asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] event to check for Registered users

[Stefan R. <sr...@re...>]

Aparna,

please have a look at the SipPeersAction. It generates PeerEntryEvents
that contain a status field that you can probably use for your purposes.

=3DStefan

On Mon, 2005-12-19 at 05:59 -0800, Aparna Ramakrishnan wrote:
> Hello,
> Is there an event in Manager API, which can detect the users that have
> been registered with asterisk?
> I now use the PeerStatusEvent which denotes every time a user
> registers with asterisk.=20
> Now at times when i need to re-run my program again, the already
> registered users are not visible. What event can help me find also the
> already resgitered users, which will be displayed after i re-run my
> program?
> =20
> Kindly help.
> =20
> Thanks n Regards
> Aparna
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around=20
> http://mail.yahoo.com=20
>=20

[Asterisk-java-users] event to check for Registered users

[Aparna R. <apa...@ya...>]

Hello,
  Is there an event in Manager API, which can detect the users that have been registered with asterisk?
  I now use the PeerStatusEvent which denotes every time a user registers with asterisk. 
  Now at times when i need to re-run my program again, the already registered users are not visible. What event can help me find also the already resgitered users, which will be displayed after i re-run my program?
   
  Kindly help.
   
  Thanks n Regards
  Aparna

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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[Asterisk-java-users] re: Problem with ConnectEvent

[Brett S. <bs...@no...>]

Here's a technique I use to get arround the problem of trying to send an 
Action when you are in an event handler.
If you don't know; its simply not possible.

The call to 'new RecieveFax(event)' creates its own connection to the 
API Manager.

I do wonder whether it would be better if the MangerEventHandler method 
was called in a seperate thread from that which the ManagerReader runs 
in however that may have other undesirable side affects.
Alternatively the call to SendAction should probably throw an 
IllegalStateException rather than just timing out. As it stand its hard 
to diagnose why the call to SendAction fails.

Basically It's necessary to send the subsequent action in a different 
thread from the the eventHandler otherwise the ManagerReader locks up 
and nothing works. The subsequent actionn should also be sent using a 
separate instance of the ManagerConnection.

You could potentionally send the subsequent action in the event handler 
thread (using a separate ManagerConnection) but you will block the 
ManagerReader from notifying your application of other events and I'm 
not certain, but I suspect that there is little to be gained from 
holding the ManagerReader up whilst you do other work as I doubt that 
Asterisk is waiting on the ManagerReader and as such you really arn't 
doing  the operation synchronously (with respect to Asterisk).

So here is a sample class demonstating one possible solution.

I hope this of some use.

Regards,
Brett.

import java.io.IOException;
import java.util.TreeSet;
import java.util.Vector;
import java.util.concurrent.LinkedBlockingQueue;

import net.sf.asterisk.manager.AuthenticationFailedException;
import net.sf.asterisk.manager.ManagerConnection;
import net.sf.asterisk.manager.ManagerConnectionFactory;
import net.sf.asterisk.manager.ManagerEventHandler;
import net.sf.asterisk.manager.TimeoutException;
import net.sf.asterisk.manager.action.StatusAction;
import net.sf.asterisk.manager.event.ManagerEvent;
import net.sf.asterisk.manager.event.NewChannelEvent;
import net.sf.asterisk.manager.event.NewStateEvent;

import org.apache.commons.configuration.ConfigurationException;
import org.apache.log4j.Logger;


<http://sourceforge.net/mailarchive/message.php?msg_id=13870642>public 
class FaxManagerInbound extends Thread implements ManagerEventHandler
{
    static Logger logger = 
Logger.getLogger(FaxManagerInbound.class.getName());

    private static FaxManagerInbound self;

    private boolean running;

    private ManagerConnection managerConnection;

    private Vector<Channel> channels;

    private static final LinkedBlockingQueue<NewChannelEvent> queue = 
new LinkedBlockingQueue<NewChannelEvent>();

    static public FaxManagerInbound getInstance()
        throws ConfigurationException,
            IOException,
            AuthenticationFailedException,
            TimeoutException
    {
        if (self == null)
        {
            self = new FaxManagerInbound(Configuration.getInstance()
                    .getChannels());
            self.start();
        }
        return self;
    }

    private FaxManagerInbound(TreeSet<Channel> channels)
        throws IOException,
            AuthenticationFailedException,
            TimeoutException
    {
        super("FaxManager Inbound");
        managerConnection = connect();

