Gordon Henderson wrote:
On Wed, 30 Mar 2011, Terry Brummell wrote:
Yah, sounds simple, how do you set it up to do this? Fail2Ban was
pretty easy, if it's that easy, why was F2B even created?
It's easy for me because I read an undestand how things work, and deal
with Linux firewalling in a
Roger Marquis wrote:
Vitelity seems to be offline to both IP and voice traffic. Is there any
place to find out what their status is?
09:30 PDT, Inland Northwest (Spokane, WA; Hayden, ID).
I went directly to their website -- http://www.vitelity.net.
Then called my business number and got
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found
when looking at virtualisation at the start of the year. I'm
using LXC and
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Roderick A. Anderson wrote:
Gordon Henderson wrote:
On Tue, 24 Aug 2010, Paul Belanger wrote:
On Tue, Aug 24, 2010 at 11:02 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
I thought OpenVZ was 'depreciated'? That's sort of what I found
I have a rather simple setup running under Asterisk 1.4. I'd like to
move it to a new install of 1.6. Before I bring it online are there any
gotchas I should look for? A Gotcha README would be better but
searching with Google and the forums, for me, gets hits that deal with
hardware issues
Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
Anderson
Subject: [asterisk-users] Using a 1.4 config with 1.6
I have a rather simple setup running under Asterisk 1.4. I'd like to
move
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
... ERROR[25658] codec_dahdi.c: Failed to open
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
--
--
_
-- Bandwidth and Colocation Provided
On 07/02/2010 10:10 AM, Gordon Henderson wrote:
I've just posted this to another list where we were talking about the same
old issues we've been plagues with recently - I'd already posted some
iptables rules, but added more to it for this...
This script probably isn't compatable with
On 06/29/2010 06:53 AM, bruce bruce wrote:
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many
SSH profiles to be saved and allows tunneling etcbut it's not very
good when it comes to scrolling up and down, colors, text size, and
specially it doesn't give a
OT = Old Topic.
Any suggestions for free DID/SIP accounts?
The local Linux User Group would like to see how to set up an Asterisk
system. And in case I can't find or remember who I loaded my TDM400
card to (and because it makes sense) I'd like do a (SIP) connection to a
DID provider. Heck
Steve Edwards wrote:
On Fri, 11 Jun 2010, Roderick A. Anderson wrote:
OT = Old Topic.
Any suggestions for free DID/SIP accounts?
The local Linux User Group would like to see how to set up an Asterisk
system. And in case I can't find or remember who I loaded my TDM400
card
khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
instead of this:
exten
Darrick Hartman wrote:
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
snip /
Randy,
That only addresses EC2 (and assumes that Amazon has any interest in
protecting their reputation). What about attacks that come
Gordon Henderson wrote:
Interesting thread recently about virtual servers...
I'm thinking of doing something similar - right now looking at Containers
(lxc) rather than proper virtualisation though, however it got me
thinking of a poor mans virtualisation solution...
This would assume
Alex Balashov wrote:
It does not appear that you have PostgreSQL set up to listen on a TCP
socket, but only UNIX domain socket. You have this line commented out:
#listen_addresses = 'localhost'
It is required in order to listen on TCP. You should uncomment it:
David A. Bandel wrote:
Folks,
I know this must be a configuration problem. Just changed servers
last nite -- an interim server running 1.6.1.6. Copied all of
/etc/asterisk to the new server and fired it up.
Local or remote PostgreSQL server?
Is the new Asterisk server allowed to connect
Tilghman Lesher wrote:
On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble
Steve Edwards wrote:
2009/9/5 Steve Edwards asterisk@sedwards.com
snip
You can, but I don't. I do it like this:
snip
On Sun, 6 Sep 2009, Olivier wrote:
snip /
To avoid that, have you tried tools like phpLogCon ?
Nope. Never heard of phpLogCon before.
Since it hasn't come up
Roderick A. Anderson wrote:
Olivier wrote:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
snip /
Not specific to Asterisk but there is Config::Std which, in Damian's
blurb for the module, is simple and limited. Still it could
Olivier wrote:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
[section1]
key1=value1
key2=value2
[section2]
key1=value1
key2=value2
...
To increase coherence when running custom-made application in Perl,
Java, PHP, ...)
Is it possible to make a call from a SIP/IAX softphone, say Zoiper, on
one computer to an Asterisk system without having an extension/account?
