asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] ExtensionStateAction on Asterisk 1.4

[Martin S. <ma...@be...>]

Hello Timothy,

When you say there's a timeout, does Asterisk-Java log an exception, or
just take a while? What version of Asterisk? What version of
Asterisk-Java? You aren't using a manager proxy, I don't think. If you
connect to the manager interface with telnet, and paste the action in,
does it take awhile?

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of Timothy Creswick
> Sent: Sunday, August 19, 2007 3:30 PM
> To: ast...@li...
> Subject: [Asterisk-java-users] ExtensionStateAction on Asterisk 1.4
>=20
> Dear All,
>=20
> I'm having trouble with the ExtensionStateAction,=20
> specifically that I'm
> getting timeouts.
>=20
> I've watched the data on the port and the Asterisk server is=20
> responding
> correctly, but it looks like asterisk-java is not parsing the response
> (malformed by Asterisk perhaps?).
>=20
> Here's a sample of what the Asterisk server is sending back:
>=20
> Response: Success
> ActionID: 13754931_22#
> Message: Extension Status
> Exten: 7001100
> Context: sip-phones
> Hint: SIP/7001100
> Status: 0
>=20
> Is there a patch available for either Asterisk or asterisk-java? What
> *should* the response be looking like?
>=20
> Thanks,
>=20
>=20
> Timothy Creswick
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

Re: [Asterisk-java-users] Making sure meetme records.

[Martin S. <ma...@be...>]

Hello Jan,
=20
I couldn't find a good way to ask Asterisk what channels are being
MixMonito'd except by patching app_mixmonitor.c in Asterisk itself (I
looked at the CLI and AMI). My intuition and current Asterisk experience
tells me that the MixMonitor application rarely "fails" -- may I ask
what has happened for you to cause failure to record? It wouldn't
surprise me if there was an easier problem to solve, like avoiding
recording failure in the first place. In a single day, our call center
is sometimes recording 60+ calls at once, and we never fail, AFAIK.
=20
Also, I'd suggest asking on Asterisk-Uiser too.
=20
Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20
=20


________________________________

	From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Jan Du Toit
	Sent: Monday, August 20, 2007 6:38 AM
	To: ast...@li...
	Subject: [Asterisk-java-users] Making sure meetme records.
=09
=09
	Hi.
=09
	This question is probably not appropriate for this list, sorry,
but was hoping somebody could give some insight.
	We have a commercial application that uses the Asterisk meetme
module. The conferences held, must be recorded and we use the 'r' option
for that. In the past we have encountered some technical problems which
caused the conferences not to be recorded. As a result the client has
asked to provide them with a mechanism that checks whether its actually
recording and keeps on recording correctly - in the event of failure the
user, conducting the conference via our software, must be alerted
immediately. How will one go about this? Any ideas?=20
=09
	Thanks you.
	Regards, Jan.
=09

Re: [Asterisk-java-users] Making sure meetme records.

[Wirsansky, E. <eya...@ve...>]

Perhaps you can launch a separate thread to monitor the size of the
file(s) that hold the active recording, and alert if there is no change
in the size of the file over a certain period.

=20

Eyal

=20

________________________________

From: ast...@li...
[mailto:ast...@li...] On Behalf Of
Jan Du Toit
Sent: Monday, August 20, 2007 6:38 AM
To: ast...@li...
Subject: [Asterisk-java-users] Making sure meetme records.

=20

Hi.

This question is probably not appropriate for this list, sorry, but was
hoping somebody could give some insight.
We have a commercial application that uses the Asterisk meetme module.
The conferences held, must be recorded and we use the 'r' option for
that. In the past we have encountered some technical problems which
caused the conferences not to be recorded. As a result the client has
asked to provide them with a mechanism that checks whether its actually
recording and keeps on recording correctly - in the event of failure the
user, conducting the conference via our software, must be alerted
immediately. How will one go about this? Any ideas?=20

Thanks you.
Regards, Jan.

[Asterisk-java-users] Making sure meetme records.

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

Hi.

This question is probably not appropriate for this list, sorry, but was
hoping somebody could give some insight.
We have a commercial application that uses the Asterisk meetme module. The
conferences held, must be recorded and we use the 'r' option for that. In
the past we have encountered some technical problems which caused the
conferences not to be recorded. As a result the client has asked to provide
them with a mechanism that checks whether its actually recording and keeps
on recording correctly - in the event of failure the user, conducting the
conference via our software, must be alerted immediately. How will one go
about this? Any ideas?

Thanks you.
Regards, Jan.

[Asterisk-java-users] ExtensionStateAction on Asterisk 1.4

[Timothy C. <ti...@th...>]

Dear All,

I'm having trouble with the ExtensionStateAction, specifically that I'm
getting timeouts.

I've watched the data on the port and the Asterisk server is responding
correctly, but it looks like asterisk-java is not parsing the response
(malformed by Asterisk perhaps?).

Here's a sample of what the Asterisk server is sending back:

Response: Success
ActionID: 13754931_22#
Message: Extension Status
Exten: 7001100
Context: sip-phones
Hint: SIP/7001100
Status: 0

Is there a patch available for either Asterisk or asterisk-java? What
*should* the response be looking like?

Thanks,


Timothy Creswick

Re: [Asterisk-java-users] Termination Status

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

Hey,

> I think there's an issue with thread synchronization in Asterisk-Java a=
t
> the moment.

could you provide some kind of isolated test case to reproduce this?
If it's really a sync issue within A-J we should fix it rather quickly :-=
)

=3DStefan

Re: [Asterisk-java-users] Little problem with asterisk-java-0.3

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

Hi Pedro,

first, please ask for support on the asterisk-users mailing list so
others can benefit from your question, too.
http://asterisk-java.org/development/mail-lists.html

Regarding your question:
You are sending actions from the manager event thread which causes a
deadlock that's why you get the TimeoutException.
You must either send the action from a different thread or just use the
live API (look for DefaultAsteriskServer) that also allows watching for
events and reading variables at a higher level that takes care of the
low level threading issues.

