I have always had excellent service from Atacomm. Not always the absolute lowest price, but they pretty much ship when they say they will.On 8/10/06, Don Tasker
[EMAIL PROTECTED] wrote:Had the same issue here as well, our company was in a
dire need for some digium cards due to an internalsystem
If I have a call coming in on one line and then I am using Dial to send it out to another outside line, is there a good way to get separate CDR for both call legs?
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Is there a simple way in extensions (or elsewhere) to select among multiple outbound routes for a given call,
not so much based on cost as on current route availability. I.e. dial out using provider X unless our connection
to them is down, or lagged, or jittery, or whatever way out of tolerance.
opencall.org seems to be off air since yesterday. I am wondering if
anyone has a private cache of the most current spandsp they would be
willing to share...
regards,
-dorn
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Are there any predictive dialers for Asterisk which will
do answering machine detection, etc?
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On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote:
You can modify and/or link to GPLed code with commercial code and get
away with it as long as you don't distribute the stuff. That's the
story with G.729, with nVidia drivers etc etc etc
I suppose it's even possible to
On Thu, Jan 13, 2005 at 11:09:28AM -0500, Ken D'Ambrosio wrote:
I'd dearly love to be able to give an Asterisk demo by just toting my
notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way
to do this? Or should I look for a small-profile box with PCI slots,
instead?
I've
I have a nice used zhone channel bank I want to experiment with, but need
a T1 interface for my * box to do so. The TE410P looks nice, but more
money than I want to spend to experiment, and I don't need 4xT1, only
1xT1.
Are there any good 1xT1 PCI cards that are recommended?
Regards,
-Dorn
On Sat, Jan 08, 2005 at 11:01:25PM +0800, David Liu wrote:
Well there is nothing much you can do if you don't own all the routes. But in
concept you can, and this is purely just theoritical and a very unhealthy
thing for the Internet, is to write a program running on your router that
On Wed, Jan 05, 2005 at 03:56:39PM +0100, Roy Sigurd Karlsbakk wrote:
hi
some time ago, I asked the list of a good book for learning ISDN and
SS7. I don't need to know how to write a channel driver or something; I
just want to know more about the possibilities and what's really sent
back
On Mon, Jan 03, 2005 at 07:11:46PM -0500, Garrett Smith wrote:
I wouldn't consider it an advertisement. There was no price, etc. I was
simply telling those that ordered from me previously I have more. It is
easier to send one email to 100's, then 100's of emails. If you do not want
to save
I have a couple of capacity related questions for which I am hoping
to find answers (or at least hints) derived from real-world experience.
asterisk as a trunking gateway; bunch of sip phones in location
one need to access other non-* sip PBX device in location two over
constrained
On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
[...]
What if, for example, the TDM400 issues were a cumulative thing? If you
had over 6dB of attenuation on the PSTN loop, coupled with greater than
5V potential on the neutral-ground of your elecrical receptacle,
compounded
On Thu, Dec 30, 2004 at 11:16:01PM -0800, Alspach Family wrote:
I don't want to sound like a TV evangelist from the 80's and 90's but if
you have it to give, please do. We have operators standing by to accept
your donation. All you have to do is PayPal it to [EMAIL PROTECTED]
mailto:[EMAIL
On Tue, Dec 28, 2004 at 11:18:42PM -0600, Matthew Boehm wrote:
Hey gang,
I was successful in recompiling my 2.4.20 kernel to support HDLC. I was
successful in hooking up our T1 line into the zap card. I was successful in
being able to ping equipment on the other end of the T1. I was
On Tue, Dec 28, 2004 at 11:37:42PM -0600, Me wrote:
What sort of chipset is your SATA controller interface? Intel
ICH6R?
Adaptec ICH5R SATA controller according to SuperMicro which makes the Mobo.
The board has an Intel® E7501 main chipset.
That should probably work. You may need to
All,
Please consider trimming off the bottom of the message you are
replying to. It usually take only a few seconds and saves
everyone reading the list from extra bloat in their mailbox :)
Happy Holidays!
-Dorn
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On Tue, Dec 28, 2004 at 09:13:02AM -0800, Geoff Nordli wrote:
[EMAIL PROTECTED] scribbled on :
On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote:
Seth Ueland Chancy wrote:
Probably your best bet is Debian + 2.4 kernel + X100P card + apt-get
install asterisk
On Tue, Dec 28, 2004 at 10:02:12PM +0200, David Norton wrote:
Hi,
I have been running asterisk for about a week though on a debian system
through apt-get. I am now trying to compile it use the CVS and im getting
this error.
