asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] Re: Controlling the Queue

[Gerwin B. <ger...@gm...>]

Hi,

is there a certain reason why you try to use the manager API for Call =20=

handling?


Just extend the BaseAGIScript and do your call handling there.
More Info can be found here:
http://asterisk-java.sourceforge.net/tutorial.html

look at the AGI section.

Gerwin



On Sep 27, 2005, at 10:13 AM, Jan Ekholm wrote:

> On Tuesday 27 September 2005 11:00, In=E9s wrote:
>
>> hello,
>> i'm a student, and i'm a beginner in the asterisk world !!!
>> i've got a few questions, i hope you can help me :
>> - i'm trying to do an application in java, wich receives a call, =20
>> palys
>> a audio file, wait for digits, then hangs up.
>> but i can't find how to receive the events, and how to asnwer the =20
>> phone.
>>
>
> RedirectAction can be used to answer the phone, but then you'll =20
> need to have
> the caller call a queue or similar. I'm not sure if Asterisk-Java =20
> can work as
> a softphone by itself, I don't think so. The AGI command =20
> GetDataCommand seems
> to be able to read DTMF stuff, but I haven't used it myself.
>
>
>
>> voila mon petit code :
>> public void run () throws IOException, AuthenticationFailedException,
>> TimeoutException, InterruptedException
>>     {
>>         //boolean x=3Dtrue;
>>
>>         //register for events
>>         managerConnection.addEventHandler(this);
>>
>>         // connect to Asterisk and log in
>>         managerConnection.login();
>>
>>
>>         wait();
>>
>>         //and finally log off and disconnect
>>         managerConnection.logoff();
>>
>>     }
>>     public void handleEvent(ManagerEvent event)
>>     {
>>         // just print received events
>>         System.out.println("this is an event: "+event);
>>         String source =3D event.getSource().toString();
>>         System.out.println("Source : "+source);
>>         //print type of event
>>         System.out.println("Type of the event :
>> "+event.getClass().getSimpleName());
>>
>>     }
>>
>> On 9/26/05, Alex <ab...@ga...> wrote:
>>
>>> Hi:
>>>
>>> My system has the following architecture (in the server side):
>>>
>>> - An app server (connected to the asterisk console)
>>> - An AGI Server (developed with AsteriskJava)
>>> - An AGI Script (executed by the above AGI Server)
>>>
>>> In the client side (Agents answering call center calls):
>>>
>>> - A softphone
>>> - A client program (used to search and register call details)
>>>
>>> Here is the thing:
>>>
>>> - =46rom AGI Server I detect that a call is coming from PSTN and =20
>>> launch the
>>> AGI Script
>>> - =46rom AGI Script I put the call in the queue and I loose the =20
>>> control of
>>> the call (here is my first confusion)
>>> - The agent answer the call (using his/her softphone) and I get =20
>>> the event
>>> from the Asterisk Console with my App Server.
>>>
>>> Now, I need to play something (TTS, wav, etc) to the caller based =20=

>>> on the
>>> client application wich is connected to my App Server. What I =20
>>> want you to
>>> know is that the information to be played to the caller comes =20
>>> from an
>>> external source.
>>>
>>> So, my two big questions/confusions are:
>>>
>>> - How can I get the entire control of the call depending on the =20
>>> status of
>>> the call, for example, if the call is in the queue and I need to =20
>>> play or
>>> do something with it, where and how I have the control? until =20
>>> now, when I
>>> put the call in the queue I loss the control until the caller or the
>>> agent hangs the call.
>>>
>>> - Once the call is answered by the Agent, how can I unlink the two
>>> channels (releasing the agent) to let the caller hear the text =20
>>> that the
>>> agent sent.
>>>
>>>
>>> Thanks in advance,
>>>
>>> Alejandro
>>>
>>
>> --
>> In=E9s MOUSSA
>> =C9l=E8ve ing=E9nieur g=E9nie logiciel
>> 21 rue Henri Marrou
>> Chatenay-Malabry 92290
>> Tel : 00 33 613984578
>> ine...@gm...
>> ine...@ec...
>>
>>
>> -------------------------------------------------------
>> SF.Net email is sponsored by:
>> Tame your development challenges with Apache's Geronimo App Server.
>> Download it for free - -and be entered to win a 42" plasma tv or =20
>> your very
>> own Sony(tm)PSP.  Click here to play: http://sourceforge.net/=20
>> geronimo.php
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>
> --=20
> Jan Ekholm
> jan...@pl...
>
>
> -------------------------------------------------------
> SF.Net email is sponsored by:
> Tame your development challenges with Apache's Geronimo App Server.
> Download it for free - -and be entered to win a 42" plasma tv or =20
> your very
> own Sony(tm)PSP.  Click here to play: http://sourceforge.net/=20
> geronimo.php
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

Re: [Asterisk-java-users] Re: Controlling the Queue

[Jan E. <jan...@pl...>]

On Tuesday 27 September 2005 11:00, In=E9s wrote:
> hello,
> i'm a student, and i'm a beginner in the asterisk world !!!
> i've got a few questions, i hope you can help me :
> - i'm trying to do an application in java, wich receives a call, palys
> a audio file, wait for digits, then hangs up.
> but i can't find how to receive the events, and how to asnwer the phone.

