On 16 May 2008, at 00:26, Julian Lyndon-Smith wrote:
I have a lot of recordings from asterisk in a .gsm format. I would
like
to play these files from a web browser (IE, firefox and opera)
What do I need to do in order to achieve this goal ?
Sorry to catch up late on this, but I have a
Update on this one.
I finally went back to AMI only for implementing this particular feature, but
ofcourse I had to make an addition of a couple of lines for my particular
requirement.
On Dial, the 'dial' event is sent over AMI which I capture. Unfortunately the
event didn't have any field
excellent contribution to the asterisk community andy congratulations Nicolas
rickygm ...
2008/5/16 Nicolás Gudiño [EMAIL PROTECTED]:
Hello,
I have finally released the queue stats package to the public.. please go to:
http://www.asternic.org/stats
To get it or see the online demo.
--
Howdy all,
The Asterisk-Java project has included some rudimentary parsing related to
dialplans and extensions.conf. I've done a blog post at
http://asterisk-java.org/ related to it, and giving a demo of some dialplan
visualizations. It could eventually get fleshed out into an open-source
visual
Hello All
Is it possible to implement and deploy Video Conferencing using Asterisk ?
Has anyone done it before ?
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
Regards
Bilal
Bilal,
I linked a store and product for you in the thread
already. A simple
google search will turn up hundred if not thousands of
suppliers.
Just
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
Regards
Bilal
Bilal,
I linked a store and product for you in the thread
already. A simple
google search will turn up hundred if not thousands of
suppliers.
Just
On Sat, 17 May 2008, bilal ghayyad wrote:
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
The new machines that I buy come with legacy power connectors. The flash
IDE drives I buy need legacy power connectors, and since
I'm implementing a simple calling card feature for testing purpose. I have a
DID number, when I called my DID number and enter the phone number to call,
Asterisk would dial the number for me but the sound was only one way.
After hours of struggling with the problem, I found out that I need to add
So no way to discover the status of FXO if a cable
pluged or not?
Regards
Bilal
-
2008/5/2 Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, May 02, 2008 at 09:06:01AM +0200, Vinz486
wrote:
2008/4/30 Tzafrir Cohen
[EMAIL PROTECTED]:
On Wed, Apr 30, 2008 at 09:07:48PM +0200,
On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:
On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good
On Sat, May 17, 2008 at 8:51 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:
On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
It seems any constructive
Is there any reason why I should be experiencing such bad line
quality on inbound calls from PSTN? Call quality is perfect when
plugging in a regular analogue phone.
Do you have other phone lines you can try the A200 with? Have you
asked Sangoma support?
Ditto on Sangoma support - they
On Sat, May 17, 2008 at 05:00:43AM -0700, bilal ghayyad wrote:
So no way to discover the status of FXO if a cable
pluged or not?
What specific card do you use?
What version of Zaptel?
Did you actually read my message you were responding to?
--
Tzafrir Cohen
icq#16849755
On Sat, May 03, 2008 at 12:19:03PM -0400, Dean Collins wrote:
I think it would be great for someone to write a small 'anonymous
collection module' that an Asterisk sys-admin could download and install
on their asterisk server which uploaded the stats to a community website
where the data was
On Fri, May 16, 2008 at 06:32:30PM -0500, James Sneeringer wrote:
On Fri, May 16, 2008 at 3:04 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
First of all, thanks Philipp, Alan, Tzafrir and James for your valuable
comments. I have listed below the exact list of commands to run for
On Sat, May 17, 2008 at 6:08 AM, Kashif Naeem [EMAIL PROTECTED] wrote:
Hello All
Is it possible to implement and deploy Video Conferencing using Asterisk ?
Has anyone done it before ?
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92
Alan Lord wrote:
Sherwood McGowan wrote:
snip /
Hrm...I have encountered this before and sometimes doing an explicit
Answer() then a Wait(2), then calling the service can help.
Hope this is helpful
Sherwood McGowan
Bingo!
Thanks a bunch. That sorted it.
Al
Hi,
Can anyone confirm if calls placed via sipbroker have their NUM CLI
changed by sipbroker??
I'm testing between two asterisk servers in seperate locations. When I
place a call directly, the CLI is fine. When the call is placed via
sipbroker lookup, the NAME stays the same, but the NUM is
All,
Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
When I put calls via sipbroker, half the time the calls fail. An enum
lookup shows 3 URIs listed, none of them seem to be google directly, and
I think 1 of them fails 100%, and the remaining one fails at other
random
Al, Randy, (and others):
What Al calls one very weak area for Asterisk is IMHO a difference
in market perceptions.
