similar to: gxp-2000 tftp cfg

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

I own and operate a number of franchised Sylvan Learning Centers where I recently upgraded to an all VOIP phone system (Asterisk) with one VS-1 and about 25 extensions scattered around the country. I had originally setup a Dell 420 SC but the Dell had incurable buss issues with single span and quad span T1 cards. I wasted a LOT of time trying to get the Dell to work and finally gave up on

Hi, i got my Grandstream GXP-2000 phone today and want to configure it with TFTP. I downloaded the firmware 1.1.0.13 and put it into my tftp-server directory. Then I downloaded the template from: http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_Unix/Grandstream_Configuration_File_Template_1.0.6.x.txt renamed it to cfg<mac-address> Did the configuration in the new file and

Mitel ICP 3300 & Asterisk, Is possible that integration? (C?sar) -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Jueves, 25 de Mayo de 2006 03:00 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 22, Issue 147 Send

Any reviews/comments out there on this server? Looks solid.. But would like to know if anyone has purchased one of these before. Any other companies out there offer pre-built * servers that someone would like to comment on? http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_As terisk_Voice_Server.html -------------- next part -------------- An HTML attachment was scrubbed...

Hello All, I am going to try and install Asterisk on a server machine that I have and wanted to get the opinion of experienced people as to the best method. Initially I was going to purchase the Asterisk Starter Kit at: http://www.thevoipconnection.com/store/catalog/product_16215_Asterisk_Starter_Kit.html But from what I can find out in the reviews, the book that comes with the kit is not

Hey All, I have a user whose Cisco 7960 died the other day. When you disconnect the Ethernet cable the phone boots (as far as it can w/o network connectivity) but as soon as you plug in the CAT5 it goes dead. (no lights, no sound, no display, nothing) I tried to contact Cisco but they say I have to contact the vendor. I have sent an email to the vendor (www.thevoipconnection.com) and I am

Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff.

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

Can anyone here have experience using G.722 on the Grandstream GXP-1200? It's supposed to support the codec, but I wonder if the handset does it justice? The older BT-200 also supported the codec, but the handset was not good enough. You could only hear the improved call quality using a headset. Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at

Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?????? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soooooo where is it???? Any light on this?

Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? Thanks Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org

I've finally got a chance to play with 1.0.1.9, and the wiki has been updated. At the moment, I don't know of _any_ bugs with it. I'm yet to play with complex things like early dial, and will update the wiki as I find information. http://aussievoip.com.au/wiki-GXP-2000 I've just been given official permission to offer the firmware as a download, so if you visit

Greetings all I was wondering if any of you have found this problem. I have a setup with 80 GXP-2000s which work great until I decide to stop now my asterisk server. When it comes back up half of the phones (not always the same ones) fail to register with Asterisk. I set up the phones to register and send keep alives (register every 4 minutes, send keep alives every 20 seconds) to no avail.

Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press "Send") Thanks, -- "Computers are useless. They can only give answers." - Pablo Picasso

>From the usual place, http://www.grandstream.com/BETATEST/GXP2000/ Note, there are two (and it took me a bit of a while to figure out) images to be loaded. Copy the ...a.bin's and the .bin's to your http provisioning directory, and reboot. The phone _must_ load the .bin files before it understands the ..a.bin files. After it loads the first one, the phone does lock up with the

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

Does anyone know the secret to get the GXP-2000 Message waiting lamp to illuminate? Or can point me toward some docs that might explain it? Thanks! --Jeffrey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060501/7ffc668f/attachment.htm

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

How can you store "pauses" in speed dials for the GXP-2000? I used something like 8005551212,,,1,7890 to dial the toll free number, wait 6 seconds (I'm used to the commas being a 2 second delay), pressing 1, waiting 2 more seconds and then entering 7890. However, when I press the speeddial button, the phone freezes. Thanks, Waldo