Thomas Kenyon wrote:
Thomas Kenyon wrote:
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a
Thomas Kenyon wrote:
> I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
>
> Is anyone else getting the following error in the asterisk console:
>
> [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
>
> every couple of seconds when a handset is i
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a handset is in a call?
I didn't notice this hap
On Wed, Sep 20, 2006 at 01:31:47PM -0700, Christopher Corn wrote:
>Michiel van Baak <[EMAIL PROTECTED]> wrote:
>On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote:
>> On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
>> > michael,
>> > at my real job, the phones display p
hmm, no we have about 7000 employees.Michiel van Baak <[EMAIL PROTECTED]> wrote: On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote:> On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:> > michael,> > at my real job, the phones display peoples names when calling out from> > your phone. how is
On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote:
> On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
> > michael,
> > at my real job, the phones display peoples names when calling out from
> > your phone. how is this done?
Maybe they put the names in the phones internal addressbook
?
--
On Tue, Sep 19, 2006 at 10:33:28PM -0400, Jay R. Ashworth wrote:
> So, that wiki page not having answered my question: is there in fact
> enough power in the not-really-a-programming-language that is the
> dialplanning syntax anyway to capture the appropriate information in a
> "variable" so it can
On Wed, Sep 20, 2006 at 12:12:58AM +0200, Michael Neuhauser wrote:
> On Tue, 2006-09-19 at 16:46 -0500, Lacy Moore - Aspendora wrote:
> > It does matter if they are using Asterisk. I think we'd all like to
> > know how they are doing it.
> >
> > When I was using the chan_sccp driver, I was able
On Tue, Sep 19, 2006 at 04:46:13PM -0500, Lacy Moore - Aspendora wrote:
>On 9/19/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote:
> > What phones/system do they use?
> I'm not sure it matters.
>
>It does matter if
On Tue, 2006-09-19 at 16:46 -0500, Lacy Moore - Aspendora wrote:
> It does matter if they are using Asterisk. I think we'd all like to
> know how they are doing it.
>
> When I was using the chan_sccp driver, I was able to display that
> information on the screen. I'm not sure the SIP protocol i
On Tue, 2006-09-19 at 17:24 -0400, Jay R. Ashworth wrote:
> On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote:
> > What phones/system do they use?
>
> I'm not sure it matters.
>
> Many commercial key systems display the destination extension user's
> name on the originating set on inte
On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
> michael,
> at my real job, the phones display peoples names when calling out from
> your phone. how is this done?
And the use Asterisk there? SIP phones? Legacy PBX? Cisco Call Manager?
--
Dr. Michael Neuhauser mailto:[EMA
It does matter if they are using Asterisk. I think we'd all like to know how they are doing it.
When I was using the chan_sccp driver, I was able to display that information on the screen. I'm not sure the SIP protocol in Asterisk supports this (and don't know if the SIP protocol itself support
On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote:
> What phones/system do they use?
I'm not sure it matters.