RedirectAction can be used to answer the phone, but then you'll need to hav=
e=20
the caller call a queue or similar. I'm not sure if Asterisk-Java can work =
as=20
a softphone by itself, I don't think so. The AGI command GetDataCommand see=
ms=20
to be able to read DTMF stuff, but I haven't used it myself.


> voila mon petit code :
> public void run () throws IOException, AuthenticationFailedException,
> TimeoutException, InterruptedException
> 	{
> 		//boolean x=3Dtrue;
>
> 		//register for events
>         managerConnection.addEventHandler(this);
>
>         // connect to Asterisk and log in
>         managerConnection.login();
>
>
>         wait();
>
>         //and finally log off and disconnect
>         managerConnection.logoff();
>
> 	}
> 	public void handleEvent(ManagerEvent event)
> 	{
> 		// just print received events
> 		System.out.println("this is an event: "+event);
> 		String source =3D event.getSource().toString();
> 		System.out.println("Source : "+source);
> 		//print type of event
> 		System.out.println("Type of the event :
> "+event.getClass().getSimpleName());
>
> 	}
>
> On 9/26/05, Alex <ab...@ga...> wrote:
> > Hi:
> >
> > My system has the following architecture (in the server side):
> >
> > - An app server (connected to the asterisk console)
> > - An AGI Server (developed with AsteriskJava)
> > - An AGI Script (executed by the above AGI Server)
> >
> > In the client side (Agents answering call center calls):
> >
> > - A softphone
> > - A client program (used to search and register call details)
> >
> > Here is the thing:
> >
> > - From AGI Server I detect that a call is coming from PSTN and launch t=
he
> > AGI Script
> > - From AGI Script I put the call in the queue and I loose the control of
> > the call (here is my first confusion)
> > - The agent answer the call (using his/her softphone) and I get the eve=
nt
> > from the Asterisk Console with my App Server.
> >
> > Now, I need to play something (TTS, wav, etc) to the caller based on the
> > client application wich is connected to my App Server. What I want you =
to
> > know is that the information to be played to the caller comes from an
> > external source.
> >
> > So, my two big questions/confusions are:
> >
> > - How can I get the entire control of the call depending on the status =
of
> > the call, for example, if the call is in the queue and I need to play or
> > do something with it, where and how I have the control? until now, when=
 I
> > put the call in the queue I loss the control until the caller or the
> > agent hangs the call.
> >
> > - Once the call is answered by the Agent, how can I unlink the two
> > channels (releasing the agent) to let the caller hear the text that the
> > agent sent.
> >
> >
> > Thanks in advance,
> >
> > Alejandro
>
> --
> In=E9s MOUSSA
> =C9l=E8ve ing=E9nieur g=E9nie logiciel
> 21 rue Henri Marrou
> Chatenay-Malabry 92290
> Tel : 00 33 613984578
> ine...@gm...
> ine...@ec...
>
>
> -------------------------------------------------------
> SF.Net email is sponsored by:
> Tame your development challenges with Apache's Geronimo App Server.
> Download it for free - -and be entered to win a 42" plasma tv or your very
> own Sony(tm)PSP.  Click here to play: http://sourceforge.net/geronimo.php
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

=2D-=20
Jan Ekholm
jan...@pl...

[Asterisk-java-users] Re: Controlling the Queue

[<ine...@gm...>]

thank's in advance

have a nice day

ines

On 9/27/05, In=E9s <ine...@gm...> wrote:
> hello,
> i'm a student, and i'm a beginner in the asterisk world !!!
> i've got a few questions, i hope you can help me :
> - i'm trying to do an application in java, wich receives a call, palys
> a audio file, wait for digits, then hangs up.
> but i can't find how to receive the events, and how to asnwer the phone.
>
> voila mon petit code :
> public void run () throws IOException, AuthenticationFailedException,
> TimeoutException, InterruptedException
> =09{
> =09=09//boolean x=3Dtrue;
> =09=09=09
> =09=09//register for events
>         managerConnection.addEventHandler(this);
>
>         // connect to Asterisk and log in
>         managerConnection.login();
>
>
>         wait();
>
>         //and finally log off and disconnect
>         managerConnection.logoff();
>
> =09}
> =09public void handleEvent(ManagerEvent event)
> =09{
> =09=09// just print received events
> =09=09System.out.println("this is an event: "+event);
> =09=09String source =3D event.getSource().toString();
> =09=09System.out.println("Source : "+source);
> =09=09//print type of event
> =09=09System.out.println("Type of the event :
> "+event.getClass().getSimpleName());
> =09=09=09=09
> =09}
>
> On 9/26/05, Alex <ab...@ga...> wrote:
> > Hi:
> >
> > My system has the following architecture (in the server side):
> >
> > - An app server (connected to the asterisk console)
> > - An AGI Server (developed with AsteriskJava)
> > - An AGI Script (executed by the above AGI Server)
> >
> > In the client side (Agents answering call center calls):
> >
> > - A softphone
> > - A client program (used to search and register call details)
> >
> > Here is the thing:
> >
> > - From AGI Server I detect that a call is coming from PSTN and launch t=
he
> > AGI Script
> > - From AGI Script I put the call in the queue and I loose the control o=
f
> the
> > call (here is my first confusion)
> > - The agent answer the call (using his/her softphone) and I get the eve=
nt
> > from the Asterisk Console with my App Server.
> >
> > Now, I need to play something (TTS, wav, etc) to the caller based on th=
e
> > client application wich is connected to my App Server. What I want you =
to
> > know is that the information to be played to the caller comes from an
> > external source.
> >
> > So, my two big questions/confusions are:
> >
> > - How can I get the entire control of the call depending on the status =
of
> > the call, for example, if the call is in the queue and I need to play o=
r
> do
> > something with it, where and how I have the control? until now, when I
> put
> > the call in the queue I loss the control until the caller or the agent
> hangs
> > the call.
> >
> > - Once the call is answered by the Agent, how can I unlink the two
> channels
> > (releasing the agent) to let the caller hear the text that the agent
> sent.
> >
> >
> > Thanks in advance,
> >
> > Alejandro
> >
> >
> >
>
>
> --
> In=E9s MOUSSA
> =C9l=E8ve ing=E9nieur g=E9nie logiciel
> 21 rue Henri Marrou
> Chatenay-Malabry 92290
> Tel : 00 33 613984578
> ine...@gm...
> ine...@ec...
>