If so what are the terms I need to search for to figure out how to do
it? So far anonymous SIP has got me the closest I think but no brass
ring.
Wilton Helm wrote:
If life were only that simple. A lot of hacking passes through
unsuspecting intermediary computers, precisely to hide their tracks, not
to mention IP spoofing. People have offered for sale access to 10,000
computers to use for propagating mischief. That's a lot of
Steve Edwards wrote:
On Tue, 24 Feb 2009, David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
In the past 30 days, you've asked questions about
configuring Apache to process PHP files,
Vicidial,
Michael wrote:
On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote:
On Wed, 18 Feb 2009, michel freiha wrote:
I suggest please if someone advice to me a free PDF book just dedicated
for AGI and nothing else
It takes a rare individual to put the effort required to write a book and
then
Doug Lytle wrote:
Josiah Bryan wrote:
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
The first place I usually start is with memtest86
Here, here!
Every time I have had problems with a system (not just Asterisk)
crashing and there is nothing in
Roderick A. Anderson wrote:
And if that ain't confusing I don't know what would be.
I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago
and ended up never using it. Passed it along to a friend who is having
some problems with it. (He isn't on this list.)
We've both
And if that ain't confusing I don't know what would be.
I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago
and ended up never using it. Passed it along to a friend who is having
some problems with it. (He isn't on this list.)
We've both tried searches using Google but
Tzafrir Cohen wrote:
On Wed, Nov 19, 2008 at 11:07:57AM -0800, Roderick A. Anderson wrote:
And if that ain't confusing I don't know what would be.
I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago
and ended up never using it. Passed it along to a friend who is having
Jared Smith wrote:
On Wed, 2008-11-19 at 11:07 -0800, Roderick A. Anderson wrote:
The TDM400 works taking inbound calls and gives a dial tone when the
phone is picked up but as soon as a key is pressed the line (Asterisk
says) hangs up. Asterisk is configured based on a working system
FYI/Heads up,
I /just/ received what looks like a phishing attempt for information
about Open Source PBX usage. It says it comes from Digium but all the
links (including the one for digium.com) point elsewhere.
Rod
--
___
-- Bandwidth and
Steve Totaro wrote:
On Wed, Nov 5, 2008 at 10:26 AM, Roderick A. Anderson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
FYI/Heads up,
I /just/ received what looks like a phishing attempt for information
about Open Source PBX usage. It says it comes from Digium
it looked more like it came from digium.
Rod
--
Alex
On Nov 5, 2008, at 5:26 PM, Roderick A. Anderson wrote:
FYI/Heads up,
I /just/ received what looks like a phishing attempt for information
about Open Source PBX usage. It says it comes from Digium but all the
links (including the one
John Todd wrote:
On Nov 5, 2008, at 7:26 AM, Roderick A. Anderson wrote:
FYI/Heads up,
I /just/ received what looks like a phishing attempt for information
about Open Source PBX usage. It says it comes from Digium but all the
links (including the one for digium.com) point elsewhere
Meftah Tayeb wrote:
my friend i have a problem with linux accessibility
i dont have (not found) a screen reader for Gnome or KDE
this is the reason tha i use windows
but linux is realy best / fast / easy
thanks
A quick search using Google gave me
http://live.gnome.org/Orca
Sound
Darren Severino wrote:
Well, after very quickly making a test call it's not Vitelity. It could
be something with your account? Might want to try opening a support
ticket. If you want, create a sub account and e-mail me off list the
username and password and I'll test it with my box or vice
Tzafrir Cohen wrote:
On Wed, Sep 24, 2008 at 03:23:52PM -0700, Roderick A. Anderson wrote:
Let me know if I should post this on the asterisk-dev list instead.
I am building a Linux-Vserver (http://www.linux-vserver.org) host system
that will have several guests running Asterisk. Since
Let me know if I should post this on the asterisk-dev list instead.
I am building a Linux-Vserver (http://www.linux-vserver.org) host system
that will have several guests running Asterisk. Since the guests can't
load kernel modules or do other dangerous stuff, but can access them I
built
Klaus Darilion wrote:
Hi!
Does somebody know a Thunderbird extension to playback the voicemail
(wav attachment) directly in Thunderbird? (or another neat workaround?)
I didn't see a reply so I'll make a pass at it. First comes the
question: what platform? UNIX/Linux, MAC, or Windows?
As
Karl Fife wrote:
Anybody care to muse on Wi-SIP vs. SIP-DECT?
