are you using
Asterisk::AMIhttp://search.cpan.org/~greenbean/Asterisk-AMI/lib/Asterisk/AMI.pm
for
this script?
On Wed, Apr 6, 2011 at 10:04 AM, Danny Nicholas da...@debsinc.com wrote:
Yes – I do it that way because I run the module this is included in on
about 10 different Asterisk servers.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, April 06, 2011 10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi voicemail callback
are you
using the realtime functions for voicemail solve this problem.
you can insert a query from your agi to add new voicemail box.
is it what you need ?
On Tue, Apr 5, 2011 at 10:17 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 5 Apr 2011, vip killa wrote:
Is it possible to create a
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.
For instance, a caller leaves a voicemail, the voicemail will then call the
owner of the voicemailbox determined by a database look up.
--
_
Is it possible to create a voicemail box using AGI? How does asterisk know
about mailboxes when using Asterisk with pure AGI?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
vip killa wrote:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI
I don't have an AGI, but I do have dial-plan code.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
On Tue, 5 Apr 2011, vip killa wrote:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.For instance, a caller leaves a voicemail, the
voicemail will then call the owner of the voicemailbox determined by a
database look up.
Use 'mailcmd' in
On Tue, 5 Apr 2011, vip killa wrote:
Is it possible to create a voicemail box using AGI?
An AGI executes as a child process when a channel executes agi() via the
dialplan.
Are you intending to call into Asterisk and let the caller create
mailboxes?
All the AGI needs to do is add a line
On 02/18/2011 07:30 PM, Mike Diehl wrote:
Hi all,
I've got a perl agi script that exec()'s the FFA version of receivefax
to... receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi-exec(receivefax,/tmp/${$}.tiff|fs);
Hi all,
I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi-exec(receivefax,/tmp/${$}.tiff|fs);
$main::agi-verbose(FAX COMPLETE,1);
I never see the FAX COMPLETE
On Fri, 18 Feb 2011, Mike Diehl wrote:
I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi-exec(receivefax,/tmp/${$}.tiff|fs);
$main::agi-verbose(FAX
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Tuesday 01 February 2011 23:43:34 Charles Solar wrote:
Hey guys I was hoping I could get a few pointers on a problem I have
been trying to debug for the last couple of months regarding asterisk
AGI scripts and unexpected termination.
I have this agi script that accepts incoming faxes using
Thanks for the useful information, I had forgotten about SIGHUP since I
usually work with asterisk 1.6.
I think however that it would be more acurate to say that the channel is
hanging up due to the script crash. I tried moving the command around in
the script and it crashes exactly on the system
Hey guys I was hoping I could get a few pointers on a problem I have been
trying to debug for the last couple of months regarding asterisk AGI scripts
and unexpected termination.
I have this agi script that accepts incoming faxes using RxFax on the latest
asterisk 1.4 branch. Its written with perl
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because the
callee was busy ?
(I'm using
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Wednesday, January 19, 2011 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi dial termination cause
On Wed, 19 Jan 2011 17:03:03 +0100
Thorsten Göllner t...@ovm-group.com wrote:
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for
-users] AGI-Macro w/Agruments
OK, I need to dial a macro from AGI and needs to pass an argument.
Ok, I found an bug report, but it was stated un fixable? really after 5
years?
https://issues.asterisk.org/view.php?id=2470
I found this email in the archive, but no solution other
OK, I need to dial a macro from AGI and needs to pass an argument.
Ok, I found an bug report, but it was stated un fixable? really after 5
years?
https://issues.asterisk.org/view.php?id=2470
I found this email in the archive, but no solution other then the dodgy work
around?
hello,
First of all i am using Asterisk 1.6.2.9-2
The following problem seem like a bug to me but im not sure.
Any help or comment will be great..
We are trying to implement our own billing software with AGI - Php Scripts.
When a hangup received, i am calling a script to calculate the
On 16 September 2010 22:23, Barry Miller asterisk-us...@notanet.net wrote:
For an interim fix, setting res_agi=1.4 in the [compat] section of
asterisk.conf should work. See UPGRADE-1.6.txt .