--
In=E9s MOUSSA
=C9l=E8ve ing=E9nieur g=E9nie logiciel
21 rue Henri Marrou
Chatenay-Malabry 92290
Tel : 00 33 613984578
ine...@gm...
ine...@ec...

[Asterisk-java-users] Re: Controlling the Queue

[<ine...@gm...>]

hello,
i'm a student, and i'm a beginner in the asterisk world !!!
i've got a few questions, i hope you can help me :
- i'm trying to do an application in java, wich receives a call, palys
a audio file, wait for digits, then hangs up.
but i can't find how to receive the events, and how to asnwer the phone.

voila mon petit code :
public void run () throws IOException, AuthenticationFailedException,
TimeoutException, InterruptedException
=09{
=09=09//boolean x=3Dtrue;
=09=09=09
=09=09//register for events
        managerConnection.addEventHandler(this);

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


        wait();

        //and finally log off and disconnect
        managerConnection.logoff();

=09}
=09public void handleEvent(ManagerEvent event)
=09{
=09=09// just print received events
=09=09System.out.println("this is an event: "+event);
=09=09String source =3D event.getSource().toString();
=09=09System.out.println("Source : "+source);
=09=09//print type of event
=09=09System.out.println("Type of the event : "+event.getClass().getSimpleN=
ame());
=09=09=09=09
=09}

On 9/26/05, Alex <ab...@ga...> wrote:
> Hi:
>
> My system has the following architecture (in the server side):
>
> - An app server (connected to the asterisk console)
> - An AGI Server (developed with AsteriskJava)
> - An AGI Script (executed by the above AGI Server)
>
> In the client side (Agents answering call center calls):
>
> - A softphone
> - A client program (used to search and register call details)
>
> Here is the thing:
>
> - From AGI Server I detect that a call is coming from PSTN and launch the
> AGI Script
> - From AGI Script I put the call in the queue and I loose the control of =
the
> call (here is my first confusion)
> - The agent answer the call (using his/her softphone) and I get the event
> from the Asterisk Console with my App Server.
>
> Now, I need to play something (TTS, wav, etc) to the caller based on the
> client application wich is connected to my App Server. What I want you to
> know is that the information to be played to the caller comes from an
> external source.
>
> So, my two big questions/confusions are:
>
> - How can I get the entire control of the call depending on the status of
> the call, for example, if the call is in the queue and I need to play or =
do
> something with it, where and how I have the control? until now, when I pu=
t
> the call in the queue I loss the control until the caller or the agent ha=
ngs
> the call.
>
> - Once the call is answered by the Agent, how can I unlink the two channe=
ls
> (releasing the agent) to let the caller hear the text that the agent sent=
.
>
>
> Thanks in advance,
>
> Alejandro
>
>
>


--
In=E9s MOUSSA
=C9l=E8ve ing=E9nieur g=E9nie logiciel
21 rue Henri Marrou
Chatenay-Malabry 92290
Tel : 00 33 613984578
ine...@gm...
ine...@ec...

[Asterisk-java-users] Controlling the Queue

[Alex <ab...@ga...>]

Hi:

My system has the following architecture (in the server side):

- An app server (connected to the asterisk console)
- An AGI Server (developed with AsteriskJava)
- An AGI Script (executed by the above AGI Server)

In the client side (Agents answering call center calls):

- A softphone
- A client program (used to search and register call details)

Here is the thing:

- From AGI Server I detect that a call is coming from PSTN and launch =
the AGI Script
- From AGI Script I put the call in the queue and I loose the control of =
the call (here is my first confusion)
- The agent answer the call (using his/her softphone) and I get the =
event from the Asterisk Console with my App Server.

Now, I need to play something (TTS, wav, etc) to the caller based on the =
client application wich is connected to my App Server. What I want you =
to know is that the information to be played to the caller comes from an =
external source.

