asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] fax event question

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

Stefan,

Thanks for the help.  I have patched app_txfax and created the class,  
but I have a couple of questions.

First, I used the FaxReceivedEvent as a template for creating  
FaxSentEvent.  There is a UID variable in the class:

     private static final long serialVersionUID =  
-1409738380177538949L; // this is from FaxReceivedEvent

I assume this is a unique value.  My question is, where do I get a  
value for this variable?

Second, I wanted to verify how to register the class with  
ManagerConnection.  I edited EventBuilderImpl.java to import and  
register the class.  Is there anything else I need to do?

Jonathan

On Oct 7, 2006, at 1:58 PM, Stefan Reuter wrote:

> Jonathan Augenstine wrote:
>> I have a question about manager fax events.  I see documented an
>> event that notifies the arrival of an incoming fax.  Is there a event
>> to notify that txfax has successfully sent a fax (am I just missing
>> it in the documentation)?
>
> No app_txfax does not currently send an event in that case.
>
>> If not, would it be difficult to implement
>> this feature?
>
> No not really. It involves patching app_txfax, creating a class for  
> the
> event an registering it with the ManagerConnection.
>
> =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...
>
> ---------------------------------------------------------------------- 
> ---
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to  
> share your
> opinions on IT & business topics through brief surveys -- and earn  
> cash
> http://www.techsay.com/default.php? 
> page=join.php&p=sourceforge&CID=DEVDEV________________________________ 
> _______________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] help setting properties file in fastagi

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

Yelson Vivas wrote:
> Hi Guys
> I'm developing a fastagi program using asterisk-java 0.2 version and i
> need to change the port and pool size, but i don't know how, can
> somebody give a example about how does it look to add
> fastagi.properties file on the classpath.

just put it into the same directory as your fastagi-mapping.properties.

=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...

Re: [Asterisk-java-users] fax event question

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

Jonathan Augenstine wrote:
> I have a question about manager fax events.  I see documented an =20
> event that notifies the arrival of an incoming fax.  Is there a event  =

> to notify that txfax has successfully sent a fax (am I just missing =20
> it in the documentation)?=20

No app_txfax does not currently send an event in that case.

> If not, would it be difficult to implement =20
> this feature?

No not really. It involves patching app_txfax, creating a class for the
event an registering it with the ManagerConnection.

=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...

[Asterisk-java-users] fax event question

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

I have a question about manager fax events.  I see documented an  
event that notifies the arrival of an incoming fax.  Is there a event  
to notify that txfax has successfully sent a fax (am I just missing  
it in the documentation)?  If not, would it be difficult to implement  
this feature?

Jonathan

[Asterisk-java-users] help setting properties file in fastagi

[Yelson V. <yv...@gm...>]

Hi Guys
I'm developing a fastagi program using asterisk-java 0.2 version and i
need to change the port and pool size, but i don't know how, can
somebody give a example about how does it look to add
fastagi.properties file on the classpath.
Thanks for your help
Br
Yelson

-- 
Yelson E Vivas C
MPC
(571) 6500-800

Re: [Asterisk-java-users] MP3Player via AGI Interface?

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

Moxie Marlinspike wrote:

> Has anyone dealt with this before, and does anyone know if there's any
> kind of workaround for this?  Either a way to install an mp3 codec so
> that asterisk will play mp3s from streamFile(),
iirc there once was an mp3 codec and format but both were removed due to
licensing issues.

> or a way to make
> asterisk run MP3Player() via fastagi?
you can run any asterisk application through exec()

=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...

[Asterisk-java-users] MP3Player via AGI Interface?

[Moxie M. <mo...@th...>]

I'm trying to stream an mp3 file via the asterisk-java FastAGI
interface, but I'm having some trouble.  It seems like if I were just
using the Asterisk dial-plan directly, I could use the MP3Player()
command -- however there doesn't seem to be an analog for the AGI
interface?  The closest thing is the streamFile() command, but I think
that only works for codecs that are installed into asterisk.

