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 Grandstream GXP-2020
* back lit 320x160 high resolution graphic LCD with multi-level gray scales;
* four XML programmable soft-keys;
* seven programmable hard keys;
* 5-way conferencing;
* 6 lines with indicators for the individual SIP account profiles;
* multi-line support of up to 13 call appearance lines with dual-color LED indicators;
* dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af);
* support NAT traversal (STUN, etc);
* support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), PPPoE, TFTP, NTP;
* Open Standards Compatible;
* multi-language support (MLS);
* DHCP (both client and server);
* client high fidelity wide band audio (G.722);
* secure central configuration with AES encryption;
* SRTP and TLS (pending) for privacy protection;
* server fail-over;
* XML phone book;
* wall-mount;
* three adjustable positioning angles;
* headset jack (2.5 mm and RJ11);
* full duplex speaker-phone with advanced audio echo cancellation;
1. Dedicated buttons 2. Line 1-6 - Line keys with LED, which can be configured to different SIP profiles 3. Graphical LCD screen 4. Softkeys 5. Message Waiting Indicator 6. "Voice Message" button 7. "Menu" Key - Enter Keypad Configuration "MENU" mode when phone is IDLE, and stands for "ENTER" button. 8. "Phone book" button 9. "Hold" Button 10. "Hand" Free Button 11. "Send" Button 12. Standard Keypad 13. Speed Dial Buttons (Programmable Buttons)
To access the Keypad Configuration Menu, press the round "MENU" button and navigate by using the arrow keys: "up", "down", "left" and "right". Confirm a menu selection with "MENU" button again and choose the "OK" button to confirm changes or delete entry by pressing the "MUTE" button. Exit menus and the MENU mode by pressing "left" arrow key.
2. Connecting
All the connectors of the GXP-2020 are located on its bottom.
1. Ext Connection port - Connects directly to the GXP,drawing power from PoE if provided by network; 2. 10/100Mbps RJ-45 port - WAN (PC) connection; 3. 10/100Mbps RJ-45 port - WAN/LAN connection (switched or routed), draws power from spare line or signal line; 4. 5V DC power port - UL Certified Power Jack 5. Headset/handset jacks - RJ11/2.5mm Headset Jack
You may connect your phone in the following order:
1. Plug the first end of the Ethernet cable provided in the GXP-2020 package 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. The other end must be plugged 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.
WARNING : Please, avoid using any other AC adapters but this one, provided with your GXP-2020 !
3. Setup
If there is a DHCP server running in your network, when started for first time your GXP-2020 will get an IP address automatically from the DHCP server - the display will show the Grandstream logo, under which you can see the network status or the current IP address.
If there is no Internet cable plugged in, it will read "NETWORK DOWN". Other messages such as "DO NOT DISTURB" or "## MISSED CALLS" are shown there too.
If there is not a DHCP server in your network, you will have to configure the network settings manually.
1. Pick an unused IP address from your network and make sure that you know all the other stuff like Netmask and Gateway. 2. Press the round "MENU" button to enter the "MENU" mode. 3. Navigate by using the "down" arrow key to "Config" and press "MENU" button again. 4. Select "Network" and press the "MENU" button for third time.
You will see the Network Configuration center. Please enter an IP address, Netmask and Gateway here, using the "MENU" button to navigate between fields and the "up"/"down" arrow keys or the number keys to enter numbers. The "*" symbol stands for ".".
NOTE: If the IP address you choose looks like 10.1.0.106, you should enter 010.001.000.106 in the "IP" field. Same with the "Netmask" and "Gateway" fields.
NOTE: Do not forget to uncheck the DHCP option if there is no DHCP server in your network!
Press the "OK" button to confirm the changes, the "left" arrow key to exit the "Config" menu and then use the "down" arrow key to select "Reboot".
NOTE: The phone automatically exits MENU mode with an incoming call, the phone is off-hook or ifleft idle for 20 seconds.
When your phone boots, it is time to configure its SIP settings. GXP-2020 can be configured in two ways - by the Key Pad Configuration Menu on the phone or through the embedded web-configuration menu.
Using the Keypad:
1. Press the round "MENU" button to enter the "MENU" mode. 2. Navigate by using the "down" arrow key to "Config" and press "MENU" button again. 3. Select "SIP" by using the arrow keys and press the "MENU" button for third time. 4. Select "SIP User ID" by using the arrow keys and press the "MENU" button. Enter your User ID, using the number keys to enter a number, then modify it by positioning the cursor BEFORE the number and pressing the "up" or "down" arrow keys and select "OK" to confirm changes. 5. Select "SIP Auth ID" by using the arrow keys and press the "MENU" button. Enter it, using the number keys to enter a number, then modify it by positioning the cursor BEFORE the number and pressing the "up" or "down" arrow keys and select "OK" to confirm changes. 6. Select "SIP Passwd" by using the arrow keys and press the "MENU" button. Enter it, using the number keys to enter a number, then modify it by positioning the cursor BEFORE the number and pressing the "up" or "down" arrow keys and select "OK" to confirm changes. 7. Select "Sip Proxy" by using the arrow keys and press the "MENU" button to confirm. Enter your SIP server's details here. 8. Select "Save" by using the arrow keys and press the "MENU" button to confirm. 9. Press the "left" arrow key to enter the main menu and select "Reboot" so the changes will take effect after your GXP-2020 is rebooted.