=3DStefan


Pedro J. Romero wrote:
> Dear Stefan:
>=20
> =20
>=20
> First of all, thank you very much for your great job. I=E2=80=99m Pedro=
 Romero
> from Iten, S.L. in Spain. We are a little company of software developer=
s
> and asterisk integrators, and we are using asterisk-java to make a
> little interface to asterisk.
>=20
> =20
>=20
> Perhaps you can help us with a little problem we have encountered: we
> want to read a variable from a channel, and when we try to do it, Java
> throws a TimeOutException. We have no look so deep, I=E2=80=99ve though=
t the
> best would be to write to you first.
>=20
> =20
>=20
> The code is this:
>=20
> =20
>=20
>       *public* *void* onManagerEvent(ManagerEvent event){
>=20
>             String cliente=3D"";
>=20
>             ManagerResponse response=3D*new* ManagerResponse();
>=20
>             *if*(event.getClass().getSimpleName().equals("JoinEvent")){=

>=20
>                   JoinEvent ev=3D(JoinEvent)event;
>=20
>                   *try*{
>=20
>                         GetVarAction getVarAction =3D *new* GetVarActio=
n();
>=20
>                         getVarAction.setChannel(ev.getChannel());
>=20
>                         getVarAction.setVariable("CLIENTE");
>=20
>                         response =3D
> parent.getManager().sendAction(getVarAction); =C3=9F parent is the inst=
ance
> where inside which the managerConnection is declared.
>=20
>                         cliente =3D response.getAttribute("Value");
>=20
>                   }
>=20
>             *catch*(TimeoutException exc){System./out/.println("Tiempo
> de espera agotado");}
>=20
>             *catch*(IOException exc){System./out/.println("Error de I/O=
");}
>=20
>             ...
>=20
>             ...
>=20
>       }
>=20
> =20
>=20
> We have tried to increase the timeout (with a second argument to the
> method sendAction) without success. I think I=E2=80=99ve read somewhere=
 that
> there was a problem with timeout and responses coming to fast, solved i=
n
> 0.3-m1. We are trying 0.3-m2.
>=20
> =20
>=20
> If you tell me that the code is the good one, we will going into deep t=
o
> look for any bug, and we will inform you if we find something.
>=20
> =20
>=20
> Advanced thanks, and congratulations again for your work.
>=20
> =20
>=20
> Pedro J. Romero.
>=20
> Iten, S.L.
>=20


--=20
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  ste...@re...
Jabber:  ste...@re...

Steuernummern 215/5140/1791 USt-IdNr. DE220701760

Re: [Asterisk-java-users] Termination Status

[<gl...@gm...>]

Look up the DefaultAsteriskServer Class, it has an
originateToExtension method, or something like that. Which is a sync
method.

On 8/16/07, Martin Smith <ma...@be...> wrote:
> Hello Chuck,
>
> I think there's an issue with thread synchronization in Asterisk-Java at
> the moment. The Live API keeps track of a channel by unique ID, and AJ
> gets the failure event before the live API has recorded the unique of
> the channel -- the failure event ends up triggering a lookup by unique
> id, and can't find the channel yet.
>
> Hopefully that's a good explanation of the problem.
>
> I recall that you can use an OriginateAction with
> .setAsync(Boolean.FALSE) if you are using the core part of the library,
> but I don't think there is an option for that in the live API yet.
>
> I hope I've explained that correctly, though I don't have a solution.
> Perhaps Stefan can comment more.
>
> Martin Smith, Systems Developer
> ma...@be...
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221
>
>
>
> > -----Original Message-----
> > From: ast...@li...
> > [mailto:ast...@li...] On
> > Behalf Of CHUCK SYPERSKI
> > Sent: Wednesday, August 15, 2007 3:02 PM
> > To: ast...@li...
> > Subject: Re: [Asterisk-java-users] Termination Status
> >
> > Is there a way to detect if the line you just called is busy.
> >  When I dial a busy number I keep getting:
> >
> > org.asteriskjava.live.NoSuchChannelException: Channel
> > 'SIP/1xxxxxxxxxx@gafachi' is not available
> >
> > Here is part of the code, its a bite ugly:
> >
> > public void run()
> >       {
> >               fireEvent( D88CCEvent.CCTHREAD_BORN );
> >               try
> >               {
> >                       if ( login() )
> >                       {
> >                               DefaultAsteriskServer server =3D
> > new DefaultAsteriskServer( managerConnection );
> >                               fireEvent(
> > D88CCEvent.CCTHREAD_RINGING );
> >                               String sp =3D "";
> >                               if (
> > task.getAsteriskServiceProvider() !=3D null &&
> > task.getAsteriskServiceProvider().length() > 0 )
> >                                       sp =3D "@" +
> > task.getAsteriskServiceProvider();
> >
> >                               channel =3D
> > server.originateToExtension(task.getAsteriskTrunk() + "/" +
> > task.getPhone() + sp, task.getAsteriskContext(),
> > task.getExtension()+"", 1, task.getTimeOut() * 1000 );
> >
> >                               if ( channel !=3D null )
> >                               {
> >                                       handleStateChange( channel );
> >                                       ChannelState oldState =3D null;
> >                                       if ( channel.getState()
> > =3D=3D ChannelState.UP || channel.getState() =3D=3D
> > ChannelState.RINGING || channel.getState() =3D=3D ChannelState.RING )
> >                                       {
> >                                               while(
> > channel.getState() =3D=3D ChannelState.UP || channel.getState()
> > =3D=3D ChannelState.RINGING || channel.getState() =3D=3D ChannelState.R=
ING )
> >                                               {
> >                                                       sleep(
> > 100 );
> >                                               }
> >
> >                                               if ( oldState
> > !=3D channel.getState() )
> >                                               {
> >
> > handleStateChange( channel );
> >
> > oldState =3D channel.getState();
> >                                               }
> >                                       }
> >                                       else
> >                                       {
> >                                               fireEvent(
> > D88CCEvent.CCTHREAD_GENERAL_ERROR );
> >                                               lg(
> > channel.toString() );
> >                                               System.exit( 1 );
> >                                       }
> >
> >                                       handleStateChange( channel );
> >                                       finalizeCall( channel );
> >                               }
> >                               managerConnection.logoff();
> >                       }
> >
> >               }
> >               catch( Exception e )
> >               {
> >                       blComplete =3D true;
> >
> >                       if ( e instanceof NoSuchChannelException )
> >                       {
> >                               fireEvent(
> > D88CCEvent.CCTHREAD_INVALIDNUMBER );
> >                               System.err.println( e.toString() );
> >                               e.printStackTrace();
> >
> >                               if ( channel !=3D null &&
> > channel.wasBusy() )
> >                               {
> >                                       System.out.println(
> > "Was Busy" );
> >                               }
> >                               else
> >                               {
> >                                       System.out.println(
> > "Was NOT Busy" );
> >                               }
> >
> >                               System.out.println( e.getCause() );
> >                               System.out.println( e.getMessage() );
> >
> >                               System.exit( 1 );
> >                       }
> >                       else
> >                       {
> >                               intCompletionCode =3D
> > D88CCEvent.CCTHREAD_DIED;
> >                               System.err.println( e.getMessage() );
> >                               e.printStackTrace();
> >                               //fireEvent( intCompletionCode );
> >                               System.exit( 1 );
> >                       }
> >               }
> >
> >       }
> >
> > >>> ma...@be... 08/15/07 11:52AM >>>
> > Hi Chuck,
> >
> > I wish I had an answer to that question myself. I'm not sure there's a
> > way that is completely error-free, especially given an older
> > generation
> > of answering machine that may do any number of things.
> >
> > Is your project open source?
> >
> > Martin Smith, Systems Developer
> > ma...@be...
> > Bureau of Economic and Business Research
> > University of Florida
> > (352) 392-0171 Ext. 221
> >
> >
> >
> > > -----Original Message-----
> > > From: ast...@li...
> > > [mailto:ast...@li...] On
> > > Behalf Of CHUCK SYPERSKI
> > > Sent: Wednesday, August 15, 2007 12:02 PM
> > > To: ast...@li...
> > > Subject: [Asterisk-java-users] Termination Status
> > >
> > > I am in the process of testing a autodialer I wrote for a
> > > school district.  Basically what I do is make an outgoing
> > > call via a gafachi sip trunk and connect the user to an
> > > internal extension that plays a message and hangs up.
> > > Everything works great, but I wanted to know if there is
> > > anyway that I can tell if I have called a fax machine, or
> > > voicemail.  Right now I am using the live package and I am
> > > using the AsteriskChannel.getState() and using the
> > > HangupCause enumeration.  Thanks so much for any help that
> > > can be provided and thanks for writting a great api.
> > >
> > > Thanks,
> > > Chuck Syperski
> > > DuPage High School District 88
> > >
> > >
> > > --------------------------------------------------------------
> > > -----------
> > > This SF.net email is sponsored by: Splunk Inc.
> > > Still grepping through log files to find problems?  Stop.
> > > Now Search log events and configuration files using AJAX and
> > > a browser.
> > > Download your FREE copy of Splunk now >>  http://get.splunk.com/
> > > _______________________________________________
> > > Asterisk-java-users mailing list
> > > Ast...@li...
> > > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> > >
> >
> > --------------------------------------------------------------
> > -----------
> > This SF.net email is sponsored by: Splunk Inc.
> > Still grepping through log files to find problems?  Stop.
> > Now Search log events and configuration files using AJAX and
> > a browser.
> > Download your FREE copy of Splunk now >>  http://get.splunk.com/
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> >
> > --------------------------------------------------------------
> > -----------
> > This SF.net email is sponsored by: Splunk Inc.
> > Still grepping through log files to find problems?  Stop.
> > Now Search log events and configuration files using AJAX and
> > a browser.
> > Download your FREE copy of Splunk now >>  http://get.splunk.com/
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> >
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>


