Hello,
We use and develop the astGUIclient suite. It is Open-source(as in GPL) and
offers Inbound and Outbound call center functions with reports, ACD,
monitoring, recording and very basic IVR scripts. Complex IVR functions need
to be custom programmed within Asterisk but that is not really that
Hello,
We tried a Dual Processor AMD system last year and were greatly
dissapointed. A single P4 system was much cheaper and actually outperformed
the Dual AMD.
Is anyone actually running an octal AMD system out there?
In our experience having more processors doesn't really matter on the x86
With a RBS(Robbed-bit) T1(in other words, not a PRI) the CallerID(Called
ANI) is sent in the digits and come across in Asterisk as part of the
extension. It is not standard, you do need to ask for it to be enabled and
you usually have to specify how you want it.
A standard way of receiving ANI on
You
should be able to get the full channel values by doing a "Action:
Command Command: Show Channels" and
picking your SIP extension out of the list it gives you of active channels. Then
you can take that and the channel that you are currently connected to, also
taken from the "Show
Hello,
We created an areacode, country code, GMT offset, country code file for the
astGUIclient project last year. I believe it has all Mexican area codes in
it. If you find any errors we've love to hear about it.
http://astguiclient.sourceforge.net/phone_codes_GMT.txt
Hope this helps,
MATT---
I've been trying to get a test G5 in our office from Terrasoft for the last
few months. They are very interested and we have offered to give them a
deposit for the machine while we test it for a week, but they don't seem to
have a machine that they want to send us. Anyone else know of another
You're going to have to go a little more in depth into what you are doing in
this call center.
- Are you going to be doing inbound or outbound? (if so how much of each)
- What kind of phones are you planning on using?
- What is the maximum number of concurrent conversations you plan on having?
-
Hello,
I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma stated to me(with evidence I might
add)they have in the past and continue to today contribute code to GPL
Asterisk. It doesn't say so on their website but their developers
Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs into
their boards. Not sure how Sangoma would feel about this idea though.
MATT---
-Original Message-
From: Matthew Boehm [mailto:[EMAIL
.
On Thu, 31 Mar 2005 11:00:48 -0500, mattf [EMAIL PROTECTED] wrote:
Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs
into
their boards. Not sure how Sangoma would feel about this idea though.
MATT
We target Sipura because they are relatively a small company, the core
developers at Sipura used to work for Cisco and worked on their ATA product
before they started their own company. A small company is much more likely
to try something new with little lead-time. Also access to decision-makers
My Sangoma Experience in Asterisk: 2005-04-07
Having pushed my Digium Asterisk systems to their capacity many times and
figuring out the limits of the Digium hardware I decided it was time to test
an Asterisk-compatible Sangoma Quad T1/E1 card(AFT-A104u) to see if they
live
As an Asterisk-related Open-Source project developer I would very much like
this idea :)
We could have a competition that ends yearly during Astricon at which point
the application is chosen.
But who would judge which is best?
I'm pretty sure that the front runners if this was done this year
Several of these RBS T1s have been here for many years and before we moved
to Asterisk a few pieces of phone hardware we used were not PRI-compatible.
There is also the fact that we still use Channel banks which are also RBS.
We have started a long process of switching to PRIs as our RBS T1
Hello,
This would be software since I still don't see the Digium echo-cancellers
anywhere for sale and don't know how to get one. If Digium wants to send me
one I would gladly test it.
The overall machine load is also affected by the way interrupts are used and
the fact that Sangoma uses
just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.
Let me know if you would like an example and I'll whip one up.
We do this kind of thing in astGUIclient to play DTMF tones automatically in
meetme rooms.
MATT---
A sample would be great. I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
off from 127.0.0.1
So it appears that my variable ${confNo} is not being set, or at least
honored.
Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Keep on bugging the Sangoma guys, I know they are working on several RBS T1
issues right now(They called me Friday to go over a few things) They just
need help from users like you and I to find the bugs in their drivers.
Have you tried any other signalling types other than LOOP?