My limited research indicates that none of the WiSip phones will ever be
able to match the performance of DECT phones. Maybe I'm wrong but a
Wi-SIP phone seems like a DIESEL sports car.
Just for fun!
Igor Hernandez wrote:
I was thinking the same thing I believe Tzafrir just alluded to. If the
passwords are encrypted in the DB with a public key then...asterisk
needs to have the private key stored somewhere to be able to decrypt the
values to authenticate the user. In this way there is
C F wrote:
Very interesting article. I guess we won't know much more for another few
weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_article=1
Interesting! I just went there and the Check your DNS link failed.
Anyone else?
Rod
--
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
the LAN behind the firewall.
Steve Totaro wrote:
Roderick A. Anderson wrote:
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so will only
have SIP connections.
The
these.
Rod
--
On Mon, May 12, 2008 at 4:57 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build
more.
Thanks Al.
Rod
--
Roderick A. Anderson wrote:
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so
Tilghman Lesher wrote:
On Monday 12 May 2008 17:27, Roderick A. Anderson wrote:
Al Baker wrote:
Asterisk will automatically chose the best format - per ATFOT
I guess I'm not getting my head wrapped around this concept. I
understand the choosing but not how I might influence it. Probably
Sanjay Rajdev wrote:
We are looking for a female voice.
I use Callie-8KHz.
Never much cared for Alison so I tried most of them from the demo site
and found Callie to be the smoothest/calmest sounding.
You can download the demo version of the software and try them on the
system(s) they will
Steve Prior's mention of using Allison's voice with Cepstral reminded me
to ask: for a listing of the text for the built-in recordings.
I found a web page but I'd prefer not having to scrape the info out of
it. I didn't notice anything while wandering through the source code/files.
I want to
Russell Bryant wrote:
Roderick A. Anderson wrote:
Steve Prior's mention of using Allison's voice with Cepstral reminded me
to ask: for a listing of the text for the built-in recordings.
I found a web page but I'd prefer not having to scrape the info out of
it. I didn't notice anything
Steve Totaro wrote:
A quote from Tilghman Lesher from a previous post.
That's fine, but I have had the most horrid results using any distribution-
supplied ODBC drivers. The best results are obtained by source-compiling
the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51
Steve Totaro wrote:
On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Has anyone created a worksheet they can share for designing a dialplan,
extensions, voicemail, etc.
I'm making my way through the O'Reilly Book (dead tree version) and
finding
Thank you Erik. I'll use it for building my worksheets. Small number
of phones/extensions doing a very nice, minimal, batch of features.
Rod
--
[EMAIL PROTECTED] wrote:
Rod,
I've made a very basic dialplan for an Asterisk for beginners
presentati on a seminar kind of event last year.
Steve Totaro wrote:
On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Steve Totaro wrote:
On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Has anyone created a worksheet they can share for designing a dialplan,
extensions
Darren Wiebe wrote:
If you're willing to cc me a copy I'll be in your debt.
You bet.
Rod
--
Thanks,
Darren Wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Steve Totaro wrote:
On Sun, May 4, 2008
Has anyone created a worksheet they can share for designing a dialplan,
extensions, voicemail, etc.
I'm making my way through the O'Reilly Book (dead tree version) and
finding it enlightening. I have hacked at dialplans created by others
but never actually came up with a design for my own
Steve Davies wrote:
2008/4/22 Tzafrir Cohen [EMAIL PROTECTED]:
On Tue, Apr 22, 2008 at 09:55:37AM +0100, Steve Davies wrote:
Hi,
Does anyone have a clever method of doing a conditional include =
line in the dialplan?
I want to include a bunch of standard contexts, but in the
This is the first of several questions I have but to keep them thread
friendly I'll post each separately.
From reading the docs I see Asterisk will attempt to bind to all
addresses. What about all interfaces? I have a box with two NICs that
I'd like to connect to both the WAN (Internet) and
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing, my code.
This is desired so I can add/remove/augment dialplans/contexts that
have a
Tilghman Lesher wrote:
On Friday 18 April 2008 13:48:04 Roderick A. Anderson wrote:
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing, my
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Roderick A. Anderson [EMAIL PROTECTED] wrote:
Just to clarify (and tie into Moj's response) if my last 'n' works out
to a priority of 20 I _do not_ need priorities 21 through 32.
