Actually, if OP upgrades to Asterisk 10 they will get video conferencing
with app_confbridge.
I think I'm not as much updated then and definitely am going to test this
application. Paul have you ever seen this application in action ! this is
going to be great then - built-in Video conference
Hi,
On a 1.8.10 system, I've got (with cdr show status) :
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
* Registered Backends
---
csv
Greetings!
Recently I had to change the port Asterisk listens to
(non-standard, to hide from bruteforce attacks), but at the same time I
wanted to not break the system for all current users. So I needed some
way to listen to two ports for some time.
I did some research in the
Internet and
You could possibly process the hindi fonts as Unicode. As for the Google
Speech and TTS, it “should” all be covered under open source agreements.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram
Sent: Saturday, April 07, 2012
As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.
After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and
We saw some activity related to this FreePBX unpatched vulnerability
this past weekend on some hosted PBXes.
http://seclists.org/fulldisclosure/2012/Mar/234
Usually we see the typical SIP Vicious attacks, but this one is much
more involved and dangerous.
--
This is what core show applications in 10.1.3 shows
SendDTMF: Sends arbitrary DTMF digits
SendFAX: Sends a specified TIFF/F file as a FAX.
SendImage: Sends an image file.
SendText: Send a Text Message.
SendURL: Send a URL.
You are
I want to use Call Deflection with DAHDISendCallreroutingFacility
Application.
I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
my dialplan is like this:
You should always specify the switchtype and signaling parameters for
ISDN issues as well. In this case it is not
What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
the agent's phone starts
Put your Queue command In a macro like this
[agi-and-queue]
Exten = s,1,Verbose(start AGI then do queue)
Exten = s,n,AGI(queproc.sh)
Exten = s,n,queue(myqueue)
You will need to put nohup into the AGI so it can run whether the line gets
picked up or not.
From:
On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
What I am trying to accomplish is to run an AGI script each time an
agent's line starts ringing. I currently have the AGI firing when the
agent answers the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works
On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote:
This is what core show applications in 10.1.3 shows
SendDTMF: Sends arbitrary DTMF digits
SendFAX: Sends a specified TIFF/F file as a FAX.
SendImage: Sends an image file.
SendText:
Not that I'm aware of; however you can call and have the phone auto-answer
just to take the message - it's a SIP header tweak that has been discussed
here.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger
You have read this thread?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, April 10, 2012 3:15 PM
To: Asterisk Users Mailing List -
Thanks Danny.
What I am trying to do is send a popup screen to the agent. I am doing this
now using Flash Operator Panel but I am trying to get away from that. I
need to know the agent the call is being sent to when calling the AGI. So,
right now I am getting all that but only when the call is
I was trying to leave the AMI out of this because I am unsure if I can
monitor it in real time without building an external listener which Flash
Operator Panel does for me right now.
When I went down this road, I thought it would be a piece of cake to just
fire an AGI, well it is until you get
Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
this for a number of years off and on, I never post unless I have dug hard,
searching all the Asterisk resources I know of.
This is where I got most of my info but the solutions mentioned on that
page require the call to
Were this my task, I would do a PERL/C daemon to run the AGI. This is how I
do it in PERL
my $astman = new Asterisk::Manager;
$astman-user('user');
$astman-secret('secret');
my $man_addr='127.0.0.1';
this section is for if your asterisk isn't on 127.0.0.1
Thanks again Danny, Perl was the first thing I tinkered with back in the
90's but haven't messed with it for years.
Looking over what you sent, I get about 90% of what's going on there. With
a little searching and brushing up on my Perl, I think I will be able to
make this work.
This is a good
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: Monday, April 09, 2012 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, April 09, 2012 9:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, April 09, 2012 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT QUESTION
I want to use Call Deflection with DAHDISendCallreroutingFacility
Application.
I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
my dialplan is like this:
You should always specify the switchtype and signaling parameters for
ISDN issues as well. In this case it is not
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