So, my two big questions/confusions are:

- How can I get the entire control of the call depending on the status =
of the call, for example, if the call is in the queue and I need to play =
or do something with it, where and how I have the control? until now, =
when I put the call in the queue I loss the control until the caller or =
the agent hangs the call.

- Once the call is answered by the Agent, how can I unlink the two =
channels (releasing the agent) to let the caller hear the text that the =
agent sent.


Thanks in advance,

Alejandro

Re: [Asterisk-java-users] Call redirection

[Jan E. <jan...@pl...>]

On Friday 23 September 2005 14:37, Jan Ekholm wrote:
> Hi,

> I use code similar to this (version 0.2-rc1):
>
>         RedirectAction redirect = new RedirectAction ();
>         redirect.setChannel ( channel.getName() );
>         redirect.setExten ( "1001" );
>         redirect.setContext ("default");
>         redirect.setPriority (new Integer(1));

Weird is that I can do this and it works fine:

          redirect.setChannel ( channel.getLinkedChannel().getName() );	

This does what it is supposed to do, ie redirect the channel to the new "1001" 
extension. Of course, here I redirect the wrong channel (I redirect the one 
that answered the original call, not the one that did the original call), but 
why that one works and the real one doesn't is beyond me. 

However, I think this is one of those PEBKAC problems, as it seems to me that 
Asterisk-Java sends the correct action to Asterisk and generally does seem to 
do the right thing. Maybe a channel that has been already redirected once (to 
the answering operator) can't be redirected, but that doesn't add up as I can 
do the redirect on the actual hardware phone (dial "#" and "1001") and that 
works fine. So it's something I probably don't grok about extensions and/or 
contexts.

Personally I wonder why the VoIP/PBX world has got to be so complex... :)

-- 
Jan Ekholm
jan...@pl...

[Asterisk-java-users] Call redirection

[Jan E. <jan...@pl...>]

Hi,

I've had some problems getting a call to be programmatically redirected from 
one extension to another. I currently can have calls redirected from queues 
to an operator or some third party. This works fine, the call is redirected 
without any problems. However, when I want to let the operator redirect an 
call that has been answered I run into problems. The call is immediately hung 
up by Asterisk, and I don't know wether I use RedirectAction wrong or just 
don't understand something.

I use code similar to this (version 0.2-rc1):

        RedirectAction redirect = new RedirectAction ();
        redirect.setChannel ( channel.getName() );
        redirect.setExten ( "1001" );
        redirect.setContext ("default");
        redirect.setPriority (new Integer(1));

The "channel" is a Channel object. Similar code works fine for my direct 
redirect without answering in between. Redirection works fine when I do it 
using the hardware phones (Zyxel and Hitachi stuff), so there doesn't seem to 
be anything wrong on the Asterisk side either. The SIP accounts are 
configured to allow call redirections.

Anyone stumbled onto this problem?

Regards,
    Jan

-- 
Jan Ekholm
jan...@pl...

Re: [Asterisk-java-users] documentation

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

Hi,

what you probably need is FastAGI. Have a look at the short tutotrial at
http://www.asterisk-java.org/0.2-rc1/docs/tutorial.html
and the javadocs of BaseAGIScript which you should extend to provide your
functionality:
http://asterisk-java.sourceforge.net/apidocs/net/sf/asterisk/fastagi/Base=
AGIScript.html

=3DStefan

> hello everybody,
> i'm a new user of asterisk java and i need some help and some
> documentation on asterisk java
> my task is to do an application in java wich enables you to pick up
> the phone, play a audio file, to get the digits typed then hang up.
> thank's a lot
>
>
> -------------------------------------------------------
> SF.Net email is sponsored by:
> Tame your development challenges with Apache's Geronimo App Server.
> Download it for free - -and be entered to win a 42" plasma tv or your v=
ery
> own Sony(tm)PSP.  Click here to play: http://sourceforge.net/geronimo.p=
hp
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

[Asterisk-java-users] documentation

[<ine...@gm...>]

hello everybody,
i'm a new user of asterisk java and i need some help and some
documentation on asterisk java
my task is to do an application in java wich enables you to pick up
the phone, play a audio file, to get the digits typed then hang up.
thank's a lot

Re: [Asterisk-java-users] still not getting CDR events

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

On Wed, 2005-09-14 at 23:27 +0500, SYED ADEEL ALI wrote:
> I m using asterisk 1.07 and asterisk-java 0.1 n i've given every
> permission (read/write) to the manager as describe in the tutorial as
> well as in manager.c
> =20
> I m getting all other events .... the CDR has been recorded in MYSQL
> database as usual but i m unable to capture it's event .... help me
> plzzz
>=20

lets have a look if the event is actually sent by asterisk or not.
telnet to your asterisk box on port 5038 and send a login action.
that should look like:

srt@endorsed:~$ telnet pbx0 5038
Trying 10.13.0.100...
Connected to pbx0.cologne.reucon.net.
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
Username: manager
Secret: obelisk

Response: Success
Message: Authentication accepted

[...]