That's true. Just remember the dialplan won't fall
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of GUI!,
and We recommend using Firefox.
Does this mean that it is known NOT to work under IE7, or just that it is
Tzafrir Cohen wrote:
On Thu, Apr 03, 2008 at 06:06:19AM -0700, Roderick A. Anderson wrote:
IE was still on the desktop because they had to support a lot
of customers that used IE but for in-house stuff it slowly became a
Firefox place.
That's no excuse.
That's what IETab's
I wanted to try AsteriskNOW plus a few others to see which I can wrap my
head around the quickest.
The issue so far is I can't figure out how to use my Vitelity account
with it. I went so far as to put their Asterisk configuration in the
sip.conf file but still no joy.
Any pointer as to
Eric Bishop wrote:
Do you have step by step instructions on how you created these RPMs. I
would like to create a few of my own but compiled for my own custom
kernel and patchea and am not very familiar with RPM packaging
A good starting point is to download and install the source RPMs in:
Adam Robins wrote:
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
http://www.coraid.com/ for a slightly different approach to large
storage capacity.
bam wrote:
How or when is the voicemail name actually played?
I've recorded my name message and can see that the voicemail directory
now has two new greet files and the original greet.gsm has been overwritten.
# ls /var/spool/asterisk/voicemail/default/4100/INBOX/ -l
-rw-r--r-- 1 root root
Jean-Denis Girard wrote:
Roman Zhovtulya a écrit :
Where did you get it?
I was looking on the internet and couldn't find any link to install this
Mozilla extension.
Have a look at:
http://taina.sysnux.pf:8080/cps/sections/telephonie/copy_of_mozphone/view
Just clic on the link in the install
Dave Cotton wrote:
On Wed, 2005-03-02 at 14:16 -0500, skamp wrote:
Yes thats the link i tried that failed gives me an install error...
soo.
This link gave me no problems, for those who don't speak French don't
worry it's in English.
Colin Anderson wrote:
LESSON FOR NOOB: Don't take it personally. Ask for help in the fashion that
you would be asked. Make sure you have Linux 101 under your belt.
Here, here and I would add ( since I made this mistake ) some basic
telephony understanding.
2. The guy who flames (I was going to
Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A
basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I
Muhammad Muzzamil Luqman wrote:
I have been googling for the RPM kernel-source-2.4.25-040218.i386.rpm or
kernel-source-2.4.25-040218.i686.rpm for the last 59 hrs and couldn't
succeed.
Can someone suggest me some good Redhat Linux 9.0 rpm repositories.
http://fedoralegacy.org/
and
So I've got it installed and running (?) except for one error message
and I haven't had time research it yet but I'd like to get a quick reply
or pointer to my next step to getting [EMAIL PROTECTED] working.
The error is during boot ( Linux ) and comes from ztcfg ( I think?
Memory going
[EMAIL PROTECTED] wrote:
And is there a specific _next_ place ( URL/URL/Wiki ) to continue to get
[EMAIL PROTECTED] configured? Actually I'm testing at home since it is not
considered a good thing to experiment with our business' lines. :-)
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error invalid
compressed format (err=2) system halted message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
Here is an
Daniel Eboa wrote:
I downloaded the iso file of the last release, but unable to burn it on CD. Got
error at 90%. Did anyone experience the same problem ?
Maybe the iso file is corrupted.
Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last
stage of the install ( compiling *
Just a little pun there!
I've been mostly lurking for a couple of weeks and realize how little I
know and understand about this PBX and phone stuff. I did a little
looking about and came across a glossary but they terms are -- for me --
kind of out of context. I'm wondering if there is (much as
Richard Cook wrote:
Hello Rod,
Have you checked out the Wiki. There's lots of information in there:
http://www.voip-info.org/wiki-Asterisk
Obviously not enough. I found it and looked around a bit a few days ago
but never really got into it. With a slap of a clue-stick and a
John Williams wrote:
OK, I have RH9 up and running ... respecting the advice that I should know my way
around Linux before plunging into Asterisk, can anyone comment on the things I
should be able to do with Linux before I attempt to bring up Asterisk? Thanks Much!
P.S. By way of
Well I suspect this list will pick up quite a bit since the article in
Linux Magazine. So I'll try to get my questions in early.
I looked at Asterisk about a year ago but the TPTB wanted a prebuilt
solution (at that time) and I couldn't find anyone local enough to
satisfiy them. We still don't
83 matches
Mail list logo
