similar to: GXP-2000 and Configdownload via TFTP

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker

Hello, At a client site yesterday I installed a dozen GrandStream GXP-2000's with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX and phones: network access for users windoze PC's through the phone's switch port was unbearably slow, making it almost impossible to work. When plugging back PC's directly to the LAN speed was normal again. On my test setup

>hi, > >can you someone post tftp template for gxp-2000? >like >http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_ Configuration_File_Template_1.0.6.x.txt > >thanks > >--------------------------------------- >Marek Cervenka >======================================= It's here:

Hi Jose, Am 11.10.2017 um 13:13 schrieb Jose Celestino: >> Before I go down the route of upgrading our main Solr instance from 6 to 7, >> just checking if anyone is using Solr 7 with Dovecot. > SOLR 7 works ok with dovecot. We did not upgrade but indexed from scratch with > 7.0.0. > > From our experience: > > Make sure that you set the default queryResponseWriter to

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

On 22 January 2018 at 23:10, Matthias Fechner <idefix at fechner.net> wrote: > Dear Odhiambo, > > Am 22.01.2018 um 19:58 schrieb Odhiambo Washington: > > ...and I am still unable to successfully compile 2.3RC on FreeBSD 8.4 and > 9.3 > ....and my reports were ignored, so should I assume support for those has > been dropped? > > Support for FreeBSD 8.4 stopped

On 2/12/19 6:03 PM, Matthias Fechner via dovecot wrote: > Am 12.02.2019 um 17:05 schrieb Robert Moskowitz via dovecot: >> I have trying to find how to set the dovecot-sql.conf for using >> SHA256/512.? I am going to start clean with the stronger format, not >> migrate from the old MD5.? It seems all I need is: > you maybe would like to have a look to the hashing algo

On 2/13/19 1:23 AM, Matthias Fechner via dovecot wrote: > > > Am 13. Februar 2019 00:34:15 schrieb Robert Moskowitz > <rgm at htt-consult.com>: > >> On 2/12/19 6:03 PM, Matthias Fechner via dovecot wrote: >>> Am 12.02.2019 um 17:05 schrieb Robert Moskowitz via dovecot: >>>> I have trying to find how to set the dovecot-sql.conf for using

On 11 January 2018 at 14:29, Aki Tuomi <aki.tuomi at dovecot.fi> wrote: > > > On 11.01.2018 13:28, Hauke Fath wrote: > > On Thu, 11 Jan 2018 13:22:07 +0200, Aki Tuomi wrote: > >> Can you try if it works if you concatenate the cert and cert-chain > >> to single file? We'll start looking if this is misunderstanding or bug. > > This is a production

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>

On 2/13/19 8:30 AM, Aki Tuomi wrote: > On 13.2.2019 15.18, Robert Moskowitz via dovecot wrote: >> >> On 2/13/19 1:23 AM, Matthias Fechner via dovecot wrote: >>> >>> Am 13. Februar 2019 00:34:15 schrieb Robert Moskowitz >>> <rgm at htt-consult.com>: >>> >>>> On 2/12/19 6:03 PM, Matthias Fechner via dovecot wrote:

Hello, I'm using 0.99.14 with debian 2.6.8-2-386 #1 Tue Aug 16 12:46:35 UTC 2005 i686 GNU/Linux and exim4. I use the shared folder with symlinks, created by the script create_dovecot_shares-1.06. I have set the permissions and use umask 0007 on dovecot and in exim in maildir_home: directory_mode = 0770 mode = 0660 every thing works well, but if I'm moving a mail from my inbox

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, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the

Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani

I have about 30 GXP-2000 phones running 1.0.1.9 which have all been configured using the provisioning feature so the configuration is all identical. The problem I am having is that they randomly seem to stop registering with asterisk. When they stop registering they can still make calls but oviously asterisk cannot ring the phone so all incoming calls go to voicemail. Has anyone else had similar

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -------------- next part