NOTE: "SIP User ID" and the "SIPAuth ID" are case sensitive!
As this is a very complicated, time consuming procedure to enter all user details using the keypad, you'd better use the faster Web Configuration Menu.
Configuration with Web browser.
The GXP-2020's embedded Web server responds to HTTP/HTTPS GET/POST requests. You can configure the IP phone through your favorite Web browser.
The Web Configuration Menu can be accessed by typing your phone's IP address in the web browser of a computer in the same network.
For example, if your phone's IP address is 10.1.0.106, you should type in your web browser: http://10.1.0.106
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.
NOTE:The default factory password for the end-user is "123".The default factory password for the administrator is "admin". They can be changed anytime through the Web Configuration Menu.
You will see a login form. Enter the end-user or the administrator's password and login. As the end-user's account is limited to see only the "STATUS" and "BASIC SETTINGS" pages, enter the administrator's password.
There are nine pages : "STATUS", "BASIC SETTINGS", "ADVANCED SETTINGS"and"ACCOUNT 1" to "ACCOUNT 6"
STATUS" and "BASIC SETTINGS" are almost the same for the end-user and the administrator.
The end-user can modify date and time settings, multi-purpose keys and LCD settings.
STATUS
At "STATUS" you can see the current status of your device.
The same you can see if you press its round "MENU" button to enter the "MENU" mode, navigate by using the "down" arrow key to "Status" and press "MENU" button again.
"MAC Address" - shows your device's MAC Address;
"IP Address" - shows your device's IP Address;
"Product Model " - shows its model;
"'Software Version" - shows its software and boot loader version, as they can be upgraded;
"System Up Time" - shows system up time since the last reboot;
"Registered" - shows user accounts 1-6 status;
"PPPoE Link Up" - shows the status of your PPPoE Link;
BASIC SETTINGS
You are allowed to change the End User Password, IP Address and the other network settings, adjust multi-purpose keys and LCD settings, the timezone settings, etc. here.
WARNING: Passwords are case sensitive with a maximum length of 25 characters
ADVANCED SETTINGS
Here you can change the Admin Password, codec settings, NAT and STUN configuration, upgrade settings, XML configuration settings, ring-tones, call waiting, language and etc. settings of your GXP-2020.
ACCOUNT 1 - ACCOUNT 6
This is the place where you can register these users, who will use the phone.
"Account Active" - mark the corresponding field to activate/deactivate the account; "Account Name" - enter account name here. After reboot it will be displayed on the LCD.
"SIP Server " - here you must type the address of the Asterisk server you want to use. I am using 10.2.2.68.
"Outbound Proxy" - enter it here if you are using one;
"SIP User ID" - it can be different from the Authenticate ID;
"Authenticate ID" - this is the ID which your user uses to login.
"Authenticate Password" - this is the password which your user uses to login.
"Name" - enter his/her name here. "office manager" or "Jane Smith" ?
NOTE: Authenticate ID and Authenticate Password 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", or that account will not login automatically.
Here you can also modify DNS SRV, some SIP settings, NAT, STUN, proxy and DTMF settings, dial-plan, VoCoder and etc. settings. They can be unique for each account.
NOTE: Please do not forget to press the "UPDATE" button on the bottom of the page and reboot the phone to have the changes take effect.
4. Setting up an Asterisk account to be used with your GXP-2020 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. Now, use your favorite text editor to make some changes. First edit 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:
I added the following to my sip.conf :
[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 only use "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" ;).
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
To execute shell command without exiting Asterisk, type "!", then press the [ENTER] key.
Check if there are any sample sounds in your /var/lib/asterisk/sounds/:
#ls /var/lib/asterisk/sounds/
If everything is all right, you will see a long list of files.
I picked up "Hello World.gsm" for this example.
Let us make a new extension in our context.
exten => 111,1,Playback(hello-world) means that if someone from the "conf" context dials 111, he will hear this file played.
Save your extensions.conf file, and press CTRL+D to return to the Asterisk CLI console.
Reload extensions now: # extensions reload
and exit Asterisk CLI console by typing:
# exit
Now you can dial 111 to test if you will hear the "Hello World" greeting. :)
NOTE: ALL characters after ";" are interpreted by Asterisk as a comment !
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 and 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.
Now you can make a call from the hardphone to Zoiper.
6. Factory reset
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone, so it's a good idea to have them stored in a file or printed, before restoring factory default settings.
Disconnect all network cables before resetting factory default settings.
1. Press "OK" key to bring up the key pad configuration UI menu, select "Config", press "OK" to enter the submenu, select "Factory Reset"
2. Look at the 12-digit MAC address printed on the bottom of the phone and encode it using this mapping:
0-9: 0-9 A: 22 (press "2" twice and "A" will show on the LCD) B: 222 C: 2222 D: 33 E: 333 F: 3333
Example: if the MAC address is 000b8200e395, it should be encoded as "0002228200333395".
3. Press "OK" again to move the cursor to "OK" button. Then press "OK" key again to confirm. If you've entered the correct MAC address, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.
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