(Thread moved from -dev)
Hi Ma-
You probably can do this with a 3GHz+ processor, but I don't have experience
with such a busy voicemail server - perhaps someone on here will...
Regards,
Scott Stingel
www.evtmedia.com
of performance and reliability.
-Scott Stingel
www.evtmedia.com
Andrew Kohlsmith wrote:
I respectfully disagree. Sangoma's voice capabilities are no less and no more
mature than Digium's voice capabilities.
I use cards from both Sangoma and Digium. Both seem to work well but (and it
does pain me
would be in order?
It's just bad PR for the company when these things happen.
Please keep us up to date on what happens.
regards
Scott Stingel
www.evtmedia.com
[EMAIL PROTECTED] wrote:
Hey gang,
Just pulled out our brand new $1,500 TE405P and put it into a Dell Poweredge
6450. Nothing. Card
spans, but heavy
transcoding might cut that number down somewhat. Test before you commit!
Hope this gets you started.
Regards
Scott Stingel
www.evtmedia.com
Wai Wu wrote:
Hi all,
My first post here. We will be implementing an * solution with Digium
Pri cards. I have a few questions. First, what
to me
that they are at a point of growth where a significant investment is
required to allow growth to the next level.
Regards
Scott Stingel
www.evtmedia.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
Good question. I've had good luck with the Digium TE405P recently in a
multiple T1 install.
Don't let the discussion scare you too much!
-Scott Stingel
cmould wrote:
Where is this discussion going. I am about to do an installation that
will require t1 interfaces. I am new to the telephone
Bravo - nice writeup Matt!
It concisely captures both the pros and cons.
Seems that we really do have (or are close to having) a second source
now - and all asterisk users will benefit in my opinion!
Cheers
Scott Stingel
www.evtmedia.com
mattf wrote:
My Sangoma Experience in Asterisk
of alphabetical order as the list
requests. My edit was promptly re-edited, so I gave up!
Anyway, just one of the risks of an open Wiki!
regards
Scott Stingel
Emerging Voice Technology
www.evtmedia.com
Sean Kennedy wrote:
Hi folks,
I recently registered with the wiki site to fix a few things I've
THAT ztcfg
commented out, Junghanns issue
#sleep 3
startast
sleep 3
GOOD LUCK
Scott Stingel
www.evtmedia.com
-
Victor Alvarez wrote:
Hello,
I have a machine with two cards installed, one digium that gives e1
connectivity and one quadBri for the ISDN line
office?
Do you have another line you can switch to and try the same card?
Does the Red alarm occur at the moment the call is disconnected, or
afterward?
Regards
Scott Stingel
www.evtmedia.com
Yusuf Iqbal wrote:
Previously I have posted the same mail but no one answered me...Sorry
for resending
LeeLee-
Try configuring all 4 spans first and then the single channel (125)
above that - works for me.
Modprobe in the same order, then ztcfg.
Regards
Scott Stingel
www.evtmedia.com
Lee Lee wrote:
Hi all
I newly added a X101P into my asterisk that already have a TE410P
running 2 E1s namely
telephone jacks wherever you want (called inside wiring), but usually
at quite a high hourly rate.
Regards
Scott Stingel
www.evtmedia.com
Manjit Riat wrote:
Thanks but I am pretty sure they won't do it. So there has to be a better
way.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent
!
Regards
Scott Stingel
_www.evtmedia.com
_
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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for perfection and just use a very
fast processor when I want to handle 120 channels with no errors.
Best regards
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
---
[EMAIL PROTECTED] wrote:
Dear All,
we have installed a TE410P card
: NOT on this mailing list):scott at
evtmedia.com,
ie: do not reply to this, just send me a new email and please put:
asterisk consulting or something in the subject line so I can see it
among the spam!
Thanks
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
on the other end
Pin 2 -- Pin 5
Pin 4 -- Pin 1
Pin 5 -- Pin 2
You should get green's on both the connected channels if your zaptel and
zapata configurations are ok, and if you've run both modprobe and ztcfg
as documented.
