.
Local RTP port This parameter defines the local RTP-RTCP port pair used to 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.
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Use Random
Port
This parameter, when set to “Yes”, will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple BT200s are behind the same
NAT. Default is No.
Keep-alive
interval
This parameter specifies how often the BT200 sends a blank UDP packet to the SIP
server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server
IP address or Domain name of the STUN server. STUN resolution result will display in
the STATUS page of the Web UI.
Firmware
Upgrade and
Provisioning
Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on
network environment. Do not interrupt the firmware upgrading process.
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not
blank, the BT200 will attempt to retrieve a new configuration file or new code image from
the specified TFTP server at boot time. It will make up to 3 attempts before timeout and
then it will start the boot process using the existing code image in the Flash memory. If a
TFTP server is configured and a new code image is retrieved, the new downloaded
image will be verified and then saved into the Flash memory.
Note: Grandstream strongly recommends that the user upgrade firmware locally in a
LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade
process (especially the power supply) as this will damage the device.
Via HTTP
Server
The HTTP server URL used for firmware upgrade and configuration via HTTP. For
Here “:6688” is the specific TCP port that the HTTP server is using; omit if using default
port 80.
Note: If Auto Upgrade is set to No, BT200 will only perform HTTP download once at
boot up.
Config Server
Path
IP address or domain name of firmware server.
Firmware File
Prefix/Postfix
Default is blank. If configured, BT200 will request the firmware file with the prefix/postfix.
This setting is useful for ITSPs. End user should keep it blank.
Config File
Prefix/Postfix
Default is blank. End user should keep it blank.
Allow DHCP
Option 66 to
override server
Default is Yes. This allows device gets provisioned automatically.
Authenticate
Conf File
Default is “No”. If set to “Yes”, configuration file would be authenticated before
acceptance. End user should use default setting.
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Automatic
Upgrade
This function is used by ITSP. End user should NOT touch these parameters.
Default is No. Choose “Yes” to enable automatic HTTP upgrade and provisioning.
In “Check for upgrade every” field, enter the number of minutes to check the HTTP
server for firmware upgrade or configuration changes. When set to “No”, the phone will
only perform HTTP upgrade and configuration check once at boot up.
Offhook Auto
Dial
To configure a User ID/extension to dial automatically when the phone is taken offhook.
DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833. Default is 101.
Onhook
Threshold
Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent
unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time
value.
Syslog Server
Syslog Level
The IP address or URL of System log server. This feature is especially useful for ITSPs.
Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,
INFO, WARNING or ERROR. Syslog messages are sent based on the following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the
following components: GS_LOG: [device MAC address][error code] error message
For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000].
Ethernet link is up.
NTP server
This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve.
It is used to display the current date/time.
Distinctive Ring Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a particular
Tone
Caller ID. The BT200 will ONLY use selected ring tones for particular Caller IDs. For all
other calls, the BT200 will use System Ring Tone. When selected and no Caller ID is
configured, the selected ring tone will be used for all incoming calls.
System Ring
Tone
System ring tone. Default is North American standard.
Adjust system ring tone frequencies and cadences based on local telecom standard.
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Call Progress
Tones
Using these settings, users can configure ring or tone frequencies based on parameters
from local telecom. By default, they are set to North American standard.
Frequencies should be configured with known values to avoid uncomfortable high pitch
sounds.
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order
to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause
of OFF ms and then repeat the pattern. Up to three cadences are supported.
Disable Call
Waiting
Default is No. If set to Yes, the call waiting will be disabled.
Use Quick IP
Call Mode
Dial an IP address under the same LAN/VPN segment by entering the last octet in the IP
address.
In the Advanced Settings page there is an option "Use Quick IP-call mode". Default
setting is No. When set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255,
phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the
local IP address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See Quick IP Call Mode
for details.
Lock keypad
update
If set to “Yes”, the configuration changes via keypad are disabled.
Table 12: SIP Account Settings
Account Name
SIP Server
The name associated with each account - displayed on LCD.
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy, Media Gateway, or Session
Border Controller. Used for firewall or NAT penetration in different network
environment. If the system detects symmetric NAT, STUN will not work. ONLY
outbound proxy can provide solution for symmetric NAT.
