It consists of a MySql database using triggers, and a PHPAGI
script that does the calling.
http://www.voip-info.org/wiki/view/MySql+trigger+based+alarm+response+system+for+AlarmReceiver%28%29
Any comments or fixes are welcome.
Ill work on
a web setup front end so people can
any programs out there that do this ?
Dovid
- Original Message -
From: Stefan Reuter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 27, 2006 8:22 PM
Subject: Re: [asterisk-users] Manager interface
Dear All,
After doing the test everything went fine, Thanks Anthony for putting
me on the right direction.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file extensions.conf but
I have a macro that bridges a call or sends it to
VM based on what the called user presses. If he just hangs up the call gets
dissconnected. This is the error I get
WARNING[7598]: res_features.c:1381
ast_bridge_call: Bridge failed on channels SIP/11011-9ea135e0 and
SIP/11010-007d0780
Is
Hii All
I have problem about zap channel.
When I am trying to make 2 sip call in same time first
is succesful but second is fail
Sip debug is on and received message for failed call
is Unable to create channel of type 'Zap' failed
Cause 0 unknow
My Extensions conf;
[nortel]
exten =
I'm having real problems getting my Sangoma A200 card with FXO board in to
detect hangup at all.
Basically if the remote end hangs up the call, Asterisk does not seem
to detect a
hangup.
A month or so ago I was running a system with 2 x X101P cards in, this
detected hangup fine.
Since switching
Emre BALCI wrote:
[nortel]
exten = _1XXX,1,Dial,Zap/1/${EXTEN}
exten = _5XXX,1,Dial,Zap/1/BYEXTENSION
These should be:
exten = _1XXX,1,Dial(Zap/g1/${EXTEN})
exten = _5XXX,1,Dial(Zap/g1/${BYEXTENSION})
You may want to read further on the dial command:
From looking at the wiki it dosent seem like the option is available. You
may want to edit app_dial.c yourself or contact the developers list.
Dovid
- Original Message -
From: snlee [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 25, 2006 9:33 PM
Subject:
There is no real way to give an answer. It depends
on many factors such as Codc type, the server specs, what else you are running
etc. There are too many variables to take in to account. The only real to know
is to test, test and test again.
- Original Message -
From:
This post is from three days ago. Dont know if you
found a solution or not. It sounds look to be your provider. What comes up in
the CLI ?
- Original Message -
From:
Crazy
Boy
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 26, 2006 3:39
AM
Subject:
Does anyone know if there is such a solution to
listen to XM radio's service thru thier site ?
- Original Message -
From:
Alex
Robar
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, July 25, 2006 12:06
PM
Subject: Re:
On 7/30/06, Dovid Bender [EMAIL PROTECTED] wrote:
any programs out there that do this ?
Dovid
You can use FOP: http://www.asternic.org
--
Nicolás Gudiño
Buenos Aires - Argentina
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Matt Riddell (NZ) wrote:
Tom Vile wrote:
Did you look on the site?
http://www.4psa.com/products/voipnow/demo.php
Man that looks nice. Kinda reminds me of the Plesk.
Anyway, I've put up a screenshot with the original post at:
http://www.sineapps.com/news.php?rssid=1399
Looks nice,
Tzafrir Cohen wrote:
On Fri, Jul 28, 2006 at 02:04:10PM +0200, Administrator TOOTAI wrote:
Morning everybody,
I try to install an asterisk test server with trunk branch and get this
error when compiling zaptel. Asterisk core compile fine as well as SVN
1.2 branch. It's a Debian SARGE
On Sun, Jul 23, 2006 at 04:24:04PM +1000, Devraj Mukherjee wrote:
Hi Everyone,
I am running Asterisk 1.2.7 Zaptel 1.2.5 on CentOS 4.3 on a Dell
PowerEdge SC420. I was running an older version of Asterisk (can't
remember what, but was using the wcfxs kernel module) under Gentoo
Linux and
On Mon, Jul 24, 2006 at 02:41:51PM +0100, Dean @ INKnBITs wrote:
I'm getting:
354+Enter+message,+ending+with+.+on+a+line+by+itself 0 0 54 0 4437
SMTP - - - -
550+Administrative+prohibition 0 0 30 0 4781 SMTP - - - -
What MTA do you use? (e.g: sendmail, postfix)?
