asterisk-java-users — Asterisk-Java User List

[Asterisk-java-users] ExtensionStatusEvent - not receiving any from asterisk

[Aaron E. <aar...@gm...>]

Hey all,
I suppose this may be more of an asterisk question and maybe I need to
get on their mailing list, but I was just wondering if any of you have
played much with ExtensionStatusEvent objects.

Basically, I have an extension set up in asterisk and I have a hint in
the dial plan identifying the peer associated with that extension:

exten => 0179,hint,SIP/aaron-0179
exten => 0179,1,Dial(SIP/aaron-0179,20,r)
exten => 0179,2,Hangup


But when I use my handset (registered with aaron-0179) to make a call,
no ExtensionStatusEvents seem to get fired.

Anyone know if there is more to it than this? The documentation on
voip-info.org is no help...

thx,
aaron

Re: [Asterisk-java-users] CVS and SVN: what's the valid?

[lumen <lu...@bu...>]

On Fri, Sep 14, 2007 at 10:33:26AM -0400, Martin Smith send:
> SVN is what is currently used. The project was originally hosted on
> SourceForge, I think, so it's possible that CVS was used at some point
> in the past.
> 
> Upon checking, it looks like CVS is only mentioned in the 1.0 docs --
> the most recent docs only mention SVN --
> http://asterisk-java.org/development/index.html.
> 
> If you find references to CVS in the latest docs, let me know and I can
> update them.

The name of the project commits mailing list (that's not really important)
http://asterisk-java.org/0.2/mail-lists.html

Documentation about how to do a patch are based on cvs:
http://asterisk-java.org/0.2/patch.html

'Source repository' page in the 'Project Info' section:

http://asterisk-java.org/0.2/cvs-usage.html

Re: [Asterisk-java-users] CVS and SVN: what's the valid?

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

SVN is what is currently used. The project was originally hosted on
SourceForge, I think, so it's possible that CVS was used at some point
in the past.

Upon checking, it looks like CVS is only mentioned in the 1.0 docs --
the most recent docs only mention SVN --
http://asterisk-java.org/development/index.html.

If you find references to CVS in the latest docs, let me know and I can
update them.

Thanks :)

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 Joan Ginard
> Sent: Friday, September 14, 2007 10:33 AM
> To: Asterisk Java Users Mlist
> Subject: [Asterisk-java-users] CVS and SVN: what's the valid?
>=20
> Hi:
>=20
> I'm diving into the project api and source.
>=20
> First of all thank all of you for your work.
>=20
> I can see in the project page that in on place you mention a cvs
> repository and a svn repository in other place.
>=20
> Whats the valid?
>=20
> svn co http://svn.reucon.net/repos/asterisk-java/trunk asterisk-java
>=20
> cvs -d=20
> :pserver:ano...@cv...:/cvsroot/asterisk-java login=20
> cvs -z3 -d=20
> :pserver:ano...@cv...:/cvsroot/asterisk-java
> co asterisk-java
>=20
>=20
> Thanks.
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

[Asterisk-java-users] CVS and SVN: what's the valid?

[Joan G. <lu...@bu...>]

Hi:

I'm diving into the project api and source.

First of all thank all of you for your work.

I can see in the project page that in on place you mention a cvs
repository and a svn repository in other place.

Whats the valid?

svn co http://svn.reucon.net/repos/asterisk-java/trunk asterisk-java

cvs -d :pserver:ano...@cv...:/cvsroot/asterisk-java login 
cvs -z3 -d :pserver:ano...@cv...:/cvsroot/asterisk-java
co asterisk-java


Thanks.

Re: [Asterisk-java-users] how to play dtmf in agi?

[jeff l. <ll...@gm...>]

Ops.. we can use the built-in files or...