SIP User ID
User account information provided by VoIP service provider (ITSP); either an
actual phone number or formatted like one.
Authenticate ID
SIP service subscriber’s Authenticate ID used for authentication. It can be
identical to or different from SIP User ID.
Authenticate Password SIP service subscriber’s account password for BT200 to register to (SIP) servers
of ITSP.
Name
SIP service subscriber’s name that is used for Caller ID display.
Use DNS SRV
Default is No. If set to “Yes”, the client will use DNS SRV to look up server.
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User ID is Phone
Number
If the phone has an assigned PSTN telephone number, this field should be set to
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request
SIP Registration
This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.
Un-register on Reboot
Register Expiration
Default is No. If set to “Yes”, the SIP user’s registration information will be
cleared on reboot.
This parameter allows user to specify the time frequency (in minutes) that BT200
refreshes its registration with the specified registrar. The default interval is 60
minutes. The maximum interval is 65,535 minutes (about 45 days).
Local SIP Port
This parameter defines the local SIP port used to listen and transmit. The default
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
Account 4 respectively.
SIP Registration Failure Retry registration if the process failed. Default is 20 seconds.
Retry Wait Time
SIP T1 Timeout
SIP T2 Interval
SIP Transport
RFC 3261 SIP T1 timer. Default is 1 second.
RFC 3261 SIP T2 timer. Default is 0.5 seconds.
Choose SIP Transport between UDP and TCP. Default is UDP.
Use RFC3581
Symmetric Routing
This option allows a SIP client to request that the server send the response back
to the source IP address and port from which the request originated.
NAT Traversal (STUN)
This parameter activates the NAT traversal mechanism. If activated (by choosing
“Yes”) and a STUN server is also specified, the phone performs according to the
STUN client specification. Using this mode, the embedded STUN client detects if
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped
public IP address and port in all of its SIP and SDP messages. If the NAT
Traversal field is set to “Yes” with no specified STUN server, the BT200 will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the “hole” on the NAT open.
Subscribe for MWI
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail UserID
When configured, user can access messages by pressing “MSG” button. This ID
is usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: in audio which means DTMF is combined in audio signal (not
very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Early Dial
Default is No. Use only if proxy supports 484 response.
Sets the prefix added to each dialed number.
Dial Plan Prefix
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Delayed Call Forward
Wait Time
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Enable Call Features
Disable Call Log
Default is Yes. If set to “Yes”, all star code call features will be supported locally.
User can choose to disable Call Log.
Session Expiration
The SIP Session Timer extension enables SIP sessions to be periodically
“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVITE message, the session is
terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Min-SE
Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer
If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request Timer
Force Timer
If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
If set to “Yes”, the phone will use session timer even if the remote party does not
support this feature. If set to “No”, the session timer is enabled only when the
remote party supports this feature. To turn off Session Timer, select “No” for
Caller Request Timer, Callee Request Timer, and Force Timer.
UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
Force INVITE
Enable 100rel
Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
PRACK (Provisional Acknowledgment) method enables reliability to SIP
provisional responses (1xx series). This is required to support PSTN inter-
networking..
Account Ring Tone
There are 4 uniquely defined ring tones:
•
One (1) System Ring Tone: when selected, all calls will ring with system
ring tone.
•
Three (3) Customer Ring Tones: when selected, incoming calls from
designated account will play selected ring tone.
Send Anonymous
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.
Anonymous Method
Whether to use “sip:[email protected]>” in the From Header or P-
Asserted-Identity header.
Anonymous Call
Rejection
Default is NO. If set to YES, anonymous call will be rejected
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Auto Answer
Default is No. If set to “Yes”, BT200 will automatically switch on speaker to
answer the incoming call. Set to Intercom/Paging mode, it will answer the call
based on the SIP info header from the server.
Allow Auto Answer by
Call-Info
If the Call-Info header contains answer-after=0, the call be answered
automatically (so called paging mode).
Turn off speaker on
remote disconnect
When BYE is received, the phone will turn off its speaker automatically.
Check SIP User ID for
incoming INVITE
Check the SIP User ID in Request URI. If they don’t match, the call will be
rejected.