Has anyone dealt with this before, and does anyone know if there's any
kind of workaround for this?  Either a way to install an mp3 codec so
that asterisk will play mp3s from streamFile(), or a way to make
asterisk run MP3Player() via fastagi?

Thanks,

- moxie

-- 
Personal Site:         http://www.thoughtcrime.org
Audio Anarchy:         http://www.audioanarchy.org
Anarchist Yacht Clubb: http://www.blueanarchy.org

Re: [Asterisk-java-users] Getting started

[Moxie M. <mo...@th...>]

Hey Eric, seems like you've already discovered the tutorial.  If you
want to use the manager API, just follow the directions under "The
Manager API" section for configuring Asterisk and writing your code.

- moxie

--
Personal Site:         http://www.thoughtcrime.org
Audio Anarchy:         http://www.audioanarchy.org
Anarchist Yacht Clubb: http://www.blueanarchy.org


Eric C. wrote:
> What is required to get started using java with Asterisk?
> So far, I've installed Asterisk on Linux and it is working fine.
> I have not edited any Asterisk files such as extensions.conf, sip.conf or 
> manager.conf.
> 
> Can I just write a java class and pass these parameters in or do I have to 
> do something else?
> 
> I would like to use the HelloManager example as show on this page: 
> http://asterisk-java.org/0.3-m1/tutorial.html
> 
> Please advise.
> Thanks,
> Eric.
> 
> 
> 
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share your
> opinions on IT & business topics through brief surveys -- and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
> 

[Asterisk-java-users] Getting started

[Eric C. <zip...@ho...>]

What is required to get started using java with Asterisk?
So far, I've installed Asterisk on Linux and it is working fine.
I have not edited any Asterisk files such as extensions.conf, sip.conf or 
manager.conf.

Can I just write a java class and pass these parameters in or do I have to 
do something else?

I would like to use the HelloManager example as show on this page: 
http://asterisk-java.org/0.3-m1/tutorial.html

Please advise.
Thanks,
Eric.

[Asterisk-java-users] PHoexARMA

[Aranrhod H. <au...@fi...>]

Good day,

VALttIUM

VIAttGRA

CIAttLIS

AMBttIEN

Save 50 % with http://www.defunkionmdesunjade.com
=20
  _____ =20


Grrr! it growled and retracted its legs, then zipped off at a great
outside influences. So it hadnt been emotion but plain old chemistry
The automated kitchen produced another stale sandwich, the machine

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Well, I learned a lot of Local channel behaviour.
I thought it would be ok to share, let me know if this is not ok with
this list.

Local channel origination does indeed what I expected, but in a cool
(read non-obvious to me) way. It actually creates 2 channels which work
sort of pty/ttyp pair (master/slave) where the master connects to
whatever extension you provide as channel id (i.e. Local/exten@context)
and once the master is up, the slave does the origination.
When the slave gets into an UP state, some channel renaming happens so
the Local pair vanishes.

For my need (automatic origination to real external number and then
connect to local app) the needed steps are:
0) originate from Local/<realnumber>@<context> to <local extension>
1) locate Local session being used, it's a prefix for the two
	local channels, with suffix ",1" for master and ",2" for slave
2) track UniqueIDs for slave (not using it now)
3) locate "real" (remote) channel being instantiated for call
4) track remote state

The piece of trick is to use a fake callerID in the originate to be able
to locate the channels (mind there might be lots of calls going at once)
The originate returns as soon as the master local channel is UP.

In my test version (1.2.0beta) originating from
Local/<realnumber>@<context> does lock the manager connection (i.e. no
other events are reported untill originate succeeds or fails, so
originating from a "fake" works always extension and redirecting to the
app once the slave connects to the real number sounds a better thing to do.

Demo code:
...
        originateAction = new OriginateAction();
        originateAction.setChannel("Local/1000@local");
        originateAction.setContext("local");
        originateAction.setExten("8502");
        originateAction.setPriority(new Integer(1));
        originateAction.setCallerId("call1000");
        originateAction.setTimeout(new Integer(15000));

        // connect to Asterisk and log in
        managerConnection.login();
        managerConnection.addEventListener(new
ManagerEventListenerProxy(this));
	originateResponse = managerConnection.sendAction(originateAction, 30000);
...

