. Virtual
here means - no hardware.
In short, you can install or otherwise use any hardware cards, like
Zaptel, with Asterisk when running on CoLinux and generally, I'll advise
you to not use Astwind for anything other then playing. It's a nice toy,
but that is all.
Gilad
--
Gilad Ben-Yossef
]
___
I'm prbably stupid, but wont this do what you want?
exten = 1,1,Goto(bye,s,1)
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Gilad Ben-Yossef [EMAIL PROTECTED]
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David Brodbeck wrote:
-Original Message-
From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED]
I'm prbably stupid, but wont this do what you want?
exten = 1,1,Goto(bye,s,1)
No, because I wanted to match on D, not 1.
I am stupid - I thought you meant the DTMF for the D button (aka 3DEF
Jay Milk wrote:
That's a known, yet not feasible work-around over accessing an
SMS-center directly. But the question remains how to accept IMCOMING
messages with *.
It's very simple - you register withj your Telco to receive SMS messages.
When an SMS message arrives, your Telco line will ring
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way to provide answer
supervision based upon the received audio only from the FXO interface
(from a public PSTN switch that does not have battery reversal, or CPC).
In zapata.conf use either
busydetecgt=yes
busycount=6
(it
Shoval Tomer wrote:
Hi all.
Can anyone comment why shouldn't we use FC 3 for an * production system?
For the same reason you should not use Fedora Core line for ANY
production system, as it designers intend it to be an experimental
branch. In particular, FC3 has the NSA's SELinux patches
Matt Schulte wrote:
Had a good question for the list, it seems whenever I work in an
Asterisk console or on the machine normally I get jitters on any audio
going through it. Especially if you did file copies or a 'ps ax' for
example. I was wondering if there was a proper way to 'nice' the
asterisk
Justin Carlson wrote:
what is wrong with running asterisk with the -pg flags at startup?
Which is exactly what I suggested:
Since VoIP is a real time activity, simple nice really isn't enough.
What you should do is mark the Asterisk proccess as a real time task for
the Linux kernel to schedule
lines aren't the same as used for digital
circuit. I'm not surprised you had trouble.
The X101P is a FXO interface card for a POTS line, not digital.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel
Damon Estep wrote:
Any PC platform is only as stable
as the sum of what you run on it, put a single analog interface in a red
hat ES on $10,000 worth of hardware and you will have to reboot every 3
days.
I'm not seeing these problem with X101P, nor does any of my (not so
many) clients. And all
Adi Linden wrote:
On Thu, 30 Dec 2004, Lyle Giese wrote:
Is your X100P set for loop start or Kewl Start? I am betting loop start,
try changing to ks instead.
Lyle
This is what I have in /etc/asterisk/zapata.conf so it should be Kewl
Start.
It might be that your local telco does not supply
Greg - Cirelle Enterprises wrote:
Are you running a stable (v 1.0 - 1.0.3) or cvs
Asterisk CVS-v1-0-10/03/04
I've upgraded two months ago to get a feature I wanted (SMS support). It
should be round about Asterisk 1.0.2
Gilad
___
Asterisk-Users mailing
Rich Adamson wrote:
The only issue I have with that is there are several people with digium
T1 and TDM cards in their systems, and its always the TDM that goes out
to lunch; not the T1. No doubt there are less then desirable mobos
around (and probably lots of them), but that doesn't explain why
... don't ask) and 2 VoIP termination/origination lines.
Of course, your mileage may very, but at least here there is no nightly
restart script.
Hope that helps you in any way.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP
Shahed wrote:
Hi All,
I am wondering if I will be breaking the GPL,
if I write for example, a channel driver or
make some modifications to the astrisk source code,
to interface at RUN TIME, through sockets, with
a proprietary system.
First, a warning - I am NOT a lawyer. I don't play one on TV,
Rich Adamson wrote:
While trying to apply the low-latency kernel patches to our RHv9
Linux 2.4.20-31.9, the patches would not apply. In comparing one
of the first patch files (lowlatency.h) to that already on the system,
it would appear the low latency patches were already applied by RH.
The
dealing with the problem in network cards. If I
can help in any way...