        // Get the list of channels we are interested in
        for (Channel channel : channels)
        {
            if (channel.type == ChannelManager.ChannelType.Inbound
                    || channel.type == ChannelManager.ChannelType.Both)
            {
                this.channels.add(channel);
            }
        }
    }

    public void run()
    {
        running = true;
        logger.info("Inbound FaxManager started");
        while (running)
        {
            try
            {
                if (!managerConnection.isConnected())
                {
                    logger
                            .info("Inbound Manager API Connection lost; 
reconnecting.");
                    managerConnection = connect();
                    logger.info("Inbound Manager API: reconnected");
                }

                NewChannelEvent event = queue.take();
                new RecieveFax(event);

            }
            catch (Throwable e)
            {
                // We don't like the manager shuting down just becuase of a
                // problem with a single fax.
                logger.error(e.getMessage(), e);
            }
        }
        logger.info("Inbound FaxManager Stopping");
    }

    public void shutdown()
    {
        running = false;
        queue.notify();
        notify();
    }

    ManagerConnection connect()
        throws IOException,
            AuthenticationFailedException,
            TimeoutException
    {
        ManagerConnection connection = null;
        ManagerConnectionFactory factory = new ManagerConnectionFactory();

        connection = factory.getManagerConnection(Configuration
                .getAsteriskHost(), Configuration.getAsteriskUsername(),
            Configuration.getAsteriskPassword());

        // register for events
        connection.addEventHandler(this);

        // connect to Asterisk and log in
        managerConnection.login();

        // request channel state
        managerConnection.sendAction(new StatusAction());

        return connection;
    }

    public void handleEvent(ManagerEvent managerEvent)
    {
        if (managerEvent instanceof NewChannelEvent)
        {
            NewChannelEvent event = (NewChannelEvent) managerEvent;
            System.out.println(event.getState());
            if (event.getState().compareToIgnoreCase("Ringing") == 0
                    || (event.getState().compareToIgnoreCase("Ring") == 0))
            {
                queue.add(event);
            }

        }
        else if (managerEvent instanceof NewStateEvent)
        {
            NewStateEvent event = (NewStateEvent) managerEvent;
            System.out.println("State:" + event.getState());
            System.out.println(event);

        }
        else

            // just print received events
            System.out.println("Event: " + managerEvent);
    }

}

Re: [Asterisk-java-users] How to Answer the phone and call an application.

[Stefan R. <sr...@re...>]

Have a look at RedirectAction it allows you to redirect a channel to an
extension. That extension should then call the application.
The other option is to not use the ManagerAPI at all for this stuff but
implement an AGI script and map the fax extensions to that AGI script.

=3DStefan

On Sat, 2005-12-17 at 10:54 +1100, Brett Sutton wrote:
> I'm trying to write an application that can monitor specific channels=20
> and when a call comes in answer the call and call an application (rx_fax)=
.
>=20
> I think I've worked out how to detect that a channel is ringing but I'm=20
> don' now how to answer it and call an application.
> I've looked at the OriginateEvent but that seems to be for makeing=20
> outbound calls.
>=20
> Any help would be appreciated.
>=20
> Regards,
> Bret.
>=20
>=20
> -------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc. Do you grep through log fi=
les
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> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] How to Answer the phone and call an application.

[Brett S. <bs...@no...>]

I'm trying to write an application that can monitor specific channels 
and when a call comes in answer the call and call an application (rx_fax).

I think I've worked out how to detect that a channel is ringing but I'm 
don' now how to answer it and call an application.
I've looked at the OriginateEvent but that seems to be for makeing 
outbound calls.