Many commercial key systems display the destination extension user's
name on the originating set on internal/intercom calls, and some
display the destination user's name on pho
- Original Message -
From: Christopher Corn
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, September 19, 2006 4:48 PM
Subject: Re: [asterisk-users] grandstream gxp 2000 does not display
nameswhencalling out
im sorry if im not being clear. when im calling
im sorry if im not being clear. when im calling from my gxp to the grandstream 100.the gxp doesn't pullup the users name from grandstream 100. someone in another mail mentioned that this is not the way its supposted to work. my office does this, when i dial someones number, it displays thei
michael, at my real job, the phones display peoples names when calling out from your phone. how is this done?Michael Neuhauser <[EMAIL PROTECTED]> wrote: On Tue, 2006-09-19 at 11:53 -0700, Christopher Corn wrote:> ... i dial now from SIP phone to SIP phone. one being a grandstream> gxp >>> grands
- Original Message -
From: Christopher Corn
marco,
can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp >>> grandstream
100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe thi
On Tue, 2006-09-19 at 11:53 -0700, Christopher Corn wrote:
> ... i dial now from SIP phone to SIP phone. one being a grandstream
> gxp >>> grandstream 100. but the gxp, when dialing, doesn't see the
> name of the person at the grandstream 100. i believe this should be
> picked up from the asterisk
marco, can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp >>> grandstream 100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe this should be picked up from the asterisk server, but im no
test it with someextension and using a sipphone like xlite or something else and you will be able to understand if it is a grandstream issue!hope it helpsOn 9/19/06,
Christopher Corn <[EMAIL PROTECTED]> wrote:
i have trixbox running, the latest version and when i make an outgoing call from this ph
i have trixbox running, the latest version and when i make an outgoing call from this phone it doesn't pick up the user's name, but instead display the number. is this a grandstream problem? or asterisk? i did specify the user name from 'extension" within trixbox. in matter of face, if i call
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
l Discussion
> Subject: Re: [Asterisk-Users] Grandstream GXP-2000
>
>
> Make sure you have the DTMF mode set to RFC.
>
> On Fri, 2006-04-28 at 15:20, Johnny Stork wrote:
> > I seem to be having a problem with my GXP-2000. No matter
> how carefully I type in the mailbox
Your problem could be DTMF-related. Make sure that both your sip peer
has dtmfmode=rfc2833 and the GXP-2000 is configured for RFC2833 as well.
- Waldo
On Apr 28, 2006, at 10:20 AM, Johnny Stork wrote:
I seem to be having a problem with my GXP-2000. No matter how
carefully I type in the mail
Make sure you have the DTMF mode set to RFC.
On Fri, 2006-04-28 at 15:20, Johnny Stork wrote:
> I seem to be having a problem with my GXP-2000. No matter how carefully I
> type in the mailbox number and password when calling the mailbox (*98), it
> keeps complaining that the password is not corr
I seem to be having a problem with my GXP-2000. No matter how carefully I type
in the mailbox number and password when calling the mailbox (*98), it keeps
complaining that the password is not correct? I can use any other phone to
check the same mailbox and it works fine, just not with the GXP-20
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Waldo Rubinstein
>
> Make sure dtmf-mode is set to rfc2833 in both sip.conf as
> well as in the GXP-2000.
OK, thanks, but problem is more general: often phone doesn't send even a
packet to the Ast
Thanks - rfc2833 setting for DTMF in the in the phone GFUI did the trick.
- Original Message -
From: "Waldo Rubinstein" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 27, 2006 10:50 AM
Subject: Re: [A
Make sure dtmf-mode is set to rfc2833 in both sip.conf as well as in
the GXP-2000.
- Waldo
On Apr 27, 2006, at 12:28 PM, dataman wrote:
We are having trouble getting the GrandStream GXP-2000 (1.0.2.13)
to work with the Asterisk (1.2.6) voice mail prompts. We access
voice mail but extens
On 10:28, Thu 27 Apr 06, dataman wrote:
> We are having trouble getting the GrandStream GXP-2000 (1.0.2.13) to work
> with the Asterisk (1.2.6) voice mail prompts. We access voice mail but
> extension and password dial tones are not accepted. An the voice mail times
> out. Dial tones work fin
We are having trouble getting the GrandStream GXP-2000 (1.0.2.13) to work
with the Asterisk (1.2.6) voice mail prompts. We access voice mail but
extension and password dial tones are not accepted. An the voice mail times
out. Dial tones work fine with voice menus and dialing but not voice mai
Cisco does.> -Original Message-> From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]> Sent: Monday, February 20, 2006 11:22 AM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [Asterisk-Users] Grandstream GXP-2000
>> On Mon, 20 Feb 2006, Richar
L PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 20 February 2006 19:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
On Mon, 20 Feb 2006, Richard Amerman wrote:
> One thing to keep in mind with PoE is that you can simply use an
On Mon, 20 Feb 2006, mustardman29 wrote:
Kinda like what Cisco does.
well i think most people know with cisco the power cube is sold
separately. just because so many people have ciscos. the number of peoples
exposure to aastra is pretty small in comparison, and even then it is only
the 480i
Kinda like what Cisco does.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Monday, February 20, 2006 11:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Grandstream GXP-2000
>
>
At 11:21 AM 02/20/2006, you wrote:
Aastra does not really make it clear that the 480i is poe _only_. A
lot of people are very suprised when I explain to them that the 480i
is poe only.