Asterisk is positioned for CPE - PBX - Appliance market which needs
feature-rich appeal
and mass-market focus.
Using asterisk for large scale does not mean that I have used it as
On 14:42, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
Hi
what about asterisk virtualization on VMWARE XS infrastructure?
The system installed will manage a call center with 50 operator,
queues, CDR logging on external database.
Hi Cohen;
I am using TDM22 (2 fxo and 2 fxs) digium card.
I am using zaptel 1.4.10.1
I readed, but not sure if readed all, as alot of
messages were going and coming.
Can u help?
Regadrs
Bilal
--
On Sat, May 17, 2008 at 05:00:43AM -0700, bilal
ghayyad wrote:
So no way to
Today I have been messing around with updating my residential
phonesystem (it was running a 1.0 version from years ago). I have
downloaded the last source packages for zaptel-1.4.10.1and
asterisk-1.4.19.2. Zaptel doesn't want to build. After a long time of
making this is the output that
At 11:44 AM 5/16/2008, you wrote:
Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
that a particular machine could handle, but from a support perspective, it
doesn't matter how many the machine could theoretically handle, it matters
how many it could handle in the particular
Hi all,
Would anyone be able to point me in the right direction as far as the
pros/cons of using a local loopback with a T1 provider, or just
peering with a company using SIP/IAX2 or my small office asterisk
setup? I've seen setups in both scenarios. The only potential pro of
the T1 that I can
On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote:
All,
Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
Yeah, I suppose a direct SIP connection would be nice.
An enum lookup shows 3 URIs listed, none of them seem to be google
directly,
No, they are SIP-PSTN
On Sat, May 17, 2008 at 3:11 PM, Mike Trest - On Travel [EMAIL PROTECTED]
wrote:
At 11:44 AM 5/16/2008, you wrote:
Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
that a particular machine could handle, but from a support perspective, it
doesn't matter how many the machine
Hi guys,
My asterisk server is connected to a pstn gateway using SIP. When I
receive a call and use the Hangup command the pstn seems to not
correctly see the request and the caller gets a 'number unknown message.
Below are the debug message printed on the CLI :
-- Executing [EMAIL
On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 14:42, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
Hi
what about asterisk virtualization on VMWARE XS infrastructure?
The system installed will manage a
On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.
I dunno, Steve; I wouldn't call Digium needs
On May 17, 2008 06:59:43 am Gordon Henderson wrote:
On Sat, 17 May 2008, bilal ghayyad wrote:
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
The new machines that I buy come with legacy power connectors. The flash
IDE
On 16:18, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 14:42, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
Hi
what about asterisk virtualization on VMWARE XS
On 16:20, Sat 17 May 08, Steve Totaro wrote:
On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good
On Sat, May 17, 2008 at 4:54 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 16:18, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 14:42, Sat 17 May 08, Steve Totaro wrote:
On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL
On Fri, May 16, 2008 at 08:18:46PM -0500, Tilghman Lesher wrote:
Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
wrong with that... unless you're an *employee*, and you're going out of
your way to hide it.
I'm been a member of this community far longer than I've
On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
It is about the money, like it or not. You are going to an Avaya type
licensing scheme, everything is charged per port. The box is capable
of doing more but you turn it off until you get more money. It's like
the Definity G3s I
On Sat, May 17, 2008 at 09:06:15AM -0400, Steve Totaro wrote:
Anyways, isn't Asterisk 1.2.x and FC6 EOL?
1.2 better not be EOL. :-)
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think
On Sat, May 17, 2008 at 10:56:21PM +0200, Michiel van Baak wrote:
He is an employee and he does not post from a Digium account or
include that fact in his signature. Not that it is to hide the fact,
but it certainly is obfuscated.
I think it just shows that his opinions are his, and in
On Sat, May 17, 2008 at 09:01:26PM +0200, Erik de Wild: Tripple-o wrote:
Today I have been messing around with updating my residential
phonesystem (it was running a 1.0 version from years ago). I have
downloaded the last source packages for zaptel-1.4.10.1and
asterisk-1.4.19.2. Zaptel
On Sat, May 17, 2008 at 04:45:00PM -0400, Matt Watson wrote:
well as 5V and 12V, molex only gives 5V and 12V. The actual physical
connector that molex uses also does not lend itself very well to
hot-plugging.
IME, it doesn't lend itself very well to *cold*-plugging, either.
