Jarga Jallow wrote:
> I need help with my grand stream GXP2000 phones they keep freezing
> randomly. Any ideas?
What firmware revision?
Want to buy a used one from me? I'm trying to standardize on Sipura 841s,
and I have
one GXP2000.
> Jarga
--
Chris 'Xenon&#
I started out a few years ago with some SPA-841 sets, because the
Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more
call appearances, and I didn't want just the 4 max that the SPA offered. As
it turns out, with the greater flexibility of VOIP, I don't need 'dedicated'
CAs
Hi all. Newbie to the list, been using VOIP with Sipura & Grandstream
hardphones for a few years, via a VOIP service provider (who I won't name
here). I haven't stepped up to running my own Asterisk box yet, because of
poor reliability of our Internet connection during non-business hours, but
I'm
Dinesh Nair wrote:
On 02/17/06 08:51 BJ Weschke said the following:
On 2/15/06, Kevin Hanson <[EMAIL PROTECTED]> wrote:
I am using an Asterisk box as a mini-softswitch and have run into a
minor (hopefully) road block. The far end switch requires CIC (Carrier
Identification Code) in t
I am using an Asterisk box as a mini-softswitch and have run into a
minor (hopefully) road block. The far end switch requires CIC (Carrier
Identification Code) in the SIP invite like:
INVITE sip:+18001234567;[EMAIL PROTECTED];user=phone SIP/2.0
^^^
I
Ben Higley wrote:
I do not see the '1' in front of the number you are trying to dial
I didn't have the ownership set to asterisk, so I changed it- but it
still doesn't work.
[EMAIL PROTECTED] tmp01]# nano 1.call
[EMAIL PROTECTED] tmp01]# chown asterisk:asterisk 1.call
[EMAIL PROTECTED] tmp
Viggiani Domenico wrote:
You can't *copy* the file into the outgoing directory. You
must *mv* it.
First, remember also to check that call file is owned by the asterisk user.
Mimmus
___
--Bandwidth and Colocation provided by Easyn
Peter Svensson wrote:
On Mon, 24 Jan 2005, Andrew Kohlsmith wrote:
As far as integrating with a website or database -- that is a piece of cake.
Your backend logic just determines when a call is needed and gerates the
approprate .call file. Just remember to create it in /tmp or
I've sent a message to the list Asterisk-user 6 hours ago and it's still
not showing up.
I've seen others with questions about the availability of the list.
It may be something the moderators want to check out.
___
--Bandwidth and Colocation provided
How would one go about setting up the wakeup feature of Asterisk to NOT
call an extension, but to call a phone number?
My setup works great for wakeup on local extensions, but I'd like to set
it up to call external phone numbers automatically and play a specific
sound file (to remind people of
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the same.
i agree with you, it seems to be a bug which i've submited before (bug
#5281) but it's now closed by bug marshals!
It's not closed. It's suspended waiting input from you:
"Closing until
Alvaro Parres wrote:
Hi list...
I have been testing the hint extension. And i detect
that when i have in the sip.cfg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.
If this is a bug or not ??
Thanks.
---
Joseph Rothstein wrote:
Greetings to all,
I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.
Does anyone know for sure if the hint function works properly in 1.0.9?
If anyone has gotten this
Mike McMullen wrote:
Hi All,
This is my first posting and if am asking dumb questions please
let me apologize. I'm in no way a telephony or pbx expert. I have
tried googling for answers but can't seem to find the answers I need.
Probably because I'm not using the right words.
I have installed
Kevin Ragsdale wrote:
Has anyone tried the newest Polycom firmware? The release notes
indicate they have added support for a new BLA draft.
TIA,
Kevin
Does anyone know if this new firmware support watching more than 7
buddies at a time?
Cheers,
Kevin
--
Optimacy Communications, LLC
http
Tom Vile wrote:
Everyone,
I have a TDM400 REV I Ver 1 board and am having an issue with 1 of the
4 FXO channels. FXO 1 always has clicks, pops and echo but the others
are crystal clear all of the time. The card is on its own IRQ zztest
shows 100% to 99.98% and is getting 1000 int per second.