--=20
Guilherme Loch G=F3es

MSN:gl...@gm...
(48) 99115299

Re: [Asterisk-java-users] Termination Status

[Martin S. <ma...@be...>]

Hello Chuck,

I think there's an issue with thread synchronization in Asterisk-Java at
the moment. The Live API keeps track of a channel by unique ID, and AJ
gets the failure event before the live API has recorded the unique of
the channel -- the failure event ends up triggering a lookup by unique
id, and can't find the channel yet.

Hopefully that's a good explanation of the problem.=20

I recall that you can use an OriginateAction with
.setAsync(Boolean.FALSE) if you are using the core part of the library,
but I don't think there is an option for that in the live API yet.

I hope I've explained that correctly, though I don't have a solution.
Perhaps Stefan can comment more.

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of CHUCK SYPERSKI
> Sent: Wednesday, August 15, 2007 3:02 PM
> To: ast...@li...
> Subject: Re: [Asterisk-java-users] Termination Status
>=20
> Is there a way to detect if the line you just called is busy.=20
>  When I dial a busy number I keep getting:
>=20
> org.asteriskjava.live.NoSuchChannelException: Channel=20
> 'SIP/1xxxxxxxxxx@gafachi' is not available
>=20
> Here is part of the code, its a bite ugly:
>=20
> public void run()
> 	{
> 		fireEvent( D88CCEvent.CCTHREAD_BORN );
> 		try
> 		{	=09
> 			if ( login() )
> 			{
> 				DefaultAsteriskServer server =3D=20
> new DefaultAsteriskServer( managerConnection );
> 				fireEvent(=20
> D88CCEvent.CCTHREAD_RINGING );
> 				String sp =3D "";
> 				if (=20
> task.getAsteriskServiceProvider() !=3D null &&=20
> task.getAsteriskServiceProvider().length() > 0 )
> 					sp =3D "@" +=20
> task.getAsteriskServiceProvider();
> 				=09
> 				channel =3D=20
> server.originateToExtension(task.getAsteriskTrunk() + "/" +=20
> task.getPhone() + sp, task.getAsteriskContext(),=20
> task.getExtension()+"", 1, task.getTimeOut() * 1000 );
>=20
> 				if ( channel !=3D null )
> 				{	=09
> 					handleStateChange( channel );
> 					ChannelState oldState =3D null;
> 					if ( channel.getState()=20
> =3D=3D ChannelState.UP || channel.getState() =3D=3D=20
> ChannelState.RINGING || channel.getState() =3D=3D ChannelState.RING )
> 					{
> 						while(=20
> channel.getState() =3D=3D ChannelState.UP || channel.getState()=20
> =3D=3D ChannelState.RINGING || channel.getState() =3D=3D =
ChannelState.RING )
> 						{
> 							sleep(=20
> 100 );				=09
> 						}
> 					=09
> 						if ( oldState=20
> !=3D channel.getState() )
> 						{
> 						=09
> handleStateChange( channel );=09
> 						=09
> oldState =3D channel.getState();
> 						}
> 					}
> 					else
> 					{
> 						fireEvent(=20
> D88CCEvent.CCTHREAD_GENERAL_ERROR );
> 						lg(=20
> channel.toString() );
> 						System.exit( 1 );
> 					}
> 				=09
> 					handleStateChange( channel );=09
> 					finalizeCall( channel );
> 				}
> 				managerConnection.logoff();
> 			}=09
> 				=09
> 		}
> 		catch( Exception e )
> 		{=09
> 			blComplete =3D true;
>=20
> 			if ( e instanceof NoSuchChannelException )
> 			{
> 				fireEvent(=20
> D88CCEvent.CCTHREAD_INVALIDNUMBER );
> 				System.err.println( e.toString() );
> 				e.printStackTrace();
> 			=09
> 				if ( channel !=3D null &&=20
> channel.wasBusy() )
> 				{
> 					System.out.println(=20
> "Was Busy" );
> 				}
> 				else
> 				{
> 					System.out.println(=20
> "Was NOT Busy" );			=09
> 				}			=09
> 			=09
> 				System.out.println( e.getCause() );
> 				System.out.println( e.getMessage() );
> 			=09
> 				System.exit( 1 );
> 			}
> 			else
> 			{
> 				intCompletionCode =3D=20
> D88CCEvent.CCTHREAD_DIED;
> 				System.err.println( e.getMessage() );
> 				e.printStackTrace();
> 				//fireEvent( intCompletionCode );=09
> 				System.exit( 1 );=09
> 			}
> 		}
> 	=09
> 	}
>=20
> >>> ma...@be... 08/15/07 11:52AM >>>
> Hi Chuck,
>=20
> I wish I had an answer to that question myself. I'm not sure there's a
> way that is completely error-free, especially given an older=20
> generation
> of answering machine that may do any number of things.
>=20
> Is your project open source?
>=20
> Martin Smith, Systems Developer
> ma...@be...=20
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221=20
>=20
> =20
>=20
> > -----Original Message-----
> > From: ast...@li...=20
> > [mailto:ast...@li...] On=20
> > Behalf Of CHUCK SYPERSKI
> > Sent: Wednesday, August 15, 2007 12:02 PM
> > To: ast...@li...=20
> > Subject: [Asterisk-java-users] Termination Status
> >=20
> > I am in the process of testing a autodialer I wrote for a=20
> > school district.  Basically what I do is make an outgoing=20
> > call via a gafachi sip trunk and connect the user to an=20
> > internal extension that plays a message and hangs up. =20
> > Everything works great, but I wanted to know if there is=20
> > anyway that I can tell if I have called a fax machine, or=20
> > voicemail.  Right now I am using the live package and I am=20
> > using the AsteriskChannel.getState() and using the=20
> > HangupCause enumeration.  Thanks so much for any help that=20
> > can be provided and thanks for writting a great api.
> >=20
> > Thanks,
> > Chuck Syperski
> > DuPage High School District 88
> >=20
> >=20
> > --------------------------------------------------------------
> > -----------
> > This SF.net email is sponsored by: Splunk Inc.
> > Still grepping through log files to find problems?  Stop.
> > Now Search log events and configuration files using AJAX and=20
> > a browser.
> > Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...=20
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users=20
> >=20
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...=20
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

