On Sun, Aug 12, 2007 at 11:12:33PM -0600, Stephen Bosch wrote:
No -- sorry, my mistake. I got the name wrong. The card is actually from
PhonicEQ; there's a description of the card at quadbri.phoniceq.com.
The right URL:
http://quadbri.pbxhardware.com/
Drivers:
On Sun, 12 Aug 2007, MOSBAH ABDELKADER wrote:
Hello all,
have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to
a .gsm audio file to use it as a voicemail file with Asterisk.
'sox' should be able to do this, AFAIK.
--
Alex Balashov
Evariste Systems
Web:
On Sat, 11 Aug 2007, Jay R. Ashworth wrote:
On Sat, Aug 11, 2007 at 01:25:32AM -0400, Alex Balashov wrote:
If it ever happens, it will certainly take Asterisk's usefulness to a
whole, whole new level, at least as a good cheap media gateway.
I dunno; my instinct is usually not to use PCI
Hi,
I am having a strange codec issue on Asterisk 1.4.10. It is a small test
scenario running on Fedora 5 and Pentium 4. Recently I have downloaded the
test G729 codec so that I can test Firefly 3010ATA. I am using an IAX
termination provider for PSTN calls , which uses ilbc codec . The issue is
On Sat, 11 Aug 2007, Jay R. Ashworth wrote:
On Fri, Aug 10, 2007 at 05:20:38PM -0400, Alex Balashov wrote:
It might be possible to glue something together with it and OpenSER and
a media gateway control protocol like H.248 and a few of these SS7-IP
appliances, but it would have all the
Hello Olivier,
I don't have this answer but would be curious to know its price for
reseller.
Any clue ?
the price for the OpenStage 20 seems to be in the range of the Snom
360. The Openstage 40 seems to be in the range of the Snom 370. The
Openstage 60 is 50% higher than the 40, with the 80
Hi all,
I am having problem in using SIP gateway in a very high volume
environment (4+ concurrent dial-in) and phone calls in every seconds.
There are totally 12 phone lines. Currently I am using 6x VIP-450 for
12 FXS and 12 FXO. However, in this high volume environment, problems
encountered:
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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I'm wondering if anyone has seen (heard!) this before. I have a site which
has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice
and I was quite angry when I heard they'd been installed )-: They have an
asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to
Olivier wrote:
Hello,
My goal is to provision C450IP or S450IP models.
Has anyone a hint to provision them from configuration files ?
Usually, we use dedicated menu embedded inside Gigaset handset.
An http server also exists but I couldn't find any dhcp-tftp combination
to configure
There's a script here that does various conversions... locate the line
converting to gsm: http://astrecipes.net/index.php?n=153
Bye
l.
On Sun, 12 Aug 2007 22:02:06 +0200, Andrew Joakimsen [EMAIL PROTECTED]
wrote:
On 8/12/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
have anyone an idea
Hi,
I got one link where I got libraries required to configure R2MF.
Their are 4 libraries required
spandsp, libsupertone, libmfcr2 and libunicall . On searching for the
tar
balls One link is given and that is not opening up.( i.e.
http://www.soft-switch.org/downloads.html )
Hi,
I have already configured the digium card and it is successfully
communicating with another E1 card running application. I did it for ISDN
lines.
Can anybody tell me what changes I have to make to configure asterisk
for R2MF protocol support.
Thanks and regards
sanchal
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Steve Totaro wrote:
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
OCOSA ListAcct wrote:
I apologize if this question has already been answered / asked. I was
searching
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
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I've also built routers small NAS boxes out of the same
motherboard...
Personally I've not used C3 systems at all so I can't comment on them,
I've only been using C7 systems.
I compile up a custom kernel for them don't use the on-board audio
hardware (nor printer port, but I use the serial
Darrick Hartman wrote:
Just because someone is using an old kernel or doesn't know what they
are doing doesn't mean the hardware is bad. I've had very good
success
with dozens of different VIA boards (from the original mini-itx board
up
to current C7 models, the Jetway boards included).
