Chris Mason (Lists) wrote:
The only thing I'd probably lose is the ability to do faxes! So I am
going
to investigate that further first!
Havn't doen that in years - an online fax service sends me my faxes by
email and I sent out faxes through them, not that I ever do that.
The
Lee Jenkins wrote:
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and
then issue the no argument warning on
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk
Sorry, part of the email got chopped of.
Hardware specs are:
AMD A64/3500+ CPU: Desktop Athlon64
Asus A8N-SLI Deluxe Athlon™ 64 S939 NVIDIA nForce(r)4 SLI™ PCI
Corsair TWINX1024-3200XL 2x 512MB 1024MB Dual Channel Optimised Pair
Western Digital WD1200JD/PD-SATA 120GB Serial ATA 150Mbps 7200rpm
On Tue, 2007-05-15 at 22:29 +0200, Francesco Peeters (Asterisk) wrote:
In NL you actually can ditch the telephony and keep the ADSL...
My ISP even gives emergency access if you transfer your main number to
their SIP service.
Here in France you can also move to ADSL only, what I found really
Adam Lovegrove wrote:
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with
On Wed, 16 May 2007, Stephen Bosch wrote:
Chris Mason (Lists) wrote:
The only thing I'd probably lose is the ability to do faxes! So I am
going
to investigate that further first!
Havn't doen that in years - an online fax service sends me my faxes by
email and I sent out faxes through them,
-- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in
new
stack
-- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(mISDN/2-2,
recordingcheck|20070516-140757|1179288477.1037) in
new stack
-- Launched AGI
Hi,
has anyone managed to get hudlite server working on a Debian Etch based
installation of Asterisk 1.4?
So far I managed to eliminate all error messages, but the process is
killed directly after starting the hudlite server without showing any
error messages.
I would be very happy if
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems with
older versions of Xen, but only with ztdummy timing and on DomU virtual
On Wednesday 16 May 2007 03:47, C F wrote:
I use asterisk -rx database put value if you are trying to batch it
from windows you can use plink
This will be VERY slow. Other options might be writing to the asterisk socket
(I heard it's not that reliable). But again, this will be a problem on
Hi all,
I have 1 Card Digium TE412P and 2PRI E1.
I have more problems with drops lines. The asterisk log is this:
May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event:
Alarm (4) on Primary D-channel of span 1
May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan:
On Wed, 16 May 2007, Stephen Bosch said something to this effect:
The fax-to-e-mail services charge as much as the telco does for a
business line, sometimes more (at least, the ones I can deal with in this
area). Better to set-up hylafax, IMHO.
Not necessarily, except perhaps in cases of
Hi,
Is it possible to use Asterisk to record or monitor all conversation
on standard PSTN PBX ?
ASLAY
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Stephen Bosch wrote:
The fax-to-e-mail services charge as much as the telco does for a
business line, sometimes more (at least, the ones I can deal with in
this area). Better to set-up hylafax, IMHO.
http://www.maxemail.com/fax/fax-lite.html
$24/annum.
--
Chris Mason
(264) 497-5670 Fax:
On Wed, 16 May 2007, [EMAIL PROTECTED] said something to this effect:
Is it possible to use Asterisk to record or monitor all conversation on
standard PSTN PBX ?
I imagine with enough work you could use it as a pass-through tap by
taking FXO trunks, barging into the media and then turning
Hello all,
I want to use Asterisk with the SRTP patch from
http://bugs.digium.com/view.php?id=5413 .
I'm confused to create the certificates for it.
Can anybody help in such question?
P. S. I've created the pem files and renamed it to
* ${astetcdir}/asterisk.crt
* ${astetcdir}/asterisk.key
*
Yes.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, 16 May 2007 6:06 AM
To: asterisk-users@lists.digium.com
Thank you all, I've got it.
On 5/16/07, Alexandr Olekhnovich [EMAIL PROTECTED] wrote:
Hello all,
I want to use Asterisk with the SRTP patch from
http://bugs.digium.com/view.php?id=5413 .
I'm confused to create the certificates for it.
Can anybody help in such question?
P. S. I've created the
Hi,
Connect you Asterix box in the middle of call-flow and you will be able to
record all calls.