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack
(customized drivers
for the tulip chip set based cards and the Linux NAPI framework).
However, it is simply much easier to just grab a Giga card then research
which 100Mb chip and which driver you need to get ;-)
Hope this helps,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you
(if-this-really-is-a-human-please-press-1)
exten = fax,2,Hangup
You must also have:
faxdetect=incoming
In zaptel.conf for this to work.
Hope this helps,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475
Nguyen Quang Hoa wrote:
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the
B G wrote:
My intention is to setup Asterisk to be a message center to receive
from and send SMS to fixed phones. Can it be possible? My fixed phone
can dial to Asterisk and send SMS to Asterisk, but I cannot setup the
other way: make Asterisk dial to fixed phone and send SMS to fixed
phone.
Ah, I
willing to upgrade the firmware to get that extra
feature but not if it breaks something that already works... :-)
Thanks!
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax
/Asterisk distribution program that includes an
auto-install and special auto-configuration features. It quickly and
effortlessly converts any PC to a functioning Asterisk PBX...
http://www.voip-info.org/wiki-Xorcom+Rapid
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm
the hook.
Hope this helps,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
on.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
TELUX wrote:
Redhat 9 is running 100% cpu usage. I had a couple boxes doing this.
upgraded to Fedora and its ok.
I would try running asterisk with LD_ASSUME_KERNEL=2.4.1 if it isn't
already.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http
and confused,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
[1] Well, I lied - there is a way, which is the callprogress feature.
it's far from perfect (or in some states, working) though. Look for
callprogress on voip-info.org
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED
option in the Dial command.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
Martin List-Petersen wrote:
You may or may not be aware that to contribute code to FSF owned and
maintained software one needs to assign copyright to the FSF in much the
same way one is required to assign copyrights to Digium.
True, I don't think the FSF are going to sell licenses to those
Joe Greco wrote:
Hey All,
Isn't it possible that part of the commercial licenses that is offered is
that you (the buyer) are not required to advertise, disclose, or even admit
that your products offerings are based on an open source project?
What other reason would one have for buying a commercial
to FSF owned and
maintained software one needs to assign copyright to the FSF in much the
same way one is required to assign copyrights to Digium.
True, I don't think the FSF are going to sell licenses to those programs
under different terms and Digium does.
FYI,
Gilad
--
Gilad Ben-Yossef [EMAIL
Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
) is
not to survive in court. The purpose of a good license is to be so
iron clad clear that it never ever gets into court in the first place.
And this, my friend, is something the GPL has done *very well*.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http
Matt Riddell wrote:
1. The strings prove it is Asterisk
2. They deny it is
3. We have multiple sources who can confirm this
So, let's all post the article on GPL violation to Slashdot. All in
favour?
Danger Will Robinson.
I actually heard Welte give a talk about GPL violation in Germany in the
George Gardiner wrote:
I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting.
A: Because otherwise we don't understand what you're replying to.
Q: Why top posting is so frowned upon?
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL
.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
modulation.
I'm guessing that if you're using ulaw/alaw codecs for the call you
shoudln't have a problem.
You might also have to shut off echo cancelation.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED
Benjamin on Asterisk Mailing Lists wrote:
On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote:
The syntax for the register command is
register=username:secret:[EMAIL PROTECTED]:port/extension
Trouble is though that this does not have any effect on the username
in the digest.
this is not the change you're
looking for.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club
WipeOut wrote:
Following on from the message below I have discovered that the X100P
causes the SNR on my ADSL line to drop even with the Asterisk box
**switched off** and the power unplugged... This seems very strange..
Why should a card in a switched off PC cause noise on a line meaning
that
All you really need is a list of 1-800 numbers in various countries.
Most multi-national corporations have a list buried somewhere on their
web site.
For example:
http://www.microsoft.com/resources/howtotell/ww/windows/what.aspx
Gilad ;-)
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence
that it isn't all that hard to do in
Asterisk (since it's done in those cheap machines) but I would be very
glad to hear some tips from someone that knows a little better then me.
Thanks,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http
this as a solution, but rather as a
better kludge then my current method of using session time limit which
is good enough for voice mail but is not good enough with conference bridge.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com
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