I have tried this but it still complains about the pipe not being a comma.
Regards
Jon
--
Jon
On 09/21/2010 04:22 PM, Jon Farmer wrote:
On 16 September 2010 22:23, Barry Millerasterisk-us...@notanet.net wrote:
For an interim fix, setting res_agi=1.4 in the [compat] section of
asterisk.conf should work. See UPGRADE-1.6.txt .
I have tried this but it still complains about
Hi,
I fixed it in the end by adding the sip headers I was interested in as extra
x headers in the openser config. Then just capturing these in the asterisk
dialplan as variables. Simples.
Regards
Jon
On 21 Sep 2010 16:03, Jonas Kellens jonas.kell...@telenet.be wrote:
On 09/21/2010 04:22 PM,
Hi
I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 1:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI Delimiter in 1.6
Hi
I am currently using
On 16 September 2010 19:50, Danny Nicholas da...@debsinc.com wrote:
Two suggestions;
#1. escape the , as \,
#2. quote the string so 1,2,3 is 1,2,3
I have thought about both of those ideas.
Is it possible to escape the string in the dialplan?
Applying quotes didn't seem to work, however I
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter
On 16 September 2010 19:50, Danny Nicholas da...@debsinc.com wrote:
If you make the string into a dialplan Variable, you can do pretty much
anything with it. Let's say your dialplan is like this
- exten = 1234,1,blah
- exten = 1234,n,AGI(myagi.xx,1234)
Change line 2 to
- exten =
16, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter in 1.6
On 16 September 2010 19:50, Danny Nicholas da...@debsinc.com wrote:
If you make the string into a dialplan Variable, you can do pretty much
anything with it. Let's
On Thu, Sep 16, 2010 at 07:44:23PM +0100, Jon Farmer wrote:
Hi
I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan
] *On Behalf Of *Ashik Ali
*Sent:* Tuesday, September 14, 2010 2:27 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] agi playback to execute say.conf settings
Hi danny,
Shall we take it as agi bug ?
Thanks,
Ashik
snip
For lack
:* Thursday, September 09, 2010 2:06 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] agi playback to execute say.conf settings
hi,
any response ?
thanks,
Ashik
snip
exec playback in AGI expects to find a file or set of files in
/var/lib
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali
Sent: Tuesday, September 14, 2010 2:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi playback to execute say.conf settings
Hi
-boun...@lists.digium.com] *On Behalf Of *asteriskguru
asteriskguru
*Subject:* [asterisk-users] agi playback to execute say.conf settings
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number reading.
snip
but when I write it in agi does not working. Here
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali
Sent: Thursday, September 09, 2010 2:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi playback to execute say.conf
it in
dialplan.
Thanks,
Ashik
On Thu, Sep 2, 2010 at 7:08 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asteriskguru
asteriskguru
*Subject:* [asterisk-users] agi playback to execute
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number reading.
In the extension.conf:
--
[number-to-voice]
exten = 8765,1,playback(num:344345,say)
exten = 8765,n,hangup
It executes corresponding say.conf script and produces good results
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asteriskguru
asteriskguru
Subject: [asterisk-users] agi playback to execute say.conf settings
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number
I am trying this approach to see who picked the line:
Here is what i am doing:
EXEC DIAL SIP/ vaso Zap/35||M(testing^30086)
Macro:
[macro-testing]
exten = s,1,DumpChan()
exten = s,2,AGI(whopicked.rb)
exten = s,3,Hangup()
From console:
-- SIP/ vaso -e26c answered Zap/14-1
In theory this snippet will do the trick
Save as updatech.pl
#!/usr/local/bin/perl -w
$ENV{PATH} = '/usr/sbin:/:/usr/bin:/usr/local/apache/bin'; # reasonable path
$ENV{ENV} = /etc/bash.bachrc;
use strict;
use warnings;
use File::Find;
use DBI;
use Date::Calc qw(:all);
use Asterisk::AGI;
On Fri, 30 Jul 2010, Zarko Zivanovic wrote:
I need simple whopicked.agi (instead of .rb) which will simply take the
value 30086 (that I pass to macro)
While .rb suggests a Ruby source file, .agi suggests nothing.