Re: [Asterisk-java-users] Termination Status

[CHUCK S. <CSY...@DU...>]

Is there a way to detect if the line you just called is busy.  When I dial =
a busy number I keep getting:

org.asteriskjava.live.NoSuchChannelException: Channel 'SIP/1xxxxxxxxxx@gafa=
chi' is not available

Here is part of the code, its a bite ugly:

public void run()
	{
		fireEvent( D88CCEvent.CCTHREAD_BORN );
		try
		{	=09
			if ( login() )
			{
				DefaultAsteriskServer server =3D new =
DefaultAsteriskServer( managerConnection );
				fireEvent( D88CCEvent.CCTHREAD_RINGING );
				String sp =3D "";
				if ( task.getAsteriskServiceProvider() =
!=3D null && task.getAsteriskServiceProvider().length() > 0 )
					sp =3D "@" + task.getAsteriskServic=
eProvider();
				=09
				channel =3D server.originateToExtension(tas=
k.getAsteriskTrunk() + "/" + task.getPhone() + sp, task.getAsteriskContext(=
), task.getExtension()+"", 1, task.getTimeOut() * 1000 );

				if ( channel !=3D null )
				{	=09
					handleStateChange( channel );
					ChannelState oldState =3D null;
					if ( channel.getState() =3D=3D =
ChannelState.UP || channel.getState() =3D=3D ChannelState.RINGING || =
channel.getState() =3D=3D ChannelState.RING )
					{
						while( channel.getState() =
=3D=3D ChannelState.UP || channel.getState() =3D=3D ChannelState.RINGING =
|| channel.getState() =3D=3D ChannelState.RING )
						{
							sleep( 100 );		=
		=09
						}
					=09
						if ( oldState !=3D =
channel.getState() )
						{
							handleStateChange( =
channel );=09
							oldState =3D =
channel.getState();
						}
					}
					else
					{
						fireEvent( D88CCEvent.CCTHR=
EAD_GENERAL_ERROR );
						lg( channel.toString() );
						System.exit( 1 );
					}
				=09
					handleStateChange( channel );=09
					finalizeCall( channel );
				}
				managerConnection.logoff();
			}=09
				=09
		}
		catch( Exception e )
		{=09
			blComplete =3D true;

			if ( e instanceof NoSuchChannelException )
			{
				fireEvent( D88CCEvent.CCTHREAD_INVALIDNUMBE=
R );
				System.err.println( e.toString() );
				e.printStackTrace();
			=09
				if ( channel !=3D null && channel.wasBusy()=
 )
				{
					System.out.println( "Was Busy" );
				}
				else
				{
					System.out.println( "Was NOT Busy" =
);			=09
				}			=09
			=09
				System.out.println( e.getCause() );
				System.out.println( e.getMessage() );
			=09
				System.exit( 1 );
			}
			else
			{
				intCompletionCode =3D D88CCEvent.CCTHREAD_D=
IED;
				System.err.println( e.getMessage() );
				e.printStackTrace();
				//fireEvent( intCompletionCode );=09
				System.exit( 1 );=09
			}
		}
	=09
	}

>>> ma...@be... 08/15/07 11:52AM >>>
Hi Chuck,

I wish I had an answer to that question myself. I'm not sure there's a
way that is completely error-free, especially given an older generation
of answering machine that may do any number of things.

Is your project open source?

Martin Smith, Systems Developer
ma...@be...=20
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of CHUCK SYPERSKI
> Sent: Wednesday, August 15, 2007 12:02 PM
> To: ast...@li...=20
> Subject: [Asterisk-java-users] Termination Status
>=20
> I am in the process of testing a autodialer I wrote for a=20
> school district.  Basically what I do is make an outgoing=20
> call via a gafachi sip trunk and connect the user to an=20
> internal extension that plays a message and hangs up. =20
> Everything works great, but I wanted to know if there is=20
> anyway that I can tell if I have called a fax machine, or=20
> voicemail.  Right now I am using the live package and I am=20
> using the AsteriskChannel.getState() and using the=20
> HangupCause enumeration.  Thanks so much for any help that=20
> can be provided and thanks for writting a great api.
>=20
> Thanks,
> Chuck Syperski
> DuPage High School District 88
>=20
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...=20
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users=20
>=20

-------------------------------------------------------------------------
This SF.net email is sponsored by: Splunk Inc.
Still grepping through log files to find problems?  Stop.
Now Search log events and configuration files using AJAX and a browser.
Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
_______________________________________________
Asterisk-java-users mailing list
Ast...@li...=20
https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Termination Status

[CHUCK S. <CSY...@DU...>]

Right now it isn't released at all to the public, as soon as it is =
complete and functional I was going to release it under GPL v2.

>>> ma...@be... 08/15/07 11:52AM >>>
Hi Chuck,

I wish I had an answer to that question myself. I'm not sure there's a
way that is completely error-free, especially given an older generation
of answering machine that may do any number of things.

Is your project open source?

Martin Smith, Systems Developer
ma...@be...=20
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of CHUCK SYPERSKI
> Sent: Wednesday, August 15, 2007 12:02 PM
> To: ast...@li...=20
> Subject: [Asterisk-java-users] Termination Status
>=20
> I am in the process of testing a autodialer I wrote for a=20
> school district.  Basically what I do is make an outgoing=20
> call via a gafachi sip trunk and connect the user to an=20
> internal extension that plays a message and hangs up. =20
> Everything works great, but I wanted to know if there is=20
> anyway that I can tell if I have called a fax machine, or=20
> voicemail.  Right now I am using the live package and I am=20
> using the AsteriskChannel.getState() and using the=20
> HangupCause enumeration.  Thanks so much for any help that=20
> can be provided and thanks for writting a great api.
>=20
> Thanks,
> Chuck Syperski
> DuPage High School District 88
>=20
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...=20
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users=20
>=20

-------------------------------------------------------------------------
This SF.net email is sponsored by: Splunk Inc.
Still grepping through log files to find problems?  Stop.
Now Search log events and configuration files using AJAX and a browser.
Download your FREE copy of Splunk now >>  http://get.splunk.com/=20
_______________________________________________
Asterisk-java-users mailing list
Ast...@li...=20
https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] resolution: question about getting peersusing SipShowPeerAction (AMI)

[Robert P. <ro...@ro...>]

Hey Martin,

No worries about timing :)

I've been thinking a bit about what might help ... while my first
inclination is to say "of course a response should be specific (i.e.,
class/type) to the event it's a response to", that would mean a new
hierarchy/set of subclasses, and could raise the learning curve that
much more for beginners.  It might just be unnecessarily clutter in
the response package.

