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3.2.3. Grandstream GXP-2000 SIP hardphone

PREREQUISITIVES
Before going further reading this tutorial make sure you have asterisk server and you are familiar with adding users to asterisk. If you are not aware of this issue you can read the tutorial explaining how to add new users to asterisk.


CONNECTING

To connect the phone you need two Ethernet and one power-supply cable. One of the Ethernet cables has to connect your PC with the IP phone (plug it in the pc labeled input at the back of the phone). Now your Internet LAN connection should be connected to the LAN intput of the phone using the second cable. Plug in the power supply and the phone is ready for use.


ADJUSTING
This is an IP hardphone that uses SIP protocol to register users. Here is how the phone looks:



You can adjust the phone in two ways:

Using the phones interface
Using the phones web interface

Using the phones interface

By clicking the Menu button you can see the menu items that can be adjusted.

1. DHCP Mode
Available options are: enabled/disabled

2. IP Address
IP address of the phone default is 192.168.1.14

3. Subnet Mask
Phones subnet mask

4. Default Gateway
IP address for the gateway

5. DNS Server
IP address for the DNS server

6. TFTP Server
IP address for the TFTP server

7. Audio Codec (default is G.711u)
Available codecs are: G.711u, G.711A, G.722, G.723, G.726, G.728, G.729 and iLBC

8. SIP server
SIP server

9. Firmware Version
Date and version for the software packets the phone is using

10. MAC Address
MAC address of the phone

11. Ring Tone
Ringing tones that are available Ring 0, Ring 1, Ring 2 and Ring 3

12. Ethernet - LB test
Shows the network status

13. Audio LB
Available options are: enabled/disabled

14. Diagnostic Mode
Diagnostic mode shows every action you do with the phone

15. RESET

The scheme used to type the most common characters is the following:

Input                         Encoding   
00                              0                   
01 				1
02				2
03				3
04		 		4
05 				5
06 				6
07 				7
08 				8
09 				9
*0 				. (dot character)
*1 				_ (underscore character)
*2 				- (hyphen character)
*3 				@
*4 				: (column character)
21 				A
22 				B	
23 				C
31 				D
32 				E
33 				F
41 				G
42 				H
43 				I
51 				J
52 				K
53 				L
61 				M
62 				N
63 				O
71 				P
72 				Q
73 				R
74 				S
81 				T
82 				U
83 				V
91 				W
92 				X
93 				Y
94 				Z

Note: This scheme is according to the Grandstream GPX-2000 IP phone providers, but for me it was not working even with the most recent firmware. So according to me it is better to use the phones web interface.

Using the phones web interface

I used the web interface to adjust the phone. Look the IP address of your phone from the menu. If it is not a valid IP address for your local network enable the DHCP and reboot the phone. This way the DHCP will give the phone an IP address which can be accessed within your local network. Type in your browser http://Phones-IP, where Phones-IP is the IP address of your phone.

This will appear in the browser:


The password for regular end users is 123 and for administrators it is admin.
Here is the main menu. As you can see you can have up to four accounts registered.

Every change you make to the phone will take effect after you click update and after you restart your phone.

1. Status

Here you can see some main features concerning the phone. Model, accounts registered and network settings.

2. Basic Settings

I prefer to set my IP address statically, so I set IP address, subnet mask and DNS server for the phone. I will also change the time zone and leave the speed dial settings untouched this feature allows you to set speed dial buttons.

3. Advanced Settings

In this menu you can change the pasword for administrators (the default one is 'admin') and the audio codec. If you want to upgrade the firmware version via TFTP server type the IP address of the TFTP server that you are going to use in firmware upgrade. I will explain later on how to do this. You can also choose another timeserver to synchronize the date and time for your phone.

4. Acount1



In order to add a user you have to register it in asterisk. The path is /etc/asterisk and the file name is sip.conf. Then you have to set it on the phone's web interface too. Let us set the user here and then we will set it on asterisk.

Make sure you check yes for account active otherwise even if your settings are correct the account will be inactive. My asterisk server (SIP server) is 10.3.3.34. I register Grandstream user with the same password. You can have a look at all the other settings and change them if you want but the default settings are good enough if you are unfamiliar with what you are doing.

When you update some changes you see that the changes will take effect after restarting the phone:

So finally when you reboot it the changes are now updated.



REGISTERING USERS ON ASTERISK

We will now register the Grandstream user in sip.conf and another user in iax.conf and will create extensions for them.

Here we create a Grandstream user with Grandstream password. Type friend (can call and can be called), host is dynamic (the changes in the IP address of the phone will not matter) and add the sip user the test context.