This should be simple – no ruby - just agi.
You are confusing a language
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Subject: Re: [asterisk-users] agi macro problem
You are confusing a language with a protocol. An AGI is a program that
complies with the AGI protocol. It can be written
On Fri, 30 Jul 2010, Zarko Zivanovic wrote:
I need simple whopicked.agi (instead of .rb) which will simply take the
value 30086 (that I pass to macro)
On Fri, 30 Jul 2010, Steve Edwards wrote:
While .rb suggests a Ruby source file, .agi suggests nothing.
On Fri, 30 Jul 2010, Danny
Hello,
I'm currently developing a simple asterisk application using SFS (Skype For
SIP) which tries to call to an outbound number, play a message and read DMTF
digits. My first approach used the Manager to originate calls and then
called an
agi script to deal with the rest. Anyway, this ended up
Try with something like
action.setChannel(SIP/99051000xxx...@yourtrunkname);
On Sat, Jul 17, 2010 at 10:19 PM, Felipe Kurkowski
felipekurkow...@gmail.com wrote:
Hello,
I'm currently developing a simple asterisk application using SFS (Skype For
SIP) which tries to call to an outbound
It appears that there's no way to get the return value from a GOSUB into
an AGI script. Is that correct?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On Friday 16 July 2010 23:35:01 Richard Kenner wrote:
It appears that there's no way to get the return value from a GOSUB into
an AGI script. Is that correct?
No.
GET VARIABLE GOSUB_RETVAL
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig
Dear All,
Please anyone help me to solve the following problem.
Thanks,
Velusamy
On Thu, Jul 8, 2010 at 4:19 PM, velusamy Krishnan
velu.techni...@gmail.comwrote:
Dear All,
I have get full variable AGI call to get the ANSWEREDTIME channel
variable. I have originated the call to one
Dear All,
I have get full variable AGI call to get the ANSWEREDTIME channel
variable. I have originated the call to one extension, once answered I have
called DeadAGI to control the call.
I have problem that after hangup the call AGI GET FULL VARIABLE returns
-1 for ANSWEREDTIME channel
On Mon, 21 Jun 2010 16:10:12 +, Edwin Quijada
listas_quij...@hotmail.com wrote:
Uhmmm.. remember for each channel you run perl or php interpreter so with that
amount of memory maybe this can be a problem.
For that kind of project I'd use C or java as fastagi protocol
Thanks Edwin. In my
Hello
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware limitations?
Ideally, I'd rather use a rich
On Mon, 21 Jun 2010, Gilles wrote:
Hello
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware
On Mon, 21 Jun 2010, Gilles wrote:
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
You could always type
asterisk blackfin
into google and see what it suggests.
Here, I'll save you the effort:
Thanks but I already know this (uCasterisk is deprecated). And can't
stand Perl ;-)
--
8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [AGI] What scripting language for embedded
hardware?
On Mon, 21 Jun 2010, Gilles wrote:
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application. If it
takes too slow to initialize - try to find some embedded versions.
openwrt, for instance, has
On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus Jaktys
desired@gmail.com wrote:
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application. If it
takes too
Hey Gilles,
for whatever reason your messages appear twice twice on this list.
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Subject: Re: [asterisk-users] [AGI] What scripting language for embedded
hardware?
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application
On Sun, Jun 13, 2010 at 2:59 PM, Vieri rentor...@yahoo.com wrote:
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
compatible).
I've taken a look at CAGI and QUIVR but their latest code releases date back
to 2006.
I've also seen a more recent project (wildpbx)
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
compatible).
I've taken a look at CAGI and QUIVR but their latest code releases date back to
2006.