   String myCallerId = "call1000";
    String myLocalChannel = null;
    String myRemoteChannel = null;
    String myUid1 = null; //Local master side
    String myUid2 = null; //Local slave side
    String myUid3 = null; //Remote

    public void onManagerEvent(ManagerEvent event) {
        if (myLocalChannel == null
        	&& event.getClass().equals(NewCallerIdEvent.class)) {
        	if
(((NewCallerIdEvent)event).getCallerIdName().equals(myCallerId)) {
        		// #1: locate our base channel name, i.e., Local session (prefix)
        		myLocalChannel = ((NewCallerIdEvent)event).getChannel();
        		if (myLocalChannel.length() > 2)
        			myLocalChannel = myLocalChannel.substring(0,
myLocalChannel.length()-2);
        		myUid1 = ((NewCallerIdEvent)event).getUniqueId();
        		System.out.println("#1: " + myLocalChannel + "/" + myUid1);
        	}
        } else if (myLocalChannel != null
        			&& myUid2 == null
        			&& event.getClass().equals(NewStateEvent.class)) {
        	if
(((NewStateEvent)event).getChannel().equals(myLocalChannel+",2")) {
        		// #2: determine slave Uid
        		myUid2 = ((NewStateEvent)event).getUniqueId();
        		System.out.println("#2: " + myLocalChannel + "/" + myUid2);
        	}
        } else if (myLocalChannel != null
    			&& myUid3 == null
    			&& event.getClass().equals(NewChannelEvent.class)) {
        	if (((NewChannelEvent)event).getCallerIdName().equals(myCallerId)
        			&&
!((NewChannelEvent)event).getChannel().startsWith(myLocalChannel)) {
        		// #3: locate our remote channel name
        		myRemoteChannel = ((NewChannelEvent)event).getChannel();
        		myUid3 = ((NewChannelEvent)event).getUniqueId();
        		System.out.println("#3: " + myRemoteChannel + "/" + myUid3);
        		System.out.println(((NewChannelEvent)event).getState());
        	}
        } else if (myUid3 != null) {
        	if (event.getClass().equals(NewChannelEvent.class)
        			&& ((NewChannelEvent)event).getUniqueId().equals(myUid3)) {
        		// #4: Change of state
        		System.out.println(((NewChannelEvent)event).getState());
        	} else if (event.getClass().equals(NewStateEvent.class)
        			&& ((NewStateEvent)event).getUniqueId().equals(myUid3)) {
        		// #4: Change of state
        		System.out.println(((NewStateEvent)event).getState());
        	}
        }
    }


Stefan Reuter @ 29/09/2006 11:58 -0300 dixit:
> Carlos G Mendioroz wrote:
>> Yes, and it sort of works, but I don't get access to the failure
>> conditions. (i.e. Busy becomes channel unavailable) :(
>>
>> Hmm, I thought it was going to be an easy one ...
> 
> no. detecting the cause for a failure is one of the most complicated
> things with the originate stuff and the current implementation in
> Asterisk-Java is probably far from excellent in this regard.
> You can do me a favor and provide me with a detailed description of the
> situations (i.e. what you did in your code and whether the caller/callee
> is busy an), a dump of the events received through the Manager API (see
> http://asterisk-java.org/page/aj?entry=debugging_manager_api) and the
> logs. Then I can further investigate and possibly enhance the detection.
> 
> =Stefan
> 
> 
> 
> ------------------------------------------------------------------------
> 
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share your
> opinions on IT & business topics through brief surveys -- and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Stefan,
I've been testing this, mainly using originateToExtensionAsync()
because it's easier to tell apart what's going on.
"success" is returned as soon as the first (originating) channel
is instantiated.