PSTN - Asterisk - Legacy PBX - Phones
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP ServicesSolutions
MOR - FREE Open Source billing for Asterisk PBX
-Original Message-
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten = 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten = _X.,1,Set(CALLERID(name)=Hotline)
exten =
There are many past posts stating that AMI is not stable when multiple
client computers are allowed to connect, particularly when connections
are dropped.
Has much progress been made on this? Is it more stable now than in the
past?
As of what versions were these issues improved?
Is it
I have a remote user with Eyebeam on a laptop. Internet connectivity
seems good, there is no packet loss to that location from the PBX.
Everytime the user starts eyebeam, the application tries to register.
Asterisk accepts the registration but the reply never gets to the client
application, so
Hi all,
I am seeing a strange problem with Asterisk queue. I am not sure if it's my
configuration which is wrong or there's something with Asterisk.
I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i
tried to call the extension number directing to the queue, the MGCP
Thank you!
I solved my problems and thanks to your useful link I have now a better
understanding of how Local channels work.
Federico
--
Federico Cabiddu RD Software Engineering
Abbeynet S.p.A. - www.abbeynet.com http://www.abbeynet.com
phone: +39 070 2339349
Hello Jeremy,
We have implemented HA systems in the past for numerous clients,
call centers, etc, where reliability matters. We can definitely offer
you a bid on this, but I would like to speak with you a bit first to
nail down the requirements. What would be a convenient time and
Hi,
I have a problem with busy tone detection.
the problem is busy tone with the different length tone and silence! Means:
Busy tone = 400/400,0/345,400/230,0/520
400 on
345 off
230 on
520 off
Repeat
I try in Zapata.conf to enable busy tone detection by this way
busydetect=yes
Hello,
I've a Problem with my CDRs. The clid and src Fields are empty, if a
hangup inside my DeadAGI Program occurs. It makes no difference if the
AGI Program or the caller initiated the hangup.
The Problem doesn't exists, when i use the normal AGI Application.
Further, this Problem started
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote:
Hi all,
I have 1 Card Digium TE412P and 2PRI E1.
I have more problems with drops lines. The asterisk log is this:
May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event:
Alarm (4) on Primary D-channel of span 1
May 16
Per Jessen wrote:
Lee Jenkins wrote:
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and
then issue the no
Hi
I've bought the Digium g729 codec and have installed it correctly (I think)
voip*CLI show g729
0/0 encoders/decoders of 3 licensed channels are currently in use
if I do an echo test either from a sip Cisco 7960 or another hard phone
(unbranded)
using g729 it sometimes works and sometimes
Anyone have any ideas how this might work? Any experience doing this?
Is there any way to play a file on a channel from the Manager API
(other than from Originate)?
This question was asked by someone else on the ast-dev list and the
only advice given was that Redirect was the solution. I
On 15 May 2007, at 20:05, Matthew Fredrickson wrote:
Are you sure that you set the T1/E1 jumpers on the board correctly
for E1 mode?
Matthew,
Ah .. that old chestnut ! , thanks - I did not actually install the
card myself (however I should have checked - schoolboy error.)
Thanks again.
If it's not stable what needs to be done to improve this? What are the
issues? What are the alternatives (eg is Adhearsion an alternative here)
I am about to start looking into a project that requires every user to
have AMI access so looking to fund development in this space.
Regards,
2007/5/15, George Pajari [EMAIL PROTECTED]:
If you have the clipping issue, make sure you get HPEC version 8.2
from Digium.
Note, however, that we have observed stability issues with HPEC 8.2 (two
kernel panics in two weeks since installing HPEC). (The 9.00 version had
such severe
Hi,
I am currently building a 1.4.4 Asterisk box for a client and they
are interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have seen the
Junghanns* version but I am not keen on the
Hi List,
Ive got a few * boxes connecting together, one box is doing
AgentCallBackLogin() and then the 2nd box is holding some phones at a remote
site. I have users login to the main box and * shows the user is logged into
a extension that resides on the other box, problem is, when I go to make
Hi,
has anyone managed to get hudlite server working on a Debian Etch
based installation of Asterisk 1.4?
