* 3-way conferencing;
* full-duplex hands-free speakerphone;
* voicemail with indicator for new messages;
* custom ring tones;
* advanced SRTP/TLS security features.
BudgeTone-200 happily supports popular codecs including G711 (a-law and u-law), G.723.1 (6.3K), G.729A/B and GSM, Dynamic negotiation of codec and voice payload length, as well as SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP/SNTP, TFTP are supported too.
NOTE: To make calls through Asterisk PBX, you must have previously installed and working Asterisk.
Please do not forget to configure it and register some users before setting up GrandstreamBudgeTone-200.
You may connect your BudgeTone-200 by simply plugging the ends of the Ethernet cable provided in the BudgeTone package in the following order:
1. Plug the first end to your PC LAN card and the second - to the "PC"port on the phone; 2. Having the end of the Internet cable free - plug it to the phone's"LAN"port. 3. Plug the other end where your Internet is coming from. 4. Plug the provided in the package AC adapter in its jack and connect it to the electricity network.
NOTE: You can use either straight through or twist cable, because BudgeTone-200 is able to auto detect Ethernet polarity.
WARNING: Please, avoid using any other AC adapters but this one, provided with your BudgeTone-200!
3. Setup
BudgeTone-200 can be configured directly by the keypad.
It also has an embedded Web server that responds to the HTTP GET/POST requests and embedded HTML pages that allow one to configure the hardphone by a web browser.
In this tutorial we will discuss the second way - configuring BudgeTone-200 using the web interface.
Every single setting you can make to the phone by the keypad interface can also be made from the Web configuration Menu.
Depending on your network and especially whether there is a DHCP server running or not, when started for first time your BudgeTone-200 will prompt for an IP address - the display will show message, saying "No IP".
NOTE: If you use static network configuration in your network, please skip to the next paragraph.
3.1. Configuring using DHCP
If you know that there is a DHCP server, check if the DHCP client of your hardphone is turned on by pressing the "Menu" button once.
You will see "[1]DHCP ON" or "[1]DHCP OFF".
If it is "OFF" - press the "Menu" button again and then the "Down" button once. Now it will show "- DHCP On".
When ready - press the "Menu" button again to confirm changes.
Now please restart your hardphone and wait until it receives a DHCP response.
When it happens the display will stop showing "No IP"and you will see date and time instead.
You are ready to access the Web Configuration Menu.
First - let's find out what IP address your BudgeTone-200 got.
When the phone is on-hook, press the "Menu" button and then press the "Down" button until you see "[2]IP Addr" and then press the "Menu" button again. You will see something like "010.001.000.106"; (this is only an example!)
NOTE: If the IP address displayed looks like 010.001.000.106 you will have to strip the leading "0" when typing the IP address into your Web Browser.
The IP Phone Web Configuration Menu can be accessed by the following URI: http://10.1.0.106
3.2. STATIC network configuration.
Even there is not a DHCP server present in your network, you may choose some unused IP address from your network so to be able to administrate the hardphone from the Web configuration site.
To enter an IP address, Subnetmask, Router /Gateway/ manually, you have to follow these simple steps:
NOTE: When entering an IP address, Subnetmask, DNS or Router, /Gateway/ please keep in mind that you just have to type the three octets of the IP address together, without separating them with dots.
3.2.1. DHCP:
When the phone is on-hook, press the "Menu" button once. You will see "[1]DHCP OFF" or "[1]DHCP ON".
If it is "ON" - press the "Menu" button again and then the "Down" button once.
Now it will show " - DHCP OFF".
When done - press the "Menu" button again to confirm changes.
3.2.2. IP Address:
When the phone is on-hook, press "Menu" button and then press "Down" button once until you see"[2]IP Addr" and then press the "Menu" button again.
You will see some IP address present. You can overwrite it by using the [0-9]Number keys.
When done, press the "Menu" button again to confirm the changes.
