asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

[Chris M. <ch...@mr...>]

hi greg,
paste the output of freepbx's channel dump output.
what columns are available, you're looking at 'location' by the sounds of it

to expedite, i've doctored a snippet [1] for you from cli

chris


[1]
from * cli it would look something like

[cm@box5 ~]# asterisk -rx "core show channels"
Channel              Location/YourCustomDestination             State
Application(Data)
SIP/101-0000eb2d    1024@from-internal                          Up
 MeetMe(1024

then use the SIP/101-xxxxxx channel






On Thu, May 23, 2013 at 2:01 AM, Greg Horton <gre...@gm...> wrote:

> Hi Chris,
>
> When I look at the channel dump in FreePBX, it shows the destination as
> "1024@from-internal:1" while it's in the state of waiting for a PIN
> sequence.  So I tried using the A-J "sendAction(PlayDtmfAction)" interface
> with that destination, but no luck.  I get returnCode "Error".
>
> I also tried:
> 1024@from-internal
> Local/1024@from-internal
> Loca1/1024@from-internal:1
> Local/1024
>
> It seems like I am trying to pass the digit into the Dialplan at this
> point, and I am wondering if the sendAction(PlayDtmfAction) will not work
> for this case.  Maybe something else is required in A-J?
>
> Thanks,
> Greg
>
>
> On Tue, May 21, 2013 at 11:28 PM, Chris Mylonas <ch...@mr...>wrote:
>
>> When your dialer module is connected to the meetme room, issue a dialplan
>> function "core show channels" over AJ or on the asterisk CLI - you might be
>> able to see which channel your dialer module is using
>>
>> *shrug*
>>
>> HTH
>> Chris
>>
>>
>> On Wed, May 22, 2013 at 12:42 PM, GregLHorton <gre...@gm...>wrote:
>>
>>>
>>> Hi all,
>>>
>>> I have a client app with a dialer module, that can dial into a conference
>>> normally.  If my conference room is 1024, then I can pass 1024 to
>>> initiateCall() to dial into the conference.  If a password is required,
>>> the
>>> request for digits is played out by Asterisk.  At that time, digits
>>> entered
>>> in the dialer module do not get passed to the endpoint that detects
>>> the password digits.  This is because I normally use PlayDtmfAction for
>>> this
>>> purpose and that requires a channel like SIP/100-000034ab.  I do not see
>>> where this type of channel identifier is applicable to the conference
>>> room I
>>> just dialed into.  I have seen where a dummy channel ID relating to DAHDI
>>> shows up, but that is not until after I can successfully pass the PIN
>>> string.
>>>
>>> This is using MeetMe on Asterisk 10.0.
>>>
>>> I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
>>> Just wild guesses of course :).
>>>
>>> Any help appreciated!
>>>
>>> Greg
>>> --
>>> View this message in context:
>>> http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422999p35422999.html
>>> Sent from the Asterisk-Java Users mailing list archive at Nabble.com.
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Try New Relic Now & We'll Send You this Cool Shirt
>>> New Relic is the only SaaS-based application performance monitoring
>>> service
>>> that delivers powerful full stack analytics. Optimize and monitor your
>>> browser, app, & servers with just a few lines of code. Try New Relic
>>> and get this awesome Nerd Life shirt!
>>> http://p.sf.net/sfu/newrelic_d2d_may
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>>
>>
>>
>> --
>>
>> -- sent from web mail --
>>
>>
>> ------------------------------------------------------------------------------
>> Try New Relic Now & We'll Send You this Cool Shirt
>> New Relic is the only SaaS-based application performance monitoring
>> service
>> that delivers powerful full stack analytics. Optimize and monitor your
>> browser, app, & servers with just a few lines of code. Try New Relic
>> and get this awesome Nerd Life shirt!
>> http://p.sf.net/sfu/newrelic_d2d_may
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
> ------------------------------------------------------------------------------
> Try New Relic Now & We'll Send You this Cool Shirt
> New Relic is the only SaaS-based application performance monitoring service
> that delivers powerful full stack analytics. Optimize and monitor your
> browser, app, & servers with just a few lines of code. Try New Relic
> and get this awesome Nerd Life shirt! http://p.sf.net/sfu/newrelic_d2d_may
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>


-- 

-- sent from web mail --

Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

[Greg H. <gre...@gm...>]

Hi Chris,

When I look at the channel dump in FreePBX, it shows the destination as
"1024@from-internal:1" while it's in the state of waiting for a PIN
sequence.  So I tried using the A-J "sendAction(PlayDtmfAction)" interface
with that destination, but no luck.  I get returnCode "Error".

I also tried:
1024@from-internal
Local/1024@from-internal
Loca1/1024@from-internal:1
Local/1024

It seems like I am trying to pass the digit into the Dialplan at this
point, and I am wondering if the sendAction(PlayDtmfAction) will not work
for this case.  Maybe something else is required in A-J?

Thanks,
Greg


On Tue, May 21, 2013 at 11:28 PM, Chris Mylonas <ch...@mr...> wrote:

> When your dialer module is connected to the meetme room, issue a dialplan
> function "core show channels" over AJ or on the asterisk CLI - you might be
> able to see which channel your dialer module is using
>
> *shrug*
>
> HTH
> Chris
>
>
> On Wed, May 22, 2013 at 12:42 PM, GregLHorton <gre...@gm...>wrote:
>
>>
>> Hi all,
>>
>> I have a client app with a dialer module, that can dial into a conference
>> normally.  If my conference room is 1024, then I can pass 1024 to
>> initiateCall() to dial into the conference.  If a password is required,
>> the
>> request for digits is played out by Asterisk.  At that time, digits
>> entered
>> in the dialer module do not get passed to the endpoint that detects
>> the password digits.  This is because I normally use PlayDtmfAction for
>> this
>> purpose and that requires a channel like SIP/100-000034ab.  I do not see
>> where this type of channel identifier is applicable to the conference
>> room I
>> just dialed into.  I have seen where a dummy channel ID relating to DAHDI
>> shows up, but that is not until after I can successfully pass the PIN
>> string.
>>
>> This is using MeetMe on Asterisk 10.0.
>>
>> I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
>> Just wild guesses of course :).
>>
>> Any help appreciated!
>>
>> Greg
>> --
>> View this message in context:
>> http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422999p35422999.html
>> Sent from the Asterisk-Java Users mailing list archive at Nabble.com.
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Try New Relic Now & We'll Send You this Cool Shirt
>> New Relic is the only SaaS-based application performance monitoring
>> service
>> that delivers powerful full stack analytics. Optimize and monitor your
>> browser, app, & servers with just a few lines of code. Try New Relic
>> and get this awesome Nerd Life shirt!
>> http://p.sf.net/sfu/newrelic_d2d_may
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>
>
>
> --
>
> -- sent from web mail --
>
>
> ------------------------------------------------------------------------------
> Try New Relic Now & We'll Send You this Cool Shirt
> New Relic is the only SaaS-based application performance monitoring service
> that delivers powerful full stack analytics. Optimize and monitor your
> browser, app, & servers with just a few lines of code. Try New Relic
> and get this awesome Nerd Life shirt! http://p.sf.net/sfu/newrelic_d2d_may
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

[Chris M. <ch...@mr...>]

When your dialer module is connected to the meetme room, issue a dialplan
function "core show channels" over AJ or on the asterisk CLI - you might be
able to see which channel your dialer module is using

*shrug*

HTH
Chris


On Wed, May 22, 2013 at 12:42 PM, GregLHorton <gre...@gm...> wrote:

>
> Hi all,
>
> I have a client app with a dialer module, that can dial into a conference
> normally.  If my conference room is 1024, then I can pass 1024 to
> initiateCall() to dial into the conference.  If a password is required, the
> request for digits is played out by Asterisk.  At that time, digits entered
> in the dialer module do not get passed to the endpoint that detects
> the password digits.  This is because I normally use PlayDtmfAction for
> this
> purpose and that requires a channel like SIP/100-000034ab.  I do not see
> where this type of channel identifier is applicable to the conference room
> I
> just dialed into.  I have seen where a dummy channel ID relating to DAHDI
> shows up, but that is not until after I can successfully pass the PIN
> string.
>
> This is using MeetMe on Asterisk 10.0.
>
> I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
> Just wild guesses of course :).
>
> Any help appreciated!
>
> Greg
> --
> View this message in context:
> http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422999p35422999.html
> Sent from the Asterisk-Java Users mailing list archive at Nabble.com.
>
>
>
> ------------------------------------------------------------------------------
> Try New Relic Now & We'll Send You this Cool Shirt
> New Relic is the only SaaS-based application performance monitoring service
> that delivers powerful full stack analytics. Optimize and monitor your
> browser, app, & servers with just a few lines of code. Try New Relic
> and get this awesome Nerd Life shirt! http://p.sf.net/sfu/newrelic_d2d_may
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>



-- 

-- sent from web mail --

[Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

[GregLHorton <gre...@gm...>]

Hi all,

I have a client app with a dialer module, that can dial into a conference
normally.  If my conference room is 1024, then I can pass 1024 to
initiateCall() to dial into the conference.  If a password is required, the
request for digits is played out by Asterisk.  At that time, digits entered
in the dialer module do not get passed to the endpoint that detects
the password digits.  This is because I normally use PlayDtmfAction for this
purpose and that requires a channel like SIP/100-000034ab.  I do not see
where this type of channel identifier is applicable to the conference room I
just dialed into.  I have seen where a dummy channel ID relating to DAHDI
shows up, but that is not until after I can successfully pass the PIN
string. 