Any help would be appreciated.

Regards,
Bret.

Re: [Asterisk-java-users] Interrupting StreamFile with DTMF digits.

[Stefan R. <sr...@re...>]

John Melody schrieb:
> Yes, Jonathan, I am receiving DTMF happily in the rest of the application.
> However, all the rest of the DTMF input takes place in extension.conf. It
> appears to be only when the application is in the FastAGI script that it
> does not recognise the DTMF tones.
> 
> Is there anything that has to be enabled to pass DTMF digits to AGI?

no it usually "just works" :)

I would suggest you to use the latest release of Asterisk-Java (i.e.
0.2) and try to reproduce the problem with that version. And you could
try to allow other dtmf digits to be pressed like streamFile("myprompt",
"1234");

=Stefan

RE: [Asterisk-java-users] Interrupting StreamFile with DTMF digits.

[John M. <jo...@sy...>]

Yes, Jonathan, I am receiving DTMF happily in the rest of the application.
However, all the rest of the DTMF input takes place in extension.conf. It
appears to be only when the application is in the FastAGI script that it
does not recognise the DTMF tones.

Is there anything that has to be enabled to pass DTMF digits to AGI?

-----Original Message-----
From: ast...@li...
[mailto:ast...@li...]On Behalf Of
Jonathan Augenstine
Sent: 15 December 2005 16:57
To: ast...@li...
Subject: Re: [Asterisk-java-users] Interrupting StreamFile with DTMF
digits.


Have you verified that you are receiving DTMF tones.  I ran into a
similar problem and discovered that my Asterisk/VoIP configuration was
not detecting the tones.  My situation turned out to be the firewall
blocking and a wrong SIP configuration.


On Thu, 2005-12-15 at 16:09 +0000, John Melody wrote:
> Hi,
>
> I have an application where I need to interrupt the StreamFile command
with
> a DTMF "*" - I cannot seem to get it to work and I just wanted to check
that
> it does work in the following configuration.
>
> I am using asterisk-java-0.1 and Asterisk 1.0.9 and the following is the
> line of Java.
>
>         streamFile(i_channel, "prompt/a.wav","*");
>
> The file just plays like as if it was a Playback command as opposed to a
> Background command.
>
> best regards,
> John.
>
> John Melody
> SyberNet Ltd.
> Galway Business Park,
> Dangan,
> Galway.
> Tel. No. +353 91 514400
> Fax. NO. +353 91 514409
> Mobile - 087-2345847
>
>
>
> -------------------------------------------------------
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files
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> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>



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Re: [Asterisk-java-users] Eclipse IDE

[Stefan R. <sr...@re...>]

On Thu, 2005-12-15 at 12:13 -0800, Jonathan Augenstine wrote:
> This is really just an annoyance and not creating any road blocks.  But,
> I am currently using the Eclipse IDE.  One of the things that it reports
> is that the AbstractAGIScript and it incumbent methods are deprecated.
> It is not creating any problems but does anyone know why it is doing
> this and what I can do to correct this problem?

yes you should use BaseAGIScript with Asterisk-Java 0.2 :)

[Asterisk-java-users] Eclipse IDE

[Jonathan A. <jau...@st...>]

This is really just an annoyance and not creating any road blocks.  But,
I am currently using the Eclipse IDE.  One of the things that it reports
is that the AbstractAGIScript and it incumbent methods are deprecated.
It is not creating any problems but does anyone know why it is doing
this and what I can do to correct this problem?

Jonathan

Re: [Asterisk-java-users] Interrupting StreamFile with DTMF digits.

[Jonathan A. <jau...@st...>]

Have you verified that you are receiving DTMF tones.  I ran into a
similar problem and discovered that my Asterisk/VoIP configuration was
not detecting the tones.  My situation turned out to be the firewall
blocking and a wrong SIP configuration.


