Thanks for the info, Ken. I was about to research this tonight.
Todd
On Mar 12, 2007, at 12:53 PM, Ken Williams wrote:
In case it hasn't been posted before, here's instructions to get
the correct time to show up on your Grandstream GXP-2000's:
1. Login to phone
2. Go to Basic Settings
Hello Cavanna,,
* Cavanna, Richard [EMAIL PROTECTED] [27-07-06 15:59]:
The real thing that would help is a complete list of the configurable
comands on the latest firmware so I can create the config file.
try that config file, works perfectly for me.
Best regards,
Matthias
--
Programming
Daniel Salama wrote:
Dustin,
any updates on this?
Thanks,
Daniel
Hey Daniel!
Yes - just posted the link.
I appologize for the delay.
Here's the link to the forum as well, if anyone is interested. This
should compile and run on Asterisk-1.2.4 and higher.
Beautiful. Will test and give you comments.
Nice work.
- Daniel
On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote:
Daniel Salama wrote:
Dustin,
any updates on this?
Thanks,
Daniel
Hey Daniel!
Yes - just posted the link.
I appologize for the delay.
Here's the link to the forum as well,
; One Touch Record
;atxfer = *2 ; Attended Xfer
-Original Message-
From: Dustin Wildes [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000
Dustin,
any updates on this?
Thanks,
Daniel
On Jun 23, 2006, at 1:07 PM, Dustin Wildes wrote:
shadowym wrote:
That feature is called Bridged (or Shared) line appearance. That
is one of
the things Asterisk cannot do and nobody seems very interested in
making it
do that because it is
I had the same problem some time ago. Make sure call waiting is NOT
disabled. This will make the phone receive more calls on the other
lines.
- Daniel
On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne
wrote:
I have a network of GXP 2000 phones and would like to know if
That feature is called Bridged (or Shared) line appearance. That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy. There has been some talk about
implementing it but so far there does not seem to be any progress.
I
shadowym wrote:
That feature is called Bridged (or Shared) line appearance. That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy. There has been some talk about
implementing it but so far there does not seem to
Here is the latest word on SLA that I could find. Looks like it is quite a
ways off but at least it is on the radar screen.
http://lists.digium.com/pipermail/asterisk-users/2006-May/153385.html
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23,
Kristian Kielhofner wrote:
Mike Fedyk wrote:
I happen to have asterisk running as a router, so I use it doing QoS
with tc (traffic control) and wondershaper set to prioritize based on
port ranges. I sent a patch to the debian bug tracking system a
while back with a few improvements -- I
Grandstream have acknowledged that there is a problem with 1.1.0.13 on
later phones (MAC's 00:0B:82:09:xx:xx I assume) and have advised me to
wait for the next firmware release. So anyone with later phones (MAC's
00:0B:82:09:xx:xx), do not upgrade to 1.1.0.13.
On Wed, 14 Jun 2006 [EMAIL
Tim Panton wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An embedded low power system would do fine.
You might
On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:
Tim Panton wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An
On Sat, Jun 17, 2006 at 11:14:33AM +0100, Tim Panton wrote:
On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:
Tim Panton wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter,
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to do a mass update of
the phones.
On
Hi Gareth,
Gareth Blades wrote:
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to
Matthias Fechner wrote:
Hi Gareth,
Gareth Blades wrote:
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for
256k upstream you should be able to handle 8 calls but this is in ideal
conditions.
If you were to use IAX and enable trunking then you would use 30kbps for
the 1st call and 10kbps for each additional call.
See
The only issue with 1.1.0.13 which affects only certain versions of the
gxp-2000 is the display blanking issue on very early phones.
It sounds like you have a faulty phone and should return it for a
replacement.
On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote:
I have had 2 GXP-2000 for a
Thats what I thought the problem might be, so I have just now upgraded the
other phone to 1.1.0.13 and its exactly the same, no speaker phone and
hangs from a soft reboot.
I also tried the audio loopback in the factory functions menu, this
loopback's fine with the older 1.1.0.13 phones but does
, June 14, 2006 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues
Thats what I thought the problem might be, so I have just now
upgraded the other phone to 1.1.0.13 and its exactly the
same, no speaker phone and hangs
this firmware, if you mail me off-list.
Bye
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP
Hi,
I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099
GET cfgMAC
What does errorcode 4099 mean?
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.
On Wed, 2006-06-14 at 14:05, Matthias Fechner wrote:
Hi,
i got my Grandstream GXP-2000 phone today and want to configure it
with TFTP. I downloaded the firmware
On Wed, 2006-06-14 at 15:46 +0200, Matthias Fechner wrote:
Hi,
I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099
GET cfgMAC
What does errorcode 4099 mean?