Also, using Local channels to originate is not giving any real reason
for the error, I guess because the channel is lost right away.

I did many test extensions like:
; Test answer and hold
exten => 8506,1,Answer
exten => 8506,n,Wait,300
; Test N/A
exten=> 8507,1,Ringing()
exten=> 8507,n,Wait(60)
; Test congestion
exten=> 8508,1,Congestion()
; Test busy
exten=> 8509,1,Busy()

and by calling from/to different ones it's easy to see the behaviour.

I will fall back to manager API (instead of live) and see if I can
understand how to do it.

-Carlos

Stefan Reuter @ 29/09/2006 11:58 -0300 dixit:
> Carlos G Mendioroz wrote:
>> Yes, and it sort of works, but I don't get access to the failure
>> conditions. (i.e. Busy becomes channel unavailable) :(
>>
>> Hmm, I thought it was going to be an easy one ...
> 
> no. detecting the cause for a failure is one of the most complicated
> things with the originate stuff and the current implementation in
> Asterisk-Java is probably far from excellent in this regard.
> You can do me a favor and provide me with a detailed description of the
> situations (i.e. what you did in your code and whether the caller/callee
> is busy an), a dump of the events received through the Manager API (see
> http://asterisk-java.org/page/aj?entry=debugging_manager_api) and the
> logs. Then I can further investigate and possibly enhance the detection.
> 
> =Stefan
> 
> 
> 
> ------------------------------------------------------------------------
> 
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share your
> opinions on IT & business topics through brief surveys -- and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Stefan Reuter @ 29/09/2006 11:58 -0300 dixit:
> Carlos G Mendioroz wrote:
>> Yes, and it sort of works, but I don't get access to the failure
>> conditions. (i.e. Busy becomes channel unavailable) :(
>>
>> Hmm, I thought it was going to be an easy one ...
> 
> no. detecting the cause for a failure is one of the most complicated
> things with the originate stuff and the current implementation in
> Asterisk-Java is probably far from excellent in this regard.
> You can do me a favor and provide me with a detailed description of the
> situations (i.e. what you did in your code and whether the caller/callee
> is busy an), a dump of the events received through the Manager API (see
> http://asterisk-java.org/page/aj?entry=debugging_manager_api) and the
> logs. Then I can further investigate and possibly enhance the detection.
> 
> =Stefan

Ok, no problem, I'll try to do that, but first let me get clear which
way "should" I be doing this:

The originate command gets a channel and a destination, which could be a
context/extension or an application.
My idea was that the reporting would be about the action involved in
doing this connection and not in the creation of the involved channel.

But in fact, I'm interested in the "real" call leg and the "loopback"
was just that, a virtual channel to make the framework work.

So I'm going to do:
	originateToExtension("Local/dest@dialout","dialing","0",1,timeout);
redirect("service", "srv_no", 1);

Or use the asynch model and get the different completion states. Right ?

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Carlos G Mendioroz wrote:
> Yes, and it sort of works, but I don't get access to the failure
> conditions. (i.e. Busy becomes channel unavailable) :(
>=20
> Hmm, I thought it was going to be an easy one ...

no. detecting the cause for a failure is one of the most complicated
things with the originate stuff and the current implementation in
Asterisk-Java is probably far from excellent in this regard.
You can do me a favor and provide me with a detailed description of the
situations (i.e. what you did in your code and whether the caller/callee
is busy an), a dump of the events received through the Manager API (see
http://asterisk-java.org/page/aj?entry=3Ddebugging_manager_api) and the
logs. Then I can further investigate and possibly enhance the detection.

=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...