This is using MeetMe on Asterisk 10.0.

I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
Just wild guesses of course :).

Any help appreciated!

Greg
-- 
View this message in context: http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422999p35422999.html
Sent from the Asterisk-Java Users mailing list archive at Nabble.com.

[Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

[GregLHorton <gre...@gm...>]

Hi all,

I have a client app with a dialer module, that can dial into a conference
normally.  If my conference room is 1024, then I can pass 1024 to
initiateCall() to dial into the conference.  If a password is required, the
request for digits is played out by Asterisk.  At that time, digits entered
in the dialer module do not get passed to the endpoint that detects
the password digits.  This is because I normally use PlayDtmfAction for this
purpose and that requires a channel like SIP/100-000034ab.  I do not see
where this type of channel identifier is applicable to the conference room I
just dialed into.  I have seen where a dummy channel ID relating to DAHDI
shows up, but that is not until after I can successfully pass the PIN
string. 

This is using MeetMe on Asterisk 10.0.

I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
Just wild guesses of course :).

Any help appreciated!

Greg
-- 
View this message in context: http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422995p35422995.html
Sent from the Asterisk-Java Users mailing list archive at Nabble.com.

[Asterisk-java-users] Beware of Biggest Fake conference in computer science

[<hi...@co...>]

Beware of Biggest Fake conference in computer science

If you have any plans of participating in the world’s biggest 
fake computer science conference then you must look at the website https://sites.google.com/site/worlddump1

If the above link didn't work then see

http://worldcomp-fake-bogus.blogspot.com  
or
https://sites.google.com/site/dumpconf
or Google search using keywords: worldcomp, fake

Re: [Asterisk-java-users] Phone Auto Answer

[Chris M. <ch...@mr...>]

I'm not familiar enough with AJ, but maybe you could try adding a SIP
header with the SIPAddHeader dialplan function over AJ.  Google sip auto
answer asterisk snom Cisco (relevant ones) for more info on what to add.

Also check the handset config in sip settings for accepting auto answer,
this is a gotcha with snoms

HTH
Chris
 On May 21, 2013 10:36 PM, "Wayne Merricks" <way...@th...>
wrote:

> Hi all,
>
> I wrote an application using Asterisk Java a few years back but it never
> got past the prototyping stage.  Long story short we're now revisiting
> this solution with a view to implementation but I'm having trouble
> making the phones auto answer a call.
>
> I know it can be done as we've had several commercial systems that demo
> this, I just can't figure out where you do this in Asterisk Java.
>
> Doing a simple OriginateAction (or RedirectAction) will successfully
> make the phones ring but you have to manually pick up the phone to
> answer the call.  I'm not sure whether there is some sort of special
> Asterisk command I can send or whether I need to craft a special packet
> to the IP Phones themselves in order to make them answer.
>
> We currently use a whole bunch of Cisco Linksys 921s and 303s which will
> auto answer but I can't find documentation on this so any hints would be
> appreciated.
>
> Regards,
>
> Wayne
>
>
> ------------------------------------------------------------------------------
> Try New Relic Now & We'll Send You this Cool Shirt
> New Relic is the only SaaS-based application performance monitoring service
> that delivers powerful full stack analytics. Optimize and monitor your
> browser, app, & servers with just a few lines of code. Try New Relic
> and get this awesome Nerd Life shirt! http://p.sf.net/sfu/newrelic_d2d_may
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

[Asterisk-java-users] Phone Auto Answer

[Wayne M. <way...@th...>]

Hi all,

I wrote an application using Asterisk Java a few years back but it never 
got past the prototyping stage.  Long story short we're now revisiting 
this solution with a view to implementation but I'm having trouble 
making the phones auto answer a call.

I know it can be done as we've had several commercial systems that demo 
this, I just can't figure out where you do this in Asterisk Java.

Doing a simple OriginateAction (or RedirectAction) will successfully 
make the phones ring but you have to manually pick up the phone to 
answer the call.  I'm not sure whether there is some sort of special 
Asterisk command I can send or whether I need to craft a special packet 
to the IP Phones themselves in order to make them answer.

We currently use a whole bunch of Cisco Linksys 921s and 303s which will 
auto answer but I can't find documentation on this so any hints would be 
appreciated.

Regards,

Wayne

Re: [Asterisk-java-users] No event class registered or Unable to set property"......" to "......" on org.asterisk.java ..... no setter

[matthieu h. <mat...@gm...>]

Hi,

Please fin my java code,
package connection;

import java.io.IOException;

import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.ManagerEventListener;
import org.asteriskjava.manager.TimeoutException;
import org.asteriskjava.manager.action.QueueAddAction;
import org.asteriskjava.manager.action.QueueRemoveAction;
import org.asteriskjava.manager.event.ManagerEvent;
import org.asteriskjava.manager.event.QueueMemberAddedEvent;
import org.asteriskjava.manager.response.ManagerError;
import org.asteriskjava.manager.response.ManagerResponse;

public class InteractAsterisk implements ManagerEventListener
{
    private ManagerConnection managerConnection;


    private boolean _run = true;

    public  InteractAsterisk() throws IOException
    {

        ManagerConnectionFactory factory = new
ManagerConnectionFactory("10.140.7.91","cccti","cccti");
        this.managerConnection = factory.createManagerConnection();
    }

    public void run() throws IOException, AuthenticationFailedException,
            TimeoutException, InterruptedException
    {
        // register for events
        managerConnection.addEventListener(this);


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

        QueueAddAction action = new QueueAddAction("support", "SIP/1002");
       // QueueAddAction action2 = new QueueAddAction("support",
"SIP/1001");
        System.out.println(" Login de l'agent");
        ManagerResponse res = managerConnection.sendAction(action,0);
        //ManagerResponse res2 = managerConnection.sendAction(action2,0);


        if (res instanceof ManagerError) {
            System.out.println(" Agent deja logué, on force le logout");
            managerConnection.sendAction(new QueueRemoveAction("support",
"SIP/1002"),0);
            //managerConnection.sendAction(new QueueRemoveAction("support",
"SIP/1001"),0);

            System.out.println("ReLogin de l'agent");
            res = managerConnection.sendAction(action,0);
            //res2 = managerConnection.sendAction(action2,0);

        }
        System.out.println();
        System.out.println(res);
        //System.out.println(res2);
        System.out.println();



        while (this._run){
            try {
                Thread.sleep(500);
            } catch (InterruptedException e) {
                // TODO Auto-generated catch block
                e.printStackTrace();
            }
        }

    }


    public static void main(String[] args) throws Exception
    {
        InteractAsterisk helloEvents;

        helloEvents = new InteractAsterisk();
        helloEvents.run();
    }


    @Override
    public void onManagerEvent(ManagerEvent arg0)
    {
        // TODO Auto-generated method stub
        System.out.println();
        System.out.println(")===>" + arg0 +"\n");
        //System.out.println(((QueueMemberEvent) arg0).getStatus());
        if (arg0 instanceof QueueMemberAddedEvent) {
            System.out.println("\n Agent logué");

            try {
                new Thread(){
                    public void run() {
                        try {
                            System.out.println("");
                            //System.out.println("Logout de l'Agent");

//InteractAsterisk.this.managerConnection.sendAction(new
QueueRemoveAction("support", "SIP/1002"),0);
                           // managerConnection.logoff();
                            //InteractAsterisk.this._run = false;
                     } catch (IllegalStateException e) {
                            // TODO Auto-generated catch block
                            e.printStackTrace();
                        } catch (IllegalArgumentException e) {
                        // TODO Auto-generated catch block
                        e.printStackTrace();

                        e.printStackTrace();
                    }
                    }
                }.start();
            } catch (IllegalArgumentException e) {
                // TODO Auto-generated catch block
                e.printStackTrace();
            } catch (IllegalStateException e) {
                // TODO Auto-generated catch block
                e.printStackTrace();
            }
        }



    }

}


Here it is my manager.conf in Asterisk:
[cccti]
secret = cccti
deny=0.0.0.0/0.0.0.0
permit=10.140.6.214/255.255.255.0
read = system,call,log,verbose,command,agent,user,reporting,cdr
,dialplan,cc
write = system,call,log,verbose,command,agent,user,reporting,message