Now I have a harphone adjusted and with registered user. I will register an iax user in iax.conf and so I can call from iax client softphone to the Grandstream hardphone.

Here I create user iedfisk that can call and can be called with password idefisk and dynamic host IP address and join it to the test context.

Here I create test context and create extension for each user. Now when 2222 is dialed user Grandstream will be contacted and 3333 will call the iax user idefisk.

For more information about how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk

Note: Remember to reload the asterisk if you want the changes you made to take effect.

Now you can call from the iax client for example Idefisk (you can download it from http://www.asteriskguru.com/ and have a look at the tutorial there) to the Grandstream harphone by just dialing 2222. And you can dial Idefisk from the hardphone by dialing 3333. You can add more users in the way shown above.

More information for the Grandstream hardphone can be found on their site http://www.grandstream.com/.

UPGRADING FIRMWARE VIA TFTP

You can upgrade the software of your phone using tftp server. For this purpose you have to download and install TFTP server on your computer. You can use Solar Winds for example it is a good TFTP server. You can download it from http://www.solarwinds.net/ for free. After installing it looks like this:

You have to download the firmware and store it in the TFTP-Root directory in our case it is C:/TFTP-Root. You can have the latest firmware version usually from the phones producer http://www.grandstream.com/y-firmware.htm
Download the firmware and unachieved it in your TFTP root directory.

Change the TFTP IP address from the web interface to the IP address where your TFTP server is installed and reboot the phone. Before booting the phone will get the new firmware from the TFTP server and will upgrade itself.

 
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ryan (ryanrockin at gmail dot com)
06 January 2009 11:59:15
what does the config file path do in web interface settings in the advanced
immi (imran_imee85 at yahoo dot com)
14 October 2008 21:17:31
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Benedek (nekeress at gmail dot com)
15 November 2007 16:43:18
I have several older GXP 2000s at my company. We just bought some new ones. Old ones work with NTP, new ones not. Yes, I tried fixed IP. And I used the same settings, even the same cable for both phones. Doesnt helps. Any advice?
maman (amann50 at yahoo dot com)
15 October 2007 16:21:03
i have the phone Gxp2000 but i want to put my pin or to register pin how can i do please thanks
SOHAIL (SOHAIL dot HASNAIN at AFTECH dot COM dot PK)
09 March 2007 12:05:16
I NEED TO RESET THE PHONE TO ITS FACTORY DEFAULT. USER HAS SET THE PASSWORD AND FORGOTTEN. pLEASE HELP US OUT AS IT IS URGENT.
Chris Ball (chris dot ball at smtnet dot co dot uk)
26 January 2007 15:26:47
I have just bought a Grandstream GXP2000, and am generally pleased with the performance of the phone.
I am having difficulty with the call parking feature (#70). The trouble is that the # key is ignored by the system and passed as a tone to the person calling.
I have tried the "Use # as Dial Key" setting in both yes and no, but with no change. I have upgraded the phone to Software Version: 1.1.2.25.

Does anyone have any idea how I can get this feature to work?

Paul Torres (ptorr at childshome dot org)
03 November 2006 20:07:08
We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Fedora Core 4, loaded on a Dell PowerEdge SC1425 Server... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to "hold" and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the "hold" in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on!!!!

We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Linux (Fedora Core) upgraded from 2 to 4 and the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any...

Thanks,
ptorr
Matthew Kleinmann (matthew_kleinmann at moldflow dot com)
27 October 2006 22:25:14
Some info that may be of some interest.

This is with Program-- 1.1.0.14
(Under Status -> Software Version)

I set up all four accounts to be identical, that is I used the same SIP authenticate ID, and the same SIP User ID, and the same SIP authenticate password on all four accounts.

As far as I can tell, the account name and SIP user ID need to be set the same for an account to register with Asterisk.

For the record: The Account name is put on the LCD in big bold letters and the SIP user ID is put in smaller letters below it. The Authentication ID does not show up at all, but the phone will not register unless it is set the same as the SIP USer ID.

As far as configuration ont he Asterisk end goes, The SIP USer ID and the Authentication ID is the name that corosponds to the name in brackets in the sip.comf file.

The Authentication Password corosponds to the secret for that phone in the sip.conf file.

You need to put in the IP or DNS locatable hostname of your Asterisk server in the SIP Server line.

You need to put in the IP or the DNS locatable hostname of your Aserisk servr followed by :5060 in the Outbound Proxy line (ie 192.168.0.17:5060)

Hint on the above two lines: Get it working with the IP addresses first. Much less to go wrong. Then once you have a phone working, try using hostnames and DNS.