On 9/14/07, Maciek Tokarski <mlo...@gm...> wrote:
>
> Hi,
> I think that the fastest solution will be a AGI Playback command and
> playing recodred DTMF tones on a channel.
>
> 2007/9/14, jeff li <ll...@gm... >:
> >
> > Hi all,
> >
> >   Is there a way to play dtmf sound(the signal) in fast agi? coz the
> > playDtmf() in manager api needs asterisk 1.4 , I just don't  want to
> > upgrade it for the test case.
> >
> >
> >
> > --
> > Regards,
> > Jeff
> >
> > -------------------------------------------------------------------------
> > This SF.net email is sponsored by: Microsoft
> > Defy all challenges. Microsoft(R) Visual Studio 2005.
> > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> >
> >
>
>
> --
> Pozdrawiam
> Maciek Tokarski
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>


-- 
Regards,
Jeff

Re: [Asterisk-java-users] how to play dtmf in agi?

[Maciek T. <mlo...@gm...>]

Hi,
I think that the fastest solution will be a AGI Playback command and playing
recodred DTMF tones on a channel.

2007/9/14, jeff li <ll...@gm...>:
>
> Hi all,
>
>   Is there a way to play dtmf sound(the signal) in fast agi? coz the
> playDtmf() in manager api needs asterisk 1.4 , I just don't  want to
> upgrade it for the test case.
>
>
>
> --
> Regards,
> Jeff
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>


-- 
Pozdrawiam
Maciek Tokarski

[Asterisk-java-users] how to play dtmf in agi?

[jeff l. <ll...@gm...>]

Hi all,

  Is there a way to play dtmf sound(the signal) in fast agi? coz the
playDtmf() in manager api needs asterisk 1.4 , I just don't  want to upgrade
it for the test case.



-- 
Regards,
Jeff

Re: [Asterisk-java-users] parsing dial plan extension patterns

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

> -----Original Message-----
> From: ast...@li...=20
> [mailto:ast...@li...] On=20
> Behalf Of Aaron Evans
> Sent: Thursday, September 13, 2007 3:48 PM
> To: ast...@li...
> Subject: Re: [Asterisk-java-users] parsing dial plan=20
> extension patterns
>=20
> Hey Martin,
>=20
> Thanks for the info.  If I do come up with something, I will=20
> certainly share it.

Glad to help. My interest in it was to replace SIPTAPI with a Java-based
solution. I have Windows applications that only speak the Microsoft
Telephony API, and one happened to speak TAPI very poorly, and
deferenced null pointers from SIPTAPI that it shouldn't have (like when
SIPTAPI said the structure was zero bytes). I was looking to make
outgoing calls from our Windows application in this fashion:

Windows app --> GJTAPI --> Asterisk-JTAPI --> Asterisk

Unfortunately, GJTAPI mostly implemented the other way around, i.e.:

Java app --> GJTAPI --> MS TAPI --> Telephony Device

> Speaking of Asterisk JTAPI, there doesn't seem to be a lot out there
> in terms of asterisk specific examples.

Yes, I ended up going through the sources myself to find out what to do.
What it lacks in documentation, it makes up for in runtime exceptions
that complain when you don't give the right commandline arguments.

> I would like to know for instance how to implement a simple Originate
> action in JTAPI.  It is unclear to me how the terminals, addresses and
> so on map to the asterisk nomenclature.

>From what I can tell, the point of JTAPI is that (like TAPI), everyone
writes to the JTAPI API and then you don't need Asterisk-specific
examples. This is very similar to Microsoft Tapi -- you have clients
that know the API, and people who provide TAPI-like devices make a
service-provider (a TSP) that speaks the API from the other side.
=20
> Do you know of any good resources out there for JTAPI?

I think I remember discovering that I needed JTAPI 1.3 to get GCJTAPI
working, which made me go looking for the Java API files (they aren't in
Java SE). I got them at http://java.sun.com/products/jtapi/, which does
provide some HTML and PDFs about the specification. Beyond that, it was
Google and trying to run some examples and call some methods in a blank
main method :). I got it working, but discovered the chain (the one I
mentioned above) was in the wrong direction.

> Also, is it
> even compatible with asterisk-java 0.3?

JTAPI and AJ-3.0 are only related through Asterisk-JTAPI, and last I
looked, it wasn't able to use any Asterisk-Java version that was new (it
expected the org.sf... Packages from Asterisk-Java).