I don't know but it looks
You need to encode txt configuration file using tool provided on GS site.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthias Fechner
Sent: Wednesday, June 14, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi Gareth,
Gareth Blades wrote:
You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.
thx that was the problem. Now it works fine.
Best regards,
Matthias
--
Programming today is a race between software engineers
Wow! 22Kbps of overhead? Are you sure? That sounds like way too much
overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any
other suggestion?
Thanks,
Daniel
On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so
Welcome to the wonderful world of VoIP, where people are eager to move
from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in
voice quality, and then throw over 20kbps of RTP, IP and related
overhead on top of them. Isn't IP wonderful? :-)
Regards,
Steve
Daniel Salama wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An embedded low power system would do fine.
You might even get away with an
That may not be such a bad idea. I've read people trying to put
Asterisk on a WRTG54 or something like that. Would that be good? I
guess I could do SIP in the office and trunk via IAX2 and save on
bandwidth plus internal calls would be local.
I tried to upgrade them to 512K but because
Would you mind telling me how to setup the GXP-2000's VLAN/QoS
settings with the DES-1226G? I just purchased the DES-1226G and want
to make sure I setup it up right. I don't have the ability to run
separate wiring for the PC and the phone and that's why I need this
help.
Thanks,
Daniel
Wow! Awesome. This template is much more complete than the one on
GS's download page.
Thanks,
Daniel
On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote:
Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.
For future reference, I think the Grandstream config files can program
any parameter that's included in the web interface. If you want to set
something that isn't in the template, you can use view source on the
web form to figure out the name of the option: the field names in the
HTML are the
That's great. GS support people are great, but I had asked him how to
set other parameters that I see on the web and they told me they
didn't know. That I should look through the wiki or other web sources.
Anyway, that's great to know.
Thanks,
Daniel
On Jun 10, 2006, at 5:16 AM, Phil
Hi,
is it possible to update the phonebook of the gxp-2000 via tftp?
So I can maintain the phonebook central or using ldap etc.?
Best regards,
Matthias
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To
good question! I'd like to know too, so keep it public please !:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Friday, June 09, 2006 9:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000 MultiPurpose
Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file?
Yes you can if you are running 1.0.2.13 or later. I have the template
which I tried posting here as an attachment but it has not arrived yet.
If it does not arrive you can email me directly or contact grandstream
support.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to
Is the 94x any better? seems without backlighting, any are
next to useless.
The SPA-9x2 have backlit displays.
Nabeel
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To UNSUBSCRIBE or update options visit:
Mike Fedyk wrote:
I have heard good things about the D-Link DES-1226G switch ($150 at
newegg). If you can run a separate cable to the computer and phone. If
you can't run the extra cables, then configure your phone to tag itself
as part of the voip vlan and let the switch tag everything else
I'm willing to bet the phones that are stalling have the most active computer users attatched to them. I wouldn't advise having the computer running through the phones port. To me that is asking too much out of the $100 phone.Run each device from it's own port on your switch.On Jun 7, 2006, at
On Thu, 2006-06-08 at 13:21 -0400, list mail wrote:
I'm willing to bet the phones that are stalling have the most active
computer users attatched to them. I wouldn't advise having the
computer running through the phones port. To me that is asking too
much out of the $100 phone.
Run each
Erick Baum wrote:
The worst ongoing issue has been the echo and the really crappy
speakerphone. The customer is pretty much used to it now. But we're
slowly replacing them with Polycom's as new people come on and as
others just get fed up. Unfortunately one of the phones met it's
doom by
I am running 1.1.0.13 and there are no issues which are causing a
problem for us. The speakerphone is not much use but we can live with
that.
1.0.1.9 would stop registering after a while causing incoming calls to
go straight to voicemail.
1.0.2.13 fixed this but had a bug where sometimes
I have a client who has about six of these phones. Luckily (for me, not
for them) they were purchased before I came into the picture.
Daniel Salama wrote:
I have heard complaints from my client about the speakerphone and they
are now
You don't notice any problems when using the speaker-phone,
They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another
Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648]
What specifically were the voice quality complaints about the spa-841
phones? The only thing I have noticed is calls can be louder than
expected. What else have you seen?
Daniel Salama wrote:
They don't all go down at the same time, or at least, my client hasn't
noticed. I just added the
The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically
Daniel Salama wrote:
snip
As for the SPA-841, I have a client with a few of them and he cannot
stop complaining about the bad audio quality.
Latest/last firmware upgrade?
Handset?
speaker phone?
headset?
I find the handset quite acceptable
Speaker phones are a can of worms, with so many
Did you try setting the RTP packet time size to 0.020? Also I would
look at the trunk, provider or internet connection before the phones I
started suspecting the phones.
I have had the same problems with providers, and the conversations sound
great from one location to another over the
Latest firmware installed and problem with handset. They don't use
headset nor speakerphone.