Good luck
Scott Stingel
President
EVT, Inc.
www.evtmedia.com
Sid wrote:
Hi list
You can read all about it, and find out where to download at:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Yes, it supports both SIP and H.323
Cheers
Scott Stingel
sai latha wrote:
Hello,
Happy New Year
where u r downloaded the asterisk server please
tell me.Iam searching the asterisk
Andrew-
Thanks for posting your update and troubleshooting checklist.Most
people on the forum don't take the time to re-post when a problem has
been resolved - but that's the thing that helps people the most!.
regards
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
-48m volt power is often used in telco central office environments,
where the C.O. provides a huge amount of battery-backed up power to the
switches and to power the local loops in the event of an AC power failure.
Regards
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com
in. If
you're used to Outlook, thought, you may miss the nice integration with
your calendar, but you certainly won't miss paying for it!
-Scott Stingel
www.evtmedia.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
!
regards,
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
Sören Malchow wrote:
Hi all,
i am stuck with the configuration of asterisk
- modules are loaded ( zaptel and wct4xxp )
- i have zaptel.conf configure, output of ztcfg -vv
--- snip --
/rapid:~# ztcfg -vv/
/Zaptel
you may not be aware of the asterisk-biz mailing list, which is probably
more appropriate for a discussion like this.
you'll find many VoIP termination vendors hang out there too.
Regards,
Scott Stingel
Mohit Muthanna wrote:
Have you used them before?
Do they provide commercial grade service
an
IVR call to a live operator without holding up channels in the IVR.
Regards,
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com
Alistair Cunningham wrote:
Eric,
E1 is a physical layer protocol, like ethernet. It defines a 2Mbps
pipe, which can be used for data, or can be split
to send calls from one to the other. By the
way, to initiate outbound calls use the .call file facility (see the Wiki)
Good luck!
Scott Stingel
President
Emerging Voice Technology, Inc.
Palo Alto California and London England
www.evtmedia.com
Johan Bilien wrote:
I guess I need some special equipment
From looking at the description, it seems that the Sangoma card (at
least the quad version) *may* have a more robust hardware buffering
mechanism than the TE4xxP series. If so, this might help solve some of
the load-related issues that my customers have experienced in very large
systems.
Just to confirm that you also powered down and up?
I've no experience with the TE110, but this is a known problem with the
TE405 and TE410. They apparently can get locked up, and only a power
cycle will clear it.
Regards
Scott Stingel
www.evtmedia.com
Alfredo Sola wrote:
Hi,
I have
what they
say.
Do you have any other errors in your /var/log/asterisk/messages flle?
regards
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ali
in /var/log/asterisk/messages.
Thanks
Scott Stingel
Knut Bakke [EMAIL PROTECTED] said:
Hi,
I'm running a simple test from asterisk towards a
public telco switch (AXE10) over E100P.
Here is the test case:
1) 30 calls are setup simultainously, 20 sec ringing
time.
2) no calls answers
to sell
- that's why the board price can get so high. Dialogic's D600-2E1 JCT
boards etc cost well over US$1. The whole point of the asterisk/digium
exercise is to move the complex software to the PC and take advantage of the
economies of scale that it brings.
Regards,
Scott Stingel
Scott M
I think several digests a day is normal! I find the digest option unusable.
The format is difficult to read, the threads are hard to follow etc.
Wish someone would improve this feature!
Regards
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
URL:
should
work fine with Perl or PHP.
Please read through the archives as this topic has been discussed a number
of times.
Regards,
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
URL:www.evtmedia.com
-Original Message
The TE410P should run fine under both Redhat 9 and Fedora Core 1. My first
question is: Are you running a new CVS version? Maybe there have been bugs
introduced with all of the recent changes. I'm running under December
versions - works ok, except for problems experienced under very heavy
Hi Matt-
Thanks for posting all of that! I was just starting to look into using this
interface, and now maybe have some second thoughts after reading your post.
You're right, it would be great for someone to fix the buffer/deadlock
problems.