Thanks for help,

Regards,
Matthieu


2013/5/16 Yves A. <yv...@gm...>

>  hi,
> post your code and we will see....
>
> yves
>
> Am 16.05.2013 14:14, schrieb matthieu hiel:
>
>    Thank you for your response.
>
>  I try with asterisk java 1.0.0-m1. It is working almost perfectly. I have
> still some errors like these:
>
> Unable to set property 'context' to 'default' on
> org.asteriskjava.manager.event.NewChannelEvent: no setter. Please report at
> http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.NewChannelEvent[dateReceived=Thu May 16
> 13:54:59 CEST
> 2013,privilege='call,all',callerid='null',state='Down',channelstate='0',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',accountcode='null',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Down',systemHashcode=23910357]
>
> Unable to set property 'connectedlinename' to 'Matthieu Hiel' on
> org.asteriskjava.manager.event.NewStateEvent: no setter. Please report at
> http://jira.reucon.org/browse/AJUnable to set property 'connectedlinenum'
> to '1001' on org.asteriskjava.manager.event.NewStateEvent: no setter.
> Please report at http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.NewStateEvent[dateReceived=Thu May 16
> 13:54:59 CEST
> 2013,privilege='call,all',callerid='null',state='Ringing',channelstate='5',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Ringing',systemHashcode=2917593]
>
> Unable to set property 'stateinterface' to 'SIP/1002' on
> org.asteriskjava.manager.event.QueueMemberStatusEvent: no setter. Please
> report at http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.QueueMemberStatusEvent[dateReceived=Thu May
> 16 13:54:59 CEST
> 2013,privilege='agent,all',queue='support',membership='dynamic',location='SIP/1002',status='6',paused='false',timestamp='null',lastcall='0',actionid='null',internalactionid='null',name='SIP/1002',server='null',membername='SIP/1002',penalty='0',callstaken='0',systemHashcode=28405330]
>
>  I will try to deal with it.
>
>  Thanks again,
>  Regards,
>  Matthieu
>
>
>
>  2013/5/16 Yves A. <yv...@gm...>
>
>>  hi,
>>
>> first of all I would recommend using the most recent version of
>> asterisk-java.
>> surely aj 1.x is older than asterisk 11.x but I am using it too and do
>> not face these errors... give it a try!
>>
>> regards,
>> yves
>>
>> Am 15.05.2013 13:59, schrieb matthieu hiel:
>>
>>  Hi Guys,
>> I am using Asterisk 11.2.1 with Asterisk-java-0.3.1.jar
>>
>> I manage to connect my Asterisk with my java program.
>> I keep getting these errors all the time and things seem to work, but it
>> is hard to follow my logging because every action that is done throws them :
>>
>> No event class registered for event type 'musiconhold', attributes:
>> {state=Start, class=default, uniqueid=1368617231.116, event=MusicOnHold,
>> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null
>>
>> No event class registered for event type 'musiconhold', attributes:
>> {state=Stop, uniqueid=1368617231.116, event=MusicOnHold,
>> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null
>>
>> No event class registered for event type 'rtcpreceived', attributes:
>> {highestsequence=18068, dlsr=0.0000(sec), fractionlost=0, iajitter=0,
>> sequencenumbercycles=0, from=10.140.6.214:16289, senderssrc=0,
>> lastsr=0.0000000000, packetslost=0, event=RTCPReceived,
>> privilege=reporting,all, pt=200(Sender Report),
>> receptionreports=1}buildEvent returned null
>>
>> No event class registered for event type 'varset', attributes:
>> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>> value=ssrc=471130521;themssrc=416812302;lp=0;rxjitter=0.000064;rxcount=233;txjitter=0.000000;txcount=241;rlp=0;rtt=0.000000,
>> variable=RTPAUDIOQOS, channel=SIP/1001-00000074}buildEvent returned null
>> No event class registered for event type 'varset', attributes:
>> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>> value=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;,
>> variable=RTPAUDIOQOSJITTER, channel=SIP/1001-00000074}buildEvent returned
>> nul
>>
>> lNo event class registered for event type 'varset', attributes:
>> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>> value=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;,
>> variable=RTPAUDIOQOSLOSS, channel=SIP/1001-00000074}buildEvent returned null
>>
>> No event class registered for event type 'varset', attributes:
>> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>> value=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;,
>> variable=RTPAUDIOQOSRTT, channel=SIP/1001-00000074}buildEvent returned null
>>
>> No event class registered for event type 'hanguprequest', attributes:
>> {uniqueid=1368617231.116, event=HangupRequest, privilege=call,all,
>> channel=SIP/1001-00000074}buildEvent returned nullDispatching event:
>>
>> org.asteriskjava.manager.event.QueueCallerAbandonEvent[dateReceived=Wed
>> May 15 13:27:18 CEST
>> 2013,privilege='agent,all',timestamp='null',position='1',queue='support',uniqueid='1368617231.116',originalposition='1',holdtime='5',count='null',channel='null',systemHashcode=28050664]
>>
>> No event class registered for event type 'varset', attributes:
>> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, value=1,
>> variable=QUEUEPOSITION, channel=SIP/1001-00000074}buildEvent returned
>> nullNo event class registered for event type 'softhanguprequest',
>> attributes: {cause=16, uniqueid=1368617231.116, event=SoftHangupRequest,
>> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned
>> nullUnable to set property 'connectedlinename' to '<unknown>' on
>> org.asteriskjava.manager.event.HangupEvent: no setterUnable to set property
>> 'connectedlinenum' to '<unknown>' on
>> org.asteriskjava.manager.event.HangupEvent: no setterDispatching event:
>>
>> org.asteriskjava.manager.event.HangupEvent[dateReceived=Wed May 15
>> 13:27:18 CEST
>> 2013,privilege='call,all',causetxt='Unknown',callerid='1001',cause='0',state='null',calleridname='MatthieuHIEl',timestamp='null',uniqueid='1368617231.116',calleridnum='1001',channel='SIP/1001-00000074',systemHashcode=2548785]
>>
>>
>> And I hava theese error too:
>>
>> Unable to set property 'position' to '1' on
>> org.asteriskjava.manager.event.LeaveEvent: no setter
>> Dispatching event:
>> org.asteriskjava.manager.event.LeaveEvent[dateReceived=Wed May 15
>> 13:27:18 CEST
>> 2013,privilege='call,all',timestamp='null',queue='support',uniqueid='1368617231.116',count='0',channel='SIP/1001-00000074',systemHashcode=7754385]
>>
>> It is always complainig about not having a setter or there is no class
>> registered for an event. Is there a way to fix the issue?
>> Do not hesitate if you want more information!
>>
>> Could anyone please help me on this.
>>
>> Thanks,
>> Matthieu
>>
>>
>>
>>
>>  ------------------------------------------------------------------------------
>> AlienVault Unified Security Management (USM) platform delivers complete
>> security visibility with the essential security capabilities. Easily and
>> efficiently configure, manage, and operate all of your security controls
>> from a single console and one unified framework. Download a free trial.http://p.sf.net/sfu/alienvault_d2d
>>
>>
>>
>> _______________________________________________
>> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> AlienVault Unified Security Management (USM) platform delivers complete
>> security visibility with the essential security capabilities. Easily and
>> efficiently configure, manage, and operate all of your security controls
>> from a single console and one unified framework. Download a free trial.
>> http://p.sf.net/sfu/alienvault_d2d
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.http://p.sf.net/sfu/alienvault_d2d
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.
> http://p.sf.net/sfu/alienvault_d2d
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] No event class registered or Unable to set property"......" to "......" on org.asterisk.java ..... no setter

[Yves A. <yv...@gm...>]

hi,
post your code and we will see....

