On Fri, Apr 4, 2008 at 4:42 PM, Jonn R Taylor [EMAIL PROTECTED]
wrote:
I made some install scripts based on centos 4 or 5 like trixbox but
without all the junk. It does have some fax setup stuff in it that I use on
our production servers that's been working for over a year. I you need any
At 01:50 PM 4/5/2008, you wrote:
Well, my $21 is still there and all of my calls are being declined.
Over a year ago, I requested a refund and regardless of all promises
that I would receive one, Jed never followed through. I'd use up
the credit if the calls would only complete.
Well, keep
Any know what Digium hasn't released the DS3 card?
It was supposed to be out a while ago.
-Matt
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On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
Any know what Digium hasn't released the DS3 card?
It was supposed to be out a while ago.
There was a fundamental problem with the chipset used, which precluded the
card from being useful. Specifically, the chipset only permitted the first
ok - but has a new release date be announced ?
or
Has Digium officially dropped the product ?
Steve Totaro wrote:
On Sun, Apr 6, 2008 at 3:47 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
Any know what Digium hasn't released the DS3
I don't use Lumenvox, yet.
I have a large client who has expressed interest in having something
like it deployed.
For such a revolutionary product there is very little chatter in the
list about it.
I really had hoped to hear from a bunch of people who had deployed it
and how it worked out for
On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
You need to have the kernel compiled specially for it to work.
Are you sure? What exactly is needed?
I think you need to rebuild the kernel on Centos, but
On 4/6/08, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
Any know what Digium hasn't released the DS3 card?
It was supposed to be out a while ago.
There was a fundamental problem with the chipset used, which precluded the
card from being
On Sun, Apr 6, 2008 at 3:47 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
Any know what Digium hasn't released the DS3 card?
It was supposed to be out a while ago.
There was a fundamental problem with the chipset used, which
Check page 38 of 74. A real pain. Hopefully either Tzafrir is
correct with a different distro (Debian)vor Sangoma makes it simple.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Apr 06, 2008 at 05:37:03AM -0400, Steve Totaro wrote:
There was a fundamental problem with the chipset used, which precluded the
card from being useful. Specifically, the chipset only permitted the first
255 channels to be addressed (instead of the full 672). Since that time,
and
OK! I'm also in the process of testing the engine Lumenvox, could not
yet approve a solution to the Portuguese language(testing with Spanish
Colombian), but will be testing this week.
Thank you for your attention, when achieve test I return follow up,ok?
Best Regards
Josué
2008/4/6 Al Baker
On Sunday 06 April 2008 04:48:19 Al Baker wrote:
ok - but has a new release date be announced ?
or
Has Digium officially dropped the product ?
Once again, Digium does not announce products until they are ready to ship,
drivers included. Therefore, I cannot say, either way.
--
Tilghman
I think they are using a specially programmed version of Asterisk to do
this.
Don't you mean:
I am using a specially programmed version of Asterisk to do this.
?
domain: dialaway4free.com
created: 16-Jan-2008
last-changed:
Sorry,
I cannot find the link to the actual Digium link but here are examples
from the wiki:
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
http://www.voip-info.org/wiki-Asterisk+Data+Configuration
Tomorrow, I will see if data T1 on a Sangoma card is much more simple.
If I find
Sorry for all the replies, I found the Digium PDF on Data mode.
http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
Good luck getting them to support it though ;)
I will post my Sangoma results tomorrow.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 10:49 AM, Steve Totaro
[EMAIL
On Sun, Apr 06, 2008 at 09:44:30AM -0500, Tilghman Lesher wrote:
On Sunday 06 April 2008 04:48:19 Al Baker wrote:
ok - but has a new release date be announced ?
or
Has Digium officially dropped the product ?
Once again, Digium does not announce products until they are ready to ship,
Another reason I am sure that Digium has not released a DS3 TDM card is the
fact that asterisk currently cannot handle that many channels. I am speaking
from experience on this. We have build before a predictive dialer with 16
PRIs. In order to do this and not have audio quality issues we had to
On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Sunday 06 April 2008 04:48:19 Al Baker wrote:
ok - but has a new release date be announced ?
or
Has Digium officially dropped the product ?
Once again, Digium does not announce products until they are ready
Should have read, I cannot recommend the Atran MX2800 M13 *enough*
Two controller cards, two power supplies, battery backup, this is a
nice little box.
It will probably be the most solid piece of equipment in your data center.