So far I managed to eliminate all error messages, but the process is
killed directly after starting the hudlite server without showing any
error messages.
I would be very happy if anyone
On 5/16/07, TienSen Chong [EMAIL PROTECTED] wrote:
Hi all,
I am seeing a strange problem with Asterisk queue. I am not sure if it's my
configuration which is wrong or there's something with Asterisk.
I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i
tried to call the
On 5/16/07, William Moore [EMAIL PROTECTED] wrote:
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote:
Hi all,
I have 1 Card Digium TE412P and 2PRI E1.
I have more problems with drops lines. The asterisk log is this:
May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got
Hi,
New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with
which I could easily edit Asterisk config files.
It seems Kate provide this type of service but I couldn't find anything
specific to Asterisk (unlike vim)
What's your advice ?
Best regards
The issue has more to do with the sheer amount of data passed to the client
from within the Asterisk application when you have 50-100+ clients connected
to the AMI on full output mode. Running a system with FreePBX/Trixbox
especially generates vast amounts of output that has to be generated on
On Wed, 16 May 2007, Matt Brown wrote:
Hi,
I am currently building a 1.4.4 Asterisk box for a client and they are
interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM
modules or higher) for use in the UK. I have seen the Junghanns*
I feel I'm doing something obviously wrong here and will kick myself when I see
the answer!!!
The scenario:
SIP phone - Asterisk1 - IAX - Asterisk2 - IAX - Asterisk3 - PSTN
So I place a call from the SIP phone. A1 picks it up and forwards it to A2
which forwards it to A3. A3 sends the call
It will take me a few hours to write a syntax highlighter for kate. But if I
do, I can commit this for KDE 3.5.8 and KDE4.
On Wednesday 16 May 2007 17:12, Olivier wrote:
Hi,
New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with
which I could easily edit Asterisk config
On May 16, 2007, at 7:43 AM, William Moore wrote:
On 5/16/07, Oscar Atienza [EMAIL PROTECTED] wrote:
Hi all,
I have 1 Card Digium TE412P and 2PRI E1.
I have more problems with drops lines. The asterisk log is this:
May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got
the forst problem you have, you need to los the meetme module, and second
one is a timer, for that you can use ztdummy, compiling the zaptel driver.
Regards
On 5/7/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi,
I'm not sure, but MeetMe needs some timer module from zaptel project.
Try read about
why not do it via an snmp interface?
If you spend the time building an solid snmp base you would open up
for an easier world of custom gui's as well as possibly some cleaner
ties into an nms infrastructure.
On May 16, 2007, at 10:14 AM, Matt Florell wrote:
The issue has more to do with
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone =
We use instead sip gsm gateway, with 4 gsm modules from 2N. It should
work in UK as well.
Jorge
Matt Brown wrote:
Hi,
I am currently building a 1.4.4 Asterisk box for a client and they are
interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad
Chris Mason (Lists) wrote:
Stephen Bosch wrote:
The fax-to-e-mail services charge as much as the telco does for a
business line, sometimes more (at least, the ones I can deal with in
this area). Better to set-up hylafax, IMHO.
http://www.maxemail.com/fax/fax-lite.html
$24/annum.
I
Quoting Bryan Laird [EMAIL PROTECTED]:
why not do it via an snmp interface?
If you spend the time building an solid snmp base you would open up
for an easier world of custom gui's as well as possibly some cleaner
ties into an nms infrastructure.
you have my vote on that implementation
On Wed, May 16, 2007 at 02:17:11PM +0100, Matt Brown wrote:
I am currently building a 1.4.4 Asterisk box for a client and they
are interested in GSM functionality.
Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have
This might be of some assistance:
http://www.voip-info.org/wiki/view/vim+syntax+highlighting
- sf
Olivier wrote:
Hi,
New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor
with which I could easily edit Asterisk config files.
It seems Kate provide this type of service but
J. David Bavousett wrote:
Problem A: Dialing in. If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context. So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute
François Delawarde wrote:
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems with
older versions of Xen, but only with ztdummy
Olivier wrote:
2007/5/15, George Pajari [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
If you have the clipping issue, make sure you get HPEC version 8.2
from Digium.
