Hello:
We are LINUX Services Ltd. in NZ.
We can help you out.
We are local resellers for digium hardware and we can also help you with
the installation.
Best regards,
Carlos Hernandez
http://www.linuxservices.co.nz
Wayne Methorst wrote:
Can anyone point me in the direction of a Asterisk
Is there any way to setup callback (for DISA) without going through writing
an AGI script ?
I have tried to use
exten = h,1,System(callback)
but this is what I get:
Feb 12 10:09:29 WARNING[1082809536]: asterisk.c:255 listener: Select retured
error: Interrupted system call
Feb 12 10:09:29
Since I got a few requests for this here it is:
http://www.rami.info/software.php?softwareid=5
On 02/11/04 12:40PM or some time around that time, Rami AlZaid wrote:
This is the same case in Kuwait. I've tried the Artech EX200 Caller ID
converter with no use. What I ended up doing is making a
On Tuesday 10 February 2004 17:11, Tim Sailer wrote:
Feb 10 09:57:37 WARNING[98311]: db.c:46 dbinit: Unable to open Asterisk
database
I'm seeing this in my logs, 0.7.2 (debian package). I looked through the
source, and thought it was looking for a RDBMS (mysql/postresql), so
I set both of
I know this is not a perl user list but it has to do with something I'm
trying to get working with Asterisk. I'm trying to create an AGI script
that can play all files in a directory. Hopefully get it to the point
where the user can hit '2' to continue and 'anyotherkey' to exit.
This is what I
Title: Message
Here
is some config that I cooked up. It may be a little rough around the
edges, and it incorporates multiple users.
exten
= *801,1,Answerexten =
*801,2,SetVar(temp=${loggedin${CALLERIDNUM}})exten =
*801,3,GotoIf($[${temp} = 1]?50:)exten = *801,4,GotoIf($[${CALLERIDNUM}
=
As there are no official links to this resource yet, here's another
reminder for those that don't know. There's a new search engine for this
list now located here:
http://asterisk.linkx.net/cgi-bin/asterisk
The indexes are updated once an hour.
- Kim
I too have seen a couple of system freezes for no apparent reason. I am *
on a RH9 box with kernel 2.4.20-28.9.
B. J.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Biggs
Sent: Monday, February 09, 2004 12:05
To: [EMAIL PROTECTED]
Subject:
I have the MySQL CDR working, and, being a long-time DBMS programmer,
I'm looking to put other things into the database. I see that Voicemail
*used* to be supported in MySQL, but that seems deprecated. Is there
anything else that has DB support? I'd sure like to be able to do
more from scripts,
I had noticed that the jitterbuffer settings under Asterisk didn't seem to
work very well, then I noticed that there was a typo in my iax.conf file
where I had:
maxexccessbuffer=750
which should have been
maxexcessbuffer=750
I have just realised that I didn't make this typo, it is actually a
As a followon to my previous unanswered question.
Why does my DG104S keep sending RTP packets?
Asterisk has hung up, but the DG is blasting away:
SRC=27343, Seq=8206, Time=1334079
379.924789 DG_IP - ASTERISK_IP RTP Payload type=ITU-T G.711 PCMU, SSRC=27343,
Seq=8207, Time=1334239
379.925240
Hi, I'm new to both asterisk and festival. I'm trying to figure out how
to change the voice festival uses. For example, I've downloaded
don_diphone to festival/lib/voices/english. I then edited
/etc/asterisk/festival.conf and changed the festival command to:
Hi all,
I am testing Audiocodes MP 104 fxs gateway with Asterisk but I already have
problems with registering. I was wondering whether anyone has used
AudioCodes fxs gateways with Asterisk and could help me out here.
SIP debug log:
to 80.54.223.79:1025
Transmitting (NAT):
SIP/2.0 407 Proxy
I was wondering if anybody was planning on attending Jeff Pulver's Spring
VON conference in Santa Clara. I am thinking about going and was hoping, if
enough Asterisk people are going to be there, if we couldn't hold some kind
of ad hoc Astericon or something. Perhaps we could rent a room at the
hello,
Is it normal that * starts its billing when
voicemail starts to prompt? can I do something like it will only start to
bill if the caller left a message? right now, im seeing that unanswered calls
that are forwared to voicemail are considered billable as well as calls to
On Thu, 2004-02-12 at 12:28, B. J. Bomar wrote:
I too have seen a couple of system freezes for no apparent reason. I am *
on a RH9 box with kernel 2.4.20-28.9.
Without wanting to sound like a RH basher that I normally am, could this
be a RH issue since I haven't noticed(maybe foggy memory) any
Hi Stan,
Thanks for the info, it helps me out loads.
JR
On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote:
This is slightly off topic so sorry for the intrusion.
I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly
Try this Perl subroutine to get an alphabetic list of all files, excluding
the dot files, and excluding sub-directories:
use DirHandle;
sub justthefiles {
my $dir = shift;
my $dh = DirHandle-new($dir) or die can't opendir $dir: $!;
return sort # sort pathnames
--- Tony Buser [EMAIL PROTECTED] wrote:
Hi, I'm new to both asterisk and festival. I'm trying to figure out
how
to change the voice festival uses. For example, I've downloaded
don_diphone to festival/lib/voices/english. I then edited
/etc/asterisk/festival.conf and changed the festival
Just a question:
everytime my asterisk make a sip call it comes out with
Specify address with base=0xN
over the linux console.
and then, after 1 second, places the call
I suppose that the error is related to zaptel cards, but I have none
in my pc and zapata.conf is competely blank.