MATT---
Hello,
The asGUIclient suite has a predictive dialer component to it(VICIDIAL) and
it can function well on multiple Asterisk servers at once using a single
MySQL server backend. It performs on par with several mid-level commercial
dialers that we have compared it to(Nobel, TripleP, DataTel,
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Predictive dialer
mattf wrote:
Hello,
The asGUIclient suite has a predictive dialer component to it(VICIDIAL)
and
it can function well on multiple Asterisk servers at once using a single
MySQL server
This was pretty much fixed several months ago, but on a heavily loaded
system you can get manager API pauses(output just stops and then floods
out all at once) of upto 20 seconds if your Asterisk system is under very
heavy load. Even in those cases the system will not deadlock, so the manager
API
We use this in the astGUIclient to transfer an active conversation(both
parties) to a meetme room:
Action: Redirect
Channel: Zap/73-1
ExtraChannel: SIP/199testphone-1f3c
Exten: 8600029
Context: default
Priority: 1
where 8600029 is a meetme room.
Works very well.
Sadly like most obscure
Hello,
You just need to send a simple Monitor command to the Manager interface to
start or stop recording(Monitor) on a channel. This can be easily
accomplished within PHP or some other basic web scripting language. But you
need to have the full channel name to make the recording work.
Are you
You need to send a Manager command(Redirect Action) to the asterisk server.
(BYou can do this by connecting to the manager API through any kind of
(Btelnet-type connector in any number of programming languages: Perl, C, PHP,
(Betc..
(B
(BTake a look at the WIKI for more info:
(B
There really isn't a solid description of every Manager API function, even
in the source code. And some of the features listed may not work the way you
think they should.
As for Monitor, there really isn't much more than to say that you send it a
channel and optionally a filename and it will
Hello,
You are correct, it's a lot of data for each client to parse through and a
lot of data for the se5rver to be sending out. It would just be easier to
use an AGI to trigger an action on the client computer, or you could just
use astGUIclient which already does what you are trying to do:
Take a look at the VICIDIAL component of the astGUIclient suite. It handles
inbound calls through an independant ACD as well as allowing outbound
dialing from a database of leads.
It can be configured to grab callerID info from inbound calls and populate a
web form that can tie it into your CRM
We record meetme rooms by sending a manager Action to place a call from the
meetme room to an extension that is defined to start recording for a
predetermined amount of time, to end that recording we just send an Action
to Hangup that channel. Been working great for over a year now with over
The 1.4.1 firmware and the 2.6.1 bootrom are also now on
http://www.freedomphones.net/polycom/files/
MATT---
-Original Message-
From: mattf
Sent: Wednesday, January 26, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom
Hello,
When I talked with the VP of VOIP phone sales at Polycom about a year ago,
he was offering a dedicated engineer for the Asterisk community that would
work through issues like people have here. BUT they would ONLY do this if a
reseller came forward and committed to be the Polycom authorized
I'm trying to get a souped-up test machine(G5 Xserve) from Terrasoft to do
some testing in a few weeks. If/when I actually get it I'll certainly post
the results here.
In theory the G5 should mop the floor with the Intel for high-volume
Asterisk Zaptel usage, and I have heard from several
Hello,
Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find
, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5
Hello,
I'm not a astGUIclient user, but I'm puzzled by the following statement:
mattf [EMAIL PROTECTED] wrote
dial
out). Or if you are using Asterisk 1.0.5 simply use the patch mentioned
before to eliminate callerid altering completely.
Thanks Mark!
MATT---
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 12:32 PM
To: 'Nicolás Gudiño'; 'Asterisk Users
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003490
apply the patch: app_dial_CID_nodelete.patch
and the deleting of the original callerid will stop in v1.0.5.
Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.
MATT---
-Original
mattf wrote:
Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.
I have to wonder why the default behavior was changed to this
non-standard usage though; in what situations do we want the CLID/CNAM
of the _recipient_ to be passed to them
Hello,
First, there is a mailing list for the astGUIclient suite:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
As for your problem, If you have everything set up correctly you should just
be able to run the AST_VDhopper.pl script from your Asterisk server to fill
your lead
Hello,
patching v1.0.5 on my system removed the problem for me. But yes it seems
strange that this feature was inserted into a final release with very little
documentation of the wide implications that are caused by the change.
This was corrected in CVS with the addition of a diabling flag for
by GUI do you mean a configuration utility or a User Interface?
MATT---
-Original Message-
From: Jon Gabrielson [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 10, 2005 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] asterisk GUI's that
Use the manager API to send a call from the meetme room to an extension that
does Monitor for a specified period of time. That is how we do it in the
astGUIclient suite and it works great.