I thought them made it really clear it was POE only and I was really
surprised when I found the wall wart in
On Mon, 20 Feb 2006, Soner Tari wrote:
For 100bit issue, I tend to believe in the data sheet, but I would also like
to hear a first-hand verification. (But I guess we have to wait, because
voipsupply accepts pre-sale orders for now, they don't ship them yet.)
The SPA-942 is $179.95, I would ra
On Mon, 20 Feb 2006, Richard Amerman wrote:
One thing to keep in mind with PoE is that you can simply use an injector at
the phone location. At least with the 480i you can easily order the phone
with the power injector.
Aastra does not really make it clear that the 480i is poe _only_. A lot of
At 03:01 AM 02/20/2006, you wrote:
also, rather disturbingly the linksys press release[1] implies the
942 is PoE only (like the aastra 480i), no external power supply.
Well, the 480i CT comes with a wall wart if you don't want to use POE
and their web site shows the optional PS available. I as
One thing to keep in mind with PoE is that you can simply use an injector at the phone location. At least with the 480i you can easily order the phone with the power injector.
Richard
On 2/20/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
On Mon, 20 Feb 2006, Soner Tari wrote:> I have Data She
: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 20, 2006 1:01 PM
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
On Mon, 20 Feb 2006, Soner Tari wrote:
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close to bottom):
On Mon, 20 Feb 2006, Soner Tari wrote:
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close to bottom):
Physical Interfaces:
2 100baseT RJ-45 Ethernet Ports (IEEE 802.3)
And that was one of the reasons I was considering 942. Do you think the data
sheet may be wrong?
: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 20, 2006 11:44 AM
Subject: RE: [Asterisk-Users] Grandstream GXP-2000
On Mon, 2006-02-20 at 01:22 -0800, [EMAIL PROTECTED] wrote:
I would still like to k
On Mon, 2006-02-20 at 01:22 -0800, [EMAIL PROTECTED] wrote:
> I would still like to know what they were smoking when they put two
> _10 meg_ ethernet ports on the linksys 942.
Probably the "let's not cannibalize the 79xx series" pipe. Wouldn't
surprise me if the Ethernet chip is capable of doing 1
On Mon, 20 Feb 2006, Lee Archer wrote:
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty
good phone. I used to run GXP-2000's, still have 10 new in a box and
another 20 in demo/test circulation, but I also run a few dozen 9133i,
480i and 9112i phones and I think Aastra are g
TECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 19 February 2006 13:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream GXP-2000
On Sat, 18 Feb 2006, Michael J. Liberatore wrote:
> Well the gxp-2000 has BLF, the polycom 501 does not correct? I ha
ailto:[EMAIL PROTECTED]]> Sent: Sunday, February 19, 2006 3:04 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'> Subject: RE: [Asterisk-Users] Grandstream GXP-2000>> > The GXP2000 firmware is not bad for features and ease of> use but still
> > bu
ing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
My GXP-2000 is currently collecting dust. I had several issues with it.
Mainly echo while on speaker.
The other person can barely mae out what you are saying. Another issue
was if the phone recieved to many calls it w
are far superior to the GXP2000 IMHO.
> -Original Message-
> From: Chris Bagnall [mailto:[EMAIL PROTECTED]
> Sent: Sunday, February 19, 2006 3:04 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Grandstream GX
My GXP-2000 is currently collecting dust. I had
several issues with it. Mainly echo while on speaker.