Not
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
___
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Hi all,
In version 1.2, there is a realtime function and it is very easy to
use with prefix.
exten = s,1,RealTime(table|name|peter|user_)
and we can easy get back the value as user_name, user_id, etc.
However, I found the the function will be depreciated in 1.4. There
is a replacement using
On Monday 13 August 2007 05:11, Darrick Hartman wrote:
VIA C3 C7 systems have been known to have DMA-related issues for some
time.
I've seen those issues, but have never experienced them myself.
I have heard those things. I have a crash I can reproduce on a VIA Esther, and
it happens even
I am originating calls through the Manager Originate API command.
I can track failures (through the OriginateResponse event)
I can track answered calls through the OriginateResponse event)
There may be occasions where I need to cancel some outbound calls whilst
they are ringing.
Here's my
I see that the TrixBox Pro website is available now:
http://www.trixbox.com/products/trixbox-pro/
From what I'm reading, there is a free version available, plus two other
versions, one at $9.99/per user/per month and the other at $19.99/per
user/per month.
They have a centralized
MOSBAH ABDELKADER wrote:
After installing Asterisk, i have installed the docs by make progdocs.
But i don't know where to locate this documentation.
Maybe /usr/src/asterisk-*/doc/api/ ?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Friday, August 10, 2007, 3:12:39 AM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
then just push 'send' on your SIP phone and the system will dial it out
for you... ??
this is working for me in .nl, and
Dermot Bradley wrote:
Darrick Hartman wrote:
Just because someone is using an old kernel or doesn't know what they
are doing doesn't mean the hardware is bad. I've had very good
success
with dozens of different VIA boards (from the original mini-itx board
up
to current C7 models, the
Hi,
(cc. asterisk-users, hope that is not a big Faux Pas)
I've had trouble with the qozap driver for a LONG time now, where it
will not recognise and ignore a missing ISDN2 line on a quad card if
one of the 4 ports is unplugged or somehow faulty.
The symptom is that is correctly recognises the
On 13:50, Mon 13 Aug 07, Steve Davies wrote:
Hi,
--- qozap/qozap.c~ 2007-08-13 13:25:38.0 +0100
+++ qozap/qozap.c 2007-08-13 13:10:18.0 +0100
@@ -827,7 +827,7 @@
if (qoztmp-spans[s].alarms != ZT_ALARM_RED) {
qoz_dfifo_tx(qoztmp,
Hello,
I have a small LAN network connected through an Asterisk Server (Trixbox). I
was looking to create my own custom made softphones, and I have been looking
into how to transmit and receive via RTP. Would anyone know how the Asterisk
RTP bridging works, and if there is any documentation
Hi All,
I am trying to install Asterisk with FreePBX
while running install_amp following error is coming
can any one help in this regards
Thanks in advance..
Linga Reddy
Connecting to database..OK
Connecting to Asterisk manager interface..OK
DB Error: no such tableGenerating AMP configs..OK
FYI to anybody who cares, here is what I did:
1. Create web page where you enter a file name and a number to call
2. Insert the file name into the *DB via Asterisk Manager
3. Through Asterisk Manager create a call file from a recording
extension to the phone number entered in #1
4. The
Does anyone have any tools to process CDR-CSV files into reports? I don't have
anything specific in mind, I'd just like some reporting examples so I don't
have to reinvent the wheel.
