Hi,
Here is my scenario.
I have a SIP call between two SIP endpoints. A calls B.
During the ringing, B disconnects (network cable is unplugged).
But A continue ringing forever (until the dial timeout) even if asterisk
detects that B is disconnected with the qualify.
Is there any setup or
Are you using asterisk from source code ?
You can run make menuselect after the ./confugure and select the
modules you want.
The interface will also help you with dependencies between modules.
So, if you select chan_sip, it will select everything needed by it.
Le 2013-11-11 02:10, s m a écrit
Hi all,
I need to build an application that will be an SIP server program that will
run on Linux and Windows.
The sip server need only some features such as be able to :
- Register sip endpoints
- Answer a call and play a local file
- Make a dial from one channel
Hi,
I setup an MeetMe conference.
So, the admin user calls and enter the conference in talk/listen mode.
(Options : dAaxs)
Then other users call the same conference and enters in muted mode
(options: dlmx)
How can the admin user decide, when he is ready to let everybody speaks ?
I didn't
I think I found it reading the code.
It is *83 to unmute everybody.
Thanks
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2013-12-04 04:07
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
a écrit :
FROM: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Ruddy
Gbaguidi
SENT: 04 December 2013 09:08
TO: 'Asterisk Users Mailing List - Non-Commercial Discussion'
SUBJECT: [asterisk-users] Asterisk SIP server on windows
Hi all,
I need
I never tought this is become a Linux vs Windows fight.
We have been using asterisk on linux from a long time now and happy with
it.
But some of our customers who has windows in their environment want to
use our call center software we developed on top of asterisk.
So, the question was :
Did
You can have your dialplan log and write a data in a specific table as soon
as you get the call.
Then send that call ID to the agent web interface. And when the agent
complete the order, you just update the table with needed information.
Ruddy Gbaguidi
Micnes
There was a nice thread on this back in April.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Monday, May 21, 2012 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk
No one have an idea ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But
-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B
Hi All
I want to be able to read some sip informations (from a database) like
username, password, host and extension number and place a Dial from
asterisk.
So basicly, I want to dial sip extensions without modifying sip.conf each
time.
I don't know, in the dialplan, what the dial string
: [asterisk-users] Full SIP dial string
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, June 11, 2011 3:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Full SIP dial
Yes, you can use the Mixmonitor command.
But if you want to have only one party on the recording, you should use the
Monitor command without the 'm' option.
http://www.astblog.com/2011/02/01/asterisk-mixmonitor-cmd/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi
In sip.conf, you generally have something like
[name]
..
username=
secret=
What is the difference between the name specified in brackets and the
username key ?
What the sip client should provide ?
What do we use in dialplan when trying to reach this client ?
--
Nobody on this ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: September-16-09 7:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] H323 RTP Transmission error of packet
Using H323 to reach
Using H323 to reach another h323 switch, I have no audio and the following
error:
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Maybe the customer hangs up during the AMD analysis or you don't have any
audio coming to asterisk through your sip channel.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Hawkin
Sent: April-23-09 11:00 AM
To:
First run
/var/lib/asterisk/agi-bin/newhire.php
From linux command line to see if you don't have any error and that your AGI
is executable.
Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi
From:
hi all
do you guys know why asterisk sometimes, in the cdr put the dst (the
extension) number in the src ??
I have 4 digit extensions (DID) and sometimes, the same values if found
in the src that usually have the calling user caller id.
Thanks
___
--
Local channel will help you send your call through the dialplan.
You can make all your decision there.
If it answers, then the specified application will be execute.
Check this example
http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/
David Klaverstyn wrote:
I have
Did you know that any commandyou type in asterisk cli starting with
exclamation point (!) is execute in the shell by asterisk ??
Example :
running
!ls
will run 'ls' in your current directory
So, be aware because your user can do whatever we want then.
Dima wrote:
On Sat, Nov 01, 2008 at
digit customer phone number.
Do you know when does this happens ??