You need to change the Send DTMF: from "In Audio" to Via RTP (RFC2833)

If you set all four lines with the same info you only need one entry in the Asterisk sip.conf file.

Here is a sample:

; grandstream gs2000
[matthew] ; Set this to the SIP USer ID and Authenticate ID on the phone!
type=friend
secret=welcome ; Set ths to the Authenticate Password on the phone!
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do

Reboot the phone, restart Asterisk and see if it shows up:

[root@mf-asterisk-1 asterisk]# /etc/init.d/asterisk restart
Shutting down asterisk: [ OK ]
Starting asterisk: [ OK ]
[root@mf-asterisk-1 asterisk]# asterisk -r
Asterisk 1.2.12.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.12.1 currently running on mf-asterisk-1 (pid = 13445)
UNIX connectionI>
Verbosity is at least 3
mf-asterisk-1*CLI> set verbose 1000
Verbosity was 3 and is now 1000
mf-asterisk-1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
699/699 192.61.4.98 D 5074 OK (5 ms)
matthew/matthew 192.61.4.63 D 5066 OK (13 ms)
gs/gs 192.61.4.19 D 5060 OK (5 ms)
3 sip peers [3 online , 0 offline]
mf-asterisk-1*CLI>
Shiju V.Joseph (devadathan at gmail dot com)
13 July 2006 12:31:37
Hi all

we are using Grandstream GXP 2000 phones with asterisk,they r working smoothly.Only difference i found apart from the configuration i found is that we are using send DTMF is via RTP (RFC2833).
it is working fine considering cost versus perfomance it is a better buy

Shiju V.Joseph
Network Engineer
Automation Experts
Mumbai
Pageus (pageus at gmail dot com)
20 April 2006 08:39:21
I don't see alot of activity on this thread, but i have a question before i go out and get a few of these. we are an office that is going all home based now creating 3 seperate locations. running aah2.8 with no problems whatsoever using voicepulse connect as my trunking agent. i will be adding in 2 digium fxs ports for local calls and the voicepulse for long distance. what i need to know is with the grandstream gxp-2000 phone, will i be able to make easy transfers (announced) like i can using good softphone packages.. this phone seems almost to good to be true with the price and features.
Any comments or suggestions would be helpful.
Thanks
Mike (webmaster at emmgee dot com)
12 April 2006 04:44:28
I am still having difficulties getting my GXP-2000 to talk with Asterisk@Home. Does anyone know a step-by-step tutorial for beginners that would specifically address the GXP-2000 and Asterisk@Home?

Thanks guys!
Mike (me at hotmail dot com)
15 February 2006 06:03:42
Chris, I dont see how what you added with make the phones operate in a Shared Line Mode. This wont allow you to pickup calls on hold on one phone from another.
Chris (CKURTIS at AOL dot COM)
15 February 2006 01:20:06
Mark, I had a similar problem. If you need to place the following lines in your sip_additional.conf setup thru config edit. incominglimit and outgoing limit.

I believe once you do that, the lines will "roll" over to each other.

100]
username=100
type=friend
secret=100
record_out=Always
record_in=Always
outgoinglimit=8
incominglimit=8
qualify=no
port=5060
pickupgroup=1
nat=never
mailbox=100@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callgroup=1
callerid=device <100>
Mark (mark dot shook1 at cox dot net)
07 February 2006 16:05:03
newbie question and i havn't found the how to on any wiki's yet. Using AAH2.5 and GPX2000 IP phone firmware 1.0.2.3 (BETA) how do i setup the (button) lines 1 -4 to map to my ZAP ports 1-4. I'm trying to make asterisk look like an hold phone system were you can just put someone on hold and go to another phone and press the line button (say line 3) and you would pickup the line that was on hold.

2nd note for Shogo. I've read somewere about some problems with windows DHCP being to slow on giving out an IP. The phone looks once to a NTP server and if it hadn't gotten an IP it won't try agian.
Try a static IP and see if that fixes it
Roman (roman at 121media dot com)
07 January 2006 01:27:53
I tried the latest 1.0.1.3 firmware and ilbc is nowhere to be found. Please change the documentation to reflect the unavailability of ilbc.
Shogo (shogonida at hotmail dot com)
31 December 2005 15:26:32
My GXP-2000 doesn't synchronize with any NTP server.
I read userguide and tried to set it up properly but it didn't work at all.
can anyone tell me how to set my phone's date and time please?
teji (a at a dot be)
02 November 2005 16:17:25
Speeddial touch :
firmware 1.0.1.2 : call directly start on push.
firmware 1.0.1.9 : first take line (or push on SPEAKEER) then push speed dial ... (i prefer 1.0.1.2 fonctionnality :-(
 
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