Can
Hi,
Any queries regarding Asterisk in New Zealand should be forwarded to myself.
I can be contacted at the email address above or:
Phone (03) 470 1641 x 818
Cell (021) 138 7245
Fax (03) 470 1645
Can anyone point me in the direction of a Asterisk developer in New
Zealand that we could
I had kphone working just fine before I (wiped everything) and installed
Fedora. If you still need it I canreinstall it and let you know how I
got it to work.
On Wednesday 11 February 2004 11:42 am, Regovich, Timothy wrote:
Not ACK'ing an invite can be problematic for the statemachine.
I have some questions for anyone that can help. I discovered an email in the
archives about someone adding an external call control router on WIndows
2003, but could not find a reference to the code. I wanted to see how far I
could go with AGI scripts before having to modify the code.
I have
I have one of my IpDialog phones giving this error about once an hour.
On the Asterisk server CLI I get this message.
Got SIP response 481 Call Leg/Transaction Does Not Exist back from
204.241.XXX.XXX
If I go to the phone and dial out it works and I no longer get the
message. Also if I check
I have perl scripts for doing voice contract recording via an extension
including 5 digit codes for each one and the ability to play them back.
Please mail me off list if you are keen.
Hi,
Does anybody know if it is possible to record a conversation with
asterisk ?
Regards
So here's a dumb question, but what's the VON conference? I would also be
interested in talking to others using Asterisk.
Chris
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 12, 2004 2:30 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I've been seeing these error messages for a couple of days but, I can't
figure out what is causing them.
Feb 12 18:22:42 WARNING[98311]: Got 200 OK on REGISTER that isn't a register
Feb 12 18:22:43 WARNING[98311]: Got 200 OK on REGISTER that isn't a register
Anyone know what I should look for?
I'm experiencing periodic beeps or screeching when I'm on a call via
the x101p card to/from PSTN. Echo cancellation seems to be working
fine. The beeps seem to happen with echo cancellation on or off. Is
there a setting I can tweak for this?
The problem does not occur if I'm
Not sure if I will attend VON, but myself and a friend would be
way into an * nerd fest.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
If there was a reasonable price on the exhibits, I would be interested. Any
vendors out there with free passes??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol
Sent: Thursday, February 12, 2004 2:30 PM
To: [EMAIL PROTECTED]
Subject:
Same here...
Usually after several of these show up in my system log:
Power alarm on module 1, resetting!
Need to unload/reload module wcfxs in order to get the dial tone back.
Happens several times a week, sometimes more frequently.
John
Hi!
Is there any way to setup callback (for DISA) without going through writing
an AGI script ?
Here is an example that might put you onto the right path:
http://ns1.jnetdns.de/jn/relaunch/asterisk/page14.html
Cheers, Philipp
___
Asterisk-Users
So here's a dumb question, but what's the VON conference? I would also be
interested in talking to others using Asterisk.
Chris
[Steven Sokol] VON is the Voice-Over-the-Network conference put on by Jeff
Pulver from Free World Dialup (FWD). It is one of several national
conferences
Chris Albertson wrote:
try adding a set of parens like this:
festivalcommand=((voice_don_diphone)(tts_textasterisk
%s'file)(quit))\n
Unfortunately that results in the following error at the asterisk console:
Feb 12 19:45:27 WARNING[409626]: app_festival.c:437 festival_exec:
Festival
Here's the web site for the convention http://www.pulver.com/von/
The convention center has conference rooms and breakout rooms. I bet if you
asked nicely, you could get one for an asterisk BOF
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol
Looks interesting - BUT Very pricey - they even charge $150 for an
exhibits-only pass, which usually would be free at most trade shows!
Here's the link - suggest feedback to the sponsors about the high pricing!
http://www.pulver.com/von/register.html
Scott M. Stingel
Emerging Voice
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able to
transfer the call directly into a persons unavailable mailbox. Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
Actually, you just need to create an extension. Here's mine:
Exten = _*55.,1,Answer()
Exten = _*55.,2,Voicemail(${EXTEN:3})
Exten = _*55.,3,Hangup()
Transfer the party to *55 (where is the mailbox to transfer into).
There may be more elegant ways to do this, but it works for me.
Sean Garland wrote:
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able to
transfer the call directly into a persons unavailable mailbox. Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
Roy wrote:
Here's the web site for the convention http://www.pulver.com/von/
The convention center has conference rooms and breakout rooms. I bet if you
asked nicely, you could get one for an asterisk BOF
Yeh, but what kind of beer do they have on tap?
--
Bob Knight
[-w] the work option
I have a basic x100p setup and several soft and hard phones that work great
until they hit the PSTN. Like a lot of the posts I've seen I've gone through
just about every echo can including mark 2, 3, and the steves with the
aggressive protection on mark 2. I am running the latest CVS source
Yeh, but what kind of beer do they have on tap?
[Steven Sokol]
For what the thing costs, they _should_ have Samuel Smiths or perhaps Chimay
(sp?) but like most hotels they probably have the usual swill plus Sam
Adams. If we have Astericon, we'll BOOB (bring our own beer).
Geert Nijpels wrote:
I run 0.7.2 and have no crashes. Do you have an error message? If this
is reproducable, please update to latest stable CVS. Generate a core
file a make a backtrace. Then post a bug at http://bugs.digium.com/
If you need help please email.
Kind regards,
Geert Nijpels
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