> I guess this is a little off topic...

Nah :)

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

Re: [Asterisk-java-users] parsing dial plan extension patterns

[Aaron E. <aar...@gm...>]

Hey Martin,

Thanks for the info.  If I do come up with something, I will certainly share it.

Speaking of Asterisk JTAPI, there doesn't seem to be a lot out there
in terms of asterisk specific examples.

I would like to know for instance how to implement a simple Originate
action in JTAPI.  It is unclear to me how the terminals, addresses and
so on map to the asterisk nomenclature.

Do you know of any good resources out there for JTAPI? Also, is it
even compatible with asterisk-java 0.3?

I guess this is a little off topic...

-aaron

On 9/13/07, Martin Smith <ma...@be...> wrote:
> Hi Aaron,
>
> I think the Asterisk JTAPI has some code that parses the dialplan and
> internally uses Asterisk-Java. Last I tried, it failed to parse the
> dialplan on one of my virtual machines, but I don't remember if that was
> Asterisk 1.4 or 1.2. I believe some of the requirements of JTAPI
> included providing information about available outgoing lines and
> detecting pre-existing calls on startup, both of which could require
> some dialplan analysis.
>
> Though I have no need for it, the idea of parsing the dialplan does
> sound intriguing to me. I would imagine others would find it useful too.
> When I first considered it (when I tried to use Asterisk JTAPI a few
> months back), I assumed the most useful part of parsing the dialplan
> would be finding syntax errors without having to fiddle with the
> dialplan on a production system.
>
> Finally, I think you'll find that Asterisk itself obviously has a parser
> for the dialplan and AEL, and that they would be trivial to port. You
> might even be able to drop those source files into a JINI wrapper and
> call them from Java, which would be a very simple solution.
>
> But yes, I'm interested. No, I have no business requirements for it, but
> do keep us informed if you attempt it :). I can't imagine there'd be a
> strong case for NOT including that kind of functionality in AJ. Besides,
> the actions to get configuration files already exist in the manager
> interface :).
>
> 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 Aaron Evans
> > Sent: Thursday, September 13, 2007 11:51 AM
> > To: ast...@li...
> > Subject: [Asterisk-java-users] parsing dial plan extension patterns
> >
> > Hi all,
> >
> > I didn't see anything in the asterisk java API for parsing dial plan
> > extension patterns and I suppose it might be beyond the scope of the
> > problems the API solves anyway.
> >
> > But I was wondering if anyone knows of any utilities out there for
> > parsing patterns.
> >
> > Specifically, I'd like to be able to parse a pattern and determine:
> >
> > 1. If the set of possible matches is bounded or not (ie.
> > fixed length).
> > 2. In the case of bounded sets, whether it is purely numeric.
> > 3. Some kind of match function to see if a given string matches the
> > pattern (perhaps by converting to a regular expression). It would be
> > cool if the match pattern could also accept a caller ID number as well
> > in case there is a filter there as well.
> > 4. In the case of bounded sets, being able to iterate through
> > the possibilities.
> >
> > I realize this is quite a laundry list. I'll need to implement
> > something myself for my specific cases if there is nothing out there
> > already...
> >
> > -aaron
> >
> > --------------------------------------------------------------
> > -----------
> > This SF.net email is sponsored by: Microsoft
> > Defy all challenges. Microsoft(R) Visual Studio 2005.
> > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> > _______________________________________________
> > Asterisk-java-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> >
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

Re: [Asterisk-java-users] parsing dial plan extension patterns

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

Hi Aaron,

I think the Asterisk JTAPI has some code that parses the dialplan and
internally uses Asterisk-Java. Last I tried, it failed to parse the
dialplan on one of my virtual machines, but I don't remember if that was
Asterisk 1.4 or 1.2. I believe some of the requirements of JTAPI
included providing information about available outgoing lines and
detecting pre-existing calls on startup, both of which could require
some dialplan analysis.