One alternative is an (abstract?) default ManagerResponse subclass
that provides a bunch of empty methods that can be ignored if you're
not interested in that data/it's not that kind of response, but you
can override those that do have the data you're interested in.  The
methods would ideally cover everything possible from any AMI response
... Also not so pretty.

In terms of documentation, it says in the method comment for
ManagerResponse.getAttributes():  "Returns a Map with all attributes
of this response."  I read that but didn't really get it.  Maybe a
paragraph or two explaining what the attributes Map is and that it's a
catch-all for all the data from the AMI in the class comments/javadoc
would help make that more explicit.

I hope this helps - thanks!


Cheers,

Robert

On 8/15/07, Martin Smith <ma...@be...> wrote:
> Hi Robert,
>
> I'm sorry I didn't answer sooner -- with most responses, the response
> subclasses (command response, getconfig response, etc) have setters that
> are called when traversing the attributes map. This means anything that
> isn't an actual data member of the class is always still sitting in the
> Map.
>
> For good measure, here's the documentation as it stands regarding the
> ManagerResponse and the getAttributes method:
> http://asterisk-java.org/development/apidocs/org/asteriskjava/manager/re
> sponse/ManagerResponse.html
>
> Would you suggest we make it more explicit in the class summary that the
> attributes are what most users would want? We'd welcome that suggestion
> and other ones for improving the javadocs :)
>
> Martin Smith, Systems Developer
> ma...@be...
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221

Re: [Asterisk-java-users] Termination Status

[Martin S. <ma...@be...>]

Hi Chuck,

I wish I had an answer to that question myself. I'm not sure there's a
way that is completely error-free, especially given an older generation
of answering machine that may do any number of things.

Is your project open source?

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of CHUCK SYPERSKI
> Sent: Wednesday, August 15, 2007 12:02 PM
> To: ast...@li...
> Subject: [Asterisk-java-users] Termination Status
>=20
> I am in the process of testing a autodialer I wrote for a=20
> school district.  Basically what I do is make an outgoing=20
> call via a gafachi sip trunk and connect the user to an=20
> internal extension that plays a message and hangs up. =20
> Everything works great, but I wanted to know if there is=20
> anyway that I can tell if I have called a fax machine, or=20
> voicemail.  Right now I am using the live package and I am=20
> using the AsteriskChannel.getState() and using the=20
> HangupCause enumeration.  Thanks so much for any help that=20
> can be provided and thanks for writting a great api.
>=20
> Thanks,
> Chuck Syperski
> DuPage High School District 88
>=20
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

Re: [Asterisk-java-users] resolution: question about getting peersusing SipShowPeerAction (AMI)

[Martin S. <ma...@be...>]

Hi Robert,

I'm sorry I didn't answer sooner -- with most responses, the response
subclasses (command response, getconfig response, etc) have setters that
are called when traversing the attributes map. This means anything that
isn't an actual data member of the class is always still sitting in the
Map.

For good measure, here's the documentation as it stands regarding the
ManagerResponse and the getAttributes method:
http://asterisk-java.org/development/apidocs/org/asteriskjava/manager/re
sponse/ManagerResponse.html

Would you suggest we make it more explicit in the class summary that the
attributes are what most users would want? We'd welcome that suggestion
and other ones for improving the javadocs :)

Martin Smith, Systems Developer
ma...@be...
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221=20

=20

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of Robert Prince
> Sent: Wednesday, August 15, 2007 12:17 PM
> To: ast...@li...
> Subject: [Asterisk-java-users] resolution: question about=20
> getting peersusing SipShowPeerAction (AMI)
>=20
> Folks,
>=20
> Well, I feel rather foolish.  It turns out the ManagerResponse already
> held all the information I was after, in its HashMap 'attributes' -
> accessible via getAttributes().  In this case, the information I'm
> after is the same that is returned at the Asterisk console by the
> command 'sip show peer [peer name]'.  So while it may have been
> unnecessary effort, at least all my fumbling around exposed me to more
> of the API and functionality, and I also learned a bit more about
> Asterisk in the process.
>=20
> So, while perhaps the API could be documented in a bit more detail, or
> maybe there could be more explicit (or just more) tutorials/docs, hats
> off for a great app.
>=20
>=20
> Cheers,
>=20
> Robert
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and=20
> a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

[Asterisk-java-users] resolution: question about getting peers using SipShowPeerAction (AMI)

[Robert P. <ro...@ro...>]

Folks,

Well, I feel rather foolish.  It turns out the ManagerResponse already
held all the information I was after, in its HashMap 'attributes' -
accessible via getAttributes().  In this case, the information I'm
after is the same that is returned at the Asterisk console by the
command 'sip show peer [peer name]'.  So while it may have been
unnecessary effort, at least all my fumbling around exposed me to more
of the API and functionality, and I also learned a bit more about
Asterisk in the process.

So, while perhaps the API could be documented in a bit more detail, or
maybe there could be more explicit (or just more) tutorials/docs, hats
off for a great app.


Cheers,

Robert

[Asterisk-java-users] Termination Status

[CHUCK S. <CSY...@DU...>]

I am in the process of testing a autodialer I wrote for a school district. =
 Basically what I do is make an outgoing call via a gafachi sip trunk and =
connect the user to an internal extension that plays a message and hangs =
up.  Everything works great, but I wanted to know if there is anyway that =
I can tell if I have called a fax machine, or voicemail.  Right now I am =
using the live package and I am using the AsteriskChannel.getState() and =
using the HangupCause enumeration.  Thanks so much for any help that can =
be provided and thanks for writting a great api.

Thanks,
Chuck Syperski
DuPage High School District 88

Re: [Asterisk-java-users] question about getting peers using SipShowPeerAction (AMI)

[Robert P. <ro...@ro...>]

Well ... I still haven't had much success.  Here's the scenario that I'm using:

First, all is set up as stated in the start of this thread.

1) I pull the plug on my ATA, causing my two peers (test1 and test2)
to time out eventually.