BJ Weschke wrote:
On 11/19/05, Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
Hi there,
as you probably know Asterisk 1.2 comes with a new Dial() behaviour on
busy. However I find conflicting documentation - which one is correct?
j - Jump to priority n+101 if all of the requested channels
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 11/16/2005 09:46:17 PM:
> Hi,
> Yes, I'm using wav for my recording and the file is quite
large.
I too am using WAV files because of the volume issue: WAV files are
shifted two bits louder than any other format. (slight detail
Jonathan k. Creasy wrote:
What context are your phones in? (context= in sip or iax config)
If your phones are in the local-users context, they will be able to dial
numbers found in local-users, extensions and local.
If your phones are in the long-users context, they will be able to dial
numb
Steven Ringwald wrote:
I apologize if this question has been asked before. Did something
change with the behaviour of the 'sip show inuse' command between
1.0.9 and 1.2-rc1? I used to be able to see a list of extensions and
the number of in/out calls. Now it just reports:
asterisk*CLI> sip s
Klaus Darilion wrote:
Hi!
I read in the archive a lot of problems using the Dell 1850 servers
and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has
tried the Dell Poweredge 850 series and can report some experiences?
btw: Does somebody knows why there are problems with 1850 bu
Andrew Kohlsmith wrote:
On Tuesday 15 November 2005 13:33, Kevin Hanson wrote:
Can you explain the earlier post in this thread that seems to imply that
Digium support thinks differently?
Yes. Digium support was misinformed. Kevin Fleming of Digium replied to you
about that in this
Andrew Kohlsmith wrote:
On Tuesday 15 November 2005 10:49, Kevin Hanson wrote:
Haven't tried swapping slots yet. And, no, bios does not allow IRQ
assignment.
Well, as Kevin Fleming and I both pointed out, lspci -v output is irrelavent,
as it is the IRQ assignments before t
Eric "ManxPower" Wieling wrote:
Kevin Hanson wrote:
Forrest W Christian wrote:
Someone mentioned IO-APIC in this thread, and it lit up a different
part of my brain for me to be able to search around the net and find
that at least under CentOS, you have to be running a SMP ker
Rusty Dekema wrote:
How do you get your system to use IO-APIC style interrupts? I am
running linux-2.6.14 and have enabled "Local APIC support on
uniprocessors" and "IO-APIC support on uniprocessors" in the kernel
options, but /proc/interrupts says that everything is using XT-PIC. I
am runnin
Forrest W Christian wrote:
Someone mentioned IO-APIC in this thread, and it lit up a different
part of my brain for me to be able to search around the net and find
that at least under CentOS, you have to be running a SMP kernel (even
on a UP machine) to be able to get the IO-APIC functional
Sean Cook wrote:
To Change the Messages Button for 1 touch dial:
For one touch messages you also need to set up.oneTouchVoiceMail=1 like
the following in sip.cfg:
up.oneTouchVoiceMail="1" up.welcomeSoundEnabled="1"
up.welcomeSoundOnWarmBootEnabled="1" up.localClockEnabled="1
Tom Rymes wrote:
On Nov 15, 2005, at 10:02 AM, Kevin Hanson wrote:
Kevin Hanson wrote:
If what you say is true, then I'm hosed. I've got six things
sharing IRQ 255 according to lspci -vb:
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI
Controller #1
Intel C
Kevin Hanson wrote:
If what you say is true, then I'm hosed. I've got six things sharing
IRQ 255 according to lspci -vb:
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI
Controlle
Piotr A. Sygula wrote:
Having an issue with shared interrupts with a setup with 3 TDM400P cards,
and dealing with Digium support, I'd like to share with the list the fact
that Digium claims the following:
The following output from "lspci -vb" (shows IRQ from PCI-bus perspective,
rather than the
harry gaillac wrote:
you mean than when the status of your subscribers
change they are notified
busy, away, ...
harry
Yes. When an extension is in use, the light on the receptionist phone
next to the speed dial position holding that buddy lights up and the
buddy icon changes to a "do not
harry gaillac wrote:
Do you configure BLA bridged line appearence for
presence in asterisk ?