3.2.3. Subnetmask:
When the phone is on-hook, press "Menu" button and then press "Down" button two times until you see "[3]Subnet" and then press the "Menu" button again.
You will see some address present. You can overwrite it by using the [0-9]Number keys.
When done press the "Menu" button again to confirm the changes.
3.2.4 Router:
When the phone is on-hook, press "Menu" button and then press "Down" button three times until you see "[4]Subnet" and then press the "Menu" button again. You will see some address present. You can overwrite it by using the [0-9]Number keys.
When done press the "Menu" button again to confirm the changes.
3.2.5. DNS:
When the phone is on-hook, press "Menu" button and then press "Down" button four times until you see "[5]DNS" and then press the "Menu" button again.
You will see some address present. You can overwrite it by using the [0-9]Number keys.
When done press the"Menu" button again to confirm the changes.
Now your BudgeTone-200 will act as a regular device in your network.
3.3. Web Administration Menu.
Open your favorite Web Browser on a PC that is in the same network and type the IP address you've just set up.
You will see the login screen of the Web Administration Menu.
Default factory password for End User is "123".
Default factory password for Advanced User is "admin".
NOTE: You can always change them, but please keep in mind that they are case sensitive and must have maximum length of 25 characters.
The Web Administration Menu has four sub-menus: "STATUS", "BASIC SETTINGS", "ADVANCED SETTINGS"and "ACCOUNT".
* The last two are not accessible from the End User account.
3.3.1. End User Configuration.
Enter the Default factory password for End User ("123") to access the Configuration page.
3.3.1.1. STATUS
You can easy monitor the status of your device here.
MAC Address: shows your device's current MAC address. IP Address: shows your device's current IP Address. Product Model: shows your device's product model. Software Version: shows your device's software version, as it can be upgraded. System Up Time: shows system up time since the last reboot. Registered:shows whether your device has registered to the SIP server with your account or not. PPPoE Link Up: shows your device's PPPoE connection status if you use a DSL modem. detected NAT type: shows what kind of NAT is the BudgeTone-200 is connected to via its LAN port.
3.3.1.2. The BASIC SETTINGS page
It allows you to change the End User Password, IP Address and the other network settings, to adjust the timezone settings, device mode, NAT configuration and etc. of your BudgeTone-200.
You can even change its MAC address ;)
3.3.2. The Advanced User Configuration.
3.3.2.1 The Advanced User Configuration.
The Advanced Settings page can be accessed via the login screen. This time you must type the Advanced User password, which is "admin".
Here you can change the Advanced User password, the firmware update server, etc.
3.3.2.2. The ACCOUNT page.
This is the place where you can register these users, who will use the phone.
Sip Server:here you must type the address of the Asterisk server you want to use. I am using 172.6.10.1; Username and Authenticate Password: these are the same as these in the sip.conf file on the Asterisk server.
NOTE: Please, do not forget to check "YES" for "Sip Registration".
Here you can also change the SIP and NAT settings to access the server, DTMF and VoCoder settings. The BudgeTone-200 supports up to 5 different Vocoder types including G.711 A-/U-law, GSM, G.723.1, G.729A/B.
3.3.3 Saving Changes
Once you have made all the changes you needed, please click the "Update" button in the Configuration Menu. Then you will be asked to Reboot the device, so activate all the changes.
4. Setting up an Asterisk account to be used with the Grandstream BudgeTone-200 hardphone.
4. Setting up an Asterisk account to be used with the GrandstreamBudgeTone-200 hardphone.
NOTE: If you are an advanced Asterisk user, please skip the following paragraph about setting up an Asterisk account to use with your phone.
To set up a working SIP account on your Asterisk server, you have to edit some .conf files.
Please, connect using your mostly preferred way to your Asterisk server and find where the sip.conf and extensions.conf files are located. They are usually located at the /etc/asterisk/ directory.
4.1. Adding a SIP account.
Now, use your favorite text editor to make some changes in the sip.conf file.
Adding a new user is a simple process, which consists of adding some lines, similar to these in our example here:
NOTE: ALL characters after ";" are interpreted by Asterisk as a comment !