I've also seen a more recent project (wildpbx) dated 2009:
Clark
Sent: Monday, May 10, 2010 11:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI and Severe Weather Alerts
All,
I am toying with an idea of using an AGI to be able to 'call'
my phone, or phones, in case of severe weather warnings. I have been
tinkering
, 2010 11:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI and Severe Weather Alerts
All,
I am toying with an idea of using an AGI to be able to 'call'
my phone, or phones, in case of severe weather warnings. I have been
tinkering with a script that reads from weather
On Tue, 11 May 2010, Danny Nicholas wrote:
For your application, the best (IMO) strategy would be to have an AGI
that is cronned to run every X minutes and launch a call when needed
using AMI.
If the script or executable is started by cron, it is not an AGI.
--
Thanks in advance,
-Commercial Discussion
Subject: Re: [asterisk-users] AGI and Severe Weather Alerts
On Tue, 11 May 2010, Danny Nicholas wrote:
For your application, the best (IMO) strategy would be to have an AGI
that is cronned to run every X minutes and launch a call when needed
using AMI.
If the script
All,
I am toying with an idea of using an AGI to be able to 'call'
my phone, or phones, in case of severe weather warnings. I have been
tinkering with a script that reads from weather underground for the
forecast, based off a PHP version of a weather AGI I found on the net.
It seems
Hi Luki:
于 2010年05月01日 06:03, Luki 写道:
The good news is, we run tens of thousands of calls every day through
this box and about half of them spit out this warning, but it never
caused any problems for over a year. Thus this warning is probably
safe to ignore.
We run tens of thousands of
We run tens of thousands of call every day too. Call is controlled
by AGI , and the asterisk version is 1.2.24. I find memory leak in
asterisk. After serveral weeks, the memory used by asterisk will reach
1.2 GB or higher. Each time I have to restart to asterisk, and the
memory leak will
CHEN XUEQIN wrote:
Hi Luki:
于 2010年05月01日 06:03, Luki 写道:
The good news is, we run tens of thousands of calls every day through
this box and about half of them spit out this warning, but it never
caused any problems for over a year. Thus this warning is probably
safe to ignore.
On Sat, 1 May 2010, SIP wrote:
[snip]
We run DeadAGI for a considerable number of calls since it has the
ability to run post-hangup cleanup no matter which side hangs up (unlike
AGI).
[snip]
When a channel hangs up, Asterisk sends a SIGHUP signal to the AGI. If the
AGI did not establish
On 5/2/2010 4:52 PM, Steve Edwards wrote:
On Sat, 1 May 2010, SIP wrote:
[snip]
We run DeadAGI for a considerable number of calls since it has the
ability to run post-hangup cleanup no matter which side hangs up (unlike
AGI).
[snip]
When a channel hangs up, Asterisk sends a
On 4/30/2010 6:03 PM, Luki wrote:
It is irrelevant who hangs up, you want to just use DeadAGI in the h
extension
I wish that would be the case, but at least on 1.4 I see:
[Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new
stack
[Apr 30 14:59:38]
It is irrelevant who hangs up, you want to just use DeadAGI in the h
extension
I wish that would be the case, but at least on 1.4 I see:
[Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new
stack
[Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running
Hello, i have this problem :
i phone person B .
*if i hang up*, i have this h extension : exten = h,1,AGI(ende.agi)
*if the person B hangs up* , i have this h extension : exten = h,1,
DeadAGI(ende.agi)
The problem is, i do not know where hangs up the first . How kann i combine
AGI and DeadAGI in
Hello, i have this problem :
i phone person B .
*if i hang up,* i have this h extension : exten = h,1,AGI(ende.agi)
*if the person B hangs up* , i have this h extension : exten = h,1,
DeadAGI(ende.agi)
The problem is, i do not know where hangs up the first . How kann i combine
AGI and DeadAGI in
Redouane Zerargui wrote:
Hello, i have this problem :
i phone person B .
_/*if i hang up*/_, i have this h extension : exten = h,1,AGI(ende.agi)
_/*if the person B hangs up*/_ , i have this h extension : exten =
h,1,DeadAGI(ende.agi)
The problem is, i do not know where hangs up the
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
I figured as much. ESOL=English as a Second Language. Apology accepted.