yves

Am 16.05.2013 14:14, schrieb matthieu hiel:
> Thank you for your response.
>
> I try with asterisk java 1.0.0-m1. It is working almost perfectly. I 
> have still some errors like these:
>
> Unable to set property 'context' to 'default' on 
> org.asteriskjava.manager.event.NewChannelEvent: no setter. Please 
> report at http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.NewChannelEvent[dateReceived=Thu May 16 
> 13:54:59 CEST 
> 2013,privilege='call,all',callerid='null',state='Down',channelstate='0',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',accountcode='null',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Down',systemHashcode=23910357]
>
> Unable to set property 'connectedlinename' to 'Matthieu Hiel' on 
> org.asteriskjava.manager.event.NewStateEvent: no setter. Please report 
> at http://jira.reucon.org/browse/AJUnable to set property 
> 'connectedlinenum' to '1001' on 
> org.asteriskjava.manager.event.NewStateEvent: no setter. Please report 
> at http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.NewStateEvent[dateReceived=Thu May 16 
> 13:54:59 CEST 
> 2013,privilege='call,all',callerid='null',state='Ringing',channelstate='5',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Ringing',systemHashcode=2917593]
>
> Unable to set property 'stateinterface' to 'SIP/1002' on 
> org.asteriskjava.manager.event.QueueMemberStatusEvent: no setter. 
> Please report at http://jira.reucon.org/browse/AJDispatching event:
> org.asteriskjava.manager.event.QueueMemberStatusEvent[dateReceived=Thu 
> May 16 13:54:59 CEST 
> 2013,privilege='agent,all',queue='support',membership='dynamic',location='SIP/1002',status='6',paused='false',timestamp='null',lastcall='0',actionid='null',internalactionid='null',name='SIP/1002',server='null',membername='SIP/1002',penalty='0',callstaken='0',systemHashcode=28405330]
>
> I will try to deal with it.
>
> Thanks again,
> Regards,
> Matthieu
>
>
>
> 2013/5/16 Yves A. <yv...@gm... <mailto:yv...@gm...>>
>
>     hi,
>
>     first of all I would recommend using the most recent version of
>     asterisk-java.
>     surely aj 1.x is older than asterisk 11.x but I am using it too
>     and do not face these errors... give it a try!
>
>     regards,
>     yves
>
>     Am 15.05.2013 13:59, schrieb matthieu hiel:
>>     Hi Guys,
>>     I am using Asterisk 11.2.1 with Asterisk-java-0.3.1.jar
>>     I manage to connect my Asterisk with my java program.
>>     I keep getting these errors all the time and things seem to work,
>>     but it is hard to follow my logging because every action that is
>>     done throws them :
>>     No event class registered for event type 'musiconhold',
>>     attributes: {state=Start, class=default, uniqueid=1368617231.116,
>>     event=MusicOnHold, privilege=call,all,
>>     channel=SIP/1001-00000074}buildEvent returned null
>>     No event class registered for event type 'musiconhold',
>>     attributes: {state=Stop, uniqueid=1368617231.116,
>>     event=MusicOnHold, privilege=call,all,
>>     channel=SIP/1001-00000074}buildEvent returned null
>>     No event class registered for event type 'rtcpreceived',
>>     attributes: {highestsequence=18068, dlsr=0.0000(sec),
>>     fractionlost=0, iajitter=0, sequencenumbercycles=0,
>>     from=10.140.6.214:16289 <http://10.140.6.214:16289>,
>>     senderssrc=0, lastsr=0.0000000000, packetslost=0,
>>     event=RTCPReceived, privilege=reporting,all, pt=200(Sender
>>     Report), receptionreports=1}buildEvent returned null
>>     No event class registered for event type 'varset', attributes:
>>     {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>>     value=ssrc=471130521;themssrc=416812302;lp=0;rxjitter=0.000064;rxcount=233;txjitter=0.000000;txcount=241;rlp=0;rtt=0.000000,
>>     variable=RTPAUDIOQOS, channel=SIP/1001-00000074}buildEvent
>>     returned null
>>     No event class registered for event type 'varset', attributes:
>>     {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>>     value=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;,
>>     variable=RTPAUDIOQOSJITTER, channel=SIP/1001-00000074}buildEvent
>>     returned nul
>>     lNo event class registered for event type 'varset', attributes:
>>     {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>>     value=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;,
>>     variable=RTPAUDIOQOSLOSS, channel=SIP/1001-00000074}buildEvent
>>     returned null
>>     No event class registered for event type 'varset', attributes:
>>     {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>>     value=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;,
>>     variable=RTPAUDIOQOSRTT, channel=SIP/1001-00000074}buildEvent
>>     returned null
>>     No event class registered for event type 'hanguprequest',
>>     attributes: {uniqueid=1368617231.116, event=HangupRequest,
>>     privilege=call,all, channel=SIP/1001-00000074}buildEvent returned
>>     nullDispatching event:
>>
>>     org.asteriskjava.manager.event.QueueCallerAbandonEvent[dateReceived=Wed
>>     May 15 13:27:18 CEST
>>     2013,privilege='agent,all',timestamp='null',position='1',queue='support',uniqueid='1368617231.116',originalposition='1',holdtime='5',count='null',channel='null',systemHashcode=28050664]
>>
>>     No event class registered for event type 'varset', attributes:
>>     {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
>>     value=1, variable=QUEUEPOSITION,
>>     channel=SIP/1001-00000074}buildEvent returned nullNo event class
>>     registered for event type 'softhanguprequest', attributes:
>>     {cause=16, uniqueid=1368617231.116, event=SoftHangupRequest,
>>     privilege=call,all, channel=SIP/1001-00000074}buildEvent returned
>>     nullUnable to set property 'connectedlinename' to '<unknown>' on
>>     org.asteriskjava.manager.event.HangupEvent: no setterUnable to
>>     set property 'connectedlinenum' to '<unknown>' on
>>     org.asteriskjava.manager.event.HangupEvent: no setterDispatching
>>     event:
>>
>>     org.asteriskjava.manager.event.HangupEvent[dateReceived=Wed May
>>     15 13:27:18 CEST
>>     2013,privilege='call,all',causetxt='Unknown',callerid='1001',cause='0',state='null',calleridname='MatthieuHIEl',timestamp='null',uniqueid='1368617231.116',calleridnum='1001',channel='SIP/1001-00000074',systemHashcode=2548785]
>>
>>     And I hava theese error too:
>>     Unable to set property 'position' to '1' on
>>     org.asteriskjava.manager.event.LeaveEvent: no setter
>>     Dispatching event:
>>     org.asteriskjava.manager.event.LeaveEvent[dateReceived=Wed May 15
>>     13:27:18 CEST
>>     2013,privilege='call,all',timestamp='null',queue='support',uniqueid='1368617231.116',count='0',channel='SIP/1001-00000074',systemHashcode=7754385]
>>
>>     It is always complainig about not having a setter or there is no
>>     class registered for an event. Is there a way to fix the issue?
>>     Do not hesitate if you want more information!
>>     Could anyone please help me on this.
>>
>>     Thanks,
>>     Matthieu
>>
>>
>>
>>     ------------------------------------------------------------------------------
>>     AlienVault Unified Security Management (USM) platform delivers complete
>>     security visibility with the essential security capabilities. Easily and
>>     efficiently configure, manage, and operate all of your security controls
>>     from a single console and one unified framework. Download a free trial.
>>     http://p.sf.net/sfu/alienvault_d2d
>>
>>
>>     _______________________________________________
>>     Asterisk-java-users mailing list
>>     Ast...@li...  <mailto:Ast...@li...>
>>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>     ------------------------------------------------------------------------------
>     AlienVault Unified Security Management (USM) platform delivers
>     complete
>     security visibility with the essential security capabilities.
>     Easily and
>     efficiently configure, manage, and operate all of your security
>     controls
>     from a single console and one unified framework. Download a free
>     trial.
>     http://p.sf.net/sfu/alienvault_d2d
>     _______________________________________________
>     Asterisk-java-users mailing list
>     Ast...@li...
>     <mailto:Ast...@li...>
>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.
> http://p.sf.net/sfu/alienvault_d2d
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] No event class registered or Unable to set property"......" to "......" on org.asterisk.java ..... no setter

[matthieu h. <mat...@gm...>]

Thank you for your response.

I try with asterisk java 1.0.0-m1. It is working almost perfectly. I have
still some errors like these:

Unable to set property 'context' to 'default' on
org.asteriskjava.manager.event.NewChannelEvent: no setter. Please report at
http://jira.reucon.org/browse/AJDispatching event:
org.asteriskjava.manager.event.NewChannelEvent[dateReceived=Thu May 16
13:54:59 CEST
2013,privilege='call,all',callerid='null',state='Down',channelstate='0',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',accountcode='null',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Down',systemHashcode=23910357]

Unable to set property 'connectedlinename' to 'Matthieu Hiel' on
org.asteriskjava.manager.event.NewStateEvent: no setter. Please report at
http://jira.reucon.org/browse/AJUnable to set property 'connectedlinenum'
to '1001' on org.asteriskjava.manager.event.NewStateEvent: no setter.
Please report at http://jira.reucon.org/browse/AJDispatching event:
org.asteriskjava.manager.event.NewStateEvent[dateReceived=Thu May 16
13:54:59 CEST
2013,privilege='call,all',callerid='null',state='Ringing',channelstate='5',calleridname='Jean-Michel',timestamp='null',uniqueid='1368705299.36',server='null',calleridnum='null',channel='SIP/1002-00000024',channelstatedesc='Ringing',systemHashcode=2917593]

Unable to set property 'stateinterface' to 'SIP/1002' on
org.asteriskjava.manager.event.QueueMemberStatusEvent: no setter. Please
report at http://jira.reucon.org/browse/AJDispatching event:
org.asteriskjava.manager.event.QueueMemberStatusEvent[dateReceived=Thu May
16 13:54:59 CEST
2013,privilege='agent,all',queue='support',membership='dynamic',location='SIP/1002',status='6',paused='false',timestamp='null',lastcall='0',actionid='null',internalactionid='null',name='SIP/1002',server='null',membername='SIP/1002',penalty='0',callstaken='0',systemHashcode=28405330]

I will try to deal with it.