Note, however, that we have observed stability issues with HPEC 8.2 (two
kernel panics in
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have 50-100+
clients connected to the AMI on full output mode. Running a system with
FreePBX/Trixbox especially generates vast amounts of output that
Do you mean nobody has ever done this before (as I thought before asking
this question to the list) ?
So which tool KDE users are using for this ?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
hi Sir,
Thank very much for your suggestion. But I hope you don't mind giving
me more detail information. Imagine that I have a pbax with
3 incoming PSTN line and I have 10 extentions using stand phone.
The requirement is to capture all conversation either internal or
external
Regards
ASLAY
Octasic SoftEcho works very well for me.
_
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 16, 2007 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HPEC audio clipping
2007/5/15, George Pajari [EMAIL PROTECTED]:
If
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject:
On Wed, 16 May 2007, Stephen Bosch said something to this effect:
Would this still be possible? (All these services have numbers in remote
area codes or have 800 numbers).
Can anybody suggest one that will take a ported number (in Canada)?
That's just something you have to contact the
Hi Chong,
I have no experience with MGCP, but do you see anything in the Asterisk
CLI or the full log while the terminal is supposedly being called by the
ACD?
Another thing you could do is to create a Local/[EMAIL PROTECTED] channel, whee
as the queue memebre you have Local/[EMAIL
I have a customer that has a campground.
Wants to see who's at the gate, remotely, via camera, and talk to that
person through a traditional squawk box and be able to open the gate
remotely from that phone.
He doesn't want to have a separate camera feed, etc, he wants to do it
all on one
in zaptel.conf use fxsks
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not
I use Bluefish, and have developed a syntax-highlighting template for
Asterisk conf files, if you're interested.
CP
Steve Finkelstein wrote:
This might be of some assistance:
http://www.voip-info.org/wiki/view/vim+syntax+highlighting
- sf
Olivier wrote:
Hi,
New to Kubuntu and Linux,
Hi,
Can any one suggest if asterisk-users is the best mailing list for questions
on Digium Iaxy (S101I) hardware, or a different one if not?
I found this link on Digium's site:
http://kb.digium.com/entry/15/120/
However, I assume that if this was the case, all of my Iaxys would click,
and only
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Regards
ASLAY
___
--Bandwidth and
From c|net News:
On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office Communicator 2007.
Hi Olivier,
I would guess that most people aren't running any type of GUI on their
Asterisk box. Running an X server plus some type of window manager adds a
lot of overhead that's completely unnecessary for a server. I just SSH into
the server and use VI to edit the files - The server doesn't
On Wed, May 16, 2007 at 06:21:30AM -0700, Ron McCarthy wrote:
Hi List,
Ive got a few * boxes connecting together, one box is doing
AgentCallBackLogin() and then the 2nd box is holding some phones at a remote
site. I have users login to the main box and * shows the user is logged into
a
Olivier wrote:
Do you mean nobody has ever done this before (as I thought before
asking this question to the list) ?
So which tool KDE users are using for this ?
I use vi(m).
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user -
On Wednesday 16 May 2007 11:47 am, Olivier wrote:
Do you mean nobody has ever done this before (as I thought before asking
this question to the list) ?
So which tool KDE users are using for this ?
I am a KDE user, although on Slackware. Have been for many, many years.
Typically you will find
On Wednesday 16 May 2007 11:43, Lee Jenkins wrote:
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have 50-100+
clients connected to the AMI on full output mode. Running a system with
Stephen Bosch wrote:
François Delawarde wrote:
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems with
older versions of Xen, but
2007/5/16, Stephen Bosch [EMAIL PROTECTED]:
Um -- Digium is need of sound captures and debug output for this
problem, because they are having trouble reproducing it in the
laboratory. Perhaps you could help out by making a capture of the
clipping phenomenon using the latest Zaptel (1.2.17) and
You would need two 4-port FXO cards. One to take the 3 outside POTS lines,
and one to generate the 3 FXO lines toward the legacy PBX pretending to be
the far end. Produce a simple dial plan that basically forwards nearly
everything in and out indiscriminately and run MixMonitor() on all of
You guys all sound like you're talking about AstManProxy.