; extensions.conf entry:
; this is used for recording conference calls, the client app sends the
filename
;
stop the recording if it is set for a period of time? Eg if
set the period as 30 minutes and the call finishes early will it cease
recording or hold up the line for 30 mins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Wednesday, February 16
Hello,
I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1
line and currently have echocancel=yes and echocancelwhenbridged=yes on
those spans in zapata.conf. I was discussing CPU load with another Asterisk
user and he mentioned that PRIs don't need echo cancelation and that
Hello,
We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
Zap(call comes in) - Asterisk - Zap(call dials out)
The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release
Would enabling Busydetect really help if Asterisk thinks it detects an
On-Hook?
MATT---
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap
We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they
all are running just fine in production environments each handling thousands
of calls a day.
I suppose reliability depends upon what you are using, but for our purposes
they all are very stable. I could do without the
The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/
To convert to/from GSM on Win32 you can use DBpowerAMP:
http://www.dbpoweramp.com/dmc.htm
And for Linux or Win32 you could use Sox of course:
http://sox.sourceforge.net/
MATT---
Hello,
A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at
least not use rpms or the preinstalled perl on the OS. RedHat has done a lot
to screw up how perl works in the last several versions and there are a lot
of angry perl developers that have just given up on the
Hello,
The astGUiclient suite has it's own mailing list for questions like this:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
The easy fix is for you to set PHP globals to on and see if it works like
that first, also you could try making that directory writable.
MATT---
Hello,
You can do either, you can send multiple Originate actions in a long line
without waiting for a response back(although the responses do usually come
back very fast) or you can open multiple connections using each one to
Originate a new call. We use the multiple connection method in the
ActionID does not return in all events related to an
Action sent, sometimes it will just send you a success message and nothing more.
Just try Originating a call from a meetme room over an outside line. You will
get about 150 lines of output and only one message will have the ActionID in it,
as possible) if the calee
phone number is ringing.
Thanks, Tom
--- mattf [EMAIL PROTECTED] wrote:
ActionID does not return in all events related to an
Action sent, sometimes
it will just send you a success message and nothing
more. Just try
Originating a call from a meetme room over an
outside line
We use astGUIclient suite, it has this functionality. Hard or soft phones
SIP, IAX or Zap
http://astguiclient.sf.net
MATT---
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 4:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial
The best way to figure out the manager protocols is through looking at the
manager.c source code and trial and error.
Some things just don't behave the way you think they should, some things are
not fully documented and some actions do not work in certain cercumstances
while others will.
And
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.0
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
Hello,
I think you need to do a little more looking around on the Asterisk
resources and on Google. What you are trying to do is mostly possible if you
have the time, patience and money to follow through with it.
One thing you need to learn is that a great many on this list despise
Hello,
I noticed this as well on a supira connected phone through Asterisk when I
upgraded from CVS 10-10-2003 to CVS 01-10-2004, nothing else changed in the
setup.
MATT---
-Original Message-
From: Steve Dolloff [mailto:[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 12:57 PM
To:
Hello,
I was using the doublehash.patch that Iain Stevenson had created back in
August to change the transfer key from a single hash # to a double-hash
#. It always patches properly, but when I went from CVS 2004-01-12 to
Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to
Hello,
I've had Asterisk installed on HT capable machines in both HT mode(with SMP)
and non HT mode (with non-SMP) and did not notice any differences
functionally between them. The processor load was always less in HT SMP mode
than non HT and I have experienced Asterisk deadlocks in both modes so
? Like, customers
sending me H323 or SIP fax calls and the Asterisk will pass through to
another gateway? Anyone successful in doing that?
Tommy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of mattf
Sent: Monday, January 19, 2004 8:32 AM
To: '[EMAIL PROTECTED
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of mattf
Sent: Monday, January 19, 2004 6:21 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] RE: Latest version of asterisk
Hello,
Our max for a single machine is 40 concurrent SIP - Zap conversations for
about a 12 hour period and over 5000 total
Hello,
You can try the doublehash.patch linked on this bug:
http://bugs.digium.com/bug_view_page.php?bug_id=885
it makes it so you have to dial two hashes ## in quick succession to trigger
a transfer. It works very well, but the problem is that it won't work with
asterisk 0.7.1 and no one
Hello,
Ditto here, it seems to be the worst 9am to 5pm in the USA, any other time
than that messages get posted right away.