Have you tried creating the file on the windows server, running sox to your
*---*
Date: Sat, 17 Apr 2010 18:19:53 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada listas_quij
*---*
--
Date: Sat, 17 Apr 2010 18:19:53 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Sat, Apr 17, 2010 at 1:48 PM
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada
listas_quij...@hotmail.comwrote:
Why don’t you use sox to transform the windows audio file into the
asterisk format – I do this with pretty good results.
I did. But my problem is not conversion my problem is that I dont know how
play the
On Fri, Apr 16, 2010 at 4:59 PM, Edwin Quijada
listas_quij...@hotmail.comwrote:
Hello!
I have developed an IVR using AGI and so far it works great. I'm using
Cepstral voices, but now want to use the voices from AT T that are on a
Windows server to be heard best. With cepstral what I do is
*---*
Date: Sat, 17 Apr 2010 13:23:22 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada listas_quij...@hotmail.com
wrote:
Why don’t you
Just a shot in the dark, have you tried ExternalIVR? It was originally
developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed
up on this one.
This option NO.
Another option would be FastAGI to your windows server. You write an app for
the windows box that
On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada
listas_quij...@hotmail.comwrote:
Just a shot in the dark, have you tried ExternalIVR? It was originally
developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer
teamed up on this one.
This option NO.
Another option would be
On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada
listas_quij...@hotmail.comwrote:
Just a shot in the dark, have you tried ExternalIVR? It was originally
developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer
teamed up on this one.
This option NO.
Another option would be
:22 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada listas_quij...@hotmail.com
wrote:
Why don’t you use sox to transform the windows audio
Hello!
I have developed an IVR using AGI and so far it works great. I'm using Cepstral
voices, but now want to use the voices from AT T that are on a Windows server
to be heard best. With cepstral what I do is to generate audio files from
shipping and this text I reproduce this method it has
To: Asterisk Asterisk
Subject: [asterisk-users] AGI, FASTAGI or Windows Voice Server
Hello!
I have developed an IVR using AGI and so far it works great. I'm using
Cepstral voices, but now want to use the voices from AT T that are on a
Windows server to be heard best. With cepstral what I do
Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with pretty good results.
I did. But my problem is not conversion my problem is that I dont know how play
the file from windows server or copy this to asterisk without my AGI continue
and
Why don't you copy the files to your asterisk box and play them from there?
-- Sent from my Android device
On Apr 16, 2010 5:03 PM, Edwin Quijada listas_quij...@hotmail.com wrote:
Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with ...
I did.
Hi all,
I am running an AGI script in a command dial, or call a SIP trunk.
I want to execute after 10 minutes a voice message (stream file) on the
channel to warn the person that the call is about to end. How to do that?
Thank you,
Mickael.
--
Use L() option in Dial application while originating the call.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Wed, Apr 7, 2010 at 7:00 PM, Mickael MONSIEUR mickael.monsi...@gmail.com
wrote:
Hi all,
I am running an AGI script in a command dial, or call a SIP trunk.
I want to
There is a parameter L which you can use in the dial command. More about
it you can see on voip-info.org, but it'll be something like this:
Dial(SIP/223,60,L(11000:1))
The first 11000 means 11 minutes allowed duration of the call and after 10
minutes it'll play message You have one minute.
Thank you Godson Zeeshan ! :-)
Mickael.
Zeeshan Zakaria a écrit :
There is a parameter L which you can use in the dial command. More
about it you can see on voip-info.org http://voip-info.org, but
it'll be something like this:
Dial(SIP/223,60,L(11000:1))
The first 11000 means 11
Wow, can't believe I missed that.
Thanks so much!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command? I scanned the CLI help but
found nothing similar. Both my 1.6 boxes do not have the
On Sat, Feb 13, 2010 at 10:50 PM, Alejandro Recarey
alexreca...@gmail.com wrote:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command?
201 - 300 of 1104 matches
Mail list logo