On Thu, 2005-12-15 at 16:09 +0000, John Melody wrote:
> Hi,
> 
> I have an application where I need to interrupt the StreamFile command with
> a DTMF "*" - I cannot seem to get it to work and I just wanted to check that
> it does work in the following configuration.
> 
> I am using asterisk-java-0.1 and Asterisk 1.0.9 and the following is the
> line of Java.
> 
>         streamFile(i_channel, "prompt/a.wav","*");
> 
> The file just plays like as if it was a Playback command as opposed to a
> Background command.
> 
> best regards,
> John.
> 
> John Melody
> SyberNet Ltd.
> Galway Business Park,
> Dangan,
> Galway.
> Tel. No. +353 91 514400
> Fax. NO. +353 91 514409
> Mobile - 087-2345847
> 
> 
> 
> -------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc. Do you grep through log files
> for problems?  Stop!  Download the new AJAX search engine that makes
> searching your log files as easy as surfing the  web.  DOWNLOAD SPLUNK!
> http://ads.osdn.com/?ad_id=7637&alloc_id=16865&op=click
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> 

[Asterisk-java-users] Interrupting StreamFile with DTMF digits.

[John M. <jo...@sy...>]

Hi,

I have an application where I need to interrupt the StreamFile command with
a DTMF "*" - I cannot seem to get it to work and I just wanted to check that
it does work in the following configuration.

I am using asterisk-java-0.1 and Asterisk 1.0.9 and the following is the
line of Java.

        streamFile(i_channel, "prompt/a.wav","*");

The file just plays like as if it was a Playback command as opposed to a
Background command.

best regards,
John.

John Melody
SyberNet Ltd.
Galway Business Park,
Dangan,
Galway.
Tel. No. +353 91 514400
Fax. NO. +353 91 514409
Mobile - 087-2345847

Re: [Asterisk-java-users] Maximum number of calls

[Stefan R. <sr...@re...>]

Ines schrieb:
> does someone know how many calls can asterisk manage at the same time ?

this question is more related to Asterisk than to Asterisk-Java.
Maybe you would like to have a look at
http://www.voip-info.org/wiki/view/Asterisk+dimensioning

=Stefan

[Asterisk-java-users] Maximum number of calls

[Ines <mou...@ya...>]

hello,
  
  does someone know how many calls can asterisk manage at the same time ? 
  
  thanks
  
  inés
  


		
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Re: [Asterisk-java-users] Answer call.

[Stefan R. <sr...@re...>]

Jan du Toit schrieb:
> Hi.
> 
> Let's say we originate a call by using AsteriksManager.originateCall()
> to a given extension.
> 
> The way it works (just looking from a user perspective) is that asteriks
> originate the call, it calls me and when I answer it calls the
> extensions as you configured it via the Originate object which the
> originateCall method receives.
> 
> Now the thing is that my installed SoftPhone starts ringing when the
> call is originated and I need to pick it up from the softphone.
> But I want to pick it up from my java application, not the softphone.
> 
> How can I go about to do that?

you cant do that via asterisk. maybe your softphone has an api that
allows you to do so. i do this with my cisco 7960 that allows to switch
on the speaker via a telnet based api.

=Stefan

[Asterisk-java-users] Answer call.

[Jan du T. <jan...@de...>]

Hi.

Let's say we originate a call by using AsteriksManager.originateCall() 
to a given extension.

The way it works (just looking from a user perspective) is that asteriks 
originate the call, it calls me and when I answer it calls the 
extensions as you configured it via the Originate object which the 
originateCall method receives.

Now the thing is that my installed SoftPhone starts ringing when the 
call is originated and I need to pick it up from the softphone.
But I want to pick it up from my java application, not the softphone.

How can I go about to do that?

Thanks in advance.

PS. (To Stefan) thanks for all the help thus far. Do appreciate it.

Re: [Asterisk-java-users] MonitorAction, audio formats and mixing.

[Stefan R. <sr...@re...>]

Jan Gerryt du Toit schrieb:
> Hi.
> 
> Can somebody please supply me with a list of all surported recording audio
> formats that MonitorAction supports?

you can use any of the audio formats for monitoring that asterisk
supports. type

show file formats

on the asterisk console to get a list.