2) I see the unreachable events in the Asterisk console:
*CLI> [Aug 10 19:59:58] NOTICE[9540]: chan_sip.c:15360
sip_poke_noanswer: Peer 'test1' is now UNREACHABLE!  Last qualify: 94
[Aug 10 20:00:05] NOTICE[9540]: chan_sip.c:15360 sip_poke_noanswer:
Peer 'test2' is now UNREACHABLE!  Last qualify: 93

3) I see the PeerStatusEvents from the code (forgive my **** and &&&&,
it's to help me identify specific debugging statements):
**** PeerStatusEvent:
org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Fri Aug 10
20:09:46 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test2',time='-1',cause='null',peerstatus='Unreachable',systemHashcode=7359402]
**** peer name: test2
&&&&&&&&& SipShowPeerAction action complete event class: class
org.asteriskjava.manager.event.PeerlistCompleteEvent
*** Manager Response:
org.asteriskjava.manager.response.ManagerResponse:
actionId='actionIdtest2'; message='null'; response='Success';
uniqueId='null'; systemHashcode=12577309
**** PeerStatusEvent:
org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Fri Aug 10
20:09:47 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test1',time='-1',cause='null',peerstatus='Unreachable',systemHashcode=27475272]
**** peer name: test1
&&&&&&&&& SipShowPeerAction action complete event class: class
org.asteriskjava.manager.event.PeerlistCompleteEvent
*** Manager Response:
org.asteriskjava.manager.response.ManagerResponse:
actionId='actionIdtest1'; message='null'; response='Success';
uniqueId='null'; systemHashcode=813251

But, I don't see PeerlistCompleteEvents, or PeerEntryEvents.

What I expect from reading the javadoc is that when I send a
SipShowPeerAction to the AMI, it should result in the AMI sending me
back a PeerEntryEvent, and possibly also a PeerlistCompleteEvent (?not
sure about that).

But, I don't seem to get either.  I've posted the (very simple) code
I'm using on Google docs to see if it will help point out where I'm
going wrong - please, feel free to take a look and if there's
anything, let me know!

http://docs.google.com/Doc?id=dcpvc479_33gqzn2s


Cheers,

Robert

On 8/10/07, Robert Prince <ro...@ro...> wrote:
> Hello Stefan,
>
> Thanks very much for your help - I'll give that a try!
>
>
> Cheers,
>
> Robert
>
> On 8/9/07, Stefan Reuter <ste...@re...> wrote:
> > Hi Robert,
> >
> > first thanks for your very well prepared question, it's a pleasure to
> > reply to it :)
> >
> > > **** PeerStatusEvent:
> > > org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
> > > 20:16:31 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test2',time='97',cause='null',peerstatus='Reachable',systemHashcode=22618484]
> > > ** Manager Response: org.asteriskjava.manager.response.ManagerError:
> > > actionId='null'; message='Peer SIP/test2 not found.';
> > > response='Error'; uniqueId='null'; systemHashcode=10703525
> >
> > The problem is that SipShowPeer requires only the name of the peer
> > without the channel prefix ("SIP/"). Unfortunately the PeerStatus event
> > returns the name with the channel prefix and Asterisk calls both
> > attributes just "peer". I will have a look at the javadocs to make this
> > more clear.
> > For you this means you just have to strip the channel prefix before
> > passing the peer name to the SipShowPeerAction.
> >
> > Hope that helps,
> >
> > Stefan
> >
> > --
> > reuter network consulting
> > Neusser Str. 110
> > 50760 Koeln
> > Germany
> > Telefon: +49 221 1305699-0
> > Telefax: +49 221 1305699-90
> > E-Mail:  ste...@re...
> > Jabber:  ste...@re...
> >
> > Steuernummern 215/5140/1791 USt-IdNr. DE220701760
> >
> >
> > -------------------------------------------------------------------------
> > This SF.net email is sponsored by: Splunk Inc.
> > Still grepping through log files to find problems?  Stop.
> > Now Search log events and configuration files using AJAX and a browser.
> > Download your FREE copy of Splunk now >>  http://get.splunk.com/
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> >
> >
> >
>

[Asterisk-java-users] Bandwidth usage

[Mike <mi...@si...>]

-Greetings-

    Any thoughts about bandwidth or server utilization as it would apply =
to an increasing number of clients which queried events from an Asterisk =
Server?  I personally do not see potential loading issues ...However, =
just in case I am overlooking something -any words of cautions?


-Mike-

Re: [Asterisk-java-users] question about getting peers using SipShowPeerAction (AMI)

[Robert P. <ro...@ro...>]

Hello Stefan,

Thanks very much for your help - I'll give that a try!


Cheers,

Robert

On 8/9/07, Stefan Reuter <ste...@re...> wrote:
> Hi Robert,
>
> first thanks for your very well prepared question, it's a pleasure to
> reply to it :)
>
> > **** PeerStatusEvent:
> > org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
> > 20:16:31 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test2',time='97',cause='null',peerstatus='Reachable',systemHashcode=22618484]
> > ** Manager Response: org.asteriskjava.manager.response.ManagerError:
> > actionId='null'; message='Peer SIP/test2 not found.';
> > response='Error'; uniqueId='null'; systemHashcode=10703525
>
> The problem is that SipShowPeer requires only the name of the peer
> without the channel prefix ("SIP/"). Unfortunately the PeerStatus event
> returns the name with the channel prefix and Asterisk calls both
> attributes just "peer". I will have a look at the javadocs to make this
> more clear.
> For you this means you just have to strip the channel prefix before
> passing the peer name to the SipShowPeerAction.
>
> Hope that helps,
>
> Stefan
>
> --
> reuter network consulting
> Neusser Str. 110
> 50760 Koeln
> Germany
> Telefon: +49 221 1305699-0
> Telefax: +49 221 1305699-90
> E-Mail:  ste...@re...
> Jabber:  ste...@re...
>
> Steuernummern 215/5140/1791 USt-IdNr. DE220701760
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>

Re: [Asterisk-java-users] MeetMeJoinEvent

[Miked <mi...@si...>]

OK ... I know you were all waiting for me to 'get it' on my own... stupid 
syntax crap :-)

This works!
if (event instanceof MeetMeJoinEvent) {

System.out.println(((MeetMeJoinEvent)event).getMeetMe()); }

THANKS!!!