Event: Cdr
AccountCode:
Source: 1310
Destination: s
DestinationContext: default
CallerID: Stefan Reuter<1310>
Channel: SIP/1310-c8b4
DestinationChannel:
LastApplication: Hangup
LastData:
StartTime: 2005-09-16 05:22:12
AnswerTime: 2005-09-16 05:22:12
EndTime: 2005-09-16 05:22:13
Duration: 1
BillableSeconds: 1
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: pbx0-cgn-808-1126848132.75
UserField:

If you get it via telnet we must investigate Asterisk-Java, if you don't
its a problem with Asterisk or its configuration.

=3DStefan

[Asterisk-java-users] still not getting CDR events

[SYED A. A. <ad...@ho...>]

<html><div style='background-color:'><DIV class=RTE>I m using asterisk 1.07 and asterisk-java 0.1 n i've given every permission (read/write) to the manager as describe in the tutorial as well as in manager.c</DIV>
<DIV class=RTE>&nbsp;</DIV>
<DIV class=RTE>I&nbsp;m getting all other events .... the CDR has been recorded in MYSQL database as usual but i m&nbsp;unable to capture&nbsp;it's event .... help me plzzz</DIV></div><br clear=all><hr>FREE pop-up blocking with the new MSN Toolbar <a href="http://g.msn.com/8HMAEN/2755??PS=47575" target="_top">MSN Toolbar</a> Get it now!</html>

[Asterisk-java-users] still not getting CDR events

[SYED A. A. <ad...@ho...>]

<html><div style='background-color:'><DIV class=RTE><BR>I m using ASterisk 1.07 and asterisk-java-0.1 ... here is my manager.conf</DIV>
<DIV class=RTE>&nbsp;</DIV>
<DIV class=RTE>[general]<BR>enabled = yes<BR>port = 5038<BR>bindaddr = 0.0.0.0</DIV>
<DIV class=RTE>&nbsp;</DIV>
<DIV class=RTE>[manager]<BR>secret=pa55w0rd<BR>permit=0.0.0.0/0.0.0.0<BR>read=system,call,log,verbose,agent,command,user<BR>write=system,call,log,verbose,agent,command,user</DIV>
<DIV class=RTE>&nbsp;</DIV>
<DIV class=RTE>i m getting all the events except the CDR event ... although the CDR has been recorded in MYSQL database as usually.</DIV></div><br clear=all><hr>Don't just search. Find. <a href="http://g.msn.com/8HMBEN/2746??PS=47575" target="_top">MSN Search</a> Check out the new MSN Search!</html>

Re: [Asterisk-java-users] CDR Event doesn't Occur

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

Hi,

On Mon, 2005-09-12 at 11:09 +0500, SYED ADEEL ALI wrote:
>              Plz tell me whether a CDR event triigers or not .... i m
> catching other events like link event, hangup event etc but unable to
> catch the cdr event ... i've enabled it in cdr_manager.conf n the cdr
> has been recorded in MySQL database ... plz tell me how can I catch a
> CDR event ????

What version of Asterisk and Asterisk-Java do you use?
What permissions did you grant to your manager user in manager.c? (this
is indicated by the corresponding read and write options and should
include the call permission to receive CDR events)

=3DStefan

[Asterisk-java-users] CDR Event doesn't Occur

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Re: [Asterisk-java-users] OriginateAction.setVariable

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

> so, in the end i think i have reached the behavior i wanted.

ok.
For more complex scenarios i like to join AGIs into a MeetMe conference.
You basically use the OriginateAction to create a channel that executes a
FastAGI and pass some kind of id via the AGI URL. Then its up to your
application to provide a script for the AGI and look up what to do based
on the id.

=3DStefan

Re: [Asterisk-java-users] OriginateAction.setVariable

[Tobias W. <tob...@ev...>]

Stefan Reuter schrieb:
> On Tue, 2005-09-06 at 17:56 +0200, Tobias Wolf wrote:
> 
> You can still use your "old" AGIs with Asterisk-Java 0.2-rc1 though you
> will get a compile time warning about using a deprecated API.
>
ok, thx for the info. i will keep that in mind ...

> 
> 
> When originating a call two channels are involved. Let's asume you
> originate a call from A to B then what happens looks like:
> 
> A channel is created to A:
> ----> A
> A is ringing
> When A is answered the second channel is created:
> ----> A ----> B
> The channel variables are set on the channel from A to B only.
> 
> Does that help in your case?
> 
thx for you explanation, it helps to improve my overall understanding of 
the concept, although i am not really happy with it, but this is 
something a have to live with, i guess :)

the main idea of what i want to do is the following:
stream an audio file (which is variable) into a conf room (which is also 
variable, of course). the problem is that i have to set something in 'A' 
and in 'B'.