I added to my sip.conf the following:
[user]: needed to indicate we are registering user "user". username=user: to determine our user's name is "user". context=conf: defines the dial context for the user "user". Asterisk divides outgoing numbers in groups called contexts in order to separate/define different needs for different uses. In this example the user "user" is in a context called "conf". type=friend: Users that can place AND receive calls are described as "friend". If your new user will only receive calls, use "peer" as type. For placing calls onlyuse "user" as type. secret=123: this is where you set up your user's password. He will use it to login/authenticate on Asterisk. It is presented in plain text, bus as only the server's administrators (you) have access to this file, there is nothing to worry about. allow=all: means that the line which this user will use supports all audio codecs. host=dynamic: You can define your users IP as static or dynamic, according to the type of network. rfc2833compensate=yes: You must have this turned on, or DTMF reception will work improperly. nat=yes: If you are placing the phone behind a NAT you must enable that.
Save the changes and continue to register an extension for the user "user" ;).
4.2 Making extensions
Now please open the file extensions.conf in your favorite editor to make some changes:
In my file I added:
[conf]: it is the name of the new context.
The first line "exten => 731,1,Dial(SIP/user)" shows that if
somebody dials the number 731 his call will be connected with the user "user" through SIP.
The second line "exten => 731,2,Hangup()" says that if the
conversation is over then Asterisk has to hang up the line. It is needed to be sure that Asterisk will free it for the next call.
NOTE: Do not forget to save the changes you just made in extensions.conf.
As your new user "user" is already configured, please start your Asterisk server:
# asterisk
enter the Asterisk CLI console by typing:
# asterisk -r
and reload SIP for the changes to take effect by using:
# sip reload
then reload the extensions:
# extensions reload
now you can check what you've done by:
# sip show peers
and exit Asterisk CLI console by typing:
# exit
TESTS
To know that everything is all right you can test it with an echo test, or by installing a softphone.
5.1 Echo test
To do an echo test you need to add few more lines in your extensions.conf file.
Open it with your favorite editor and edit it following the example:
exten => 111,1,Answer(): that means that the call will be answered; exten => 111,2,Echo(): makes possible for you to hear your own voice; exten => 111,3,HangUp(): hangs up after you've finished the call.
Now, if you dial 111 from your BudgeTone-200and say something you will hear your own voice.
5.2 Testing by a softphone
To make a test call with a softphone you need one.
Download Zoiper and test it for free from the download section on our website!
5.2.1 Installation
Zoiper on Linux
If you are using Linux, please follow these steps which will lead you to a working Zoiper phone.
1. Download the appropriate archive from our website;
2. choose a destination directory and extract it there;
3. open that directory and click the newly extracted file Zoiper with your mouse to run it. You can also open a terminal in the active X-session, navigate to the target directory by using cd and execute the file by
# ./zoiper
Zoiper on Windows
If you are using Windows, please follow these steps which will lead you to a working Zoiper phone.
1. Download the appropriate archive from our website;
2. run the newly downloaded .exe file;
3. you will see Zoiper's installation wizard. You will be expected to read a License Agreement, and maybe - agree it.
4. If you agree it, please click on the appropriate button to proceed to the next screen, to choose a destination directory for your Zoiper. You can see some information about space required by Zoiper and the available free space on your hard drive.
5. The next screen is an opportunity to choose where do you want your Zoiper Start Menu shortcuts.
6. Once again you will deal with shortcuts – this time some additional ones – for your Desktop and Quick Launch. If you do not want them, just unmark the appropriate checkboxes.
7. When you click the “Install” button the real installation will begin. You can click the “Details button to see some details, but it will finish so fast that you'd hardly have enough time.
8. The next screen is the last one. It will inform you that your Zoiper installation had finshed successfully.
Congratulations ! You can start using it now !.