----- Original Message ----- 
From: "Miked" <mi...@si...>
To: "Carlos G Mendioroz" <tr...@ac...>; 
<ast...@li...>
Sent: Thursday, August 09, 2007 9:09 PM
Subject: Re: [Asterisk-java-users] MeetMeJoinEvent


> OK Java casting got that ...google is my friend ;-)  after thinking I
> understood and applying the suggested code ... Eclipse flags error on the
> last line 'event.getMeetMe()'  "The Method getMeetMe() is undefined for 
> the
> type ManagerEvent"  However what does work is as follows:
>
> This works:
> if (event.getClass().equals(MeetMeJoinEvent.class)) {
>
> MeetMeJoinEvent mmje=(MeetMeJoinEvent)event;
>
> System.out.println(mmje.getMeetMe());
>
> }
>
>
>
> Code flaged with error "undefined for type ManagerEvent":
>
>          if (event instanceof MeetMeJoinEvent) {
>  System.out.println((MeetMeJoinEvent)event.getMeetMe()); }
>
>
>
> Not sure why???  Maybe this is Java 101 stuff, but just trying to 
> understand
> ..it would seems these bits of code are the same???.... Help is GREATLY
> APPRECIATED!
>
> ----- Original Message ----- 
> From: "Carlos G Mendioroz" <tr...@ac...>
> To: <ast...@li...>
> Cc: <mi...@si...>
> Sent: Thursday, August 09, 2007 5:40 AM
> Subject: Re: [Asterisk-java-users] MeetMeJoinEvent
>
>
>> Oops, typo :)
>> This should go into a FAQ.
>>
>> Events returned via onManagerEvent come with a common type of Event.
>> Depending on the current event, more functionality is available via
>> subclasses (of which the actual event "is a").
>>
>> Usually you access subclass functionality by casting to the subclass,
>> guarded by a type check like:
>>
>> public void onManagerEvent(ManagerEvent event)
>> {
>>        // just print received events
>>        System.out.println(event);
>>
>> if (event.getClass().equals(MeetMeJoinEvent.class)) {
>> System.out.println((MeetMeJoinEvent)event.getMeetMe());
>> ...
>>
>> HTH.
>>
>> Stefan Reuter @ 09/08/2007 05:18 -0300 dixit:
>>> try this: (and take a look at some java introduction ;)
>>>
>>>  public void onManagerEvent(ManagerEvent event)
>>>  {
>>>          // just print received events
>>>           System.out.println(event);
>>>
>>>           System.out.println((MeetMeJoinEvent) mmje.getMeetMe());
>>>  }
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> -------------------------------------------------------------------------
>>> This SF.net email is sponsored by: Splunk Inc.
>>> Still grepping through log files to find problems?  Stop.
>>> Now Search log events and configuration files using AJAX and a browser.
>>> Download your FREE copy of Splunk now >>  http://get.splunk.com/
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>> -- 
>> Carlos G Mendioroz  <tr...@ac...>
>>
>>
>> -- 
>> No virus found in this incoming message.
>> Checked by AVG Free Edition.
>> Version: 7.5.476 / Virus Database: 269.11.10/943 - Release Date: 8/8/2007
>> 5:38 PM
>>
>>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.11.10/943 - Release Date: 8/8/2007 
> 5:38 PM
>
> 

Re: [Asterisk-java-users] MeetMeJoinEvent

[Miked <mi...@si...>]

OK Java casting got that ...google is my friend ;-)  after thinking I 
understood and applying the suggested code ... Eclipse flags error on the 
last line 'event.getMeetMe()'  "The Method getMeetMe() is undefined for the 
type ManagerEvent"  However what does work is as follows:

This works:
if (event.getClass().equals(MeetMeJoinEvent.class)) {

MeetMeJoinEvent mmje=(MeetMeJoinEvent)event;

System.out.println(mmje.getMeetMe());

}



Code flaged with error "undefined for type ManagerEvent":

          if (event instanceof MeetMeJoinEvent) {
  System.out.println((MeetMeJoinEvent)event.getMeetMe()); }



Not sure why???  Maybe this is Java 101 stuff, but just trying to understand 
..it would seems these bits of code are the same???.... Help is GREATLY 
APPRECIATED!

----- Original Message ----- 
From: "Carlos G Mendioroz" <tr...@ac...>
To: <ast...@li...>
Cc: <mi...@si...>
Sent: Thursday, August 09, 2007 5:40 AM
Subject: Re: [Asterisk-java-users] MeetMeJoinEvent


> Oops, typo :)
> This should go into a FAQ.
>
> Events returned via onManagerEvent come with a common type of Event.
> Depending on the current event, more functionality is available via
> subclasses (of which the actual event "is a").
>
> Usually you access subclass functionality by casting to the subclass,
> guarded by a type check like:
>
> public void onManagerEvent(ManagerEvent event)
> {
>        // just print received events
>        System.out.println(event);
>
> if (event.getClass().equals(MeetMeJoinEvent.class)) {
> System.out.println((MeetMeJoinEvent)event.getMeetMe());
> ...
>
> HTH.
>
> Stefan Reuter @ 09/08/2007 05:18 -0300 dixit:
>> try this: (and take a look at some java introduction ;)
>>
>>  public void onManagerEvent(ManagerEvent event)
>>  {
>>          // just print received events
>>           System.out.println(event);
>>
>>           System.out.println((MeetMeJoinEvent) mmje.getMeetMe());
>>  }
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >>  http://get.splunk.com/
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
> -- 
> Carlos G Mendioroz  <tr...@ac...>
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.11.10/943 - Release Date: 8/8/2007 
> 5:38 PM
>
> 

Re: [Asterisk-java-users] question about getting peers using SipShowPeerAction (AMI)

[Carlos G M. <tr...@hu...>]

Hmm, your peer is "SIP/test1" or just "test1" ?

Asterisk is not all that clean in naming things here and there.
A bit of parsing here and there might help.

I'm fighting right now with agents and all things that are taken
for granted. Like Agent/XXXX being an agent callback channel,
which is sometimes referred to by the context/extension used to reach
it, sometimes by the channel being used by it...

HTH

Robert Prince @ 09/08/2007 21:38 -0300 dixit:
> Hello asterisk-java folks,
> 
> I've read the tutorial, and the javadoc, and browsed the mailing
> lists, but so far I can't find why what I'm trying to do does not
> work.
> 
> I have an asterisk server (1.49), and two peers configured (called
> 'test1' and 'test2').  The peers connect and register properly and all
> that good stuff (using Linksys SPA2102).
> 
> I'm trying to use asterisk-java to detect peer state changes -
> specifically, reachable and unreachable events.  The following seems
> to be basic code for how to do that:
> 
> ... (assume ManagerConnection works, we've logged in to the AMI
> successfully, etc) ...
> 
>     public void onManagerEvent(ManagerEvent event) {
>         String peerName = "";
>         if (event instanceof PeerStatusEvent) {
>                 peerName = ((PeerStatusEvent) event).getPeer();
>                 System.out.println("**** PeerStatusEvent: "+event);
>                 SipShowPeerAction showPeer = new SipShowPeerAction(peerName);
>                 managerConnection.sendAction(showPeer, new
> SendActionCallback() {
>                         public void onResponse(ManagerResponse response) {
>                                 System.out.println("** Manager
> Response: " + response);
>                         }
>                 });
>         }
>         if (event instanceof PeerEntryEvent) {
>                 System.out.println("** peer entry event - IP address:
> "+((PeerEntryEvent) event).getIpAddress());
>                 System.out.println("** peer entry event - port:
> "+((PeerEntryEvent) event).getIpPort());
>         }
> 
> What I hope this should do for me, if I'm reading the javadocs
> correctly, is poke the asterisk server into giving me (through
> asterisk-java) a PeerEntryEvent, so I can get the IP address and port.
>  However, what I get is the following:
> 
> **** PeerStatusEvent:
> org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
> 20:16:31 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test2',time='97',cause='null',peerstatus='Reachable',systemHashcode=22618484]
> ** Manager Response: org.asteriskjava.manager.response.ManagerError:
> actionId='null'; message='Peer SIP/test2 not found.';
> response='Error'; uniqueId='null'; systemHashcode=10703525
> **** PeerStatusEvent:
> org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
> 20:16:33 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test1',time='100',cause='null',peerstatus='Reachable',systemHashcode=8846886]
> ** Manager Response: org.asteriskjava.manager.response.ManagerError:
> actionId='null'; message='Peer SIP/test1 not found.';
> response='Error'; uniqueId='null'; systemHashcode=9938272
> 
> I'm a bit confused ... I see the peer names in the PeerStatusEvents,
> but then the ManagerError says they don't exist.  Is there something
> in manager.conf I need to do?  I've configured it thus:
> 
> [general]
> enabled = yes
> port = 5038
> bindaddr = 0.0.0.0
> 
> [manager]
> secret=xxxxxxxx
> permit=0.0.0.0/0.0.0.0
> read=all,event,system,call,log,verbose,agent,command,user
> write=all,event,system,call,log,verbose,agent,command,user
> 
> Honestly I don't even know if 'all,event' are proper syntax or really
> do anything, but I have been trying lots of things to see if it's e.g.
> a configuration problem.
> 
> Anyone have ideas?  Much appreciated!
> 
> 
> Cheers,
> 
> Robert
> 
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >>  http://get.splunk.com/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> 

