Hi All,
There are a lot of existing projects for configuring Asterisk via GUI,
so instead of trudging through them all, I'm hopeing to get some
guidance.
My architecture is ITSP based, we supply hosted PBX's to business
customers. A few systems are dedicated PBX's but the majority are
hi,
i am using asterisk 1.6 and i want to integrate sphinx speech engine with my
asterisk, so that i can use the generic speech API provided by asterisk 1.6...
Plz help me, how can i do that... any help will be highly appreciated...
waiting for your positive response...
Thanks Regards,Rizwan
Hello
I also tried it in begining but cant give time to it. So no success.
you can try this link
http://www.voip-info.org/wiki/view/Sphinx
http://cmusphinx.sourceforge.net/html/cmusphinx.php
http://cmusphinx.sourceforge.net/html/cmusphinx.php
hope this helps
On Tue, Dec 1, 2009 at 12:16 PM,
Hello,
I have installed iaxmodem(1.1.1) and hylafax(4.4.4) but my Asterisk(1.2)
configuration seems to be wrong (calls are destroyed by asterisk).
Does someone know how to configure Asterisk (iax.conf, sip.conf and
extension.conf)???
This is my configuration of iaxmodem:
device
Hi,
I am trying to use Asterisk PBX with T1 CAS. The setup that I am looking for
is as below
Analog phones == Asterisk T1 CAS === Integrated Access Device
IP Network for VoIP.
The Asterisk has a T1 card and I want a CAS config between Asterisk and T1
port of IAD. The
Hi all,
I would need to parse asterisk configuration files with PHP code.
Does anyone know if one already exist?
Thanks in advance
Yann JOUANIN
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On Mon, Aug 13, 2007 at 10:34:56AM +0200, [EMAIL PROTECTED] wrote:
I would need to parse asterisk configuration files with PHP code.
Does anyone know if one already exist?
Parse? In what way? What information do you want to extract?
--
Tzafrir Cohen
icq#16849755
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an
ATA? If so, could you share how you did it?
TIA
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Hello
Am starting on my Asterisk journey - am getting a single span Digium
card to connect Asterisk to our Alcatel 4400 EPABX and install about 100
VoIP instruments.
The Asterisk VoIP extensions and Alcatel digital extensions have to talk
to each other.
Am I right in understanding that
IN
On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote:
Is that broadly correct?
Yes
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Title: RE: [asterisk-users] Asterisk Configuration Complete Newbie question
Thanks very much - let me see how far I can take it now.
Best wishes
Iyer
-Original Message-
From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora
Sent: Fri 10/6/2006 03:37 PM
To: Asterisk Users
Hi
All
I am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.
I have configured two extensions in extensions.conf
exten = 228,1,Dial
exten = 234,1,Dial
and configured the xlite soft phone. when I am calling from 234 to
228 it is unable to establish
Hello Linga,
you could download and install the ESCAUX net.PBX Free Edition and
create whatever device or call flow you want with the web interfaces.
Afterwards, in the /etc/asterisk/ directory, take a look at the
generated configuration files (gen_sip.conf, gen_extensions.conf,
Hi
All
I am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.
I have configured two extensions in extensions.conf
exten = 228,1,Dial
exten = 234,1,Dial
and configured the xlite soft phone. when I am calling from 234 to
228 it is unable to establish
Hi:
First at all:
You SIP phones are right register on sip.conf file?
Cris
From: R.Linga Reddy [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
You need to tell asterisk what to dial. Check the dial command syntax and probably the sip.conf file.On 8/9/06, R.Linga Reddy
[EMAIL PROTECTED] wrote:HiAllI am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.I have configuredtwo extensions in
Hello,I'm new to Asterisk. I want to do the folloing job.I want to run Asterisk as a voip gateway to forward h323 calls to another gateway. my-gateway - Asterisk -- your-gateway h323 h323Is it possible to do this? If so, can anyone give me an idea
Hi all,
I want to
configure Asterisk using Mysql Database. But on compilation of asterisk-addons
I am getting some errors. I have pasted the errors in the pastebin.