Thanks,
Steve Tototaro
On Sun, Apr 6, 2008 at 11:12 AM, Steve
On Sun, Apr 06, 2008 at 11:12:33AM -0400, Steve Totaro wrote:
I cannot recommend the Adtran MX2800 M13, it has redundant everything
and is very easy to setup and not very expensive either.
We'll assume you meant that you can't recommend them enough. :-)
Cheers,
-- jra
--
Jay R. Ashworth
I cannot recommend the Adtran MX2800 M13, it has redundant everything
and is very easy to setup and not very expensive either.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 11:04 AM, Michael Cargile [EMAIL PROTECTED] wrote:
Another reason I am sure that Digium has not released a DS3 TDM card is
Hi all,
I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway.
Call audio is fine and all seems well but after 15 to 20 sec the call
drops
Most of them are dropped while setting up after 5 - 10 sec
This fails much more often
On Sunday 06 April 2008 10:13:43 Steve Totaro wrote:
On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher wrote:
On Sunday 06 April 2008 04:48:19 Al Baker wrote:
ok - but has a new release date be announced ?
or
Has Digium officially dropped the product ?
Once again, Digium does
On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet [EMAIL PROTECTED] wrote:
Hi all,
I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway.
Call audio is fine and all seems well but after 15 to 20 sec the call
drops
Most of
On Sun, Apr 6, 2008 at 11:58 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Sunday 06 April 2008 10:13:43 Steve Totaro wrote:
On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher wrote:
On Sunday 06 April 2008 04:48:19 Al Baker wrote:
ok - but has a new release date be announced ?
http://www.vicidial.com/On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
We've already connected ~600 analog extensions to Asterisk and we were
far from reaching the bottlenecks. We have used a machine that is hardly
a top-of-the line server (a dual-core Dell machine),
On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
http://www.vicidial.com/On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
(Off-list, and not expecting an on-list reply)
We've already connected ~600 analog extensions to Asterisk and we were
far
On Sun, Apr 06, 2008 at 10:58:56AM -0500, Tilghman Lesher wrote:
Was this policy put in place after announcing the DS3 dud?
The policy was put into place for a number of reasons, the main one being that
it's generally a good idea and is therefore pretty industry standard.
Two words: Osborne
On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
http://www.vicidial.com/On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
(Off-list, and not expecting an on-list reply)
Ooh, geee: Why
Hi there,
We get witheld caller cli from client and no cli output, but i dont
think it's the problem.
We had a test with about 200 calls and we got an ACD of about 30 sec,
while from another client with asterisk, for the same route we get
about 3 min ACD.
Beside that we get calls dropped
On Sunday 06 April 2008 11:45:51 Jay R. Ashworth wrote:
On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
http://www.vicidial.com/On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir
Cohen [EMAIL PROTECTED] wrote:
On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
http://www.vicidial.com/On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
(Off-list, and not expecting an on-list reply)
Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream gets
clipped / dropped
Sorry about that not used to gmail. :-)
Michael Cargile
Director of Consulting
The Vicidial Group
www.vicidial.com
On Sun, Apr 6, 2008 at 12:45 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
On Sun, Apr 06, 2008 at 12:12:17PM
On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
While I don't have your experince, I would still speculate that even for
trunks the average capacity is not ther same load as that of a
predictive dialer.
This is very true.
Hi,
I am feeling very frustrated with the Digium TDM400P, I have 3 x FXS
1x FXO modules and I have tried various things and different versions
of Asterisk and Zaptel to no avail.
Clearly there are issues with this card, so I am wondering - is there
a card out there that does the following
Steve Totaro wrote:
Sorry for all the replies, I found the Digium PDF on Data mode.
http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
Good luck getting them to support it though ;)
I will post my Sangoma results tomorrow.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 10:49 AM,
On Sun, Apr 6, 2008 at 1:53 PM, Michael Cargile [EMAIL PROTECTED] wrote:
On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
While I don't have your experince, I would still speculate that even for
trunks the average capacity is not ther same load as that of a
http://www.surpluscomputers.com/store/main.aspx?p=ItemDetailitem=CES11532
Ground shipping $9 to MD. Never used so I cannot comment on quality
and Linux is not listed as compatible but I suppose as long as your
audio jacks work, then the handset will too.