Ping times from both of my network connections to digium.com domains are
horrible at 300-700ms but the last hop before entering the digium.com land
is always really good
Hello,
As far as I've heard the IP400 doesn't have enough physical memory to store
all of the files necessary to run SIP. For those familiar with SIP, H323 and
MGCP protocols, SIP files are huge while MGCP takes up very little space and
H323 is in the middle. The IP400 phones were designed for
Hello,
I'd love for the doublehash option(two ## in succession to transfer) to be
available as a config flag. I would think this would be trivial to add to
Asterisk, we just need someone to do it. It would solve one of my major
problems with 0.7.1 and would give another option for those can't use
It's been added to the wiki:
http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script
MATT---
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Friday, January 23, 2004 12:38 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom Reboot Script -
Go ahead and edit the page. I've fixed several little errors on pages that I
didn't create. The voip-info.org Wiki is like the total-open-source Asterisk
manual. Although the total-access may be a problem in the future because all
someone has to do to delete everything is just to register and
Hello,
My reply posts are getting posted right away, but I tried posting a new
message and it hasn't appeared in over 6 hours. What kind of filter do you
have on new threads now?
MATT---
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 28, 2004 9:53
I've tried posting a new message 3 times since midnight last night and it
won't post, but as you can see my replys always make it on very fast. What's
up? am I doing something wrong?
I thought you just had to email a message to [EMAIL PROTECTED]
from your member email account and it would be
Hello,
After battling with the Asterisk Manager interface(and getting it to pretty
much do everything I want to do with it) I thought I'd share my experiences
with those who are developing or are thinking of developing applications
using it.
First here's a list of some of the things the manager
Hello,
After battling with the Asterisk Manager interface(and getting it to pretty
much do everything I want to do with it) I thought I'd share my experiences
with those who are developing or are thinking of developing applications
using it.
First here's a list of some of the things the manager
Hello,
After battling with the Asterisk Manager interface(and getting it to pretty
much do everything I want to do with it) I thought I'd share my experiences
with those who are developing or are thinking of developing applications
using it.
First here's a list of some of the things the manager
This was the third posting (sent at 10:11 AM), you won't see any more, I
promise.
Thanks to Brian for de-spamming my posts.
MATT---
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 29, 2004 10:11 AM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users
Hello,
Did you set the flag in the makefile for zaptel for SMP kernels?
1. I have a couple Snom200 phones on my system running redhat with a P4 HT
and haven't had any issues with horrible sound quality using 711ulaw.
2. As for the speakerphone cutout, that's to be expected, The snom200s are
Hello,
I was referring to the availability of the ExtensionState Action (see the
wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API
), even though I don't actually use it.
For my purposes of status of an extension I wrote an updater script, that
runs outside of Astrisk,
I'd love to have a non-mp3 music-on-hold option. Anybody put this as a
feature request yet?
MATT---
-Original Message-
From: James H. Cloos Jr. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 1:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: sementation fault with
What firmware and sip versions are you using? I have several Polycom phones
on my system right now and I've never had any registration problems with
them.
Instead of leaving the host as dynamic try declaring an IP address(that's
the only difference I see between your sip.conf and mine).
If you
..(repeat until phone stops registering)
David
- Original Message -
From: mattf [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 04, 2004 3:59 AM
Subject: RE: [Asterisk-Users] Minor Registration Problem With Polycom
Soundpoint IP 500
What firmware and sip versions
---
-Original Message-
From: Dustin Knuttgen [mailto:[EMAIL PROTECTED]
Sent: Friday, February 06, 2004 9:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk GUI Client - New verison 0.9
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 05
Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP
kernel and have about 30 channels in conference. Here's the bug listing:
http://bugs.digium.com/bug_view_page.php?bug_id=963
MATT---
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]
Did you have any active meetme sessions at the time of the freeze?
What Asterisk version are you using?
MATT---
-Original Message-
From: Jonathan Biggs [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System freeze
I would like to request an alternative to the mpg123-only musiconhold. I
could live with just about anything that isn't mp3. Just as an alternative
for those of us who have had nothing but problems with mpg123 and would like
to use the functionality of musiconhold.
MATT---
-Original
to use. Are the bytes being read one at a time with
programmed IO? With DMA? Is there a hardware ring buffer being
used?
How about packet filtering on the card? Take a look, the Reltek
is not a bad choise.