/usr/bin/ld: cannot find -lssl
What do I need to install to
On Tue, Dec 28, 2004 at 02:17:47PM -0600, Me wrote:
Hello, I am trying to build up a pretty meaty Asterisk box after doing our
initial testing and playing on a 1ghz system.
Right now I have decided on a prebuilt system which I normally don't do but
thought it seemed like a good deal.
I
On Tue, Dec 28, 2004 at 04:12:26PM -0600, Me wrote:
Dorn,
Can you give me some details on this linux md driver you mentioned?
Also, you say not to scrap the SATA drives, is this because you think I can
use them with FC1 or because you think I should try Debian? I really don't
want to
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I am looking for a small device with four FXO and one WAN connection
On Sat, Dec 25, 2004 at 11:12:22PM -0500, Dorn Hetzel wrote:
I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull. Is there a way to get the number
On Sun, Dec 26, 2004 at 02:22:43PM -0600, James Taylor wrote:
On Sun, 26 Dec 2004 14:17:59 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
[EMAIL PROTECTED] wrote:
On Sat, Dec 25
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
Greg Hill wrote:
I am looking for a small device with four FXO and one WAN connection.
Simple, so that the cleaning woman can make a hardware reset if
necessary. This device should be connnected to my Asterisk box. The box
On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote:
Seth Ueland Chancy wrote:
Probably your best bet is debian + 2.4 kernel + X100P card + apt-get
install asterisk
Cheers,
Jean-Michel.
I can also confirm that * works fine on Debian w/2.6.10-rc2-mm3
for the adventurous :)
I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull. Is there a way to get the number in the VM_CALLERID
string, or is there a second variable I can use
On Thu, Dec 23, 2004 at 11:45:04AM +0100, Michael Vogel wrote:
Rich Adamson schrieb:
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
Because they don't think or they love the telephone companies ...
Ok, this is probably a stupid
On Thu, Dec 23, 2004 at 02:10:12PM +0100, E. Versaevel wrote:
Qui, mais je ne parle pas français ;)
On Thu, 2004-12-23 at 00:12 +0100, Thierry wrote:
Hi
I have something like this but it's in french and it uses teh res_config
Best regards
Thierry wehr
Thierry,
If you are
Thanks! :)
On Thu, Dec 23, 2004 at 08:57:22AM -0600, Rich Adamson wrote:
As in many cases with *, there are usually multiple ways to accomplish
a task. Here's a couple that you'll need to tailor to your environment.
[in5100]
exten = s,1,Dial(SIP/sip1,20,tr)
The above assumes the pstn
On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
matter if it is a
I understand some of the basic Goto() forms,
such as Goto(context,extension,priority) and
Goto(extension,priority) [within context I presume].
Can someone Explain Goto(6275|1) as found in the
sample extensions.conf? Is this the same as
Goto(6275,1) just with a different delimiting
character?
On Thu, Dec 23, 2004 at 12:05:21PM -0600, Steven Critchfield wrote:
Close on the complete reason. There is also a licensing conflict with
Dialogic drivers and GPL software. You have to get a commercial license
for asterisk to clear the licensing issue. Beware that I think as soon
as you get
After I leave a voicemail for an extension and hangup,
my asterisk console (with debug turned up quite high)
shows two error messages like:
WARNING[7664]: app_queue.c:341 changethread: Can't change device with no
technology!
WARNING[7668]: app_queue.c:341 changethread: Can't change device with
On Thu, Dec 23, 2004 at 01:26:38PM -0600, Steven Critchfield wrote:
I seem to remember that mysql made a stink recently about the dual
licensing of asterisk. That was the cause of the mysql code getting
pulled and placed in the add-ons sections so the core didn't have any
licensing issues.