5.2.2. Setting an additional account.
Please connect/open a terminal session to your Asterisk server in order to modify your /etc/asterisk/sip.conf file and add one more user account. It will be needed to use with Zoiper.
You can copy the “user” account, and modify it a little.
[user]:change it to [user2] to determine that we are setting an account for “user2” username=user: change it too -> username=user2to determine our user's name is now "user2". context=conf: defines the dial context for the user "user2". You should leave it as it is, as “user” and "user2" can be in the same context. type=friend: leave as it is, as you will need both of your users to place and receive calls when testing. secret=123: you may use the same password or change it. allow=all: you do not need to change that. host=dynamic: you can either enter the IP address of the computer you've just installed Zoiper on or leave this as it is. rfc2833compensate=yes: You must have this turned on, or DTMF reception will work improperly. nat=yes: If you are placing the phone behind a NAT you must enable that.
Now save this file and open /etc/extensions.conf to add a new extension for that user.
5.2.3. Setting additional extensions.
You added 3 lines for your first account. Now add the new ones below:
As adding user 2 to the existing context, you do not need to specify a new one.
Just add these two lines:
exten => 732,1,Dial(SIP/user2) shows that if somebody dials the number 732 his call will be connected with the user "user2" through SIP. exten => 732,2,Hangup(): it is needed to be sure that when the conversation is over then Asterisk has to hang up the line. It is needed to be sure that Asterisk will free it for the next call.
5.2.4 Setting up Zoiper
To use your new Zoiper softphone you need an account and you have to set it up.
This you can do following my steps:
Accessing Options Form
Start your Zoiper and right-click on its interface. Click on "Options" to open the configuration menu.
There is an Options button on the Zoiper’s interface. You can start configuration form from there too.
Use "Alt+O" if you prefer using keyboard than clicking.
Adding SIP account
When the form starts you should click on the "Add new SIP account" label in the navigation menu to the left.
A new sub-form will appear. Enter a name the account and click on the "OK" button to create it.
The new entry will appear in the navigation menu to the left.
Click on it to continue the configuration.
On the right side of the navigation menu you can specify some options for the account.
Enter the hostname or the IP address of the Asterisk server (or other VoIP server) that you are going to use.
Enter the account name and secret that you put in the sip.conf file (user2 & 012). Enter a caller id and name and click on the "Apply" button to activate the changes.
There are additional options that you might want to configure. In order to view the advanced options for the current account you should enable the checkbox with label "Show advanced options" which is located on the bottom left corner.
When you do this you can set an outbound proxy, if necessary, or a voicemail extension, a STUN server, if needed. You can select if Zoiper should register this account upon application’s start-up.
When you are ready click on the "Apply" button.
Registering SIP accounts
You should select the proper account from the drop down menu in the bottom of Zoiper’s interface. When you are ready with this click on the "Register" button.
As Zoiper supports IAX accounts, you can read the Zoiper2 tutorial and find out how to set up and use an IAX account.
You can call your Zoiper from the hardphone.
6. Factory reset
WARNING: Please backup all your settings before resetting your BudgeTone-200.
6.1. Find the MAC Address of the device.
It is a12-digit number and is located on the bottom of the device.
Or, when the phone is on-hook, press "Menu" button and then press "Down" button eight times until you see "[9]Phy Addr" and then press the "Menu" button again. You will see a 12-digit number.
6.2. Encode the MAC address.
Please use the following mapping:
If the MAC address is 000b8200e395, it should be encoded as "0002228200333395".
6.3. RESET
Press the "Menu"button once, than press the "Up"button once and you will see "-- reset --" on the phone display.
6.4. Enter the encoded MAC address, then press the "Menu" button. Your GrandstreamBudgeTone-200 will reboot automatically and restore to factory default setting.
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Handey (mosehandey at gmail dot com) 24 November 2016 12:58:31 Thank you. This solved my problem.
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Shawnboy (raincoat4me at yahoo dot co dot in) 07 October 2008 09:24:26 Thank U for the procedure on settingup an budge tone 200, its very helpful.
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