Thanks,
Daniel
On Jun 7, 2006, at 3:14 PM, John Novack wrote:
Daniel Salama wrote:
snip
As for the SPA-841, I have a client with a few of them and he
cannot stop complaining about the bad
John Novack wrote:
Is the 94x any better? seems without backlighting, any are next to
useless.
Yes, I like the 941 better than the Polycom 301 and the display is much
improved (no backlight, but one of the guys at voipsupply told me that
the 942 has a backlight which sounds very promising).
No changes whatsoever. Unplugged the spa and replaced it with a gxp.
I haven't tweaked any RTP or QoS parameters for I don't have any
documentation on it :(
Thanks,
Daniel
On Jun 7, 2006, at 3:44 PM, Mike Fedyk wrote:
Did you try setting the RTP packet time size to 0.020? Also I
would
With hundreds of installed phones now, here are my choices in order
Linksys SPA-941/942
Polycom 501/601
Cisco 7960
Polycom 301
Snom 320/360
I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for
Kerry Garrison wrote:
I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for better phones very quickly.
Let me say that when suggesting the spa-841 it is only in the context of
sub-$100 phones.
What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client
They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low.They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the
I have heard good things about the D-Link DES-1226G switch ($150 at
newegg). If you can run a separate cable to the computer and phone. If
you can't run the extra cables, then configure your phone to tag itself
as part of the voip vlan and let the switch tag everything else as the
computer
We had not used these phones before, which I will admit was my first mistake. However, I did do research online to see what other peoples experiences were but the major problems with the phone started surfacing onlinealmost immediately after we installed them. Before that, there were the usual
What about Aastra 480i, 9133i?
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 07, 2006 1:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] GXP-2000
With hundreds of installed phones now
I can't say why you're having this problem, but I can tell you that my phone
can receive (and make) multiple calls easily. It might have more to do with
Asterisk than the GXP2000.
I am using the latest release firmware, not a beta.
Mike
-Original Message-
From: [EMAIL PROTECTED]
I enabled call-waiting from the tftp configuration and it now works.
What firmware are you using and where can I get it?
My client complaints that the phone stops working every once in a
while with no explanation. My client says that he could be using the
phone with no problem and a few
We setup a company with 50 of these phones and had my client not been as understanding as they were, that could have put me out of business. What an unbelievable nightmare. This was about 8 months ago when the firmware was so bad the phone was a better paper weight than anything else.
Since
Erick Baum wrote:
We setup a company with 50 of these phones and had my client not been as
understanding as they were, that could have put me out of business.
What an unbelievable nightmare. This was about 8 months ago when the
firmware was so bad the phone was a better paper weight than
Well, these are encouraging words :)You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently
I suggest you contact grandstream about this. Only thing I can suggest is look
at feature's Early Dial (I have set to no) and No Key Entry Timeout (set to
10-15 seconds). As for all these other problems of phone stop working, etc.,
we haven't come across these in office (then again we don't
I had the same problem!
You have in your PXXX in your configs that 1.1.0.11 does not support.
Took me an hour to go through my configs and the web page to find what
PXXX in my configs unset the phone :)
Once its done, the phone will be accept the configs with no problems.
-Original
On 01/05/06, Jeffrey Macko [EMAIL PROTECTED] wrote:
Does anyone know the secret to get the GXP-2000 Message waiting lamp to
illuminate?
No secret - just set a 'mailbox' line in the appropriate peer entry in
sip.conf. Later GXP-2000 firmware shows the number of messages waiting
on the LCD
Mark,
Do you have the Flash Operator Panel or anything else installed?
I only had 1 phone stop registering in the first 2 weeks that I used
them and then after I installed FOP I had 3 phones stop registering in
the next couple of days.
I have now disabled FOP and have gone just over 2 days without
Yes. Me.
I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch.
I would definitely recommend it to you.
Mark
-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED]
Sent: Monday, 10
So the bug still exists in the 1.0.2 branch?
Thanks
On Mon, 2006-04-10 at 12:14, Mark Edwards wrote:
Yes. Me.
I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch.
I would definitely recommend it
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten = 100,4,Macro(hangupcall)
so the user doesn't need to put in a
Thanks
Waldo
On Apr 9, 2006, at 2:19 PM, Tim Litwiller wrote:
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten
I sent this from the wrong address and I don't think it went through. I've just done some testing on the phone on 1.0.1.9 and 1.0.2.13. The one on
1.0.1.9 has no outbound gain issues, it is nominal with the rest of the phones in out office (Snom 320, Polcyom IP 301, and Budgetone 101). However,
I had the opposite problem, I had to set txgain
down as they were too loud and causing problems.