Though I have no need for it, the idea of parsing the dialplan does
sound intriguing to me. I would imagine others would find it useful too.
When I first considered it (when I tried to use Asterisk JTAPI a few
months back), I assumed the most useful part of parsing the dialplan
would be finding syntax errors without having to fiddle with the
dialplan on a production system.

Finally, I think you'll find that Asterisk itself obviously has a parser
for the dialplan and AEL, and that they would be trivial to port. You
might even be able to drop those source files into a JINI wrapper and
call them from Java, which would be a very simple solution.

But yes, I'm interested. No, I have no business requirements for it, but
do keep us informed if you attempt it :). I can't imagine there'd be a
strong case for NOT including that kind of functionality in AJ. Besides,
the actions to get configuration files already exist in the manager
interface :).

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 Aaron Evans
> Sent: Thursday, September 13, 2007 11:51 AM
> To: ast...@li...
> Subject: [Asterisk-java-users] parsing dial plan extension patterns
>=20
> Hi all,
>=20
> I didn't see anything in the asterisk java API for parsing dial plan
> extension patterns and I suppose it might be beyond the scope of the
> problems the API solves anyway.
>=20
> But I was wondering if anyone knows of any utilities out there for
> parsing patterns.
>=20
> Specifically, I'd like to be able to parse a pattern and determine:
>=20
> 1. If the set of possible matches is bounded or not (ie.=20
> fixed length).
> 2. In the case of bounded sets, whether it is purely numeric.
> 3. Some kind of match function to see if a given string matches the
> pattern (perhaps by converting to a regular expression). It would be
> cool if the match pattern could also accept a caller ID number as well
> in case there is a filter there as well.
> 4. In the case of bounded sets, being able to iterate through=20
> the possibilities.
>=20
> I realize this is quite a laundry list. I'll need to implement
> something myself for my specific cases if there is nothing out there
> already...
>=20
> -aaron
>=20
> --------------------------------------------------------------
> -----------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>=20

[Asterisk-java-users] parsing dial plan extension patterns

[Aaron E. <aar...@gm...>]

Hi all,

I didn't see anything in the asterisk java API for parsing dial plan
extension patterns and I suppose it might be beyond the scope of the
problems the API solves anyway.

But I was wondering if anyone knows of any utilities out there for
parsing patterns.

Specifically, I'd like to be able to parse a pattern and determine:

1. If the set of possible matches is bounded or not (ie. fixed length).
2. In the case of bounded sets, whether it is purely numeric.
3. Some kind of match function to see if a given string matches the
pattern (perhaps by converting to a regular expression). It would be
cool if the match pattern could also accept a caller ID number as well
in case there is a filter there as well.
4. In the case of bounded sets, being able to iterate through the possibilities.

I realize this is quite a laundry list. I'll need to implement
something myself for my specific cases if there is nothing out there
already...

-aaron

Re: [Asterisk-java-users] Timeout waiting for response to Originate

[jeff l. <ll...@gm...>]

Hi andy,

  Using the live api is ok. Thanks.

On 9/13/07, Andy Burns <and...@pr...> wrote:
>
> On 13/09/2007 04:40, jeff li wrote:
>
> >   I try make an Originate according the example in the tutorial, but  I
> > get TimeoutExcepion(immediately,surely not out) when the call is not
> > answered.
>
> >         originateResponse = managerConnection.sendAction
> (originateAction);
>
> have you tried as per the tutorial?
>
>     originateResponse = managerConnection.sendAction
> (originateAction,30000);
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>


-- 
Regards,
Jeff

Re: [Asterisk-java-users] How to play a voice file in the live api?

[jeff l. <ll...@gm...>]

Hi andy,

  Thanks, it works.


