Displaying 20 results from an estimated 300 matches similar to: "grandstream firmware update 1.0.5.23"
2005 Mar 25
0
CAUTION: Re: grandstream firmware update 1.0.5.23
Voicemail works fine for me.
Post console output here to let us know what went wrong.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John
Breeden
Sent: Friday, March 25, 2005 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: CAUTION: Re: [Asterisk-Users] grandstream
2005 Aug 05
3
Is this echo problem down to IP Phone hardware?
Hello
I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy.
Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more
2006 Mar 03
5
new beta Grandstream firmware HT488_496_386
http://grandstream.com/BETATEST/HT488_496_386/
2005 May 08
3
Grandstream firmware 1.0.6.2
Grandstream owners,
I just noticed that there is a new firmware release, for those that are
interested:
http://www.grandstream.com/BETATEST/
Doug
2007 Feb 02
1
Compile without GSSAPI disables SSL?
Perchance this is something unique to Red Hat -- I'm compiling/running
1.0rc?? on a CentOS 4 machine (i386), building RPMs and trimming down
things I don't need (SQL e.g.).
If I compile with the flag --with-gssapi, the resulting dovecot-auth
binary links to libcrypto and libssl, the standard OpenSSL libs. Life is
happy even though I don't need gssapi/krb5 support.
If I remove
2004 Nov 24
3
Grandstream Firmware 1.0.5.16 Attended Transfer
I've searched for a few days now without finding an answer. The
release notes for version 1.0.5.16 of the Grandstream firmware says it
supports attended transfer using replace but the docs haven't been
updated so I can't work out how to enable it, or whether it should
Just Work. I'm currently using the # attended transfer patch for *
but would like to get back to using the
2007 Feb 09
6
The High Performance Echo Canceller (HPEC)
I recently read about the following new technologies from Digium. Has
anyone tried the new HPEC or knows when it will be available?
TDM800P and HPEC
The TDM800P is an 8-port analog telephony interface card, so it fills the
gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines
are going to be with us for some time, and demand for support for them
remains high. The
2005 May 24
1
BudgeTone 101 doesn't register with FirmWare 1.5.23
Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register staement (nothing in thertereal log). With the 1.0.3.81
version, the phone register properly.
Is ther any know bug with the SW Version?
Best regards,
Daniel ANDRE
--
Daniel ANDRE (mailto:daniel.andre@iris-tech.fr)
IRIS
2005 May 19
1
Re: Grandstream ATA 286 and ilbc (Anton Krall)
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
<blockquote type="cite"><br>
<pre
2006 Jan 31
0
New GXP-2000 Beta firmware available
>From the usual place, http://www.grandstream.com/BETATEST/GXP2000/
Note, there are two (and it took me a bit of a while to figure out)
images to be loaded. Copy the ...a.bin's and the .bin's to your http
provisioning directory, and reboot. The phone _must_ load the .bin
files before it understands the ..a.bin files.
After it loads the first one, the phone does lock up with the
2007 Nov 13
0
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Hi,
I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues).
You can find a lot of info and old firmware versions at this
2004 Aug 13
1
Problem with grandstream devices and DTMF signalling
Hi,
I've got a problem with some grandstram devices (namely a couple of
budgetone 101 and an ata-486). The point is that, unless I use inband
for DTMF, asterisk ignore the first digit dialed. Inband DTMF forces me
to use A-law/Mu-law, which is not what I want.
BTW, this appens after a Playtones(), waiting for user entering an
extension.
I've tried many solutions, played around with
2004 Mar 05
3
dropped calls
Hello list,
I'm getting droped calls on an asterisk installation. When on GS phone
dials another one, the call is dropped after some (usually random) time
but most of the tome within 3 to 20 seconds.
I think the cause is stated on the logs, see bellow, and is related with
the message "Didn't get a frame from channel: SIP/3805-df43", but I
can't figure why.
asterisk logs:
2007 Sep 28
3
[LLVMdev] Bugs in Getting Started Guide
On Thursday 27 September 2007 22:51, Tanya M. Lattner wrote:
> The getting started quickly instructions say that you "cd
> where-you-want-the-C-front-end-to-live" and the unzip the
> front-end, followed by "cd where-you-want-llvm-to-live".
Maybe this adds to the confusion: that he has to make decisions
("Where do I want the c-front end to live?") --- and at
1998 Mar 11
4
Re: Towards a solution of tmp-file problems
Hi everyone,
Thanks all for your feedback. Here is a reply to most of your
comments....
Roger.
Chris Evans wrote:
> On Mon, 9 Mar 1998, Rogier Wolff wrote:
> > not to give those rights away. A non-setuid program should not have to
> > worry about buffer overruns (you can crash the program, wow!). It
> Just a reminder, that in some cases, it _should_ worry. As a
2005 Jul 07
2
Asterisk/Grandstream Budgetone disconnect issue
I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone
2005 May 19
1
no music on hold
Hello,
I am having problems with music on hold on grandstream phones.
When I press Hold button on grandstream phone this is the debug of sip.
But nothing happens, no music.
Is it problem of asterisk or grandstream budget phone?
Sip read:
INVITE sip:1105@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41
From:
2005 Feb 24
7
CallTransfer
Hi
I was wondering if there are any special settings that
I need to be able to transfer calls.
Whenever I press the 'recall' button, I just here a click,
and no ring-tone to transfer.
in my debug log I get this :
--------------------------
Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1
Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1
(index 0)
Feb 24
2005 Mar 15
2
Grandstream and Transfers
Hi all,
Just wondering if anyone's come across this issue, and what might be a fix for it:
We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The
phone can do proper supervised transfer, but _only_ once. If the user attempts
to transfer a second time, it won't work.
any suggestions/hints/tips are welcome..
Flynn
2003 Apr 29
4
thick plot lines
Dear People,
In a qqplot I am doing, I get lines/points that are very thick. I've tried
setting the lwd variable to 0.1, but it doesn't seem to have any effect.
Also, I have set the value of lty to dashed, but I still get dots. The
command looks like
qqplot(cdf.inv(seq(0,1,length=size),theta,pos,len),empmargdistvec(len,theta,pos,size),
xlim=c(-theta,theta), ylim=c(-theta,theta),