Refer-To Use Target
Contact
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header
uses the transferred target’s Contact header information.
Disable Multiple Media Default is No.
Attribute in SDP
Preferred Vocoder
BT200 supports up to 5 different Vocoder types including G.711(a/µ) (also known
as PCMU/PCMA), GSM, G.723.1, G.729A/B.
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by
choosing the appropriate option in “Choice 8”.
SRTP Mode
Enable SRTP mode based on selection. Default is No.
Special Feature
Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
SAVING THE CONFIGURATION CHANGES
After the user makes a change to the configuration, press the “Update” button in the Configuration Menu.
The web browser will then display a message window to confirm saved changes.
Grandstream recommends reboot or power cycle the IP phone after saving changes.
REBOOTING THE PHONE REMOTELY
Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in
again.
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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding
configuration settings are in the ADVANCED SETTINGS configuration page.
FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set
to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are
examples of some valid URLs.
•
•
firmware.mycompany.com:6688/Grandstream/1.1.5.15
168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web
Configuration Interface.
Key Pad Menu
Only TFTP would be used when configure the phone using key pad. To configure the Upgrade Server via
Key Pad Menu options, select “[6]tFtP” from the Main Menu, then press MENU button to enter this option.
Enter the 12-digit IP address. Then, press MENU button to save. Must reboot the phone to take effect.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the
BT200 IP address. Enter the admin password to access the web configuration interface. In the
ADVANCED SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the “Firmware Server
Path” field. Select TFTP or HTTP upgrade method. Update the change by clicking the “Update” button.
“Reboot” or power cycle the phone to update the new firmware.
During this stage, the LCD will display the firmware file downloading process. If a firmware upgrade fails
for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for
upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the
existing firmware/software.
Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet.
Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever
possible.
No Local TFTP Server
For users who do not have a 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
Support/Download section of our website to obtain this TFTP server IP address:
Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades.
A free Windows version TFTP server is available:
customerFree.cfm.
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Instructions for local TFTP Upgrade:
1. Unzip the file and put all of them under the root directory of the TFTP server.
2. The PC running the TFTP server and the BT200 should be in the same LAN segment.
3. Go to File -> Configure -> Security to change the TFTP server's default setting from
"Receive Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server, in the phone’s web configuration page
5. Configure the Firmware Server Path with the IP address of the PC
6. Update the change and reboot the unit
User can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS
web server.
NOTE:
•
When BT200 phone boots up, it will send TFTP or HTTP request to download configuration file
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the BT200 phone. This file is
for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following error
messages in a TFTP or HTTP server log can be ignored: “TFTP Error from [IP ADRESS]
requesting cfg000b82023dd4 : File does not exist. Configuration File Download”
CONFIGURATION FILE DOWNLOAD
The BT200 can be configured via Web Interface as well as via Configuration File through TFTP or HTTP.
“Config Server Path” is the TFTP or HTTP server path for the configuration file. It needs to be set to a
valid URL, either in FQDN or IP address format.
A configuration parameter is associated with each particular field in the web configuration page.
A
parameter consists of a Capital letter P and 2 to 4 digit numeric numbers. i.e., P2 is associated with
“Admin Password” in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the
corresponding configuration template of the firmware.
Once the BT200 boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”,
where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configuration file
name should be in lower cases.
Managing Firmware and Configuration File Download
When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-
scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider
can manage and reduce the Firmware or Provisioning Server load at any given time.
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Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.
Please backup or print all the settings before you restoring factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
INSTRUCTIONS FOR RESTORATION:
Step 1: Press the MENU button for Key Pad Menu options and press the Up button to see “reset”.
Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following
mapping:
0-9: 0-9
A: 22 (press the “2” key twice, “A” will show on the LCD)
B: 222
C: 2222
D: 33 (press the “3” key twice, “D” will show on the LCD)
E: 333
F: 3333
Example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to
move the cursor or wait for 4 seconds to continue to key in another “2”.
Step 3: Press the MENU button again. If the MAC address is correct, the phone will reboot. Wait for
phone reboot and the LCD backlight finish flashing.
Grandstream Networks, Inc.
BT200 User Manual
Page 33 of 33
Firmware 1.1.5.15
Last Updated: 01/2008
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