The other person can barely mae out what you are
saying. Another issue was if the phone recieved to
many calls it would just freeze up and I had to pull
out the plug. Again I have not used it in a
On Sat, 18 Feb 2006, Michael J. Liberatore wrote:
Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an
astra 480i and it was prety bad, but I was going to test the 9133i for
an inexpensive phone to compete with the gxp2000. The gxp2000 is not
bad though, the new firmware helps
On Sun, 19 Feb 2006, Michiel van Baak wrote:
I tried with one phone on both * svn head, *1.2 and *1.0.9
The exact fw version for the phone is something I cannot get
for you now as the phone is back to the shelve.
What is the MAC address of your phones? There are hardware revisions of
the gxp20
On 11:07, Sun 19 Feb 06, Chris Bagnall wrote:
> > I know it's still beta, but don't use the latest firmware in
> > production unless you can live with an empty display after
> > transferring a call.
> > Only a reboot of the phone will give you text on the display again.
> > I tested and confirmed
> I know it's still beta, but don't use the latest firmware in
> production unless you can live with an empty display after
> transferring a call.
> Only a reboot of the phone will give you text on the display again.
> I tested and confirmed this with 5 phones.
What firmware are you running? I'v
> The GXP2000 firmware is not bad for features and ease of use
> but still buggy. The hardware is junk to be quite honest and
> I don't think firmware will ever fix that. The Aastra 9133i
> hardware is 10x better.
I have a few of both here at the moment, and I'm not sure I'd agree with
that.
One thing to keep in mind with some of these phones is that physical quality can not be changed easily, why software is a different matter, though no guarantees.
I have been working with Aastra 480i's for about 14 months and at first they were fairly limited and had many issues. This all turned o
--Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, February 18, 2006 7:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream GXP-2000
On Fri, 17 Feb 2006, mustardman29 wr
If you can live without a speakerphone, the Polycom 301 is an excellent
phone and is only $30US more expensive than the GXP-2000s. I tend
to trust the Polycom brand, and they haven't really steered me wrong
yet in the IP phone hardware. I'm interested though in any
reports of success with the Aas
On Fri, 17 Feb 2006, mustardman29 wrote:
The GXP2000 firmware is not bad for features and ease of use but still
buggy. The hardware is junk to be quite honest and I don't think firmware
will ever fix that. The Aastra 9133i hardware is 10x better. The 9133i
firmware is still a work in progress
bject: Re: [Asterisk-Users] Grandstream GXP-2000
On 13:38, Fri 17 Feb 06, Clint Sharp wrote:
> I'm still having numerous echo issues, even on SIP calls, with the
> GXP-2000s. Unfortunately, they cause echo on the remote end on SIP
> calls, which does not occur on other phone models.
-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
I'm still having numerous echo issues, even on SIP calls, with the
GXP-2000s. Unfortunately, they cause echo on the remote end on SIP calls,
which does not occur on other phone models. The speaker phone is unusable
due to
Clint Sharp wrote:
I'm still having numerous echo issues, even on SIP calls, with the
GXP-2000s. Unfortunately, they cause echo on the remote end on SIP
calls, which does not occur on other phone models. The speaker phone is
unusable due to echo problems. Maybe the 1.0.2 firmware branches
On Fri, 17 Feb 2006, Clint Sharp wrote:
I'm still having numerous echo issues, even on SIP calls, with the
GXP-2000s. Unfortunately, they cause echo on the remote end on SIP calls,
which does not occur on other phone models. The speaker phone is unusable
due to echo problems. Maybe the 1.0.2 f
On 13:38, Fri 17 Feb 06, Clint Sharp wrote:
> I'm still having numerous echo issues, even on SIP calls, with the
> GXP-2000s. Unfortunately, they cause echo on the remote end on SIP calls,
> which does not occur on other phone models. The speaker phone is unusable
> due to echo problems. Maybe t
I didn´t have any problem buying it from http://www.digitnetworks.com/store/. I liked a lot their services when I bought 7 Budge Tone and 1 GXP 2000. It was fast and I got them in good conditions. The phones cost a less than 100 Euro/phones. I find it easy to work with and I am planning to buy
I'm still having numerous echo issues, even on SIP calls, with the
GXP-2000s. Unfortunately, they cause echo on the remote end on SIP
calls, which does not occur on other phone models. The speaker phone
is unusable due to echo problems. Maybe the 1.0.2 firmware branches
will help, but I'm scared
> I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.