> Also, I have set my MonitorAction mixing variable to true, but it doesn't "join"
> the in and out channels, it still produces two seperate files with the recorded
> in and out channels. I am using .wav format. Does the chosen format have
> something to do with my mixing problem?

Asterisk uses soxmix to mix the files therefore soxmix must be installed
on the Asterisk machine and must be available on the PATH.

=Stefan

[Asterisk-java-users] MonitorAction, audio formats and mixing.

[Jan G. du T. <s22...@tu...>]

Hi.

Can somebody please supply me with a list of all surported recording audio
formats that MonitorAction supports?

Also, I have set my MonitorAction mixing variable to true, but it doesn't "join"
the in and out channels, it still produces two seperate files with the recorded
in and out channels. I am using .wav format. Does the chosen format have
something to do with my mixing problem?

Thanks in advance.

Re: [Asterisk-java-users] DTMF digit collection

[Jonathan A. <jau...@st...>]

Stefan,

Thank you very much for your input.  You have confirmed some ideas I
have been working with last night which was to integrate AGI/Manager
API.  I was not yet confident that was the solution.  I wanted to make
sure I was not missing something in looking at the Manager API.  I
believe that utilization of both interfaces will provide the
capabilities I need.

Thanks again for your help.

Jonathan


On Wed, 2005-12-14 at 16:40 +0000, Stefan Reuter wrote:
> 
> On Wed, 2005-12-14 at 08:00 -0800, Jonathan Augenstine wrote:
> > The specific problem I encountered is that I make an outbound call from
> > Asterisk using the execCommand(Dial).  The problem is that the
> > execCommand call blocks and I need to have the application and move on
> > to other tasks.
> 
> Thanks for the clarification.
> I propose to use AGI for the identification as you said that already
> works well. I understand you problem with the exec dial though. For this
> there are two solutions. First you can call setContext, setExtension and
> setPriority inside your AGI script instead of exec dial. This causes the
> call to continue at the given extension in the dialplan when the AGI
> script ends. Second you could just issue a RedirectAction via the
> Manager API at the end of the AGI script.
> That way you can combine the strenghts of both interfaces (and you
> probably already use the Manager API to originate the outbound call).
> Do you think this would solve your problem?
> 
> =Stefan

Re: [Asterisk-java-users] DTMF digit collection

[Stefan R. <sr...@re...>]

On Wed, 2005-12-14 at 08:00 -0800, Jonathan Augenstine wrote:
> The specific problem I encountered is that I make an outbound call from
> Asterisk using the execCommand(Dial).  The problem is that the
> execCommand call blocks and I need to have the application and move on
> to other tasks.

Thanks for the clarification.
I propose to use AGI for the identification as you said that already
works well. I understand you problem with the exec dial though. For this
there are two solutions. First you can call setContext, setExtension and
setPriority inside your AGI script instead of exec dial. This causes the
call to continue at the given extension in the dialplan when the AGI
script ends. Second you could just issue a RedirectAction via the
Manager API at the end of the AGI script.
That way you can combine the strenghts of both interfaces (and you
probably already use the Manager API to originate the outbound call).
Do you think this would solve your problem?

=3DStefan

Re: [Asterisk-java-users] DTMF digit collection

[Jonathan A. <jau...@st...>]

Stefan,

The specific problem I encountered is that I make an outbound call from
Asterisk using the execCommand(Dial).  The problem is that the
execCommand call blocks and I need to have the application and move on
to other tasks.

Specifically, I have an inbound call to Asterisk, it does some user
id/account verification and then makes an outbound call.  I need to make
the outbound call, perform other tasks, and then disconnect the outbound
call.

I concluded that while the AGI worked great for the initial IVR portion,
some of the other actions I needed to perform had a better fit to an
event driven model.