Since variables only appear in the called channel (B) it is only useful 
to send the conf room number per variable and the problem stucks with 
the file to stream, which has to be set in A.

so i switched to use
	originateAction.setChannel("Local/"+number+"@confannounce"); 

         originateAction.setApplication("Playback");
         originateAction.setData(file);

	[confannounce]
	exten => _X.,1,NoOp(This is the conf room number in which to stream: 
${EXTEN})
	exten => _X.,2,MeetMe(${EXTEN})
	exten => _X.,3,Hangup

This approach connects to the Local Channel. I tried even to set 
variables in this szenario, but they does not apppear accordingly to the 
valuable information of Stefan. After the Channel is answered, wich is 
done by the MeetMe-Application, the Application is executed which is 
defined in the Originating-Action.

so, in the end i think i have reached the behavior i wanted.

thx for the help :)

tobias

Re: [Asterisk-java-users] OriginateAction.setVariable

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

On Tue, 2005-09-06 at 17:56 +0200, Tobias Wolf wrote:
> first of all i am still using asterisk-java-0.1 because i didn't want to=20
> disturb the AGIs presently in use :-S

You can still use your "old" AGIs with Asterisk-Java 0.2-rc1 though you
will get a compile time warning about using a deprecated API.

> The call is executed correctly and the sound is streamed into the=20
> conference, but the channel variable is not.

When originating a call two channels are involved. Let's asume you
originate a call from A to B then what happens looks like:

A channel is created to A:
----> A
A is ringing
When A is answered the second channel is created:
----> A ----> B
The channel variables are set on the channel from A to B only.

So when you change your example to

        OriginateAction originateAction =3D new OriginateAction();
        originateAction.setChannel("Local/5000@test");
        originateAction.setVariable("var1=3DTEST");
        originateAction.setExten("5001");
        originateAction.setContext("test");
        originateAction.setPriority(new Integer(1));
        originateAction.setTimeout(new Integer(30000));

and your extensions.conf to

[test]
exten =3D> 5000,1,Answer()
exten =3D> 5000,2,NoOp(Our var is not printed here: ${var1})
exten =3D> 5000,3,Playback(demo-abouttotry)
exten =3D> 5000,4,Hangup()

exten =3D> 5001,1,NoOp(Our var is printed here: ${var1})
exten =3D> 5001,2,MeetMe(1)

you will see that your channel variable is actually set on the channel
to the MeetMe room.
Does that help in your case?

=3DStefan

[Asterisk-java-users] OriginateAction.setVariable

[Tobias W. <tob...@ev...>]

Hi,

first of all i am still using asterisk-java-0.1 because i didn't want to 
disturb the AGIs presently in use :-S

I wanted to set a Variable while i am originating a call. Here is what i 
wrote:

OriginateAction originateAction = new OriginateAction();
originateAction.setChannel("Local/5000@test");
originateAction.setVariable("var1=TEST");
originateAction.setApplication("MeetMe");
originateAction.setData(number);
originateAction.setTimeout(new Integer(30000));

ManagerResponse originateResponse = 
managerConnection.sendAction(originateAction, 30000);

Here is the extension:

exten => 5000,1,Answer()
exten => 5000,2,NoOp(Our var should be printed here : ${var1})
exten => 5000,3,Playback(demo-abouttotry)
exten => 5000,4,Hangup()

The call is executed correctly and the sound is streamed into the 
conference, but the channel variable is not.

I also have tried out an call file:

Channel: Local/5000@test
SetVar: var1=100
Application: MeetMe
Data: 333

With the same result.

Can anybody point me to my fault ??

Greetings,

tobias

Re: [Asterisk-java-users] Backward compatibility and startup performance of DefaultAsteriskManager in Asterisk-Java 1.2 (was: ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released)

[Jan E. <jan...@pl...>]

On Thursday 01 September 2005 22:22, Stefan Reuter wrote:
> Thanks for your feedback.
>
> On Thu, 2005-09-01 at 14:53 +0300, Jan Ekholm wrote:
> > Seems to work just fine for me. The only thing so far that has bitten me
> > is the change from Channel.getExtension() to
> > Channel.get???().getExtension().
>
> Oh yes, it is not nice to just remove those methods ;)
> I readded these methods for 0.2, now there are shortcuts for
> - Channel.getCurrentExtension().getContext()
> - Channel.getCurrentExtension().getExtension()
> - Channel.getCurrentExtension().getPriority()
> Besides backward compatibility, thats probably handy if you are not
> interested in the channel's extension history.

Ah, well, I already use getCurrentExtension() so it's better to have the code 
updated to what will become and not keep it as what has been. Maybe add a 
small comment in the Javadoc that they are equivalent?

> > I do notice a significantly longer connect/startup time, but I haven't
> > tried to track it down yet, could well be something I do that's not
> > necessary anymore.
>
> This is due to the way the queue initialization works now. I fixed
> Asterisk 1.2 to send a QueueStatusComplete event which Asterisk-Java now
> waits for and that was not send by Asterisk 1.0.x. So when using
> Asterisk-Java with Asterisk 1.0.x you will run into a timeout (default
> is 5 seconds).
> If you don't need queue information you can get around that with
> Asterisk 1.0.x by setting skipQueues to true in DefaultAsteriskManager.

No, I rather have queue info properly on startup, even on 1.0, than a somewhat 
faster startup. I just noticed the extra delay, it doesn't bother me. My app 
isn't meant to be started frequently so 5s is no issue at all.

> > I also see a lot of fun functionality that's only available in the 1.2
> > version of Asterisk, maybe I should get someone to upgrade our internal
> > development server. :)
>
> Yes that will be worth it. Asterisk 1.2 includes several nice
> enhancements to the Manager API.