Harry
I am using hints in extensions.conf in asterisk combined with buddy
lists / buddy watch on the Polycoms. It works pretty well in Asterisk
1.2. I'm having a couple of issues that are outlined i
Michael Araba wrote:
I am having the same problems. The polycom phones the 501 or 601 or
301 will list more more than 7 buddies neither will the 601 with an
expansion module monitor more than 7 other phones.
Is there anyone out there who can explain waht is happening. My
reseller can not hel
Anton Krall wrote:
Hey Guys!
I know sip hpones can be configured to disable call waiting but this is for
all call appearances. I was wondering if there is a way to limit outgoing
calls (asterisk -> phone) to a sip phone (techonology) to 1?
Is there any other way of doing this without groups
Rich Adamson wrote:
"Brand new " cards, from a recommended Digium distributor.
What rev? It'll say "Freshmaker Rev x" on the blue board. I *think* it was
corrected around Rev E or F IIRC, but I don't know for certain.
I spoke with Digium and the distributor tech support on it -
Andrew Kohlsmith wrote:
On Saturday 05 November 2005 22:33, Gary Eck wrote:
I have popping with FSO modules only on channel 1 - the other 3 channels
are clear.
That was corrected a long time ago. You must have an older rev TDM400 carrier
card.
-A.
Well, it must be back. I had
Andrew Kohlsmith wrote:
On Wednesday 02 November 2005 11:17, Kevin Hanson wrote:
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
Do I need to set "faxdetect=both" in z
Morel Mosolff wrote:
Hi,
sorry - I know that problem is not directly related to asterisk but mabe
someone can help anyway.
After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is
mostly not possible to dial numbers with leading zeros like 0018...
If you do so you see on
Andrew Kohlsmith wrote:
On Wednesday 02 November 2005 11:17, Kevin Hanson wrote:
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
Do I need to set "faxdetect=both" in z
Robbie Hughes wrote:
Does anyone know if the intel e7230 chipset in the new dell poweredge
sc430 and poweredge 830 servers is compatible with the te110p and
tdm400p cards?
I know there were problems with previous generation dells, but I've
read that these work fine. Can anyone confirm this?
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
I believe this will work, but wanted to know if anyone has done this.
Do I need to set "faxdetect=both" in zapata.conf?
I am assuming that Asterisk will b
Ric Moseley wrote:
I have the digital receptionist answering when an incoming call comes
in to the main trunk. How do i get it to answer after so many rings
or seconds. It seems to pick up on the first ring. I know how to
modify the time it takes to go to voice mail but I do not see an
obv
Jerry Geis wrote:
Does anyone know about sending SMS messages to a sprint pcs phone.
Can you give me a few details. Thanks,
Jerry
___
-
Not sure if this is what you're looking for, but we use this in our
voicemail.conf:
@messaging.sprintpcs.com
Olle E. Johansson wrote:
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten => 1234,hint,SIP/1234 works,
exten =
Howard Leadmon wrote:
Hello all,
I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow
something is not quite right with my vMail setup. I would have sworn this was
all working, but maybe I was just dreaming.
Anyway here is what is happening, say I am on extension 2
I have a customer that has a Dell SC1420. They want to use that as
their Asterisk server w/ a Digium TE110P.
Digium's website doesn't mention this server one way or the other in
their compatibility list. Is there anyone out there using the SC1420?
If so, any problems?
Cheers,
Kevin
__
Trevor Peirce wrote:
Kevin Hanson wrote:
The Polycom 600 supports Shared Call Appearance Signaling. The
Polycom documentation states:
...
"The phone supports shared call appearances (SCA) using the
SUBSCRIBE-NOTIFY method in the 'SIP Specific Event Notification'
fram
The Polycom 600 supports Shared Call Appearance Signaling. The Polycom
documentation states:
"Incoming calls can be presented to multiple phones simultaneously.
This feature is dependent on support from a SIP server which binds the
appearances together logically and looks after the necessary
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
A client wants an incoming call to ring at the receptionists desk and
also at their desk. If the receptionist is in it will be answered there
and put on hold followed by a "Joe, you
I have a customer that wants to run Asterisk in their Italy office.
They have 3 BRI connections and 2 analog (POTS) lines.