Thanks again,
Regards,
Matthieu



2013/5/16 Yves A. <yv...@gm...>

>  hi,
>
> first of all I would recommend using the most recent version of
> asterisk-java.
> surely aj 1.x is older than asterisk 11.x but I am using it too and do not
> face these errors... give it a try!
>
> regards,
> yves
>
> Am 15.05.2013 13:59, schrieb matthieu hiel:
>
> Hi Guys,
> I am using Asterisk 11.2.1 with Asterisk-java-0.3.1.jar
>
> I manage to connect my Asterisk with my java program.
> I keep getting these errors all the time and things seem to work, but it
> is hard to follow my logging because every action that is done throws them :
>
> No event class registered for event type 'musiconhold', attributes:
> {state=Start, class=default, uniqueid=1368617231.116, event=MusicOnHold,
> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null
>
> No event class registered for event type 'musiconhold', attributes:
> {state=Stop, uniqueid=1368617231.116, event=MusicOnHold,
> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null
>
> No event class registered for event type 'rtcpreceived', attributes:
> {highestsequence=18068, dlsr=0.0000(sec), fractionlost=0, iajitter=0,
> sequencenumbercycles=0, from=10.140.6.214:16289, senderssrc=0,
> lastsr=0.0000000000, packetslost=0, event=RTCPReceived,
> privilege=reporting,all, pt=200(Sender Report),
> receptionreports=1}buildEvent returned null
>
> No event class registered for event type 'varset', attributes:
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
> value=ssrc=471130521;themssrc=416812302;lp=0;rxjitter=0.000064;rxcount=233;txjitter=0.000000;txcount=241;rlp=0;rtt=0.000000,
> variable=RTPAUDIOQOS, channel=SIP/1001-00000074}buildEvent returned null
> No event class registered for event type 'varset', attributes:
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
> value=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;,
> variable=RTPAUDIOQOSJITTER, channel=SIP/1001-00000074}buildEvent returned
> nul
>
> lNo event class registered for event type 'varset', attributes:
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
> value=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;,
> variable=RTPAUDIOQOSLOSS, channel=SIP/1001-00000074}buildEvent returned null
>
> No event class registered for event type 'varset', attributes:
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
> value=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;,
> variable=RTPAUDIOQOSRTT, channel=SIP/1001-00000074}buildEvent returned null
>
> No event class registered for event type 'hanguprequest', attributes:
> {uniqueid=1368617231.116, event=HangupRequest, privilege=call,all,
> channel=SIP/1001-00000074}buildEvent returned nullDispatching event:
>
> org.asteriskjava.manager.event.QueueCallerAbandonEvent[dateReceived=Wed
> May 15 13:27:18 CEST
> 2013,privilege='agent,all',timestamp='null',position='1',queue='support',uniqueid='1368617231.116',originalposition='1',holdtime='5',count='null',channel='null',systemHashcode=28050664]
>
> No event class registered for event type 'varset', attributes:
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, value=1,
> variable=QUEUEPOSITION, channel=SIP/1001-00000074}buildEvent returned
> nullNo event class registered for event type 'softhanguprequest',
> attributes: {cause=16, uniqueid=1368617231.116, event=SoftHangupRequest,
> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned
> nullUnable to set property 'connectedlinename' to '<unknown>' on
> org.asteriskjava.manager.event.HangupEvent: no setterUnable to set property
> 'connectedlinenum' to '<unknown>' on
> org.asteriskjava.manager.event.HangupEvent: no setterDispatching event:
>
> org.asteriskjava.manager.event.HangupEvent[dateReceived=Wed May 15
> 13:27:18 CEST
> 2013,privilege='call,all',causetxt='Unknown',callerid='1001',cause='0',state='null',calleridname='Matthieu
> HIEl',timestamp='null',uniqueid='1368617231.116',calleridnum='1001',channel='SIP/1001-00000074',systemHashcode=2548785]
>
>
> And I hava theese error too:
>
> Unable to set property 'position' to '1' on
> org.asteriskjava.manager.event.LeaveEvent: no setter
> Dispatching event:
> org.asteriskjava.manager.event.LeaveEvent[dateReceived=Wed May 15 13:27:18
> CEST
> 2013,privilege='call,all',timestamp='null',queue='support',uniqueid='1368617231.116',count='0',channel='SIP/1001-00000074',systemHashcode=7754385]
>
> It is always complainig about not having a setter or there is no class
> registered for an event. Is there a way to fix the issue?
> Do not hesitate if you want more information!
>
> Could anyone please help me on this.
>
> Thanks,
> Matthieu
>
>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.http://p.sf.net/sfu/alienvault_d2d
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.
> http://p.sf.net/sfu/alienvault_d2d
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] No event class registered or Unable to set property"......" to "......" on org.asterisk.java ..... no setter

[Yves A. <yv...@gm...>]

hi,

first of all I would recommend using the most recent version of 
asterisk-java.
surely aj 1.x is older than asterisk 11.x but I am using it too and do 
not face these errors... give it a try!

regards,
yves

Am 15.05.2013 13:59, schrieb matthieu hiel:
> Hi Guys,
> I am using Asterisk 11.2.1 with Asterisk-java-0.3.1.jar
> I manage to connect my Asterisk with my java program.
> I keep getting these errors all the time and things seem to work, but 
> it is hard to follow my logging because every action that is done 
> throws them :
> No event class registered for event type 'musiconhold', attributes: 
> {state=Start, class=default, uniqueid=1368617231.116, 
> event=MusicOnHold, privilege=call,all, 
> channel=SIP/1001-00000074}buildEvent returned null
> No event class registered for event type 'musiconhold', attributes: 
> {state=Stop, uniqueid=1368617231.116, event=MusicOnHold, 
> privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null
> No event class registered for event type 'rtcpreceived', attributes: 
> {highestsequence=18068, dlsr=0.0000(sec), fractionlost=0, iajitter=0, 
> sequencenumbercycles=0, from=10.140.6.214:16289 
> <http://10.140.6.214:16289>, senderssrc=0, lastsr=0.0000000000, 
> packetslost=0, event=RTCPReceived, privilege=reporting,all, 
> pt=200(Sender Report), receptionreports=1}buildEvent returned null
> No event class registered for event type 'varset', attributes: 
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, 
> value=ssrc=471130521;themssrc=416812302;lp=0;rxjitter=0.000064;rxcount=233;txjitter=0.000000;txcount=241;rlp=0;rtt=0.000000, 
> variable=RTPAUDIOQOS, channel=SIP/1001-00000074}buildEvent returned null
> No event class registered for event type 'varset', attributes: 
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, 
> value=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;, 
> variable=RTPAUDIOQOSJITTER, channel=SIP/1001-00000074}buildEvent 
> returned nul
> lNo event class registered for event type 'varset', attributes: 
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, 
> value=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;, 
> variable=RTPAUDIOQOSLOSS, channel=SIP/1001-00000074}buildEvent 
> returned null
> No event class registered for event type 'varset', attributes: 
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, 
> value=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;, 
> variable=RTPAUDIOQOSRTT, channel=SIP/1001-00000074}buildEvent returned 
> null
> No event class registered for event type 'hanguprequest', attributes: 
> {uniqueid=1368617231.116, event=HangupRequest, privilege=call,all, 
> channel=SIP/1001-00000074}buildEvent returned nullDispatching event:
>
> org.asteriskjava.manager.event.QueueCallerAbandonEvent[dateReceived=Wed May 
> 15 13:27:18 CEST 
> 2013,privilege='agent,all',timestamp='null',position='1',queue='support',uniqueid='1368617231.116',originalposition='1',holdtime='5',count='null',channel='null',systemHashcode=28050664]
>
> No event class registered for event type 'varset', attributes: 
> {uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, 
> value=1, variable=QUEUEPOSITION, channel=SIP/1001-00000074}buildEvent 
> returned nullNo event class registered for event type 
> 'softhanguprequest', attributes: {cause=16, uniqueid=1368617231.116, 
> event=SoftHangupRequest, privilege=call,all, 
> channel=SIP/1001-00000074}buildEvent returned nullUnable to set 
> property 'connectedlinename' to '<unknown>' on 
> org.asteriskjava.manager.event.HangupEvent: no setterUnable to set 
> property 'connectedlinenum' to '<unknown>' on 
> org.asteriskjava.manager.event.HangupEvent: no setterDispatching event:
>
> org.asteriskjava.manager.event.HangupEvent[dateReceived=Wed May 15 
> 13:27:18 CEST 
> 2013,privilege='call,all',causetxt='Unknown',callerid='1001',cause='0',state='null',calleridname='Matthieu 
> HIEl',timestamp='null',uniqueid='1368617231.116',calleridnum='1001',channel='SIP/1001-00000074',systemHashcode=2548785]
>
> And I hava theese error too:
> Unable to set property 'position' to '1' on 
> org.asteriskjava.manager.event.LeaveEvent: no setter
> Dispatching event:
> org.asteriskjava.manager.event.LeaveEvent[dateReceived=Wed May 15 
> 13:27:18 CEST 
> 2013,privilege='call,all',timestamp='null',queue='support',uniqueid='1368617231.116',count='0',channel='SIP/1001-00000074',systemHashcode=7754385]
>
> It is always complainig about not having a setter or there is no class 
> registered for an event. Is there a way to fix the issue?
> Do not hesitate if you want more information!
> Could anyone please help me on this.
>
> Thanks,
> Matthieu
>
>
>
> ------------------------------------------------------------------------------
> AlienVault Unified Security Management (USM) platform delivers complete
> security visibility with the essential security capabilities. Easily and
> efficiently configure, manage, and operate all of your security controls
> from a single console and one unified framework. Download a free trial.
> http://p.sf.net/sfu/alienvault_d2d
>
>
> _______________________________________________
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> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] No event class registered or Unable to set property"......" to "......" on org.asterisk.java ..... no setter

[matthieu h. <mat...@gm...>]

Hi Guys,
I am using Asterisk 11.2.1 with Asterisk-java-0.3.1.jar

I manage to connect my Asterisk with my java program.
I keep getting these errors all the time and things seem to work, but it is
hard to follow my logging because every action that is done throws them :

No event class registered for event type 'musiconhold', attributes:
{state=Start, class=default, uniqueid=1368617231.116, event=MusicOnHold,
privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null