Ok, I'll convince someone else to do it for us. :) Thanks for the great 
release.

-- 
Jan Ekholm
jan...@pl...

[Asterisk-java-users] Backward compatibility and startup performance of DefaultAsteriskManager in Asterisk-Java 1.2 (was: ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released)

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

Thanks for your feedback.

On Thu, 2005-09-01 at 14:53 +0300, Jan Ekholm wrote:
> Seems to work just fine for me. The only thing so far that has bitten me is 
> the change from Channel.getExtension() to Channel.get???().getExtension().

Oh yes, it is not nice to just remove those methods ;)
I readded these methods for 0.2, now there are shortcuts for
- Channel.getCurrentExtension().getContext()
- Channel.getCurrentExtension().getExtension()
- Channel.getCurrentExtension().getPriority()
Besides backward compatibility, thats probably handy if you are not
interested in the channel's extension history.

> I do notice a significantly longer connect/startup time, but I haven't tried to 
> track it down yet, could well be something I do that's not necessary anymore.

This is due to the way the queue initialization works now. I fixed
Asterisk 1.2 to send a QueueStatusComplete event which Asterisk-Java now
waits for and that was not send by Asterisk 1.0.x. So when using
Asterisk-Java with Asterisk 1.0.x you will run into a timeout (default
is 5 seconds).
If you don't need queue information you can get around that with
Asterisk 1.0.x by setting skipQueues to true in DefaultAsteriskManager.

> I also see a lot of fun functionality that's only available in the 1.2 version 
> of Asterisk, maybe I should get someone to upgrade our internal development 
> server. :)

Yes that will be worth it. Asterisk 1.2 includes several nice
enhancements to the Manager API.

=Stefan

Re: [Asterisk-java-users] ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released

[Jan E. <jan...@pl...>]

On Wednesday 31 August 2005 08:57, Stefan Reuter wrote:
> Asterisk-Java 0.2-rc1, a Java control for the Asterisk PBX, has been
> released.

Seems to work just fine for me. The only thing so far that has bitten me is 
the change from Channel.getExtension() to Channel.get???().getExtension(). I 
do notice a significantly longer connect/startup time, but I haven't tried to 
track it down yet, could well be something I do that's not necessary anymore.

I also see a lot of fun functionality that's only available in the 1.2 version 
of Asterisk, maybe I should get someone to upgrade our internal development 
server. :)

Good work Stefan!

-- 
Jan Ekholm
jan...@pl...

[Asterisk-java-users] ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released

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

Asterisk-Java 0.2-rc1, a Java control for the Asterisk PBX, has been
released.

The Asterisk-Java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-Java supports both interfaces that Asterisk
provides=20
for this scenario: The FastAGI protocol and the Manager API.

The 0.2-rc1 release candidate focuses on the new features of=20
Asterisk 1.2-beta1 though it still supports Asterisk 1.0.x.
The changes include
	* Support for the new Actions, Events and Commands of Asterisk 1.2
	* New support for event generating Actions, i.e. Actions
	  that send their result as a series of Event rather than
	  the usual ManagerResults. See the sendEventGeneratingAction()
	  methods in ManagerConnection for more information.
	* New base class for AGI scripts that allows you write cleaner=20
	  AGI scripts as you don't have to pass the channel variable
	  to all methods.
	* New convenience constructors for manager actions
	* Some minor bug fixes

Asterisk-Java is used in several commercial environments and by
the following Open Source projects:
	* Asterisk-IM
	  A plugin for the Jive Messenger XMPP (jabber) server. It provides
	  integrated presence between your IM client and phone, notification=20
	  of incoming calls by IM and originate calls from supported IM=20
	  clients.
	* Asterisk Desktop Manager (ADM)
	  A desktop application that will allow for automatic on-call volume=20
	  reduction, one click dial from clipboard, integrated phonebook
	  and more.

Asterisk-Java is available under Apache 2.0 license at
http://asterisk-java.sourceforge.net

[Asterisk-java-users] Re: Asterisk-java-users digest, Vol 1 #68 - 2 msgs

[Samant N. <sa...@ai...>]

Hi Stefan,
I modified your LogFactory class and solved the problem. Thanks

----- Original Message ----- 
From: <ast...@li...>
To: <ast...@li...>
Sent: Friday, August 19, 2005 1:15 PM
Subject: Asterisk-java-users digest, Vol 1 #68 - 2 msgs