I am looking to see if there are any success stories out there with a
similar configuration. Right now I plan on using a Digium TDM02B for
the analog lines, but I don't
Bill Ford wrote:
Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
looks
Altus Snyman wrote:
Good day all
I have a Fedora core 3 installation
Is there any hassles with asterisk?
Thanks
Altus
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See subject:
Does [EMAIL PROTECTED] support Dual-Processor installations? I didn't see
anything on the sourceforge page clarifying that. I suppose they could
leave out the SMP version of the Linux kernel to save space on the .iso?
I had trouble some time ago installing version .5 of [EMAIL PRO
Whenever I try to install [EMAIL PROTECTED], I get this error at about 43%
There was an error installing rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space, and/or hardware
problems. This is a fatal error and your install will be aborted.
Please verify your media a
- Original Message -
From: "Stuart Ford" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, March 10, 2005 8:02 AM
Subject: RE: [Asterisk-Users] Re: Grandstream Message button
>Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004)
accord
Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
Sent: Friday, February 18, 2005 6:04 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] festival weather report
This script was developed by Mark Johnson.
All I did (
My broadvoice works perfectly. I am using a standard registration
string, however. Not the funky one broadvoice says to use. I can make
outbound and receive inbound calls over broadvoice.
I'm using AMP also.
register=phonenumber:[EMAIL PROTECTED]
sip.conf:
[952XX]
username=952XX
type
- Original Message -
From: "Dinesh" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, March 08, 2005 4:51 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello roger,
Yes I can ping sip.broadvoice.com and
- Original Message -
From: "Dinesh" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, March 08, 2005 3:39 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello all,
I am sorry for posting again, but when a
- Original Message -
From: "Gabriel Gunderson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Monday, February 28, 2005 4:49 PM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
Am i not providing some helpfull info?
see bottom
- Original Message -
From: "John Millican" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 28, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:45 pm, John Millican
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, February 25, 2005 9:20 AM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
Great! It works now!! Thanks so much.
What was it that made it work? Share the
ove type=peer from [Broadvoice] in sip.conf incoming calls
work great but outgoing calls don't work. If i leave type=peer in
there, outgoing calls work great but incoming calls get routed to
Broadvoice's Voicemail . . .
Roger Hanson wrote:
- Original Message - From: <[EM
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice
I have configured asterisk with the AMP php configuration utility. I
am able to make outgoing calls through broadvoice but incoming calls
a
05 10:20 AM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on
setup.
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my do
;Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 12, 2005 4:50 PM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on
setup.
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (
I've downloaded 2x and burned 2 cds and get an error "invalid compressed
format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
- Original Message -
From:
No problems here - works fine.
- Original Message -
From: "Joseph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 09, 2005 8:17 AM
Subject: [Asterisk-Users] IAX <=> FWD down again?
Can anybody confirm if IAX on FWD is down ag
- Original Message -
From: "Siju George" <[EMAIL PROTECTED]>
To:
Sent: Monday, February 07, 2005 11:34 PM
Subject: [Asterisk-Users] Best OS for Asterisk--newbie!!!
Hi all,
Could some one tell me which OS is best suited for installing Asterisk
at present??
I had planned to install it on
BTW, I use AMP as well.
Dean, et. al.
Here's my portion of sip.conf
[952nnn]
type=friend
secret=passwordhere
regexten=952nnn
insecure=very
host=sip.broadvoice.com
fromuser=952nnn
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes
Remember, the password you n
- Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 03, 2005 8:37 PM
Subject: RE: [Asterisk-Users] BroadVoice Help
Skamp,
Can you post your configs to help out the people who cant get it to
Hi community,
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
Thanx, Hanson
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Sorry, I don't know what happened to my line returns. Here's that
message readable.
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to mudd
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to muddle my way
through getting that working (although that may be questionable) but
is
it feasable to
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to muddle my way
through getting that working (although that may be questionable) but
is
it feasable to
e asterisk tarball... edit the file, move it to
/var/spool/asterisk/outgoing and it'll dial and connect de callee with
the extension of your choice...
