Now I'm ready to begin playing with dial plans and am having a difficult
time getting started.
You may want to read the book :
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
That should help you
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Okay, I have Asterisk up and running on Fedora Core 5 with a TDM400
board with one FXO and FXS module. Zap is up and running and * is
functioning with the modules. Oh yeah, and I have some soft phones
configured and have them working as well.
Now I'm ready to begin playing with dial plans
Wanted some advice for the docs that you'd recommend someone new to
Asterisk to read. I have a good knowledge of Unix and networking, so
that part shouldn't be a problem.
Try...
http://www.asteriskdocs.org/modules/news/
The authors of Asterisk: The Future of Telephony are pleased to
Hi Guys,
Wanted some advice for the docs that you'd recommend someone new to
Asterisk to read. I have a good knowledge of Unix and networking, so
that part shouldn't be a problem.
Cheers,
Sukrit.D.
--
\|||/
(o o)
Wanted some advice for the docs that you'd recommend someone new to
Asterisk to read. I have a good knowledge of Unix and networking, so
that part shouldn't be a problem.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
Welcome to *
check http://www.asteriskguru.com/tutorials/
Diyanat
From: sukrit [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] getting started
Date: Fri, 16 Dec 2005 09:06:28
Hi sjkrit
If you are beginner of the Asterisk then i think Asterisk - Future of
Telephoy is the best book to start with the asterisk. you
can download a free copy from voip-info.org.
http://www.voip-info.org/wiki/view/Asterisk:+The+Future+of+Telephony
Thanks
Chandan
On 12/16/05, sukrit [EMAIL
Let me simplify my problem. I have a single Aastra 9133i SIP phone and
latest Asterisk from SVN source running on Fedora Core 4. The phone
currently says No Service I would like to be able to dial 1234 from
the phone and get Asterisk to play back an audio message or say some
digits. I
One more thing. I upgraded the firmware of the 9133i to 1.3.
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On Mon, 2005-12-05 at 11:15 -0500, Robert La Ferla wrote:
Let me simplify my problem. I have a single Aastra 9133i SIP phone and
latest Asterisk from SVN source running on Fedora Core 4. The phone
currently says No Service I would like to be able to dial 1234 from
the phone and get
Pete Barnwell wrote:
I wasted a lot of time getting 9112is to work with almost identical
setup. The problem I eventually found was that the 9112is look for the
config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas
the documentation says they look for lower case, so they were
On Mon, 2005-12-05 at 11:27 -0500, Robert La Ferla wrote:
Pete Barnwell wrote:
I wasted a lot of time getting 9112is to work with almost identical
setup. The problem I eventually found was that the 9112is look for the
config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas
Dave Cotton wrote:
One thing is to do a factory reset to reinit everything, I did that with
my 9112i after upgrading the firmware.
I just did that. Now Asterisk is giving me the follow error: (0.99 is
my Asterisk server and 0.111 is the phone)
Dec 5 12:04:10 NOTICE[14222]:
I solved it by registering the phone in the sip.conf.
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I have a Aastra 9133i phone and would like to do a simple test to make sure
everything works. I already assigned an IP address to the phone (I'm able to
ping it.) I have Asterisk running (installed Asterisk and Zaptel only) but not
configured. I don't have a FXS/FXO card yet but I would like
Hello people, actually I'm new on this new technology (for me). Well I'll
tell you what do I have got and what I like to do, so
if I made the mistake to post this new thread here, just forgive me.-
Let me intruduce myself and what I am working on.-
Well, I'm from Argentina and, actually I'm a
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
I can use older mdk or red
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani
[EMAIL PROTECTED] wrote:
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think
Luca Bariani wrote:
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
Luca,
I run Mandrake 10.1 Official with current updates and compile without issue.
Doug
___
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani
[EMAIL PROTECTED] wrote:
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
It compiles perfectly on MDK 10.0/10.1/Cooker, if you posted the
compilation
On Tue, Mar 08, 2005 at 04:55:30PM +0100, Luca Bariani wrote:
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk
Better yet, ditch the Mandrake box and try [EMAIL PROTECTED] for you test
machine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, March 08, 2005 10:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users
I have installed asterisk - hurray!
I want to configure, i.e.,
800#---switch-asterisk-phone
need to create members and callers. Realize I need to configure the
dialplan in extensions.conf, however isn't there a checklist that
helps coordinate the many extensions of this
Hello!