No event class registered for event type 'musiconhold', attributes:
{state=Stop, uniqueid=1368617231.116, event=MusicOnHold,
privilege=call,all, channel=SIP/1001-00000074}buildEvent returned null

No event class registered for event type 'rtcpreceived', attributes:
{highestsequence=18068, dlsr=0.0000(sec), fractionlost=0, iajitter=0,
sequencenumbercycles=0, from=10.140.6.214:16289, senderssrc=0,
lastsr=0.0000000000, packetslost=0, event=RTCPReceived,
privilege=reporting,all, pt=200(Sender Report),
receptionreports=1}buildEvent returned null

No event class registered for event type 'varset', attributes:
{uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
value=ssrc=471130521;themssrc=416812302;lp=0;rxjitter=0.000064;rxcount=233;txjitter=0.000000;txcount=241;rlp=0;rtt=0.000000,
variable=RTPAUDIOQOS, channel=SIP/1001-00000074}buildEvent returned null
No event class registered for event type 'varset', attributes:
{uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
value=minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;,
variable=RTPAUDIOQOSJITTER, channel=SIP/1001-00000074}buildEvent returned
nul

lNo event class registered for event type 'varset', attributes:
{uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
value=minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;,
variable=RTPAUDIOQOSLOSS, channel=SIP/1001-00000074}buildEvent returned null

No event class registered for event type 'varset', attributes:
{uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all,
value=minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;,
variable=RTPAUDIOQOSRTT, channel=SIP/1001-00000074}buildEvent returned null

No event class registered for event type 'hanguprequest', attributes:
{uniqueid=1368617231.116, event=HangupRequest, privilege=call,all,
channel=SIP/1001-00000074}buildEvent returned nullDispatching event:

org.asteriskjava.manager.event.QueueCallerAbandonEvent[dateReceived=Wed May
15 13:27:18 CEST
2013,privilege='agent,all',timestamp='null',position='1',queue='support',uniqueid='1368617231.116',originalposition='1',holdtime='5',count='null',channel='null',systemHashcode=28050664]

No event class registered for event type 'varset', attributes:
{uniqueid=1368617231.116, event=VarSet, privilege=dialplan,all, value=1,
variable=QUEUEPOSITION, channel=SIP/1001-00000074}buildEvent returned
nullNo event class registered for event type 'softhanguprequest',
attributes: {cause=16, uniqueid=1368617231.116, event=SoftHangupRequest,
privilege=call,all, channel=SIP/1001-00000074}buildEvent returned
nullUnable to set property 'connectedlinename' to '<unknown>' on
org.asteriskjava.manager.event.HangupEvent: no setterUnable to set property
'connectedlinenum' to '<unknown>' on
org.asteriskjava.manager.event.HangupEvent: no setterDispatching event:

org.asteriskjava.manager.event.HangupEvent[dateReceived=Wed May 15 13:27:18
CEST
2013,privilege='call,all',causetxt='Unknown',callerid='1001',cause='0',state='null',calleridname='Matthieu
HIEl',timestamp='null',uniqueid='1368617231.116',calleridnum='1001',channel='SIP/1001-00000074',systemHashcode=2548785]


And I hava theese error too:

Unable to set property 'position' to '1' on
org.asteriskjava.manager.event.LeaveEvent: no setter
Dispatching event:
org.asteriskjava.manager.event.LeaveEvent[dateReceived=Wed May 15 13:27:18
CEST
2013,privilege='call,all',timestamp='null',queue='support',uniqueid='1368617231.116',count='0',channel='SIP/1001-00000074',systemHashcode=7754385]

It is always complainig about not having a setter or there is no class
registered for an event. Is there a way to fix the issue?
Do not hesitate if you want more information!

Could anyone please help me on this.

Thanks,
Matthieu

Re: [Asterisk-java-users] asterisk 11

[Максим Б. <bui...@gm...>]

Thanks for the reply.
"SIP/1002-00000000" this is only a sample.
I look at the channel name in the asterisk log after creating a connection.

fragment of the log:
    -- Executing [2000@webrtc:4] MixMonitor("SIP/1002-00000000",
"/var/lib/asterisk/sounds/1367661071.0_20130504-135111.wav") in new stack

You are right, the name of the channel is changing, but I use the actual
name.

2013/5/3 Jhoan Orozco <jho...@gm...>

> No esta pasando nada, porque el canal que pasas como argumento,
> posiblemente ya no existe, debes buscar la forma de pasarle el canal que en
> ese momento esta activo, los canales entre llamadas siempre cambien de
> nombre.
>
> Ejemplo: SIP: 1000
>
> 1° llamada: SIP/1000-7654567d89d64d
> 2° Llamada: SIP/1000-iuytr5tyuio33i4u
> 3° Llamada: SIP/1000-09876f7890d098
>
> Conclusion: el canal nunca se va a llamar igual
>
>
> There's nothing going on, because the channel you pass as an argument,
> possibly longer exists, you must find a way to pass the channel is active
> at that time, the channels between calls always change their name.
>
> Example: SIP: 1000
>
> 1 call: SIP/1000-7654567d89d64d
> 2nd Call: SIP/1000-iuytr5tyuio33i4u
> 3rd Call: SIP/1000-09876f7890d098
>
> Conclusion: the channel will never be the same call
>
>
>
>
>
> 2013/5/3 Максим Буйлин <bui...@gm...>
>
>> I'm trying to send the StopMonitorAction using asterisk-java-0.3.1.jar.
>> But nothing happens.
>> code example:
>> dmc.sendAction(new StopMonitorAction("SIP/1002-00000000"));
>>
>> Asterisk 11.3.0
>> Ubuntu 12.04.2 LTS (GNU/Linux 3.2.0-23-generic x86_64)
>>
>>
>> ------------------------------------------------------------------------------
>> Get 100% visibility into Java/.NET code with AppDynamics Lite
>> It's a free troubleshooting tool designed for production
>> Get down to code-level detail for bottlenecks, with <2% overhead.
>> Download for free and get started troubleshooting in minutes.
>> http://p.sf.net/sfu/appdyn_d2d_ap2
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] asterisk 11

[Jhoan O. <jho...@gm...>]

No esta pasando nada, porque el canal que pasas como argumento,
posiblemente ya no existe, debes buscar la forma de pasarle el canal que en
ese momento esta activo, los canales entre llamadas siempre cambien de
nombre.

Ejemplo: SIP: 1000

1° llamada: SIP/1000-7654567d89d64d
2° Llamada: SIP/1000-iuytr5tyuio33i4u
3° Llamada: SIP/1000-09876f7890d098

Conclusion: el canal nunca se va a llamar igual


There's nothing going on, because the channel you pass as an argument,
possibly longer exists, you must find a way to pass the channel is active
at that time, the channels between calls always change their name.

Example: SIP: 1000

1 call: SIP/1000-7654567d89d64d
2nd Call: SIP/1000-iuytr5tyuio33i4u
3rd Call: SIP/1000-09876f7890d098

Conclusion: the channel will never be the same call





2013/5/3 Максим Буйлин <bui...@gm...>

> I'm trying to send the StopMonitorAction using asterisk-java-0.3.1.jar.
> But nothing happens.
> code example:
> dmc.sendAction(new StopMonitorAction("SIP/1002-00000000"));
>
> Asterisk 11.3.0
> Ubuntu 12.04.2 LTS (GNU/Linux 3.2.0-23-generic x86_64)
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

[Asterisk-java-users] asterisk 11

[Максим Б. <bui...@gm...>]

I'm trying to send the StopMonitorAction using asterisk-java-0.3.1.jar.
But nothing happens.
code example:
dmc.sendAction(new StopMonitorAction("SIP/1002-00000000"));

Asterisk 11.3.0
Ubuntu 12.04.2 LTS (GNU/Linux 3.2.0-23-generic x86_64)

Re: [Asterisk-java-users] (no subject)

[Jhoan O. <jho...@gm...>]

Eso lo puedes hacer también por medio de AMI, creando una acción Command,
y envías el comando: "sip show channelstats"


2013/5/1 Jimmy Chang <cha...@gm...>

>  I'm using ESL.
> When receive CHANNEL_CREATE, start counting.
> When receive CHANNEL_HANGUP_COMPLETE, finish counting.
>
> Between these two events, you can get the duration yourself.
>
> Regards,
> Jimmy
>
>
> 於 2013/4/2 上午 02:41, Diego Guimarães 提到:
>
> how can I get the value of the cdr duration in the channel? because
> CallDetailRecord is only created after the call is finished.
>
>  Diego Augusto Costa Guimarães
> Gerente TI - Praia Clube
> *dCAP (Digium Certified Asterisk Professional)*
> Pabx: (34) 3256-3116
> (Mobile) celular: (34) - 9268-9999
> Chat Google Talk: die...@gm... Skype: diegoacguimares MSN:
> die...@gm...
>
>
>
> ------------------------------------------------------------------------------
> Own the Future-Intel&reg; Level Up Game Demo Contest 2013
> Rise to greatness in Intel's independent game demo contest.
> Compete for recognition, cash, and the chance to get your game
> on Steam. $5K grand prize plus 10 genre and skill prizes.
> Submit your demo by 6/6/13. http://p.sf.net/sfu/intel_levelupd2d
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Introducing AppDynamics Lite, a free troubleshooting tool for Java/.NET
> Get 100% visibility into your production application - at no cost.
> Code-level diagnostics for performance bottlenecks with <2% overhead
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap1
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] (no subject)

[Jimmy C. <cha...@gm...>]

My bad, it's wrong.
That's about FreeSWITCH.