On 9/13/07, Andy Burns <and...@pr...> wrote:
>
> On 13/09/2007 06:43, jeff li wrote:
>
> >   After my application makes a originate call to a outer system, I want
> > to play something to it. In the live api , we only have playDtmf(which
> > even fails coz we're using the low version of asterisk) . What if I want
> > to play a file, a piece of music? In the manage api, no action seems to
> > be capable to do that except the CommandAction. But how do I specify
> > which channel the voice should be played to during the CommandAction?
>
> A couple of ways ...
>
> Play the sounds from the dialplan, within the context/extension you
> place the calls to.
>
> or
>
> Within the dialplan use the AGI command, which will allow invoke your
> asteriskjava service with fastagi protocol, then you can handle the
> sounds/dtmf as you like. This gives you a procedural rather than state
> table way of hadling calls.
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>


-- 
Regards,
Jeff

Re: [Asterisk-java-users] How to play a voice file in the live api?

[Andy B. <and...@pr...>]

On 13/09/2007 06:43, jeff li wrote:

>   After my application makes a originate call to a outer system, I want 
> to play something to it. In the live api , we only have playDtmf(which 
> even fails coz we're using the low version of asterisk) . What if I want 
> to play a file, a piece of music? In the manage api, no action seems to 
> be capable to do that except the CommandAction. But how do I specify 
> which channel the voice should be played to during the CommandAction?

A couple of ways ...

Play the sounds from the dialplan, within the context/extension you 
place the calls to.

or

Within the dialplan use the AGI command, which will allow invoke your 
asteriskjava service with fastagi protocol, then you can handle the 
sounds/dtmf as you like. This gives you a procedural rather than state 
table way of hadling calls.

Re: [Asterisk-java-users] Timeout waiting for response to Originate

[Andy B. <and...@pr...>]

On 13/09/2007 04:40, jeff li wrote:

>   I try make an Originate according the example in the tutorial, but  I 
> get TimeoutExcepion(immediately,surely not out) when the call is not 
> answered.

>         originateResponse = managerConnection.sendAction(originateAction);

have you tried as per the tutorial?

    originateResponse = managerConnection.sendAction(originateAction,30000);

[Asterisk-java-users] How to play a voice file in the live api?

[jeff l. <ll...@gm...>]

Hi all,

  After my application makes a originate call to a outer system, I want to
play something to it. In the live api , we only have playDtmf(which even
fails coz we're using the low version of asterisk) . What if I want to play
a file, a piece of music? In the manage api, no action seems to be capable
to do that except the CommandAction. But how do I specify which channel the
voice should be played to during the CommandAction?

-- 
Regards,
Jeff

[Asterisk-java-users] Timeout waiting for response to Originate

[jeff l. <ll...@gm...>]

Hi all,

  I try make an Originate according the example in the tutorial, but  I get
TimeoutExcepion(immediately,surely not out) when the call is not answered.
  Some codes:

   public void run() throws IOException, AuthenticationFailedException,
            TimeoutException
    {
        OriginateAction originateAction;
        ManagerResponse originateResponse;

        originateAction = new OriginateAction();
        originateAction.setChannel("SIP/mlu");
        originateAction.setContext("stdexten");
        originateAction.setExten("620");
        originateAction.setPriority(new Integer(1));
        originateAction.setTimeout(new Integer(30000));



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

//        managerConnection.addEventListener(this);

        originateResponse = managerConnection.sendAction(originateAction);

        managerConnection.logoff();
    }

And logs:

Exception in thread "main" org.asteriskjava.manager.TimeoutException:
Timeout waiting for response to Originate
        at
org.asteriskjava.manager.internal.ManagerConnectionImpl.sendAction(
ManagerConnectionImpl.java:806)
        at
org.asteriskjava.manager.internal.ManagerConnectionImpl.sendAction(
ManagerConnectionImpl.java:765)
        at org.asteriskjava.manager.DefaultManagerConnection.sendAction(
DefaultManagerConnection.java:283)
        at test.HelloManager.run(HelloManager.java:53)
        at test.HelloManager.main(HelloManager.java:110)

  I'm using asterisk 1.2, can anyone help , thanks!



-- 
Regards,
Jeff

Re: [Asterisk-java-users] Excetpion when trying use the channel of live api

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

Which version of Asterisk are you running?
Afair playDtmf is only supported in 1.4. With 1.2 you might receive this
(somewhat misleading) exception.