I'm new to asterisk, too.
* Bilal Ghayad [EMAIL PROTECTED] [2005-01-14 21:29]:
1) Where I can find the PC to Phone software to be used? Does it support
G723 codec? If not, then the used codec will need how much bandwidth? What
the client OS should be?
I found something written about
Hello,
I have been on the IRC Chat but will post here as well.
We have 1 office currently with 4 land lines.
We want to use asterisk as our call attendant with 6 lines.
We were going to use vonage but it was recommended that we look over the
voip providers in the wiki because other providers
Hi All;
I am starting to know about Asterisk and trying to use it and knowing its
functionalities, I hope that here I can find answers for my following
questions:
1) Where I can find the PC to Phone software to be used? Does it support
G723 codec? If not, then the used codec will need how much
I am starting to know about Asterisk and trying to use it and knowing its
functionalities, I hope that here I can find answers for my following
questions:
Welcome to the community! Most of the answers are here
http://voip-info.org/tiki-index.php?page=Asterisk
http://asteriskdocs.org
I am interested in learning Asterisk and have DSL (1 static IP) and a
single POTS line at home. I have an Ethernet LAN running behind a
Linksys router using NAT. My question is only about the hardware
needed at this point. The software configuration I will read about
and learn. So what
What is the minimum set of configuration files needed to operate a SIP
only Asterisk setup?
I wrote a howto that might help:
http://iheavy.com/modules.php?op=modloadname=Newsfile=articlesid=35
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[EMAIL PROTECTED]
Hello ,
Ill just started with asterisk and I would
liket to to dial between your two
phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks
and i have the ring in the phone but iam not able to place a call
Between the twe phone .
[EMAIL PROTECTED] is believed to have said:
Hello ,
I'll just started with asterisk and I would liket to to dial between your
two phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks and i have the ring in
the phone but i'am not able
On Wednesday 01 December 2004 17:11, Aldo Bergamini wrote:
[...]
I can't help as this is the same problem I have with the very same sample
configurations.
Here I am using the OSX install of Asterisk (with an older release).
What I saw poking around is that on the server machine at port 5060,
[EMAIL PROTECTED] is believed to have said:
Telnet uses TCP, SIP listens on UDP, use netstat instead.
B
Bob,
thanks for the hint! I should have imagined that SIP could not use a tcp
protocol...
Aldo
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[EMAIL
Hi all
Problem with gnophone:
I can not make a call. (just hangs)
Im am a novice to Asterisk but quite experienced Linux user. I am having
some problems with the gnophone. I have tried to isert my user/password
but nothing have changed.
I have tested the michrophone and it is working. The sound
I'm interested in
getting started with * and have some questions that I could sure use some help.
Appreciate any and all inputs
1) Regarding the
softphone client - what are the differences between an SIP client such as
available from www.xten.com or an IAX client
(there are several
What is the best way in getting started evaluating Asterisk? Are there
recommendations on the types of card I should be using for an initial eval?
Thanks
Peter
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[EMAIL PROTECTED]
download the code,
complile the code,
start bashing on configs. :)
if you want to glue to the PSTN, i'd
recommend getting a FXO (WC-X100P) and FXS (TDM-10B)
card and some cheap SIP / VoIP phones (grandstream or snom)
john brown, ceo
chagres technologies, inc
Providers of VoIP hardware
Hi,
I am a total newbie to asterisk and can't find any useful documentation
for asterisk...how are people supposed to get started?
I'd like to know, how I create User Accounts, so that a SIP UA can login
into asterisk with a password, for example.
___
Read the documentation, read the sip.conf file. And if it still doesn't
make sense try one more time through the documentation and config file.
At that point you should at least know enough to ask pointed questions
at specific problems in your configs.
On Mon, 2003-07-14 at 11:19, Johannes
Just purchased a couple of T100 and E100 cards in order to interface from
our company's proprietary system through a linux gateway. I am new to
Asterisk and Digium.
After installing the T100 card, I went looking for drivers for this card.
Are the drivers built into the Asterisk application? If
To enable any of TxxxP or ExxxP card you must compile the zaptel package
from cvs.
Then to enable the T100P or E100P you should load the wct1xxp module.
Then you can use the zttool (/sbin/zttool) to check the status of the cards
and of the spans you've configured in /etc/zaptel.conf
--
Stefano
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