? 2013/4/2 ?? 02:41, Diego Guimarães ??:
> how can I get the value of the cdr duration in the channel? because 
> CallDetailRecord is only created after the call is finished.
>
> Diego Augusto Costa Guimarães
> Gerente TI - Praia Clube
> /dCAP (Digium Certified Asterisk Professional)/
> Pabx: (34) 3256-3116
> (Mobile) celular: (34) - 9268-9999
> Chat Google Talk: die...@gm... <mailto:die...@gm...> 
> Skype: diegoacguimares MSN: die...@gm... 
> <mailto:die...@gm...>
>
>
>
> ------------------------------------------------------------------------------
> Own the Future-Intel&reg; Level Up Game Demo Contest 2013
> Rise to greatness in Intel's independent game demo contest.
> Compete for recognition, cash, and the chance to get your game
> on Steam. $5K grand prize plus 10 genre and skill prizes.
> Submit your demo by 6/6/13. http://p.sf.net/sfu/intel_levelupd2d
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] (no subject)

[Jimmy C. <cha...@gm...>]

I'm using ESL.
When receive CHANNEL_CREATE, start counting.
When receive CHANNEL_HANGUP_COMPLETE, finish counting.

Between these two events, you can get the duration yourself.

Regards,
Jimmy


? 2013/4/2 ?? 02:41, Diego Guimarães ??:
> how can I get the value of the cdr duration in the channel? because 
> CallDetailRecord is only created after the call is finished.
>
> Diego Augusto Costa Guimarães
> Gerente TI - Praia Clube
> /dCAP (Digium Certified Asterisk Professional)/
> Pabx: (34) 3256-3116
> (Mobile) celular: (34) - 9268-9999
> Chat Google Talk: die...@gm... <mailto:die...@gm...> 
> Skype: diegoacguimares MSN: die...@gm... 
> <mailto:die...@gm...>
>
>
>
> ------------------------------------------------------------------------------
> Own the Future-Intel&reg; Level Up Game Demo Contest 2013
> Rise to greatness in Intel's independent game demo contest.
> Compete for recognition, cash, and the chance to get your game
> on Steam. $5K grand prize plus 10 genre and skill prizes.
> Submit your demo by 6/6/13. http://p.sf.net/sfu/intel_levelupd2d
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] help

[vinethan j. <vin...@gm...>]

how to get channel specific dtmf digits.
my programs gets all dtmf digits pressed from multiple channels
and one more thing when i press a dtmf digits only once,it is getting
stored 2 times .


need help.

thanks in advance...!


my code

import java.io.IOException;
import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.ManagerEventListener;
import org.asteriskjava.manager.TimeoutException;
import org.asteriskjava.manager.action.ManagerAction;
import org.asteriskjava.manager.event.ManagerEvent;
import javax.net.ssl.ManagerFactoryParameters;

public class El implements ManagerEventListener
{
    private ManagerConnection managerConnection;
    String s3 = "";

    public El() throws IOException,
AuthenticationFailedException,TimeoutException,InterruptedException
    {
        ManagerConnectionFactory factory = new
ManagerConnectionFactory("x.x.x.x", "asterisk", "vinethan");
        managerConnection = factory.createManagerConnection();


    }
    public void run() throws IOException,
AuthenticationFailedException,TimeoutException,InterruptedException
    {
        managerConnection.login();
        managerConnection.addEventListener(this);

         Thread.sleep(2099999999);

         managerConnection.logoff();

    }

    public void onManagerEvent(ManagerEvent e1)
    {
         String s2 = e1.toString();
         s3=s3+s2.charAt(154);
         System.out.println(s2.charAt(154));
         System.out.println(s3);

    }



    public static void main(String[] args) throws Exception
    {

        El helloEvents = new El();
        helloEvents.run();
    }
}

Re: [Asterisk-java-users] Integrating asterisk-java in Apache Tomcat

[Oscar A. <osc...@gm...>]

Where can I find these docs and this exemples?


2013/4/16 Yves A. <yv...@gm...>

>  if you want to "react" on a servlet request and fire some action on
> asterisk, you just have to open a
> manager connection (from your servlet) to your asterisk and execute the
> appropriate action... its as easy as 1-2-3 and
> really well documented... have you read the docs and tried the examples?
> "action" can be really anything you want... and actions can be performed
> synchronously and asynchronously
>
> yves
>
> Am 16.04.2013 08:45, schrieb Oscar Alvarez:
>
> I need externalize some Asterisk actions, for exemple, I need that
> asterisk load/unload a module when recives a instrucciton from a Tomcat.
>
>
> 2013/4/11 Yves A. <yv...@gm...>
>
>>  hi,
>>
>> thats the wrong approach... the java-agi scripts are handled by the
>> agi-server, not by an app server like tomcat...
>> there is a tutorial on how to use (start / stop / mapping) the agi server
>> on the page where your link points to.
>>
>> yves
>>
>> Am 11.04.2013 10:49, schrieb Oscar Alvarez:
>>
>> Yes,
>>
>>  But I need info to create the servlet.
>>
>>  In dialplan, I put anything like this:
>>
>>  exten => 1001,1,AGI(agi://ip_Tomcat/app1)
>>
>>
>>  I like that app1 answer tha call and hangup.
>>
>>
>>  import net.sf.asterisk.fastagi.AGIChannel;
>> import net.sf.asterisk.fastagi.AGIException;
>> import net.sf.asterisk.fastagi.AGIRequest;
>> import net.sf.asterisk.fastagi.AbstractAGIScript;
>>
>> public class HelloAGIScript extends AbstractAGIScript
>> {
>>     public void service(AGIRequest request, AGIChannel channel)
>>             throws AGIException
>>     {
>>         // Answer the channel...
>>         answer(channel);
>>
>>
>> 		
>>         // ...and hangup.
>>         hangup(channel);
>>     }
>> }
>>
>> What is the mode tu mapping this??
>>
>> I wold find a tutorial like this <http://www.asterisk-java.org/0.2/tutorial.html> but running in a Tomcat.
>>
>> Thxx
>>
>>
>>
>> 2013/4/11 Yves A. <yv...@gm...>
>>
>>>  hi,
>>>
>>> you just have to put the asterisk-java jar file in the library path of
>>> your tomcat (tomcat/lib) or your web-app (WEB-INF/lib).
>>>
>>> yves
>>>
>>> Am 11.04.2013 09:29, schrieb Oscar Alvarez:
>>>
>>>  Hi,
>>>
>>>  Where can I find some info for this. I need deploy the asterisk-java
>>> in a Tomcat servlet.
>>> Do you know any tutorial?
>>>
>>>  Thxxx
>>>
>>>  --
>>> Oscar Alvarez
>>>
>>>
>>>  ------------------------------------------------------------------------------
>>> Precog is a next-generation analytics platform capable of advanced
>>> analytics on semi-structured data. The platform includes APIs for building
>>> apps and a phenomenal toolset for data science. Developers can use
>>> our toolset for easy data analysis & visualization. Get a free account!http://www2.precog.com/precogplatform/slashdotnewsletter
>>>
>>>
>>>
>>> _______________________________________________
>>> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Precog is a next-generation analytics platform capable of advanced
>>> analytics on semi-structured data. The platform includes APIs for
>>> building
>>> apps and a phenomenal toolset for data science. Developers can use
>>> our toolset for easy data analysis & visualization. Get a free account!
>>> http://www2.precog.com/precogplatform/slashdotnewsletter
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>>>
>>
>>
>>  --
>> Oscar Alvarez
>>
>>
>> ------------------------------------------------------------------------------
>> Precog is a next-generation analytics platform capable of advanced
>> analytics on semi-structured data. The platform includes APIs for building
>> apps and a phenomenal toolset for data science. Developers can use
>> our toolset for easy data analysis & visualization. Get a free account!http://www2.precog.com/precogplatform/slashdotnewsletter
>>
>>
>>
>> _______________________________________________
>> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Precog is a next-generation analytics platform capable of advanced
>> analytics on semi-structured data. The platform includes APIs for building
>> apps and a phenomenal toolset for data science. Developers can use
>> our toolset for easy data analysis & visualization. Get a free account!
>> http://www2.precog.com/precogplatform/slashdotnewsletter
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
>  --
> Oscar Alvarez
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!http://www2.precog.com/precogplatform/slashdotnewsletter
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!
> http://www2.precog.com/precogplatform/slashdotnewsletter
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>