Unfortunately, shipping goes up linearly
Tilghman Lesher [EMAIL PROTECTED] writes:
Oh, please, don't start THAT flame war. People who consider Reply-To
munging harmful have obviously failed to read the ENTIRE rfc (especially
the part where it specifically says the use of the Reply-To header is for use
on listservs). :-P
It's
On April 6, 2008 11:12:33 am Steve Totaro wrote:
I cannot recommend the Adtran MX2800 M13, it has redundant everything
and is very easy to setup and not very expensive either.
Agreed; I've set these up and they are rock effing solid. We did have a shelf
controller die and without the backup
On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen [EMAIL PROTECTED] wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
Oh, please, don't start THAT flame war. People who consider Reply-To
munging harmful have obviously failed to read the ENTIRE rfc (especially
the part where it specifically
You probably have to unlock it first. Google or voip-info.org is your friend.
On Sun, Apr 6, 2008 at 5:02 PM, Christian [EMAIL PROTECTED] wrote:
Hello all,
I need some help with my Cisco 7960 enabling TFTP. Does anyone know what
numbers to press in the menu? Or can I enable this through
Hello all,
I need some help with my Cisco 7960 enabling TFTP. Does anyone know what
numbers to press in the menu? Or can I enable this through telnet?
Many thanks,
Christian
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Hello,
I know how to unlock the phone and what the password is.
I am asking this kind of question because i am visually impaired and cannot see
the screen.
many thanks,
Christian
On 2008-04-06 at 17:05 Steve Totaro wrote:
You probably have to unlock it first. Google or voip-info.org is your
On Sun, Apr 06, 2008 at 05:02:53PM -0400, Steve Totaro wrote:
http://www.metasystema.net/essays/reply-to.mhtml
Wildefires are harmful to forests and aniaml life but are essentially
a good thing in the balance and cycles of nature. It all depends on
your viewpoint.
Yes, I've seen that, and
In that case, I guess I would ask somone with better sight to help me
out, uless they have braille on the buttons.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 5:09 PM, Christian [EMAIL PROTECTED] wrote:
Hello,
I know how to unlock the phone and what the password is.
I am asking this kind of
On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen [EMAIL PROTECTED] wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
Oh, please, don't start THAT flame war. People who consider Reply-To
munging harmful have obviously failed to read the ENTIRE rfc (especially
the part where it specifically
On Sunday 06 April 2008 15:56:22 Steve Totaro wrote:
On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen [EMAIL PROTECTED]
wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
Oh, please, don't start THAT flame war. People who consider Reply-To
munging harmful have obviously failed to read the
On Sunday 06 April 2008 16:22:58 Jay R. Ashworth wrote:
On Sun, Apr 06, 2008 at 05:02:53PM -0400, Steve Totaro wrote:
http://www.metasystema.net/essays/reply-to.mhtml
Wildefires are harmful to forests and aniaml life but are essentially
a good thing in the balance and cycles of nature.
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
Thanks.
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On Sun, Apr 6, 2008 at 1:52 PM, Alex Kauffmann [EMAIL PROTECTED] wrote:
Thank you for the replies. It was my understanding that rebuilding the
kernel was necessary in 2.4 but everything needed was already included
in 2.6 series.
*HEAVILY* dependent on the distribution!
Hi
By the past 2 months i have install a server with Asterisk with a E1 in
Axtel(Mexico). The call presented a Echo in
randoms call, inside and outside.I decide migrate for the last
version Asterisk 18.1,Zaptel 1.49 and Unicall and instalaled
a new Digim Card TE122 B (Echo
Snap,
Well, after trying to buying a TDM400P and then getting persuaded to buy
a TDM410P
because they no longer sell the 400 model I'd say I'm not impressed. It
took three 2.6
kernel builds (zaptel 1.4 won't even build with the latest kernel
release) before I finally got the
kernel to build
I had this problem before...the following appeared in a previous post...
For some reasons, the * and 1 must be pressed pretty quickly together on
the Polycom phone before it can be transmitted successfully to Asterisk.
Does anyone know if that can be tuned?
Sure... go to features.conf, and change
Thanks for that. I have the timeout set to 3000 ms and I have been pressing
the *1 within 500 ms so I don't think it is related to that. As I can do it
over SIP but not ZAP does not make much sense to me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Hi All,
For some reason One Touch Recoding does not work over ZAP but it does
work when I call another extension. Both Dial commands have the W
option for the calling party to enable recording.
Does anyone know why it works internally but not over ZAP. I have a
TE110P card on an E1
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