This e-mail, facsimile, or letter and any files or attachments transmitted
You could perhaps try to have a look on func_devstate
http://www.asterisk.org/node/48360
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Todd H
Envoyé : samedi 11 août 2007 18:52
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Hi Linga,
You will likely get a much better response by posting to the FreePBX list
(here: http://sourceforge.net/mail/?group_id=121515 ) or the FreePBX forums
(here: http://www.freepbx.org/forums/ ). FreePBX is an entirely different
animal on top of Asterisk and this group mainly focuses on
On Mon, Aug 13, 2007 at 02:57:09AM -0400, Alex Balashov wrote:
On Sat, 11 Aug 2007, Jay R. Ashworth wrote:
On Fri, Aug 10, 2007 at 05:20:38PM -0400, Alex Balashov wrote:
It might be possible to glue something together with it and OpenSER and
a media gateway control protocol like H.248
On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote:
Could everyone that has a working production Asterisk server that uses a
Digium telephony card as a BRI/PRI gateway let me know what
motherboard/processor your server uses?
Currently running a TE412P in a IBM x3650 Model 7979. I had some
Looks like it's time to fork..
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lenz
Sent: Monday, August 13, 2007 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: The trixbox
On Fri, 10 Aug 2007, Carlos Chavez wrote:
I usually have good results when using a regular fax machine connected
to a PAP2T on a small network. I have a customer that has this setup in
several offices. Lately I have noticed that recent versions of Asterisk
have worse results with this
Trevor Peirce wrote:
OCOSA ListAcct wrote:
I apologize if this question has already been answered / asked. I was
searching on Google and nothing I do will work. All that happens is that
the phones ring for 00:01:15 then voicemail kicks in.
I wonder if this is your phone deciding it
test list not working
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I am using the page command per the example in the Wiki and am having
trouble getting it to work the way I want. The call is coming from a
SipXchange system and all the phones are attached to the SipXchange.
Please let me know what config file you need. I also have the sip debug
trace available.
On Mon, 2007-08-13 at 15:25 +0530, [EMAIL PROTECTED]
wrote:
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
What you describe is doable; we have a number of device configuration
wizards.
But it is generally easier to use the device's bulk provisioning
methods, like https an XML configuration file to the device. The
provisioning settings a pretty standard and don't change very often.
The problem
We at Evariste have a lot of experience writing all sorts of custom CDR
reports and would be happy to write what you need for you--very
inexpensively, guaranteed.
On Mon, 13 Aug 2007, Jeremy Mann wrote:
Does anyone have any tools to process CDR-CSV files into reports? I
don't have anything
James FitzGibbon wrote:
On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote:
Could everyone that has a working production Asterisk server that uses a
Digium telephony card as a BRI/PRI gateway let me know what
motherboard/processor your server uses?
Currently running a TE412P in a IBM
[EMAIL PROTECTED] wrote:
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Zaptel drivers are designed to be protocol independent. The TE120P
Free and inexpensive aren't quite the same. I don't follow the -biz
list because I don't want to hear plugs from You at Evariste about
this stuff.
I sent a PHP snippet to the list maybe a year and a half ago, search
something like
site:lists.digium.com Mojo csv
or
site:lists.digium.com
I remember now, it totaled the *times* of cdr entries after grouping
them by account code.
Mojo
Mojo with Horan Company, LLC wrote:
Free and inexpensive aren't quite the same. I don't follow the -biz
list because I don't want to hear plugs from You at Evariste about
this stuff.
I sent
First, this is a non-commercial list, please do not post that stuff here.
Alex, cdr csv is less then efficient for reporting, you should drive
your cdr's to a database and then you can do some good reports based on
that.
Anthony
Alex Balashov wrote:
We at Evariste have a lot of experience
On Mon, 13 Aug 2007, Anthony Francis wrote:
First, this is a non-commercial list, please do not post that stuff here.
My apologies.
Alex, cdr csv is less then efficient for reporting, you should drive
your cdr's to a database and then you can do some good reports based on
that.
Every little bit helps, thanks!
I guess I'm actually going to look at cost/benefit analysis, trying to see
where calls are going across IAX and tallying up what would have been LD
cost(we're doing intra-office IAX calling where possible) to tag as savings
to justify the systems to our
I was wondering if anyone knew how to use hints for extension monitoring
over DUNDi
Thanks,
John Coleman
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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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Anthony Francis wrote:
First, this is a non-commercial list, please do not post that stuff here.