-- 
Oscar Alvarez

Re: [Asterisk-java-users] Integrating asterisk-java in Apache Tomcat

[Yves A. <yv...@gm...>]

if you want to "react" on a servlet request and fire some action on 
asterisk, you just have to open a
manager connection (from your servlet) to your asterisk and execute the 
appropriate action... its as easy as 1-2-3 and
really well documented... have you read the docs and tried the examples?
"action" can be really anything you want... and actions can be 
performed  synchronously and asynchronously

yves

Am 16.04.2013 08:45, schrieb Oscar Alvarez:
> I need externalize some Asterisk actions, for exemple, I need that 
> asterisk load/unload a module when recives a instrucciton from a Tomcat.
>
>
> 2013/4/11 Yves A. <yv...@gm... <mailto:yv...@gm...>>
>
>     hi,
>
>     thats the wrong approach... the java-agi scripts are handled by
>     the agi-server, not by an app server like tomcat...
>     there is a tutorial on how to use (start / stop / mapping) the agi
>     server on the page where your link points to.
>
>     yves
>
>     Am 11.04.2013 10:49, schrieb Oscar Alvarez:
>>     Yes,
>>
>>     But I need info to create the servlet.
>>
>>     In dialplan, I put anything like this:
>>
>>     exten => 1001,1,AGI(agi://ip_Tomcat/app1)
>>
>>
>>     I like that app1 answer tha call and hangup.
>>
>>
>>     import net.sf.asterisk.fastagi.AGIChannel;
>>     import net.sf.asterisk.fastagi.AGIException;
>>     import net.sf.asterisk.fastagi.AGIRequest;
>>     import net.sf.asterisk.fastagi.AbstractAGIScript;
>>
>>     public class HelloAGIScript extends AbstractAGIScript
>>     {
>>          public void service(AGIRequest request, AGIChannel channel)
>>                  throws AGIException
>>          {
>>              // Answer the channel...
>>              answer(channel);
>>
>>     		
>>              // ...and hangup.
>>              hangup(channel);
>>          }
>>     }
>>     What is the mode tu mapping this??
>>     I wold find a tutorial likethis  <http://www.asterisk-java.org/0.2/tutorial.html>  but running in a Tomcat.
>>     Thxx
>>
>>
>>     2013/4/11 Yves A. <yv...@gm... <mailto:yv...@gm...>>
>>
>>         hi,
>>
>>         you just have to put the asterisk-java jar file in the
>>         library path of your tomcat (tomcat/lib) or your web-app
>>         (WEB-INF/lib).
>>
>>         yves
>>
>>         Am 11.04.2013 09:29, schrieb Oscar Alvarez:
>>>         Hi,
>>>
>>>         Where can I find some info for this. I need deploy the
>>>         asterisk-java in a Tomcat servlet.
>>>         Do you know any tutorial?
>>>
>>>         Thxxx
>>>
>>>         -- 
>>>         Oscar Alvarez
>>>
>>>
>>>         ------------------------------------------------------------------------------
>>>         Precog is a next-generation analytics platform capable of advanced
>>>         analytics on semi-structured data. The platform includes APIs for building
>>>         apps and a phenomenal toolset for data science. Developers can use
>>>         our toolset for easy data analysis & visualization. Get a free account!
>>>         http://www2.precog.com/precogplatform/slashdotnewsletter
>>>
>>>
>>>         _______________________________________________
>>>         Asterisk-java-users mailing list
>>>         Ast...@li...  <mailto:Ast...@li...>
>>>         https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>         ------------------------------------------------------------------------------
>>         Precog is a next-generation analytics platform capable of
>>         advanced
>>         analytics on semi-structured data. The platform includes APIs
>>         for building
>>         apps and a phenomenal toolset for data science. Developers
>>         can use
>>         our toolset for easy data analysis & visualization. Get a
>>         free account!
>>         http://www2.precog.com/precogplatform/slashdotnewsletter
>>         _______________________________________________
>>         Asterisk-java-users mailing list
>>         Ast...@li...
>>         <mailto:Ast...@li...>
>>         https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>>     -- 
>>     Oscar Alvarez
>>
>>
>>     ------------------------------------------------------------------------------
>>     Precog is a next-generation analytics platform capable of advanced
>>     analytics on semi-structured data. The platform includes APIs for building
>>     apps and a phenomenal toolset for data science. Developers can use
>>     our toolset for easy data analysis & visualization. Get a free account!
>>     http://www2.precog.com/precogplatform/slashdotnewsletter
>>
>>
>>     _______________________________________________
>>     Asterisk-java-users mailing list
>>     Ast...@li...  <mailto:Ast...@li...>
>>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>     ------------------------------------------------------------------------------
>     Precog is a next-generation analytics platform capable of advanced
>     analytics on semi-structured data. The platform includes APIs for
>     building
>     apps and a phenomenal toolset for data science. Developers can use
>     our toolset for easy data analysis & visualization. Get a free
>     account!
>     http://www2.precog.com/precogplatform/slashdotnewsletter
>     _______________________________________________
>     Asterisk-java-users mailing list
>     Ast...@li...
>     <mailto:Ast...@li...>
>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> -- 
> Oscar Alvarez
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!
> http://www2.precog.com/precogplatform/slashdotnewsletter
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Integrating asterisk-java in Apache Tomcat

[Oscar A. <osc...@gm...>]

I need externalize some Asterisk actions, for exemple, I need that asterisk
load/unload a module when recives a instrucciton from a Tomcat.


2013/4/11 Yves A. <yv...@gm...>

>  hi,
>
> thats the wrong approach... the java-agi scripts are handled by the
> agi-server, not by an app server like tomcat...
> there is a tutorial on how to use (start / stop / mapping) the agi server
> on the page where your link points to.
>
> yves
>
> Am 11.04.2013 10:49, schrieb Oscar Alvarez:
>
> Yes,
>
>  But I need info to create the servlet.
>
>  In dialplan, I put anything like this:
>
>  exten => 1001,1,AGI(agi://ip_Tomcat/app1)
>
>
>  I like that app1 answer tha call and hangup.
>
>
>  import net.sf.asterisk.fastagi.AGIChannel;
> import net.sf.asterisk.fastagi.AGIException;
> import net.sf.asterisk.fastagi.AGIRequest;
> import net.sf.asterisk.fastagi.AbstractAGIScript;
>
> public class HelloAGIScript extends AbstractAGIScript
> {
>     public void service(AGIRequest request, AGIChannel channel)
>             throws AGIException
>     {
>         // Answer the channel...
>         answer(channel);
>
>
> 		
>         // ...and hangup.
>         hangup(channel);
>     }
> }
>
>  What is the mode tu mapping this??
>
>  I wold find a tutorial like this <http://www.asterisk-java.org/0.2/tutorial.html> but running in a Tomcat.
>
>  Thxx
>
>
>
> 2013/4/11 Yves A. <yv...@gm...>
>
>>  hi,
>>
>> you just have to put the asterisk-java jar file in the library path of
>> your tomcat (tomcat/lib) or your web-app (WEB-INF/lib).
>>
>> yves
>>
>> Am 11.04.2013 09:29, schrieb Oscar Alvarez:
>>
>>  Hi,
>>
>>  Where can I find some info for this. I need deploy the asterisk-java in
>> a Tomcat servlet.
>> Do you know any tutorial?
>>
>>  Thxxx
>>
>>  --
>> Oscar Alvarez
>>
>>
>>  ------------------------------------------------------------------------------
>> Precog is a next-generation analytics platform capable of advanced
>> analytics on semi-structured data. The platform includes APIs for building
>> apps and a phenomenal toolset for data science. Developers can use
>> our toolset for easy data analysis & visualization. Get a free account!http://www2.precog.com/precogplatform/slashdotnewsletter
>>
>>
>>
>> _______________________________________________
>> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Precog is a next-generation analytics platform capable of advanced
>> analytics on semi-structured data. The platform includes APIs for building
>> apps and a phenomenal toolset for data science. Developers can use
>> our toolset for easy data analysis & visualization. Get a free account!
>> http://www2.precog.com/precogplatform/slashdotnewsletter
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
>  --
> Oscar Alvarez
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!http://www2.precog.com/precogplatform/slashdotnewsletter
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!
> http://www2.precog.com/precogplatform/slashdotnewsletter
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>


-- 
Oscar Alvarez

Re: [Asterisk-java-users] need help

[Diego G. <di...@pr...>]

ok! one minute please...


Diego Augusto Costa Guimarães
Gerente TI - Praia Clube
*dCAP (Digium Certified Asterisk Professional)*
Pabx: (34) 3256-3116
(Mobile) celular: (34) - 9268-9999
Chat Google Talk: die...@gm... Skype: diegoacguimares MSN:
die...@gm...


On Mon, Apr 15, 2013 at 2:57 AM, vinethan jain <vin...@gm...> wrote:

> i need to capture dtmf digits
> tried a lot but cant capture
> can any one help me with a program code for capturing dtmf events..please!!
>
>
> ------------------------------------------------------------------------------
> Precog is a next-generation analytics platform capable of advanced
> analytics on semi-structured data. The platform includes APIs for building
> apps and a phenomenal toolset for data science. Developers can use
> our toolset for easy data analysis & visualization. Get a free account!
> http://www2.precog.com/precogplatform/slashdotnewsletter
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

[Asterisk-java-users] need help

[vinethan j. <vin...@gm...>]

i need to capture dtmf digits
tried a lot but cant capture
can any one help me with a program code for capturing dtmf events..please!!