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
Romanfaums (interesuru at i dot ua)
28 December 2017 13:00:28
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18 April 2016 04:54:27
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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
configration passward after config password tracking
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: 126.96.36.199.
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 188.8.131.52, 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...
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-- 184.108.40.206
(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!
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 220.127.116.11, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <email@example.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 18.104.22.168 currently running on mf-asterisk-1 (pid = 13445)
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 22.214.171.124 D 5074 OK (5 ms)
matthew/matthew 126.96.36.199 D 5066 OK (13 ms)
gs/gs 188.8.131.52 D 5060 OK (5 ms)
3 sip peers [3 online , 0 offline]
Shiju V.Joseph (devadathan at gmail dot com)
13 July 2006 12:31:37
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
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.
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?
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.
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 184.108.40.206 (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 220.127.116.11 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 18.104.22.168 : call directly start on push.
firmware 22.214.171.124 : first take line (or push on SPEAKEER) then push speed dial ... (i prefer 126.96.36.199 fonctionnality :-(