Please checkout this link. http://pastebin.com/499106
Also please do let me know which are packages required for
I have installed Redhat Linux 9 and Asterisk 1.2.1
on new computer. I need to know initial configuration of Asterisk i.e How to
register a sip user?. What files do I have to edit?
I am new about the Asterisk
please help me
Faheem Ahmed
___
Here is where you will find the answer to all of your questions:
http://www.asterisk.org/
http://www.voip-info.org/wiki-Asterisk
Jonathan
On Sat, 2005-12-24 at 01:34 +0500, Faheem Ahmed wrote:
I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I
need to know initial
I want to let Asterisk read its configuration from a mysql database. I
configured everything according to the wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config.
However it doesn't work. I am using 1.0.8 asterisk version and here are
my config files:
Extconfig.conf:
is your Asterisk compiled from cvs head?
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frank
Aartman
Sent: Thursday, August 18, 2005 3:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk configuration from database
Nope, I got the stable 1.08 release from cvs.
Frank
From: Wei Kun [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk configuration from database
withres_config
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL
That wiki page is old, ugly and out of date. There are many like it and
if I only knew how to delete wiki pages, I would clean it up some.
The easiest way, Frank, to do what you want is to download CVS-HEAD and
use ARA to store your config files. Also download addons from HEAD and
you can use
Dear users,
I am new to this mailing list. Can someone send me a guide
or steps to configure Asterisk on Linux box? I will highly appreciate.
Regards,
Afzaal
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On Monday 25 July 2005 14:07, Afzaal Mirza wrote:
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate.
http://www.voip-info.org
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Hi Afzall,
i am also still a beginner on *. A made best experience with the * wiki on
http://www.voip-info.org/wiki-Asterisk. Maybe you start with the introduction part.
Afzaal Mirza wrote:
Dear users,
I am new to this mailing list. Can someone send me a guide or steps to
configure
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Configuration
On Monday 25 July 2005 14:07, Afzaal Mirza wrote:
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate.
http://www.voip-info.org
Mirza [mailto:[EMAIL PROTECTED]
Sent: Monday, July 25, 2005 7:08 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Configuration
Dear users,
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate
Afzaal Mirza wrote:
Dear users,
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate.
Regards,
Afzaal
Hello everybody,
I'm newto Asterisk and I'm trying to configure the SIP side.
I use Asterisk under the following configuration:
SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A
I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to
13, 2004 10:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Hello everybody,
I'm new to Asterisk and I'm trying to configure the SIP side.
I use Asterisk under the following configuration:
SIP Proxy INTERNET | NAT FIREWALL
Alex,
The media ports are configured in rtp.conf. Also, note that Asterisk
sends RTP packets out the same ports it expects them to return on. This
has the effect of creating a NAT mapping for that 5-tuple, as well as
opening a hole in your firewall (naturally, YMMV depending on exactly
what
Brian D'Arcy wrote:
Hello all,
Im having a nightmare of a time trying to get stable results with SIP
clients on Asterisk. I cant seem to find a configuration that works!
In our office, we run a Sonicwall Pro 200, which is a sip aware,
stateful firewall.
We've discovered that certain
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre, P.E.
Sent: Saturday, April 24, 2004 5:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
Brian D'Arcy wrote:
Hello all,
I'm having a nightmare of a time trying to get
On Saturday 24 April 2004 18:39, Brian D'Arcy wrote:
Hi Russ,
On a side note, I tried IAX2 last night for the first time using
IAXPHONE. HOLY CRAP I'M IMPRESSED!!! Everything just *works*, period.
I might just use softphones until IAX hardphones are released and say
screw SIP.
I'll second
Hello all,
Im having a nightmare of a time trying to get stable
results with SIP clients on Asterisk. I cant seem to find a
configuration that works! In our office, we run a Sonicwall Pro 200,
which is a sip aware, stateful firewall.
Originally, I had configured Asterisk to run on
Hi,
Is there any patches to make asterisk to read all the conf
from mysql instead of files?
Regards,
Soragan
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