Cheers
Scott Stingel
Scott M. Stingel
Emerging
call setups very well. Framing errors etc
Regards,
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott at evtmedia.com
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED
Well, the nice things about telephones, in the US anyway, is that they are
generally backward-compatible with each other.
Why not, as a first step, connect a normal telco line to L1 and L2, and see
if you get dial tone through the receiver?
Scott M. Stingel
Emerging Voice Technology Inc.
Palo
ON second thought:
I re-read your message - missed the part about the magneto. Maybe you
shouldn't connect that phone to the PSTN after all! (or to a digium card)
Magneto-generated ring detection is a bit beyond the digium card spec I'm
sure!
Cheers!
Scott M. Stingel
Emerging Voice
] On Behalf Of Greg Boehnlein
Sent: Wednesday, February 04, 2004 12:51 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] [OT] Oldest Telephone
On Tue, 3 Feb 2004, Scott Stingel wrote:
Well, the nice things about telephones, in the US anyway, is that they are
generally backward-compatible
It would be very helpful if the digest format could be improved, ie: ONE
digest per day, and threads grouped within the digest. The format now is
not very useful.
I too am overwhelmed with 50-100 messages per day.
Thanks!
Scott M. Stingel
Emerging Voice Technology Inc.
Email:
Yes, you can certainly do something like this. We do outgoing applications
for our customers, similar to this.
Basically, you dump a triggering text file into
/var/spool/asterisk/outgoing, which asterisk checks for every second. This
causes an outgoing call to be made based on the dialplan
Hello all-
I have 3 TE410P cards in service in the field. Two of them have an regular
problem that they get stuck during a system reboot. What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the reboot.
The only thing
Try this Perl subroutine to get an alphabetic list of all files, excluding
the dot files, and excluding sub-directories:
use DirHandle;
sub justthefiles {
my $dir = shift;
my $dh = DirHandle-new($dir) or die can't opendir $dir: $!;
return sort # sort pathnames
Looks interesting - BUT Very pricey - they even charge $150 for an
exhibits-only pass, which usually would be free at most trade shows!
Here's the link - suggest feedback to the sponsors about the high pricing!
http://www.pulver.com/von/register.html
Scott M. Stingel
Emerging Voice
might consider waiting a bit before buying one, to give
them time to work the bugs out.
Cheers
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto California and London England
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:www.evtmedia.com http
. They said that a
firmware change is needed to fix.
Do you know if you have the latest firmware? If not, you might call them
directly (I know you're in Oz) - their tech support doesn't do email too
efficiently, but they're pretty good on the phone.
Regards
Scott Stingel
Scott M. Stingel
Emerging
in the UK and western europe. My specialty is
very large scale IVR, multilingual systems, and calling card platforms. If
you're looking for something like a VoIP PBX server, for example, you'll
find many good consultants on here as well.
Regards
Scott Stingel
Scott M. Stingel
Emerging Voice
Hi Bill-
I've built some load testers for asterisk, using the outgoing call facility.
It's been a little while, so you may want to test this yourself, but I
recall finding a couple of problems:
(a) I don't think it manages queuing very well if there are a limited amount
of outbound resources.
Hi Ariel-
I wonder if you could please expand on that a little? What was your
configuration for conferences when you had the problem (how big were the
conferences, were there errors in the /var/log/asterisk/messages file, etc)
I do have problems in a setup with two TE410P's, although my
choosing Digium and asterisk, they
are super accomplishments, just want you to manage your expectations a bit.
Test, test, test!
Good luck in your projects
Scott Stingel
President
Emerging Voice Technology Inc.
Email: scott at evtmedia.com
URL:www.evtmedia.com
Hi-
Malcolm and Greg are at Digium in the USA (sales/marketing I think):
Put an @ between the following words - trying to foil the email address
crawlers:
Malcolm is malcolmd digium.com
Greg is gvance digium.com
Scott M. Stingel
Emerging Voice Technology Inc.