:-)
Cheers,
Hello,
Someone told me about using a Loopback plug for RJ-45 for testing if a
Digium Card gave him 'green' Alarm (for testing if the card had been
damaged by a strange voltage surge); would this have some bad side effect?
Thanks in advice,
--
Jose P. Espinal
On Sat, May 17, 2008 at 05:50:46PM -0400, Jose P. Espinal wrote:
Someone told me about using a Loopback plug for RJ-45 for testing if a
Digium Card gave him 'green' Alarm (for testing if the card had been
damaged by a strange voltage surge); would this have some bad side effect?
Well, it's
It will go Green if a PROPER loopback plug is inserted.
Pins 1 and 2 shorted to 4 ad 5
Pin 1 to 4
Pin 2 to 5
Leave the others open...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Saturday, May 17, 2008 6:02
On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 09:06:15AM -0400, Steve Totaro wrote:
Anyways, isn't Asterisk 1.2.x and FC6 EOL?
1.2 better not be EOL. :-)
Cheers,
-- jra
--
Jay R. Ashworth Baylink
On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
It is about the money, like it or not. You are going to an Avaya type
licensing scheme, everything is charged per port. The box is capable
of doing more but
On Sat, May 17, 2008 at 11:56:47AM -0700, bilal ghayyad wrote:
Hi Cohen;
I am using TDM22 (2 fxo and 2 fxs) digium card.
I am using zaptel 1.4.10.1
I readed, but not sure if readed all, as alot of
messages were going and coming.
Can u help?
You should see (RED) in /proc/zaptel/1 for
On Saturday 17 May 2008 17:43:51 Steve Totaro wrote:
On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
It is about the money, like it or not. You are going to an Avaya type
licensing scheme, everything is
On Sat, 17 May 2008 18:32:57 -0500, Tilghman Lesher wrote:
Maybe next they will charge $250 for conference bridge capabilities.
It's a joke to cripple things that can be enabled by flicking a
switch. Your system comes with eight ports of VM but for another $250
we can give you 12..
Hi List;
In the below link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I saw this line and did not find for it explaination,
anyone can explain it?
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]/nLocal/interal
[EMAIL PROTECTED]|)
Thank you very much for your replies!,
And thanks Alexander for the T1 loopback pins schema :)
Alexander Lopez wrote:
It will go Green if a PROPER loopback plug is inserted.
Pins 1 and 2 shorted to 4 ad 5
Pin 1 to 4
Pin 2 to 5
Leave the others open...
-Original Message-
Hi Nicolas,
Thank you so very much for this!
(Also on behalf of a large group of Asterisk queue users, I'm sure!)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño
Sent: May 16, 2008 7:50 PM
To: asterisk-users@lists.digium.com
Subject:
Martin,
That's a wicked visualization tool!
Thanks for this contribution!
Thanks,
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin B.
Smith
Sent: May 17, 2008 2:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Lee,
You should probably clean it up and put it up on the wiki. I don't think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it will
be available to everybody.
Thanks for this, btw.
Mark.
-Original Message-
On Sat, May 17, 2008 at 06:34:12PM -0400, Steve Totaro wrote:
End of life date for Asterisk 1.2 was August 1, 2007.
Well, my app won't *run* on 1.4 reliably yet, so I hope they get it
fixed soon...
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL
On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
Maybe next they will charge $250 for conference bridge capabilities.
It's a joke to cripple things that can be enabled by flicking a
switch. Your system comes with eight ports of VM but for another $250
we can give you 12..
On Sat, May 17, 2008 at 07:07:51PM -0500, Michael Graves wrote:
I work in the broadcast and TV production business. Some time ago a
major company called Quantel created a hardware system called Edit
Box. It was wickedly fast and could do things with multiple streams of
uncompressed video in
On Sun, May 18, 2008 at 01:52:03AM +0300, Tzafrir Cohen wrote:
You should see (RED) in /proc/zaptel/1 for the channel if it is
disconnected. Not to mention that the channel will be in alarm (InAlarm
in zap show channel NNN).
Is that true for *all* makes of card? I know the Sangomas put it in
On Sun, May 18, 2008 at 12:35:39AM -0400, Jay R. Ashworth wrote:
On Sun, May 18, 2008 at 01:52:03AM +0300, Tzafrir Cohen wrote:
You should see (RED) in /proc/zaptel/1 for the channel if it is
disconnected. Not to mention that the channel will be in alarm (InAlarm
in zap show channel NNN).
bilal ghayyad wrote:
Hi List;
In the below link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I saw this line and did not find for it explaination,
anyone can explain it?
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
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