Greetings
On Mon, 24 Jan 2005 02:57:13 -0600, Roger Hanson <[EMAIL PROTECTED]>
wrote:
I'm trying to get a script working on a websit
I'm trying to get a script working on a website to send out automatic
email reminders to customers reminding them monthly to change furnace
filters. I haven't got one running successfully, yet.
That made me think - could it be done with a phone call using Asterisk?
A monthly automated phone ca
G, CONNECT
message which are replied by a CONNECT ACKNOWLEDGE message. =>
connection established
Please find my trace below,
thanx in advance,
Hanson
~-- Executing CallingPres("SIP/8-66ac", "1") in new stack
~-- Executing NoOp("SIP/8-66ac", "0") in new
%20oh323%20channels
Hope this helps,
Hanson
Julio Tejera wrote:
| Hello:
|
| I have some h323 gateways (Planet VIP400FO/FS) and I want to use it
| on Asterisk ...
|
| I have some questions about * and h323
|
| - Can I use Asterisk as an h323 gatekeeper ? - If so, where I can
| find any related info about
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have encountered an similar error with my E1 card. I would propose
to check your timing source!
Chris A. Icide wrote:
| I have a system with two 4 port T1 cards, with 5 PRI's configured.
| Each PRI is configured as an individual PRI and belongs to it
I'm using CentOS - which is another Red Hat Enterprise clone, like WBEL
www.centos.org
I've had no problems of any kind with the OS
- Original Message -
From: "Imran Sadiq" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 11, 2005 9:58 PM
Subject: [Asterisk-Users] What is the best and easie
Is the meeting still on for Saturday 1/8/05?
11:30am at 2375 University Av W STE120, Saint Paul.
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I saw the post on the wiki a last month stating the meeting was this
Saturday. Is that confirmed? Still on for 1/8?
Roger
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- Original Message -
From: "David" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 04, 2005 9:04 PM
Subject: [Asterisk-Users] Where to start. {Scanned}
Hello All, Yep I'm a newbe.
I'm just started to play with asterisk.
What I have
Redhat Fedora Core 2 (New install)
3 X100P cards.
I insta
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
put this in sip.conf
;externip = x.x.x.x; Address that we're going to put in outbound
SIP messages ( official ip address)
;localnet=10.0.0.0/255.255.255.0; if we're behind a NAT
add
nat=yes to every sip account which is behind of NAT
- Original Message -
From: "Tomasz Chmielewski" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Wednesday, December 01, 2004 4:42 AM
Subject: [Asterisk-Users] software phones for Asterisk - is there a
list?
Hello,
Is there a l
- Original Message -
rom: "dean collins" <[EMAIL PROTECTED]>
To: "Oliver Stone" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Wednesday, November 24, 2004 9:39 AM
Subject: RE: [Asterisk-Users] asterisk gui?
Oliver, I've offered
- Original Message -
From: "Eric Hall" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, November 23, 2004 9:18 AM
Subject: RE: [Asterisk-Users] Spandsp and Asterisk
I did that
[EMAIL PROTECTED] apps]# patch < Makefile.p
Has anyone also run Bastille on the Asterisk pbx?
Here's the link: http://www.bastille-linux.org/
It's a Linux hardening add-on. I was wondering if it'd mess up my
Asterisk installation if I also installed Bastille, if it was a good
idea to install it and work through the problems that may aris
Original Message -
From: "Chris TenHarmsel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, November 22, 2004 3:18 PM
Subject: Re: [Asterisk-Users] asterisk gui?
It kind of depends.
There's AMP, but it's set-up is non-
I have a Grandstream 101 phone on my desk. I also use a Microsoft
Wireless Optical mouse.
When I'm using the phone, the mouse doesn't work very well - herky jerky
movement. If I move the phone away from the mouse and receiver, they
work fine again - otherwise I need them very close together to
I've also seen my 101 try to display 'grandstream', but with missing
letters.
I think it showed "gra str m" once - all lowercase letters.
At least the display is capable of crude letter display. Getting it to
work the way you want is another matter altogether.
- Original Message -
- Original Message -
From: "Adam Holt" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Hi,
I have a Grandstream ATA today connected to my 750k broadband
connection via an older router / firewall that doesn't have any QoS /
ToS capability. It works fine apart from the obvious problem of wh
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