I've got a configuration with PSTN line connected to FXO
on TDM400P ringing through to a phone connected on a
Sipura SPA-3000. The phone *does* ring before the
caller-id is available. In fact, it shoes some
alternate message like waiting for caller id info
right after the first ring and then
On Thu, Dec 23, 2004 at 04:25:55PM -0600, Christopher L. Wade wrote:
Dorn Hetzel wrote:
I've got a configuration with PSTN line connected to FXO
on TDM400P ringing through to a phone connected on a
Sipura SPA-3000. The phone *does* ring before the
caller-id is available. In fact, it shoes
Are there any common silent failure modes for email
notification from the Voicemail module. I put the
email and pager email addresses in my entry in
voicemail.conf but no mail gets sent when I leave
a voicemail. No obvious error messages either,
unless I'm just not looking in the right place.
On Thu, Dec 23, 2004 at 04:51:34PM -0600, Rich Adamson wrote:
Are there any common silent failure modes for email
notification from the Voicemail module. I put the
email and pager email addresses in my entry in
voicemail.conf but no mail gets sent when I leave
a voicemail. No obvious
Are there any IAX speaking hardphones out there?
If so, can anyone offer comment on their quality?
Thanks!
-Dorn
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I can't get the link to work. Does this mean that there is
some IP phone available which if loaded with the right
firmware can do IAX? If so, where can I buy one and where
can I get the code?
-Dorn
On Wed, Dec 22, 2004 at 05:30:48PM +0100, Wilson Pickett wrote:
This just in:
Centrality
I just installed a new TDM400P with one FXO interface
in slot 4 (how it came from Digium). This box is
running Debian with a 2.6.10-rc2-mm3 kernel. After
the make linux26 and make install in /usr/local/src/zaptel,
I can see contents in /dev/zap but any attemp to
touch for example /dev/zap/ctl
Not such a stupid question :)
The paper instructions included with the card failed
to mention that needed to be done manually :)
Up and running :)
-Dorn
On Wed, Dec 22, 2004 at 10:34:17PM -0800, Shahed wrote:
I can see contents in /dev/zap but any attemp to
touch for example /dev/zap/ctl
On Sat, Dec 18, 2004 at 06:28:54PM +, Antony Stone wrote:
On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:
I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin. But, having said that, if you
can use IAX, it sucks even much than SIP does :)
Um
On Thu, Dec 16, 2004 at 05:58:08PM -0500, Gary Carr wrote:
So they offer termination via SIP for $0.013/minute?
Most of the good deals I have found are for IAX termination,
but maybe the same deal are available for SIP.
-Dorn
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I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin. But, having said that, if you
can use IAX, it sucks even much than SIP does :)
On Thu, Dec 16, 2004 at 08:53:01PM -0500, Andrew Kohlsmith wrote:
On December 16, 2004 08:47 pm, Gary Carr wrote:
Why is IAX
On Fri, Dec 17, 2004 at 07:51:00AM +0100, Wilson Pickett wrote:
I'm looking to change from a standard telephone line to a VoIP phone line at
home. I'm looking for recommendations for VoIP providers that I can use
with
Asterisk.
Don't forget about emergency services (lack of) with
On Fri, Dec 17, 2004 at 07:54:19AM +0100, Wilson Pickett wrote:
I am searching for a new PBX for the company. My choice is Astrisk. My Boss
wants background music via all the telephones. This is done in a
conventional PBX that he wants, but I can use the Asterisk PBX if it can do
What a
This is the sort of language you are likely to find from anyone offering
unlimited plans. It's just the reality of the fact that there is
no such thing as unlimited, really... Everything has it's limits :)
If you go with a per-minute plan, like from NuFone, Voipjet, etc., you
will not find
On Tue, Dec 14, 2004 at 05:20:02PM +0100, Dave Cotton wrote:
On Tue, 2004-12-14 at 18:44 +, Jean-Michel Hiver wrote:
To the list: Am I right understanding that Fritz + BRI line = no echo
issues?
I have two systems using AVM cards, one uses a C2 and the other 2 Fritz
cards neither
Would anyone care to offer opinions as to the FXO interface which sucks
the least :) I have an application in which it appears I must route
certain calls out an analog PSTN line. Presently, I am testing an
SPA-3000, but I can't seem to get the echo heard on the IP end of the
call down to a
On Tue, Dec 14, 2004 at 06:44:04PM +, Jean-Michel Hiver wrote:
Dorn Hetzel wrote:
Would anyone care to offer opinions as to the FXO interface which sucks
the least :)
So far, for me, using VoIP - PSTN termination provider has been the
solution which sucked the least.
My FXO card
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