Alex, cdr csv is less then efficient for reporting, you should drive
your cdr's to a database and then you can do some good reports based on
that.
On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote:
Does anyone have any tools to process CDR-CSV files into reports?
Throw them into a near-by spreadheet.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406
Enable mysql loggin of cdr's by installing asterisk-addons and use
asterisk-stat
http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
On 13/08/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote:
Does anyone have
On Mon, 13 Aug 2007, [EMAIL PROTECTED] wrote:
I have a small LAN network connected through an Asterisk Server
(Trixbox). I was looking to create my own custom made softphones, and I
have been looking into how to transmit and receive via RTP. Would
anyone know how the Asterisk RTP
I am working on a call-back solution where the initiating call should
never be answered.
I was doing this simply through the dial plan, sending a progress
tone, and then dumping the channel, and firing off a DeadAGI which
created a call file to make the callback.
Now I've tried extending
Hello Daryl,
See
http://www.asterisk.org/doxygen/1.4/res__agi_8c.html#c631d48f46d51d4b057
b31807baa1f10
The AGI application will answer the channel if it isn't already
answered.
You probably need to do whatever you want to do in the dialplan, and
keep using DeadAGI.
Martin Smith, Systems
On 8/12/07, C F [EMAIL PROTECTED] wrote:
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten =
Asterisk Users,
I have never done a dial plan for this scenario before. Is it possible to
have Sip Phones make outbound calls through the PSTN? What would the call
routing/dial plan would look like?
-John
_
Messenger Café
On 8/13/07, John Meksavan [EMAIL PROTECTED] wrote:
Asterisk Users,
I have never done a dial plan for this scenario before. Is it possible to
have Sip Phones make outbound calls through the PSTN? What would the call
routing/dial plan would look like?
Yes - certainly possible. There's
Not sure what all the licensing in TrixBox is but if they dump the
open, can't we always just fork. I have not played with TrixBox in
some time but most of it was just a bunch of separate, valuable
projects
meshed together. I don't really see how they can close that.
Yes! That's one of the
Doug Lytle wrote:
Kate Kretz wrote:
OpenVPN is very good in NAT (if one of your boxes is behind NAT).
otherwise, OpenVPN seems to be a bad choice, it's complicated,
non-standard (there'n no RFC on OpenVPN).
It's complicated? No more so then Asterisk.
We use, it was quite straight
I'm looking for pointers towards building and running the zaptel drivers
under Solaris 10.
Can anyone help?
Frank
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Hello,
Do you have install doxygen?
Best regards
On 8/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
MOSBAH ABDELKADER wrote:
After installing Asterisk, i have installed the docs by make progdocs.
But i don't know where to locate this documentation.
Maybe
Thanks.
In app_realtime, it is very easy to get a value of a field by only
applying the realtime application. However, in func_realtime, we need
to get the key-value pair according to the position of it by using
function CUT. After that, we need to apply another CUT to get the
value. It will
Hi everybody,
As the Asterisk community is getting larger and larger, I was wondering that
the features which are provided in Asterisk and are programmed by the open
source community under GPL, or GUIs like FreePBX which also come loaded with
wonderful features and uses same Asterisk, are they
I'm trying to define distinctive rings for lines in this gateway but don't
works.
Nothing happen when sending a call...the phone doesn't ring
The same configuration works fine for PAP2NA devices.
Adriano Almeida
Flickr agora em português. Você clica, todo mundo vê. Saiba
http://areski.net/asterisk-stat-v2/about.php
???
PaulH
On Mon, 2007-08-13 at 08:49 -0500, Jeremy Mann wrote:
Does anyone have any tools to process CDR-CSV files into reports? I
don’t have anything specific in mind, I’d just like some reporting
examples so I don’t have to reinvent the
What if it is an international call? Then your callerID won't work.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: lunes, 13 de agosto de 2007 3:21
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users]
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