> - Is it a good choice (budget limit of 100 Euro/phone is mandatory)?
> - Can be a profitable business the direct buying of 50 phones (to save other
> money) or is it a risk?
if you've never tried a phone, it's always a
Hi,
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.
- Is it a good choice (budget limit of 100 Euro/phone is mandatory)?
- Can be a profitable business the direct buying of 50 phones (to save other
money) or is it a risk?
Thanks in advance
--
Mimmus
___
> Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
>
> >From that article:
> There is an 'allpage.agi' now available at
> http://aussievoip.com.au/allpage.agi. Documentation is available in
I'm the author of that, and I've actually re-written it, because I was
pretty unhappy with
Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
>From that article:
There is an 'allpage.agi' now available at
http://aussievoip.com.au/allpage.agi. Documentation is available in
the file. This should work with Snom and Grandstream GXP2000 phones
(and possibly budgettones if they
Has anyone been
successful getting Auto-Answer by Call-Info to work with the GXP
2000
I have followed the
suggestions in
http://www.voip-info.org/wiki/view/GXP-2000
Specifically I
have:
1. Upgraded to
1.0.1.13, which supposedly supports this feature
2. Set Allow
Auto-Answer by Ca
It does appear to be the remote party that is hearing the echo as a result of the loud volume on the other end. They actually had a few people calling in from the outside report that they could hear their echo. When they turned down the volume on the Grandstream, the echo seemed to go away. So I
On Fri, 28 Oct 2005, Erick Baum wrote:
We have 50 of these phones in one location and a couple remote phones. The
problem seems to be caused by the volume settings on the phone. We have
noticed that the echo seems to be worse when the volume is very high on the
phone (not using speakerphone). We
We have 50 of these phones in one location and a couple remote phones. The problem seems to be caused by the volume settings on the phone. We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone). We're still testing, but that's what we'v
Erick Baum wrote:
We're having a rather serious echo problem using the Grandstream
GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm
overlooking that might be an easy fix. The echo seems to be worst on
internal SIP to SIP calls but you do get it every once in a while on
> Erick, we're also using 1.0.1.12, having some echo problems,
> mostly with in/out going ZAP calls (on quadBRI, w/asterisk
> 1.0.9), the internal SIP calls seem to work fine. (but you
> have to make sure your volume isn't too high) Also the
> GXP-2000 has the annoying feature that calls get di
On 10/27/05, Erick Baum <[EMAIL PROTECTED]> wrote:
We're having a rather serious echo problem using the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get
We're having a rather serious echo problem using the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through
g more. More bang for the buck than the Polycoms but slightly less feature set. These phones are more of an office standard like the Nortel or Avaya phones.
Message: 9Date: Thu, 1 Sep 2005 13:28:54 -0400From: Aaron W <
[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Grandstream GXP-
200
2005 13:28:54 -0400From: Aaron W <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Grandstream
GXP-2000 Poor sound QualityTo: Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com>Message-ID: <[EMAIL PROTECTED]>Content-Type: text/plain;
charset=&qu
On Thu, 1 Sep 2005, Jesus Mogollon wrote:
> We have all those problems and then some... after a while, the phone starts
> degrading: The ringing becomes lower and lower and there is a lot of
> stuttering in the conversation. Also, if I stop/start asterisk, half of the
> phones reconnect while t
Thanks..I am begining to agree with you about these phones. Which
poylcoms do you have? I have been looking at the polycom
soundpoint IP501. It seems like a good phone for just under
200USD.