Email: scott at
Hi Steve-
Actually, there isn't that much processing power on the
Dialogic boards.
You're right, but they were not really intended for full duplex transcoding
as you mention. I think the horsepower they had was plenty for typical IVR
apps however. If memory serves, the D600 (2 E1's) has
cool! I like the light bulbs. (and the name)
nice job
Scott M. Stingel
Emerging Voice Technology Inc.
Email: scott at evtmedia.com
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent:
Hello-
It might help to ask more specific questions.
I would suggest some research in the following places as a starting place:
www.asterisk.org
http://www.voip-info.org/tiki-index.php?page=Asterisk
and the many links referred to from these sites.
regards,
Scott M. Stingel
Emerging Voice
Hi Jason-
I was going to update the page called Dimensioning an Asterisk System, at
the request of Philipp von Klitzing, who set up that page. Maybe at the
same time, we could do the page you refer to?
As far as IVR, my customer's set up is a bit unusual, extremely high call
volume and very
] On Behalf Of Jason Boyd
Sent: Tuesday, March 09, 2004 6:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IVR setup (was Dialogic
supported well?)
On Tue, 9 Mar 2004 17:44:09 -
Scott Stingel [EMAIL PROTECTED] wrote:
I was going to update the page called Dimensioning an Asterisk
System
Hello-
I'm considering some TE405P's for a customer of mine. This is the 5 volt
version of the TE410P.
Digium is now shipping these - does anyone have production experience with
these cards in the field?
Many thanks in advance.
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto,
?
On Thu, 2004-03-11 at 03:11, Scott Stingel wrote:
Hello-
I'm considering some TE405P's for a customer of mine. This
is the 5 volt
version of the TE410P.
Digium is now shipping these - does anyone have production
experience with
these cards in the field?
yes, was there something more you
Hi Nichoas-
Are you are getting lots of frame re-transmission messages in
/var/log/asterisk/messages as well?
regards
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott at evtmedia.com
URL:www.evtmedia.com
Never - unless I must have a new feature, or need a critical bug fix!
but, seriously, mine are production systems, and I don't use many of the
VoIP features of asterisk. There is so much development going on in
asterisk, that you may want to update only an in-house, non-production
system, at
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nicolas Bougues
Sent: Thursday, March 11, 2004 8:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PRI errors blocking Asterisk
On Thu, Mar 11, 2004 at 04:48:42PM -, Scott Stingel wrote:
Hi
On Thu, Mar 11, 2004 at 08:13:48PM -, Scott Stingel wrote:
could you please post your zaptel.conf?
Here it is :
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,1,0,ccs,hdb3 # Colt est source de timing
span=4,0,0,ccs,hdb3
defaultzone=fr
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
Hi Andrew-
The unknown error 500 and the frame rejects are somewhat normal - I get
thousands of these in a busy IVR system. The underlying cause for these, I
think, is that your processor occasionally does not keep up with the frame
transmitter on the PRI board - something that will happen from
my zaptel to 2/11/04 from 3/8/04. And
kept my libpri from 3/8/04. I never had this error before
updated. I had other issues, but not this one.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Scott Stingel
Sent: Tuesday, March 16, 2004 10:38 AM
Hello Azher-
I have a similar setup in hardware, ie: TE410P running on dual-xeon system,
however I'm running IVR only. I start getting the I-frame errors above
about 80 simultaneous calls. I do not get IRQ misses at all. Also I do not
get the startup error messages. The errors I get the most
---
http://www.consulttech.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Stingel
Sent: Sunday, March 21, 2004 9:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI issues with TE410P
Hello Azher-
I have
on these issues today.
Regards
Azher
---
http://www.consulttech.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Stingel
Sent: Monday, March 22, 2004 11:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
I think Steve is referring to the following line:
export LD_ASSUME_KERNEL=2.4.1
If you put this in your command line before starting asterisk, you will get
around the RH9 problem of leaving zombies when AGI processes quit. Other
than that, I don't think it influences CPU load.
Note that the
) and dialed number
(DDI) to the software so I can see them in variables?