While you're at it: what
just have a look at http://sourceforge.net/projects/oscsta
Hi,
Can anybody tell me that is their CSTA support for asterisk
sanchal
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To UNSUBSCRIBE
Hans-Jürgen Brand wrote:
just have a look at http://sourceforge.net/projects/oscsta
looks rather empty...
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Description: OpenPGP digital signature
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asterisk-users mailing list
To
I've just read through the voip-info link and the GPL
FAQ.
My professional (but not formally legal) interpretation
would be:
As long as the none-GPL program or module is a separate
entity and not actually compiled in to another program, you can consider the two
licences seperately.
You
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dovid Bender wrote:
Does anyone know if there is such a solution to listen to XM radio's service
thru thier site ?
What format is the feed?
- --
Cheers,
Matt Riddell
___
The code in app_followme.c that's now in /trunk has a framework to
play audio to more than one channel at the same time.
On 7/30/06, Dovid Bender [EMAIL PROTECTED] wrote:
From looking at the wiki it dosent seem like the option is available. You
may want to edit app_dial.c yourself or contact
Works fine for me with 1.2.10...
In fact, I've just started using environment variables set when Asterisk
is started, setting global variables and using them in my AGI's (written
in c).
Here's a snippet from my /etc/init.d/asterisk:
source /etc/asterisk/environment.sh
nice
Tom Vile wrote:
Did you look on the site?
http://www.4psa.com/products/voipnow/demo.php
Does above means that the license for voipnow need to be paid to packet 8 as
well?
http://biz.yahoo.com/prnews/060613/sftu062.html
Senad
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Hi
On 7/27/06, Luki [EMAIL PROTECTED] wrote:
There is this old patch that does remote MWI over IAX (among other
things). I used it on earlier versions and it worked quite nicely.
This was before 1.2 so it may no longer work at all. At the very least
it will likely required some updating.
On Tue, Jul 25, 2006 at 08:53:11AM +0200, Jan du Toit wrote:
Hi.
I'm trying to install and configure a TDM01B -1 FXO card.
I'm getting the following errors when starting up asterisk:
Jul 25 08:48:40 WARNING[1775]: chan_zap.c:923 zt_open: Unable to specify
channel 1: No such device
Jul
On Sat, Jul 29, 2006 at 04:51:27PM +0300, Tzafrir Cohen wrote:
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote:
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2 type forum. It seems to be working well
for me and I think some of you
The Panasonic
will not allow you to dial an extension directly from the outside.
You need to have Asterisk setup as a Panasonic extension and use an extension
to extension format in your dial plan.
Try making
connections like this.
--
--
|
|
| |
|
SIP
Hi,
can anybody recommend HP Proliant ML110 for Asterisk and ISDN interface cards?
This Server has only two PCI 32Bit/33 MHz 3,3 Volt.
Is this OK for PRI cards?
thanks..
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asterisk-users
snip
I've never heard of that (UAIP) before !
now you do :) It's also known as UAUDP.
Do you have anything describing this protocol ?
Have not seen any official documents. There was mention of an Ethereal
dissector from Alcatel, but none of my contacts at our local Alcatel
vendors could
Hello,
We've just released our Libero Management System application, a web-based
interface to configure and manage your Asterisk-based PABX. Designed for the
not-so-novice Asterisk administrator in mind.
LMS is simple to install, has minimal requirements (no external databases or
components
Hi Guys,
I am writing a dynamic extension mapping dial plan
application so that staff can essentially log into different SIP stations and
have their calls delivered to this phone as well as having their caller id
manipulated so that the calls appear to come from the correct user. I
Hi,With freepbx I have created sip users and sip trunks. But I need to charge calls and I want to use a2billing. But in a2billing I see, that I can create sip users and trunks too. So, is possible to use FreePBX with a2billing. Let say, with FreePBX I'll create trunks and extensions and with
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