Thanks
Ruddy Gbaguidi
http://www.astblog.com
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit
Hi Guys
We have a service that can be use by our customer via a website and also
via telephone.
On the website, we already accept credit card by sending users to paypal
website where we have an account.
Now, we want to do the same with an IVR where people can call a number,
enter their credit
Yes, we can do that. But :
1. we are not too confortabe about keeping users credit card
informations in our databases
2. we are now targeting the 50, 60+ people and their are not confortable
about a website. So, we want to
be able to register people by phone, and they can make payments by phone.
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10 digit customer number or press * to register
So, I want to read up to 10
-0400 schrieb Ruddy Gbaguidi:
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10 digit customer number or press
Hi thanks for the hint.
That will works I think.
But now, if I'm in an AGI script and I want to stay in there and don't
want to jump from an extension to other in the dialplan,
how can I do it ??
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote
Ruddy,
Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi:
Thanks for the hint. Sorry about that.
If I use your soution, I cannot make any difference between a user
pressing * and a user that reach the timeout because he didn't enter any
digit.
In both cases, I will have
Thanks for your help.
This can be add to Read command as feature
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi thanks for the hint.
That will works I think.
But now, if I'm in an AGI script and I want to stay in there and don't
want
First, if you want to use that, you may want pass the call tracknum to
the myagi.agi,
so you will know which call the dialedtime and answeredtime belongs to.
But you can use the Dial 'g' option that doesn't hangup up both legs of
the call when the called party hangs up.
selmak se wrote:
_
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 19, 2008, at 10:20 PM, Igor A. Goncharovsky wrote:
Hi!
Ruddy Gbaguidi wrote:
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
script to stop
Hi all.
I'm using asterisk 1.4.21.2 and when I run
ps -ef |grep defunct,
I can see a lot of my perl agi still pending there.
The channel has been cleaned up in asterisk.
I don't have this kind of problem with python or php.
I'm using ubuntu ...
Anyone has an idea ?
I've tried export
is not catching SIGHUP, which is what Asterisk uses to tell
the AGI the channel went away.
Ruddy Gbaguidi wrote:
Hi all.
I'm using asterisk 1.4.21.2 and when I run
ps -ef |grep defunct,
I can see a lot of my perl agi still pending there.
The channel has been cleaned up in asterisk.
I don't have
Hi all
Back in the 1.2 days I think, there were some discussions about how two
asterisk
servers can share channel variables through an IAX protocol.
I don't see anything in 1.4 at least to be able to make it done.
Thanks
___
-- Bandwidth and
It doesn't seems to be working ...
What I wanted to do is on the first server, Set a channel variable...
then dial the number.
When I received the call on the remote server, use that variable ...
Is it possible ?
Richard Lyman wrote:
Ruddy Gbaguidi wrote:
Hi all
Back in the 1.2 days I
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts.
the problem is that some variables (and maybe all, not sure) like
ANSWEREDTIME does not kept if the caller hangs up.
my agi script continues to run after
Try DeadAGI and it should work..
Paradise Dove wrote:
I'm using AGI and set AGISIGHUP=no
to make it keep on running on channel hangup
On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using
!??
On Fri, Aug 8, 2008 at 11:30 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Try DeadAGI and it should work..
Paradise Dove wrote:
I'm using AGI and set AGISIGHUP=no
to make it keep on running on channel hangup
On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote
maybe you are using the L option in Dial app to limit the conversation time.
Check those channel variables (just a wild guess)
*LIMIT_PLAYAUDIO_CALLER
**LIMIT_PLAYAUDIO_CALLEE
**LIMIT_TIMEOUT_FILE
**LIMIT_CONNECT_FILE
**LIMIT_WARNING_FILE
*
Felippe Silvestre wrote:
Hi all,
Our users are
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a database table and retrieve an id
2. pass the id to test_connect and test_connect will then
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL
You can check asterisk CDR (call detail records).
You should have a csv file in /var/log/asterisk/cdr-csv/Master.csv
You can also configure it to write the CDR in a database
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
Then you can just write a script that will look at your database and
send
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