Hi all,
I'm working on a system with 4 Grandstream GP-200 Phones and the base
Asterisk install.
I have added a 5 phone which is remote to the client and located in my
office.
I can't get the phone to transfer a call or put a call on hold. This
applies to all the phones at the location.
I have a function *1 that starts and stops recording in a call. I use a
function so I can use MixMonitor.
It works well, however I would like to make it a little more integrated for
my users. We have GXP 200 hardphones. So far I've been able to configure a
softkey using the speeddial option to dial
We are connecting the GXP2000 to a Cisco POE switch 3650, it the
default mode at 15.4 for each phone the phones power up but we can only
have 24 on a 48 port switch, when we adjust the setting to 7000 (which
is what we calculate the phone to use) they don't power up ... we
tried all the power se
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 ta
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 tab
3. Change Daylight Savings Time to yes
4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change
clocks the second
I would really really appreciate it if someone would forward me
Release_GXP2000-BT200_1.1.0.16.zip for the Grandstream GXP-2000. I need
it in order to upgrade the phones to the latest stable.
Thanks,
Alex
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Hello,
We are having a hard time making the GXP-2000 work reliably with Asterisk.
We have several clients using the GXP-2000. These phones are behing NAT
and our Asterisk server has a public IP (no NAT).
The biggest problem we face is the clients complain of random, but
frequent, calls (in or out
Hello,
I have been trying to get my Grandstream busy line filter to
work for ages..
All the lights flash as they are supposed to.
If one Grandstream 7000 calls another Grandstream 7003 I can
use Grandstream 7002 to pick the call up pressing the BLF button and all works
fine.
Hi,
currently I use version 1.1.0.16 for my GXP-2000 which works really
fantastic. The only drawback I see is the addressbook.
Is the firmware 1.1.1.9 stable enough to use the phone in normal
environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000
says that there it is possible to dow
Hi,
currently I use version 1.1.0.16 for my GXP-2000 which works really
fantastic. The only drawback I see is the addressbook.
Is the firmware 1.1.1.9 stable enough to use the phone in normal
environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000
says that there it is possible to dow
I have a client with about 24 GXP-2000. Everything seems to be
working fine except one particular behavior of the blind transfer.
Whenever anyone makes an outbound call, they can transfer the call
between extensions either blind or attended with no problems.
However, whenever an incoming ca
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
--
"Programmi
Can anyone tell me how to configure the grandstream gxp-2000 for 4 line
apearances. I have the the sample conf from the website and the phone
is getting its config from my TFTP server. But it does not have any info
for the other line apearance butons
The real thing that would help is a complete
Hey everyone,
I was wondering if anyone is able to help me with a
solution.
I have a small office set up with GXP-2000 phones and the
one thing I cannot get to work is them being able to transfer a caller directly
to another person’s voicemail.
If I have a dial tone (and not on
Call
automon => *1 ; 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
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, i
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.
http://www.vecsector.com/phonecall/
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 app
, June 23, 2006 2:00 AM
> To: Non-Commercial Discussion Asterisk
> Subject: [Asterisk-Users] GXP-2000 and Shared Line Appearances
>
> I have a client with 20 GXP-2000s. Everything seems to be
> working fine. However, after a couple of weeks of use, the
> client is having a har
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 be
turn away
potential business because of the lack of that one feature. If I could
write code I'd try do it myself.
> -Original Message-
> From: Daniel Salama [mailto:[EMAIL PROTECTED]
> Sent: Friday, June 23, 2006 2:00 AM
> To: Non-Commercial Discussion Asterisk
> Subj
I have a client with 20 GXP-2000s. Everything seems to be working
fine. However, after a couple of weeks of use, the client is having a
hard time adjusting to the new IP based phone systems and only misses
one feature from their old Lucent system.
That is, they had 8 analog lines before and
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
I have a network of GXP 2000 phones and would like to know if there is a
way to configure the phones so that if there is one person talking, and
another call comes in then they can hold/hangup that call and take the
incoming call.
At the moment, when a call comes in and the phone is offhook, then
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 shoul
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 PROTECTED
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 con
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
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 eve
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 i
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
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 Thu
Hi,
is it possible to have one central phonebook and install it on the
phone or using ldap?
Best regards,
Matthias
--
"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. S
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 they
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
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:
W
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
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 striv
t; Subject: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
>
> 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 f
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: [][708][FF71][0101000D] ERROR 4099
> GET cfg
>
> What does errorcode 4099 mean?
I don't know but it looks li
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 1.1.
Hi,
I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [][708][FF71][0101000D] ERROR 4099
GET cfg
What does errorcode 4099 mean?
Best regards,
Matthias
--
"Programming today is a race between software engineers striving to
build
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_Templ
; 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 s
MAIL PROTECTED]
> Sent: Wednesday, 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
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
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 wh
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
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 http://www.voip
I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u
codec, I know they upstream bandwidth is the limiting factor and they
most likely
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
O
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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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 Blunde
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 sa
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.
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 progra
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
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 MultiPu
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?
Thanks,
Daniel
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Aster
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
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 9:3
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
> Is the 94x any better? seems without backlighting, any are
> next to useless.
The SPA-9x2 have backlit displays.
Nabeel
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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
>
> Wi
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 online almost immediately after we installed them. Before that, there were the usual l
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 vl
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
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 ha
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.
I
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 bett
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 lo
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). Th
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:
As for the SPA-841, I have a client with a few of them and he
cannot stop complaining about the bad audio qua
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 intern
Daniel Salama wrote:
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 issues
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 wer
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 qua
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] chan
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 one
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,
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 reviewin
ot a beta.
> >
> > Mike
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> > [mailto:[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>] On Behalf Of
> > Daniel Sal
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 have
am using the latest release firmware, not a beta.>> Mike>> -Original Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of> Daniel Salama> Sent: June 6, 2006 4:12 PM > To: Non-Commercial Discussion Asterisk> Subject: [Asterisk-Users] G
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 anyt
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]> [mailto:[EMAIL PROTECTED]] On Behalf Of> Danie
sing the latest release firmware, not a beta.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: June 6, 2006 4:12 PM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000
I'm using a few GXP-2000 with f
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama
Sent: June 6, 2006 4:12 PM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000
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
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 ma
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Ringwald
Sent: Wednesday, 17 May 2006 10:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware
I had provisioning via tftp working on this phone. I have
I had provisioning via tftp working on this phone. I have verified that
after the firmware upgrade, it contacts the tftp server and downloads
the cfgMACADDR file, and the ring/etc files successfully. Unfortunately,
changes made to the config file don't make it to the phone (SIP account
info/ser
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
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 di
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
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As
Hi,
is there a way to completely disable TFTP/HTTP provisioning on the
Grandstream GXP-2000?
Thanks
--
Domenico Viggiani
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ter 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 April 2006 8:49 PM
> To: asterisk-users@lists.digium.com
> I would definitely recommend it to you.
>
> Mark
>
> -Original Message-
> From: Gareth Blades [mailto:[EMAIL PROTECTED]
> Sent: Monday, 10 April 2006 8:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GXP-2000 phones stop registering
>
>
y, 10 April 2006 8:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 phones stop registering
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
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
ovious
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}
ext
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