Jonathan


On Wed, 2005-12-14 at 07:54 +0000, Stefan Reuter wrote:
> Hi Jonathan,
> 
> what problems did you encounter when you tried the AGI interface?
> The AGI interface is usually much better suited for IVR style
> applications than the Manager API.
> 
> =Stefan
> 
> On Tue, 2005-12-13 at 20:52 -0800, Jonathan Augenstine wrote:
> > I am developing an IVR application.  I have done some prototyping with
> > the AGI interface and it does not quite work the way I need.  I have
> > determined that the asynchronous behavior of the Manager API is a better
> > fit.  However, I have been unable to determine how to call CLI
> > applications.  For instance, I need to play a prompt file and collect
> > DTMF digits to allow the caller to enter an ID code.  Can someone please
> > point me in the direction of how to accomplish this?
> > 
> > Jonathan
> > 
> > 
> > 
> > 
> > -------------------------------------------------------
> > This SF.net email is sponsored by: Splunk Inc. Do you grep through log files
> > for problems?  Stop!  Download the new AJAX search engine that makes
> > searching your log files as easy as surfing the  web.  DOWNLOAD SPLUNK!
> > http://ads.osdn.com/?ad_id=7637&alloc_id=16865&op=click
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Small bug in StopMonitorAction

[Stefan R. <sr...@re...>]

On Tue, 2005-12-13 at 12:09 +0200, Jan Gerryt du Toit wrote:
> The constructor of StopMonitor which receives a String as a channel ID is=
 empty:
> I don't now if this is the correct mailing list to send to for a bug/fix =
report.
> Sorry if it isn't

thanks for the bug report.
The best way to address it is through our bug tracker available at
http://jira.reucon.com

=3DStefan

Re: [Asterisk-java-users] Muting and UnMuting in *-J

[Stefan R. <sr...@re...>]

On Tue, 2005-12-13 at 10:24 +0200, Jan Gerryt du Toit wrote:
> Can *-J handle muting? Their doesn't seem to be a MutingAction for the Ma=
nager.
> AGI maybe?

I don't think that Asterisk itself provides that capability.

=3DStefan

Re: [Asterisk-java-users] DTMF digit collection

[Stefan R. <sr...@re...>]

Hi Jonathan,

what problems did you encounter when you tried the AGI interface?
The AGI interface is usually much better suited for IVR style
applications than the Manager API.

=3DStefan

On Tue, 2005-12-13 at 20:52 -0800, Jonathan Augenstine wrote:
> I am developing an IVR application.  I have done some prototyping with
> the AGI interface and it does not quite work the way I need.  I have
> determined that the asynchronous behavior of the Manager API is a better
> fit.  However, I have been unable to determine how to call CLI
> applications.  For instance, I need to play a prompt file and collect
> DTMF digits to allow the caller to enter an ID code.  Can someone please
> point me in the direction of how to accomplish this?
>=20
> Jonathan
>=20
>=20
>=20
>=20
> -------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc. Do you grep through log fi=
les
> for problems?  Stop!  Download the new AJAX search engine that makes
> searching your log files as easy as surfing the  web.  DOWNLOAD SPLUNK!
> http://ads.osdn.com/?ad_id=3D7637&alloc_id=3D16865&op=3Dclick
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] DTMF digit collection

[Jonathan A. <jau...@st...>]

I am developing an IVR application.  I have done some prototyping with
the AGI interface and it does not quite work the way I need.  I have
determined that the asynchronous behavior of the Manager API is a better
fit.  However, I have been unable to determine how to call CLI
applications.  For instance, I need to play a prompt file and collect
DTMF digits to allow the caller to enter an ID code.  Can someone please
point me in the direction of how to accomplish this?

Jonathan

[Asterisk-java-users] MonitorAction, audio formats and mixing.

[Jan G. du T. <s22...@tu...>]

Hi.

Can somebody please supply me with a list of all surported recording audio
formats that MonitorAction supports?

Also, I have set my MonitorAction mixing variable to true, but it doesn't "join"
the in and out channels, it still produces two seperate files with the recorded
in and out channels. I am using .wav format. Does the chosen format have
something to do with my mixing problem?

Thanks in advance.