=3DStefan

jeff li wrote:
> Hey guys,
>=20
>   I'm not quite familiar with live api and a little confused by some
> concept in it.  What does the channel stand for(same as fastagi)? Why d=
o
> I get NoSuchChannelException when trying to use the channel when
> NewAsteriskChannel is fired.Some codes:
>=20
>     public void onNewAsteriskChannel(AsteriskChannel channel){
>         System.out.println("Caller ID: " + channel.getCallerId());
>         System.out.println ("Name: " + channel.getName());
>=20
>         try {
>           =20
> //            channel.startMonitoring("/tmp/mlu/live/test.wav");       =
I
> do find two audio files,so this one not failed
> //            channel.redirect("extensions-shanghai-sip","628",0);  Fai=
led
>               channel.playDtmf("1");                                   =
=20
>         Failed
>         } catch (ManagerCommunicationException e) {
>             // TODO Auto-generated catch block
>             e.printStackTrace();
>         } catch (NoSuchChannelException e) {
>             // TODO Auto-generated catch block
>             e.printStackTrace();
>         }
>     }
>   Logs:
> org.asteriskjava.live.NoSuchChannelException: Channel 'SIP/mlu-087c3870=
'
> is not
> available: Invalid/unknown command
>         at org.asteriskjava.live.internal.AsteriskChannelImpl.playDtmf
> (AsteriskC
> hannelImpl.java:648)
>         at test.HelloLive.onNewAsteriskChannel(HelloLive.java:106)
>         at
> org.asteriskjava.live.internal.AsteriskServerImpl.fireNewAsteriskChan
> nel(AsteriskServerImpl.java:730)
>         at
> org.asteriskjava.live.internal.ChannelManager.addNewChannel(ChannelMa
> nager.java:198)
>         at
> org.asteriskjava.live.internal.ChannelManager.handleNewChannelEvent(C
> hannelManager.java:372)
>         at
> org.asteriskjava.live.internal.AsteriskServerImpl.onManagerEvent(Aste
> riskServerImpl.java:838)
>         at
> org.asteriskjava.manager.ManagerEventListenerProxy$1.run(ManagerEvent
> ListenerProxy.java:77)
>         at
> java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExec
> utor.java:650)
>         at
> java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor
> .java:675)
>         at java.lang.Thread.run ( Thread.java:595)
>=20
>   Any suggestions?Thanks in advance.
>=20
> --=20
> Regards,
> Jeff
>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> -----------------------------------------------------------------------=
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>=20
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> _______________________________________________
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--=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] Connecting manager's ActionId to asterisk's UniqueID

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

Andy Burns wrote:
> I've now set my own call reference on the channel using
> OriginateAction.setVariables()
> and can then retrieve the value inside my AGI script with
> BaseAgiScript.getVariable()
>=20
> Is this a good approach?

yes.

--=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] How to Disable or Redirect log/debugging info

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

H=E9ctor Maldonado wrote:
> Just one simple question: Is there any way to disable messages that AJ
> sends to stdout or simply redirect these messages to a file? =20

Asterisk-Java uses commons-logging so you can easily configure what kind
of messages you want to log and where the messages should be written to.
Basically there are two options: By default commons-logging uses
java.util logging so you can have a look at the JDK documentation on how
to configure it. The second option is to put a log4j.jar on your
classpath along with a log4j.properties file to configure it.

=3DStefan

--=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] How to Disable or Redirect log/debugging info

[<ham...@gm...>]

Hello list:

Just one simple question: Is there any way to disable messages that AJ send=
s
to stdout or simply redirect these messages to a file?

I have a console application that uses asterisk java to capture Asterisk
event.  This app already send messages to the stdout but they are shown
mixed with those ones sent by asterisk java.

Thanks in advance.

Regards,

H=E9ctor.

Re: [Asterisk-java-users] Events blocked by Responses?

[Andy B. <and...@pr...>]

On 24/08/2007 09:54, Andy Burns wrote:

> I *am* passing a callback to sendaction rather than waiting with a 
> timeout, and no events are delivered to my listener method until after 
> the response is delivered to my callback method, then all the "buffered" 
> events get delivered at once.