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Tue, 2004-02-10 at 07:01, mattf wrote
I had the same problem, Digium sent me a new card and now all is well.
MATT---
-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 11, 2004 1:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stuck TE410P cards
Hello all-
I have 3 TE410P
I'd like to know why too.
I'm using a TE410P card in a dual Athlon XP system right now and it seems to
be playing nicely with the dual Athlons, should I worry about something
going wrong with my TE410P since it is basically the same card as the TE405P
except it runs at a different voltage?
Athlon systems
On Sun, 2004-02-15 at 00:46, mattf wrote:
I'd like to know why too.
I'm using a TE410P card in a dual Athlon XP system right now and it seems
to
be playing nicely with the dual Athlons, should I worry about something
going wrong with my TE410P since it is basically the same card
That is extremely depandant upon what you want to do. First we have to know
what the job of the asterisk server is, will it be an inbound ACD, a SIP
phone PBX, a T1-only IVR, a conference call system, a voicemail system, and
office phone system with H323 phones, etc..
Also, we need to know what
Yep, you can use the asterisk manager interface to initiate Monitor(start
recording) and StopMonitor(stop recording) Actions on any Zap channel(it
won't work on SIP-only or any other VOIP-only calls). Look at the wiki for
more info:
try putting something in for these values instead of leaving them blank in
the phone1.cfg file:
reg reg.1.displayName=2002 reg.1.address=2002 reg.1.label=2002
let us know if that clears things up.
MATT---
-Original Message-
From: James Treleaven [mailto:[EMAIL PROTECTED]
Sent:
The only adapter that I know of that allows you to modify the ring voltage
is the Sipura analog SIP adapter. I was able to get my old fax machine to
answer after jacking up the ring voltage to 90V. http://www.sipura.com
MATT---
-Original Message-
From: Nicolas Bougues [mailto:[EMAIL
I always keep a terminal window open with top running for my asterisk
servers. Since we've had Asterisk in production, for about 9 months, I've
noticed with every platform and every card we've tried that the load average
will be going along at about 0.1 to 0.5 with about 30 channels(15 SIP -
Zap
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Processor load spikes
On Mon, 2004-02-23 at 09:19, mattf wrote:
I always keep a terminal window open with top running for my asterisk
servers. Since we've had Asterisk in production, for about 9 months, I've
noticed with every platform
load spikes
On Mon, 2004-02-23 at 18:42, mattf wrote:
Thanks for the response. I plan on trying Slackware on my backup/test
asterisk server when I have a new backup server ready in a few weeks. I've
noticed in some database machine testing that Slackware starts up in about
half the time
Take a look at my GUI app:
http://sourceforge.net/projects/astguiclient/
It'll run on Linux and Windows, it's written in perl and it'll list every
channel(Zap/SIP/Local) that is active on your system updated every second.
You can also do a lot of other things with it too.
We've been using it
It actually has components that run on the Asterisk server as well as a GUI
app that runs on the desktop(Win32 or Linux).
astguiclient was really designed for a larger environment than most of the
Asterisk client apps that are out there, and it is not as easy to set up. It
was also initially
I had my first production system Asterisk crash today with no apparent
reason for the crash. This was on a production server that hasn't had
anything changed on it for 3 weeks and is rebooted every night. The load was
low when the crash occured and the logs give no indications as to what
caused
, 2004 11:45 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Core dump crash
mattf wrote:
I had my first production system Asterisk crash today with no
apparent reason for the crash. This was on a production server that
hasn't had anything changed on it for 3 weeks and is rebooted every
What processor? What distro? What kernel(SMP, non-SMP)? What are you doing
specifically with your Asterisk system?
My personal experience is that I've had high-load crashes anywhere from 6.0
to 8.0 on a SMP P4 Single-processor with HT enabled.
The load isn't the best indicator of when a crash is
compiling zaptel is enough,
I have one and it loaded up just fine on RH9, make sure you modprobe zaptel,
modprobe wct4xxp and then ztcfg -vvv before you try starting the first time.
MATT---
-Original Message-
From: Tomica Crnek [mailto:[EMAIL PROTECTED]
Sent: Friday, March 05, 2004
Hello,
We've made another release of the astguiclient suite of client GUI
interfaces for Asterisk(please note this is not a config file editor).
This release has a lot of bug fixes and includes the VICIDIAL
one-call-at-a-time dialer. We have also finished our new website complete
with a new
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