Thanks again,
Aaron On 9/1/05, Jesus Mogollon <[EMAIL PROTECTED]> wrote:
Greetings
We have all those
Greetings
We have all those problems and then some... after a while,
the phone starts degrading: The ringing becomes lower and lower and
there is a lot of stuttering in the conversation. Also, if I stop/start
asterisk, half of the phones reconnect while the rest don't. I was
using the same firm
I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the
phone is much more usable However, I still have two slight sound
quality issues:
1) There is static on the line at all times. It is not that
noticable to me, but when I make calls out the PSTN the person on the
other end hears
al Discussion
> Subject: Re: [Asterisk-Users] Grandstream GXP-2000 headset
>
> > Hi all
> >
> > What headset do people use with the GXP-2000? Any
> recommondations for
> > or against particular models?
>
> i'm sent mail to [EMAIL PROTECTED], need i
Done, I sent a message to them also.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|marek cervenka
|Sent: Lunes, 23 de Mayo de 2005 01:03 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Grandstream
Hi all
What headset do people use with the GXP-2000? Any recommondations for
or against particular models?
i'm sent mail to [EMAIL PROTECTED], need info too
btw i'm asked that will support IAX, they respond "yes, if customers
want it" -> write them
---
M
Hi all
What headset do people use with the GXP-2000? Any recommondations for
or against particular models?
Thanks
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk
On Wed, 18 May 2005, Paulo Loureiro wrote:
> Can you post the configuration template?
It is for an as yet unreleased version of the firmware. I'm sure
Grandsrteam will release the template with the new firmware. Given the
finished feel of the testing firmware we received I'd say it is close to
On Wed, 18 May 2005, Anton Krall wrote:
> Peter.. I just bought a gxp 2000 and I wanted to know, how are you
> configuring them using templates?
There is a template->binary config file compiler available from the
download page at the Grandstream web site. Fill in the template and serve
it via h
Of
|Peter Svensson
|Sent: Miércoles, 18 de Mayo de 2005 02:22 p.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Grandstream GXP-2000 and good support
|
|We recently purchased a Grandstream GXP-2000 phone and I would
|like to share our experiences with it, especially out very
|goo
On 21:25, Wed 18 May 05, Paulo Loureiro wrote:
> Can you post the configuration template?
>
> --- Paulo.
>
Or even better: Create a wiki page about it.
--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Two
Can you post the configuration template?
--- Paulo.
On May 18, 2005, at 20:22, Peter Svensson wrote:
We recently purchased a Grandstream GXP-2000 phone and I would like to
share our experiences with it, especially out very good support
experience.
The phone was easy enough to set up. The phone was
We recently purchased a Grandstream GXP-2000 phone and I would like to
share our experiences with it, especially out very good support
experience.
The phone was easy enough to set up. The phone was configured using a
configuration file served via tftp. Creating the configuration file was a
bit
2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Grandstream GXP-2000
Does anyone have any experience with this phone? I'm considering
purchasing it but wish to hear if anyone has any experience
:23 AM
Subject: [Asterisk-Users] Grandstream GXP-2000
Does anyone have any experience with this phone? I'm considering
purchasing it but wish to hear if anyone has any experience with it.
Thanks,
Daniel
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Does anyone have any experience with this phone? I'm considering
purchasing it but wish to hear if anyone has any experience with it.
Thanks,
Daniel
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http://lists.digium.com/mailman/listinfo
Message -
From: "Cory Andrews" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
; <[EMAIL PROTECTED]>
Sent: Tuesday, March 15, 2005 3:17 AM
Subject: [Asterisk-Users] Grandstream GXP-2000
FYI, spoke with Grandstream this morning, th
FYI, spoke with Grandstream this morning, the GXP-2000 release has been
delayed again. Looking like April now before these hit the street.
--
Cory Andrews
Senior Partner
VOIPSupply.com
+
V: 800.398.VOIP X22
F: 716.630.1548
E: [EMAIL PROTECTED]
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