I need this for an application that I'm proposing.
Thanks in advance,
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: [EMAIL PROTECTED]
URL
thanks
Scott Stingel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, June 26, 2003 12:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Digium E400P in EuroISDN
environment
It's either EuroISDN or EM w
on disconnect events.
No doubt I have not read some very basic documentation or user notes, so if
anyone can steer me to the proper documents to read, would be greatly
appreciated!
Thanks
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email
Hi Dan-
In Steven's defense, he did write back to me outside the mailing list and
told me what he meant. I was using reply in a new thread.
He also answered my technical questions with a brief summary.
But your other comments are interesting - it is difficult for a new user
like me to get
spec for your consideration! I
don't need a huge amount of time - maybe 5-10 hours over the next 2
weeks.
Thanks
Scott
Stingel
(probably
A-Law encoded) on a number of E1 circuits simultaneously.
Realistically, how many 30-channel E1's can I support with, for example, a
Pentium III 1.4 GHz? I want to build an 8 E1 (240 channel) system - do I
need more horsepower than this?
Thanks
Scott Stingel
Scott M. Stingel
Scott Stingel wrote:
Hi-
I'm almost embarrassed to ask the following simple question,
following John's
excellent and rigorous bandwidth analysis (see earlier thread):
I have a straightforward Asterisk application, IVR-only (no
connections
between channels). It will simply
!
Cheers
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:www.evtmedia.com http://www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
shepherd
Hi Frank-
I've had a similar experience with Asterisk over my first week, but have had
some recent breakthroughs. Asterisk/Digium will in the end save me a huge
amount of money over a typical Dialogic/high-level voice language GUI
approach, so it will all be worth it (I hope!)
But it has been
Hi-
Its not clear from your msg whether you want to detect the open-circuit
condition of the T1/E1 directly or on one of the analog circuits connected
on the other side of the channel bank.
If directly on the E1/T1, I imagine that asterisk can detect the condition
of the alarm signals for the
Hello-
What is the .gsm format? Ie: what's the encoding method and sample rate
please?
Thanks
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: [EMAIL PROTECTED]
URL:www.evtmedia.com
PROTECTED]
Subject: RE: [Asterisk-Users] .gsm voice format
Gsm is wav in 8/mono
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Scott
Stingel
Enviado el: lunes, 14 de julio de 2003 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk
bytes.
On Mon, 2003-07-14 at 06:45, Scott Stingel wrote:
Thanks - do you know the bit rate?
I'm trying to play these prompts with other voice
application software, and
so far have been unable to. I've tried: Windows Media
player, Vox Studio,
Envox prompt editor - no luck with any
Hi Tilghman-
I recently had a lot of problems getting the DevLit kit working out of the
box, even using the configurations supplied on the floppy that came with the
kit. I didn't have the problem you are experiencing though, which sounds
like some kind of hardware conflict to me. I would
I live in California, and saw one of those cube PC's in Frye's for a few
hundred dollars ($400??). Really tiny, everything contained inside. Had
one PCI slot. I thought it would be nice for demos since it was so easily
shipped. I think the processor was 800 MHz or so. Would have preferred 2
if the pricing you mention is per system,
or per-port.
Thanks,
Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto California London England
Email: [EMAIL PROTECTED]
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto
Title: Message
yes,
they are the same. And in the UK, they call it the "hash" key - just to
add to the confusion!
So I
guessthe same keymeans different things depending on where the
caller is in the transaction.
Scott M.
Stingel Emerging Voice Technology Inc.Palo Alto, California and
. Below
are samples of each output.
regards,
Scott Stingel
ON ASTERISK STARTUP SESSION: (INCOMING CALL)
*CLI
-- Starting simple switch on 'Zap/1-1'
NOTICE[1217603008]: File chan_zap.c, Line 4134 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Answer(Zap/1-1, ) in new stack
Hello-
I've been writing a number of AGI scripts in Perl, and so far everything's
working ok.