Using OriginateAction.setAsync(true) was instumental too :-)

Re: [Asterisk-java-users] Connecting manager's ActionId to asterisk's UniqueID

[Andy B. <and...@pr...>]

On 06/09/2007 17:21, Andy Burns wrote:

> I know I can set an actionId when I place an outbound call using manager
> 
> Then the fastCGI calls my script, and from the request I can retrieve 
> the uniqueID, but by that time I have had not chance to establish a link 
> back to my actionId so am not sure which of several concurrent calls 
> each script is handling.

I've now set my own call reference on the channel using 
OriginateAction.setVariables()
and can then retrieve the value inside my AGI script with 
BaseAgiScript.getVariable()

Is this a good approach?

[Asterisk-java-users] Excetpion when trying use the channel of live api

[jeff l. <ll...@gm...>]

Hey guys,

  I'm not quite familiar with live api and a little confused by some concept
in it.  What does the channel stand for(same as fastagi)? Why do I get
NoSuchChannelException when trying to use the channel when
NewAsteriskChannel is fired.Some codes:

    public void onNewAsteriskChannel(AsteriskChannel channel){
        System.out.println("Caller ID: " + channel.getCallerId());
        System.out.println ("Name: " + channel.getName());

        try {

//            channel.startMonitoring("/tmp/mlu/live/test.wav");       I do
find two audio files,so this one not failed
//            channel.redirect("extensions-shanghai-sip","628",0);  Failed
              channel.playDtmf("1");
    Failed
        } catch (ManagerCommunicationException e) {
            // TODO Auto-generated catch block
            e.printStackTrace();
        } catch (NoSuchChannelException e) {
            // TODO Auto-generated catch block
            e.printStackTrace();
        }
    }
  Logs:
org.asteriskjava.live.NoSuchChannelException: Channel 'SIP/mlu-087c3870' is
not
available: Invalid/unknown command
        at org.asteriskjava.live.internal.AsteriskChannelImpl.playDtmf(AsteriskC
hannelImpl.java:648)
        at test.HelloLive.onNewAsteriskChannel(HelloLive.java:106)
        at
org.asteriskjava.live.internal.AsteriskServerImpl.fireNewAsteriskChan
nel(AsteriskServerImpl.java:730)
        at org.asteriskjava.live.internal.ChannelManager.addNewChannel
(ChannelMa
nager.java:198)
        at
org.asteriskjava.live.internal.ChannelManager.handleNewChannelEvent(C
hannelManager.java:372)
        at org.asteriskjava.live.internal.AsteriskServerImpl.onManagerEvent
(Aste
riskServerImpl.java:838)
        at org.asteriskjava.manager.ManagerEventListenerProxy$1.run
(ManagerEvent
ListenerProxy.java:77)
        at java.util.concurrent.ThreadPoolExecutor$Worker.runTask
(ThreadPoolExec
utor.java:650)
        at java.util.concurrent.ThreadPoolExecutor$Worker.run
(ThreadPoolExecutor
.java:675)
        at java.lang.Thread.run( Thread.java:595)

  Any suggestions?Thanks in advance.

-- 
Regards,
Jeff

[Asterisk-java-users] Connecting manager's ActionId to asterisk's UniqueID

[Andy B. <and...@pr...>]

I'm using the manager interface to place outbound calls, with the 
internal leg of the call placed to an extension where they will be 
handled by fastCGI as an IVR, the bones of this is working nicely.

I know I can set an actionId when I place an outbound call using manager

The successful originate response has my actionId, but has a null 
uniqueId so I can't establish a link at that point.

When I get the NewChannel/NewState/NewExten events they do have the 
uniqueID, but not my actionID, so I can't establish the link at that 
point either.

Then the fastCGI calls my script, and from the request I can retrieve 
the uniqueID, but by that time I have had not chance to establish a link 
back to my actionId so am not sure which of several concurrent calls 
each script is handling.

Any suggestions? Have I missed a trick in the asteriskjava 
documentation, or do I need to start passing extra variables through the 
dialplan?