-- 
Carlos G Mendioroz  <tr...@hu...>  LW7 EQI  Argentina

Re: [Asterisk-java-users] question about getting peers using SipShowPeerAction (AMI)

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

Hi Robert,

first thanks for your very well prepared question, it's a pleasure to
reply to it :)

> **** PeerStatusEvent:
> org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=3DThu Aug 0=
9
> 20:16:31 EDT 2007,privilege=3D'system,all',timestamp=3D'null',peer=3D'S=
IP/test2',time=3D'97',cause=3D'null',peerstatus=3D'Reachable',systemHashc=
ode=3D22618484]
> ** Manager Response: org.asteriskjava.manager.response.ManagerError:
> actionId=3D'null'; message=3D'Peer SIP/test2 not found.';
> response=3D'Error'; uniqueId=3D'null'; systemHashcode=3D10703525

The problem is that SipShowPeer requires only the name of the peer
without the channel prefix ("SIP/"). Unfortunately the PeerStatus event
returns the name with the channel prefix and Asterisk calls both
attributes just "peer". I will have a look at the javadocs to make this
more clear.
For you this means you just have to strip the channel prefix before
passing the peer name to the SipShowPeerAction.

Hope that helps,

Stefan

--=20
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  ste...@re...
Jabber:  ste...@re...

Steuernummern 215/5140/1791 USt-IdNr. DE220701760

[Asterisk-java-users] question about getting peers using SipShowPeerAction (AMI)

[Robert P. <ro...@ro...>]

Hello asterisk-java folks,

I've read the tutorial, and the javadoc, and browsed the mailing
lists, but so far I can't find why what I'm trying to do does not
work.

I have an asterisk server (1.49), and two peers configured (called
'test1' and 'test2').  The peers connect and register properly and all
that good stuff (using Linksys SPA2102).

I'm trying to use asterisk-java to detect peer state changes -
specifically, reachable and unreachable events.  The following seems
to be basic code for how to do that:

... (assume ManagerConnection works, we've logged in to the AMI
successfully, etc) ...

    public void onManagerEvent(ManagerEvent event) {
        String peerName = "";
        if (event instanceof PeerStatusEvent) {
                peerName = ((PeerStatusEvent) event).getPeer();
                System.out.println("**** PeerStatusEvent: "+event);
                SipShowPeerAction showPeer = new SipShowPeerAction(peerName);
                managerConnection.sendAction(showPeer, new
SendActionCallback() {
                        public void onResponse(ManagerResponse response) {
                                System.out.println("** Manager
Response: " + response);
                        }
                });
        }
        if (event instanceof PeerEntryEvent) {
                System.out.println("** peer entry event - IP address:
"+((PeerEntryEvent) event).getIpAddress());
                System.out.println("** peer entry event - port:
"+((PeerEntryEvent) event).getIpPort());
        }

What I hope this should do for me, if I'm reading the javadocs
correctly, is poke the asterisk server into giving me (through
asterisk-java) a PeerEntryEvent, so I can get the IP address and port.
 However, what I get is the following:

**** PeerStatusEvent:
org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
20:16:31 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test2',time='97',cause='null',peerstatus='Reachable',systemHashcode=22618484]
** Manager Response: org.asteriskjava.manager.response.ManagerError:
actionId='null'; message='Peer SIP/test2 not found.';
response='Error'; uniqueId='null'; systemHashcode=10703525
**** PeerStatusEvent:
org.asteriskjava.manager.event.PeerStatusEvent[dateReceived=Thu Aug 09
20:16:33 EDT 2007,privilege='system,all',timestamp='null',peer='SIP/test1',time='100',cause='null',peerstatus='Reachable',systemHashcode=8846886]
** Manager Response: org.asteriskjava.manager.response.ManagerError:
actionId='null'; message='Peer SIP/test1 not found.';
response='Error'; uniqueId='null'; systemHashcode=9938272

I'm a bit confused ... I see the peer names in the PeerStatusEvents,
but then the ManagerError says they don't exist.  Is there something
in manager.conf I need to do?  I've configured it thus:

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[manager]
secret=xxxxxxxx
permit=0.0.0.0/0.0.0.0
read=all,event,system,call,log,verbose,agent,command,user
write=all,event,system,call,log,verbose,agent,command,user

Honestly I don't even know if 'all,event' are proper syntax or really
do anything, but I have been trying lots of things to see if it's e.g.
a configuration problem.

Anyone have ideas?  Much appreciated!


Cheers,

Robert

Re: [Asterisk-java-users] MeetMeJoinEvent

[David R. <dr...@in...>]

On Thu, 2007-08-09 at 06:40 -0300, Carlos G Mendioroz wrote:

> Usually you access subclass functionality by casting to the subclass,
> guarded by a type check like:
>=20
> public void onManagerEvent(ManagerEvent event)
> {
>         // just print received events
>         System.out.println(event);
> =09
> 	if (event.getClass().equals(MeetMeJoinEvent.class)) {
          if (event instanceof MeetMeJoinEvent) {
> 		System.out.println((MeetMeJoinEvent)event.getMeetMe());

This might be much more common. :)


Nitpicking,

	David
--=20
INA Service GmbH
Papenreye 63
22453 Hamburg
Germany

Mail: in...@in...
Phone: +49 (0)40 557 07-07
Fax: +49 (0)40 557 07-100

Gesch=C3=A4ftsf=C3=BChrer: Ines M. Hoerner

Amtsgericht Hamburg
HRB 96470
Ust-IdNr. DE248754961