However, yesterday I tried the AGI command RECORD FILE for the first time,
and my channel locked up. Trying to stop asterisk produced a segmentation
fault. There may be a bug here, but first let me
Title: Message
Just a
suggestion, but wouldn't it be more appropriate for Digium to host the
documentation?
I
think the missing link here is someone who will write (and illustrate) the
documentation. All of this open source software is great because it's free
- but commercial users and
Hi-
Asterisk (CVS 7/30/03) seems to leave my AGI processes (written in Perl) as
zombies, even though they exit normally with exit(0). I am running Red
Hat 9.
I tried the same AGI etc with an older CVS (7/1/03) and this does not
happen.
I think a zombie process is a process that doesn't
if the problem has been added to the new bug tracking
system. We should check.
My workaround is to run the AGI scripts on a RH7 box and forward calls
using IAX.
Scott Stingel wrote:
Hi-
Asterisk (CVS 7/30/03) seems to leave my AGI processes (written in Perl)
as
zombies, even though they exit normally
-Users] Asterisk agi interface leaveszombie processes?
On Sat, 2003-08-02 at 14:20, Scott Stingel wrote:
Yes, this concerns me too, as I'm about to install 2 big systems in
Europe.
How hard is it to roll-back to RH 8? I've only got a couple days to make
this all work in house before I have
,
irc.freenode.net) and let me login and try to debug it. That
will make it
easier by far.
Mark
On Sat, 2 Aug 2003, Scott Stingel wrote:
Yes, this concerns me too, as I'm about to install 2 big
systems in Europe.
How hard is it to roll-back to RH 8? I've only got a
couple days to make
Hi Dave-
I think it was Simon, not me, who answered your questions. I would add that
you might consider getting one of the Digium starter development kits.
Both of the non-digital kits can connect to a single analog line (you don't
need ISDN to get started). One also includes a cool USB device
you start
asterisk:
export LD_ASSUME_KERNEL=2.4.1
It works for me - I'm still checking to see if there are any other side
effects - none yet..
-Scott Stingel
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: [EMAIL PROTECTED]
URL
Hi-
Two questions:
(a) Does the CDR SQL module log both incoming and outgoing calls, or just
outgoing only?
(b) If I enable the CDR SQL module (to use mysql), does it disable the text
logging at the same time?
Thanks
Scott S
___
How do I get the asterisk compile to produce the cdr_mysql.so module,
assuming this is what I need to get CDR's into my mysql database.
Is load = cdr_mysql.so what I put in the modules.conf file as it says?
It looks like asterisk loads the modules in /usr/lib/asterisk/modules, but
there is no
I don't think that board is usable with asterisk, due to it's half-duplex
design. (unless it's a JCT model)
If it's a D/41ESC in good condition, you might be able to get $100-$200 for
it on Ebay, if just a D/41, only $50 if you're lucky!
You might check out www.digium.com - they have a
I'm pretty sure the Error 500 can be ignored. If memory serves, I think it
is related to a transmitter underrun error on the T100P framer chip. This
would occur if the processor doesn't quite keep up with the transmit data
stream on the T1 - ie: is load related.
These occur on all of my
checking with NTL ask if your circuit is ISDN 110 or Q.931e
or ETSI; these are the three names that are commonly used by UK carriers for
full spec. EuroISDN PRI.
Regards
Scott Stingel (who has also had this problem in the UK)
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto
(sticking my nose in here:) Yes, but it sounds like Gary, in addition, is
getting Red Alarms, which probably indicates a more serious problem... Red
Alarm is a loss of sync from the other end, as I recall. I have *not* found
that slow processor interrupt service or similar maladies cause
Hi-
Have you checked out the Wiki on this subject:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql
This should help you get started with MySQL, which works great with
asterisk, I've found.
You have to install the MySQL package - use the free version 4 at
www.mysql.com,
Hello-
Have you compiled and installed the proper versions of OpenH323 and PWLib?
(Before you compiled the h323 code.)
See the instructions in ~/asterisk/channels/h323/README..
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London
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