- Original Message -
From:
Clint
Sharp
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, March 02, 2006 6:56
AM
Subject: [Asterisk-Users]
Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
On Dec 31, 2005, at 7:28 AM, Ross C wrote:
Peter,
After upgrading to 1.0.1.13 I had some miscellaneous problems on one
of my GXP-2000's--it would grab an IP address, but it wouldn't get the
time/date, it wouldn't register, blah
Kristof Hardy wrote:
Was there a resolution to this issue? The GXP-2000 seems to be a very
popular phone, so I can't imagine others on the list not experiencing
this? Or is this part of a batch with unresolvable problems that I
need to send back to the seller?
Well, I'm using dozens of
Philip Edelbrock wrote:
18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144?
Gratuitous ARP
19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at
00:10:4b:96:2f:eb
20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline -
Transaction ID
Tony Hoyle wrote:
Philip Edelbrock wrote:
18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has
206.228.191.144? Gratuitous ARP
19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144
is at 00:10:4b:96:2f:eb
20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: Saturday, January 21, 2006 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
On Dec 31, 2005, at 7:28 AM, Ross C wrote:
Peter
Was there a resolution to this issue? The GXP-2000 seems to be a very
popular phone, so I can't imagine others on the list not experiencing
this? Or is this part of a batch with unresolvable problems that I need
to send back to the seller?
Well, I'm using dozens of these phones without this
On Dec 31, 2005, at 7:28 AM, Ross C wrote:
Peter,
After upgrading to 1.0.1.13 I had some miscellaneous problems on
one of my
GXP-2000's--it would grab an IP address, but it wouldn't get the
time/date,
it wouldn't register, blah blah blah. I could access the web
interface OK,
so it
I had a problem which I spoke to Grandstream about. It seemed that
around 7 seconds in it goes for time sync and if it fails it doesn't
retry. This problem was highlighted by the .12 firmware and a Windows
DHCP server we were using. Upon moving to a Linux DHCP server the
process was much
On Saturday 31 December 2005 01:57, Ross C wrote:
... and 2 Snom 320's (now discontinued I think).
No, they are not discontinued !!!
Regards,
Sven
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: Re: [Asterisk-Users] GXP-2000 any good with * ?
On Saturday 31 December 2005 01:57, Ross C wrote:
... and 2 Snom 320's (now discontinued I think).
No, they are not discontinued !!!
Regards,
Sven
___
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: Monday, January 02, 2006 2:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
On Saturday 31 December 2005 01:57, Ross C wrote:
... and 2 Snom 320's (now discontinued I think).
No, they are not discontinued !!!
Regards
-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
This doesn't seem to be correct, too...
Sven
On Monday 02 January 2006 17:43, Ross C wrote:
Sorry!!
Just discontinued @ voipsupply.com I guess.
Thx for the correction.
-Original Message-
From: [EMAIL PROTECTED
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Monday, January 02, 2006 12:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?
http://www.voipsupply.com/product_info.php?cPath=95_114produc
ts_id=883
: Monday, January 02, 2006 2:19 PM
To: Ross C
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?
My understanding is that there is currently a shortage of phones at
voipsupply (and also in other places). The 320 is selling pretty good
Original Message
From: Peter Bowyer [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 31, 2005 11:34 AM
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Hi all
Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to
Michiel van Baak wrote:
Hinting works fine for me with the latest firmware.
What version are you running?
We use 1.0.1.9 but the leds next to the speeddials wont
use latest * and latest gxp firmware, have a look here on how to do it:
http://www.voip-info.org/wiki/view/GXP-2000
Peter Bowyer wrote:
side-effect the phone won't sync with an NTP server - I've tried
different server names (time.nist.gov and pool.ntp.org) and IPs in the
config, but it refuses to update the time on the display.
No problem here. Using the 1.0.1.13 (very beta:)) also, synching with an
Peter,
After upgrading to 1.0.1.13 I had some miscellaneous problems on one of my
GXP-2000's--it would grab an IP address, but it wouldn't get the time/date,
it wouldn't register, blah blah blah. I could access the web interface OK,
so it wasn't a network issue (I don't think). Anyway...I ended
On Fri, 2005-12-30 at 15:01 -0800, [EMAIL PROTECTED] wrote:
Anyone using the GXP-2000 with * ?
Any showstopper problems?
The echo issues, is it speakerphone only?
I have a gxp and dont have the echo issue. I got it from
www.thevoipconnection.com and its a good phone I think anyway. No
On 15:01, Fri 30 Dec 05, [EMAIL PROTECTED] wrote:
Anyone using the GXP-2000 with * ?
Any showstopper problems?
The echo issues, is it speakerphone only?
Hi,
We have some of these phones in production.
They work ok.
The echo issues are fixed in recent fw versions.
The speeddial buttons
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