See:
http://www.voip-info.org/tiki-index.php?page=AstManProxy
I'm not saying it is the solution to your problem per se, but I can't
help but think of it when I read the descriptions of what people want
(you even use the word proxy!).
Alex Balashov wrote:
On Wed, 16 May 2007, Stephen Bosch said something to this effect:
Would this still be possible? (All these services have numbers in remote
area codes or have 800 numbers).
Can anybody suggest one that will take a ported number (in Canada)?
That's just something you
Olivier wrote:
Do you mean nobody has ever done this before (as I thought before asking
this question to the list) ?
So which tool KDE users are using for this ?
This KDE user is using vim :P
(What's wrong with vim?)
-Stephen-
___
--Bandwidth and
You could probably make something work, but instead of trying to pound
a nail with a wrench.buy a door entry control system.
*
Adam Moffett
Plexicomm, LLC
[EMAIL PROTECTED]
ph: 866-759-4678x104
*
Smith, Rick wrote:
I
Really, Harvey?
Lines 1-4 are FXS ports, lines 5-8 are FXO... My outside line is
plugged into port 8. Ports 1-4 are inside, and work fine.
--David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wireless
Sent: Wednesday, May 16, 2007 10:52 AM
To:
[EMAIL PROTECTED] wrote:
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Regards
ASLAY
On Thu, 17 May 2007, [EMAIL PROTECTED] wrote:
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is
not very satisfactory and volume too soft
There are dozens if not 100's to choose
[EMAIL PROTECTED] wrote:
Hi,
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Regards
ASLAY
Polycoms seem great to me and according to
On May 16, 2007, at 8:14 AM, Olivier wrote:
2007/5/15, George Pajari [EMAIL PROTECTED]:
If you have the clipping issue, make sure you get HPEC version
8.2
from Digium.
Note, however, that we have observed stability issues with HPEC 8.2
(two
kernel panics in two weeks since
George Pajari wrote:
From c|net News:
On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office
On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have 50-100+
clients connected to the AMI on full output mode. Running a system with
FreePBX/Trixbox
I like and use aastra sets. Very good quality - build and sound wise. If
you're in the GTA (Toronto) area, I could show you
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT peace of mind.
(Sent via Blackberry -
How sad, cnet misspelled Polycom and Cisco didn't make the cut.
On 5/16/07, George Pajari [EMAIL PROTECTED] wrote:
From c|net News:
On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the
On Wednesday 16 May 2007 1:00 pm, François Delawarde wrote:
Thanks again for your help, and sorry if I was not 'that' convinced on
your first answer and sent a mail to Xen user mailing list to check if
they knew that issue (no answer yet). Now I almost believe you a lot. If
I understand well I
On Wednesday 16 May 2007 1:07 pm, Alex Balashov wrote:
You would need two 4-port FXO cards. One to take the 3 outside POTS lines,
and one to generate the 3 FXO lines toward the legacy PBX pretending to be
the far end. Produce a simple dial plan that basically forwards nearly
everything in
Hi Olivier -
Our last trial was so conclusive (every call was affected), we step back to
previous situation without HPEC.
We will do our best to help to solve this (gathering audio captures for
instance) though it will be very hard for me to convince our customer to
try.
Being able to to
How does the Events cache in AstManProxy work?(is there a cache?)
MATT---
On 5/16/07, Martin Smith [EMAIL PROTECTED] wrote:
You guys all sound like you're talking about AstManProxy.
See:
http://www.voip-info.org/tiki-index.php?page=AstManProxy
I'm not saying it is the solution to your
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Polycom, Snom, Cisco, Aastra
___
--Bandwidth and
Hi Matthew -
Can any one suggest if asterisk-users is the best mailing list for questions
on Digium Iaxy (S101I) hardware, or a different one if not?
Digium support is the best resource for this. How old is the affected IAXy?
- Noah
___
From c|net News:
On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office Communicator 2007.
Now we
François Delawarde wrote:
Stephen Bosch wrote:
François Delawarde wrote:
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems
Responses to your inquiry have addressed recording external calls only.
If you want to record internal calls as well, I recommend that you replace
the existing PBX with an Asterisk system as you have a relatively small
installation.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual
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