> Send Asterisk-java-users mailing list submissions to
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> Today's Topics:
> 
>   1. Switch Off Debugging (Samant Nagpaul)
>   2. Re: Switch Off Debugging (Stefan Reuter)
> 
> --__--__--
> 
> Message: 1
> From: "Samant Nagpaul" <sa...@ai...>
> To: <ast...@li...>
> Date: Thu, 18 Aug 2005 16:57:01 +1000
> Subject: [Asterisk-java-users] Switch Off Debugging
> Reply-To: ast...@li...
> 
> This is a multi-part message in MIME format.
> 
> ------=_NextPart_000_05D9_01C5A415.DA2FC780
> Content-Type: text/plain;
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> 
> If there a way to switch off Log4JLogger?
> 
> Samant
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> Content-Type: text/html;
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> charset=3Diso-8859-1">
> <META content=3D"MSHTML 6.00.2900.2722" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>If there a way to&nbsp;switch off=20
> Log4JLogger?</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
> <DIV><FONT face=3DArial size=3D2>Samant</FONT></DIV></BODY></HTML>
> 
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> 
> 
> 
> --__--__--
> 
> Message: 2
> Subject: Re: [Asterisk-java-users] Switch Off Debugging
> From: Stefan Reuter <sr...@re...>
> To: ast...@li...
> Date: Thu, 18 Aug 2005 19:54:51 +0000
> Reply-To: ast...@li...
> 
> 
> --=-I/cB36/NwEsyb8t3XnlI
> Content-Type: text/plain
> Content-Transfer-Encoding: quoted-printable
> 
> On Thu, 2005-08-18 at 16:57 +1000, Samant Nagpaul wrote:
>> If there a way to switch off Log4JLogger?
> 
> If you don't include log4j on your classpath it is not used.
> If you have log4j on your classpath you can exclude asterisk-java's
> clases from logging by configuring log4j accordingly.
> 
> =3DStefan
> 
> --=-I/cB36/NwEsyb8t3XnlI
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> ejE1sLxoZjtIVM1HNQXwndY=
> =s2BI
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> 
> 
> 
> 
> --__--__--
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> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> 
> 
> End of Asterisk-java-users Digest

Re: [Asterisk-java-users] Re: Asterisk-java-users digest, Vol 1 #68 - 2 msgs

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

On Tue, 2005-08-23 at 10:05 +1000, Samant Nagpaul wrote:
> I do have the log4j package in the classpath but I dont want any logging=20
> done.
> So if I remove the "Log" and "LogFactory" classes from the asterisk-java=20
> package, will it work?

There is no need to change Asterisk-Java for that. Just configure log4j
to not log any Asterisk-Java stuff.
You can change the log level for Asterisk-Java to FATAL, that would do
the trick.
To do that just add a log4j.properties to your classpath (if it is not
there already) and include a line like

log4j.logger.net.sf.asterisk =3D FATAL

=3DStefan

[Asterisk-java-users] Re: Asterisk-java-users digest, Vol 1 #68 - 2 msgs

[Samant N. <sa...@ai...>]

Hi Stefan,
Thank you for your reply.
I do have the log4j package in the classpath but I dont want any logging 
done.
So if I remove the "Log" and "LogFactory" classes from the asterisk-java 
package, will it work?

Regards

----- Original Message ----- 
From: <ast...@li...>
To: <ast...@li...>
Sent: Friday, August 19, 2005 1:15 PM
Subject: Asterisk-java-users digest, Vol 1 #68 - 2 msgs


> Send Asterisk-java-users mailing list submissions to
> ast...@li...
>
> To subscribe or unsubscribe via the World Wide Web, visit
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> or, via email, send a message with subject or body 'help' to
> ast...@li...
>
> You can reach the person managing the list at
> ast...@li...
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-java-users digest..."
>
>
> Today's Topics:
>
>   1. Switch Off Debugging (Samant Nagpaul)
>   2. Re: Switch Off Debugging (Stefan Reuter)
>
> --__--__--
>
> Message: 1
> From: "Samant Nagpaul" <sa...@ai...>
> To: <ast...@li...>
> Date: Thu, 18 Aug 2005 16:57:01 +1000
> Subject: [Asterisk-java-users] Switch Off Debugging
> Reply-To: ast...@li...
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_05D9_01C5A415.DA2FC780
> Content-Type: text/plain;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> If there a way to switch off Log4JLogger?
>
> Samant
> ------=_NextPart_000_05D9_01C5A415.DA2FC780
> Content-Type: text/html;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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> <META http-equiv=3DContent-Type content=3D"text/html; =
> charset=3Diso-8859-1">
> <META content=3D"MSHTML 6.00.2900.2722" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>If there a way to&nbsp;switch off=20
> Log4JLogger?</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
> <DIV><FONT face=3DArial size=3D2>Samant</FONT></DIV></BODY></HTML>
>
> ------=_NextPart_000_05D9_01C5A415.DA2FC780--
>
>
>
> --__--__--
>
> Message: 2
> Subject: Re: [Asterisk-java-users] Switch Off Debugging
> From: Stefan Reuter <sr...@re...>
> To: ast...@li...
> Date: Thu, 18 Aug 2005 19:54:51 +0000
> Reply-To: ast...@li...
>
>
> --=-I/cB36/NwEsyb8t3XnlI
> Content-Type: text/plain
> Content-Transfer-Encoding: quoted-printable
>
> On Thu, 2005-08-18 at 16:57 +1000, Samant Nagpaul wrote:
>> If there a way to switch off Log4JLogger?
>
> If you don't include log4j on your classpath it is not used.
> If you have log4j on your classpath you can exclude asterisk-java's
> clases from logging by configuring log4j accordingly.
>
> =3DStefan
>
> --=-I/cB36/NwEsyb8t3XnlI
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> ejE1sLxoZjtIVM1HNQXwndY=
> =s2BI
> -----END PGP SIGNATURE-----
>
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>
>
>
>
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