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Yes, and it sort of works, but I don't get access to the failure
conditions. (i.e. Busy becomes channel unavailable) :(

Hmm, I thought it was going to be an easy one ...
-Carlos



Stefan Reuter @ 29/09/2006 08:11 -0300 dixit:
> Did you try it the other way round, i.e.
> originateToExtension("Local/dest_no@dialout","dialing","0",1,25000);
> 
> This would call dest_no and when answered connect to your extension.
> 
> =Stefan
> 
> Carlos G Mendioroz wrote:
>> Hi,
>> I'm new to the manager interface, and I'm stumbling on how to do
>> something that I thought was going to be much easier :(
>>
>> I've to dial a list of numbers and pass a message on connecting.
>> I have to audit # of calls, connetions and failures.
>>
>> After some thinking, I was thinking about "originatingToExtension"
>> from a "virtual" channel and on answering, redirecting that to
>> the extension which is head of the message/service.
>>
>> Like:
>> <extensions.conf>
>> ...
>> [dialing]
>> exten => 0,1,Answer
>> exten => 0,n,Wait,60
>>
>> and
>> originateToExtension("Local/0@dialing","dialout","<dest_no>",1,25000);
>>
>> But this is not behaving as I expected. It returns before the call is
>> connected, failures are not correctly typed, etc.
>>
>> What is the right way (or a working one) of doing this ?
>>
>> -Carlos
> 
> 
> 
> ------------------------------------------------------------------------
> 
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share your
> opinions on IT & business topics through brief surveys -- and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] Originating calls for automatic messaging

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

Did you try it the other way round, i.e.
originateToExtension("Local/dest_no@dialout","dialing","0",1,25000);

This would call dest_no and when answered connect to your extension.

=3DStefan

Carlos G Mendioroz wrote:
> Hi,
> I'm new to the manager interface, and I'm stumbling on how to do
> something that I thought was going to be much easier :(
>=20
> I've to dial a list of numbers and pass a message on connecting.
> I have to audit # of calls, connetions and failures.
>=20
> After some thinking, I was thinking about "originatingToExtension"
> from a "virtual" channel and on answering, redirecting that to
> the extension which is head of the message/service.
>=20
> Like:
> <extensions.conf>
> ...
> [dialing]
> exten =3D> 0,1,Answer
> exten =3D> 0,n,Wait,60
>=20
> and
> originateToExtension("Local/0@dialing","dialout","<dest_no>",1,25000);
>=20
> But this is not behaving as I expected. It returns before the call is
> connected, failures are not correctly typed, etc.
>=20
> What is the right way (or a working one) of doing this ?
>=20
> -Carlos


--=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...

Re: [Asterisk-java-users] No such channel?

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

Your logs say

chan_sip.c: Forbidden - wrong password on authentication for INVITE to
'"Unknown" <sip:052...@si...>;tag=3Das63b159c3'

so that seems to be your problem :)

=3DStefan

Javier V=E1zquez wrote:
> I found a PHP-script that does exactly what I'm trying to do with my=20
> Java-application. I implemented it and it still doesn't work. Maybe the=
=20
> PHP-code and the log-output help getting to the root of my problem.
>=20
> PHP-code (look at chapter 5, resp. webcall2.php. it's in german, but th=
e=20
> php-code should be readable anyway ;-)):
> http://www.voipsurfer.de//phpBB2//viewtopic.php?t=3D258
>=20
> Here is the log-output:
> http://195.141.78.164/webcall_log.txt
>=20
> -----------------------------------------------------------------------=
--
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share=
 your
> opinions on IT & business topics through brief surveys -- and earn cash=

> http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D=
DEVDEV
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users


--=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...

[Asterisk-java-users] No such channel?

[<vaz...@zh...>]

I found a PHP-script that does exactly what I'm trying to do with my 
Java-application. I implemented it and it still doesn't work. Maybe the 
PHP-code and the log-output help getting to the root of my problem.

PHP-code (look at chapter 5, resp. webcall2.php. it's in german, but the 
php-code should be readable anyway ;-)):
http://www.voipsurfer.de//phpBB2//viewtopic.php?t=258

Here is the log-output:
http://195.141.78.164/webcall_log.txt

[Asterisk-java-users] Originating calls for automatic messaging

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

Hi,
I'm new to the manager interface, and I'm stumbling on how to do
something that I thought was going to be much easier :(

I've to dial a list of numbers and pass a message on connecting.
I have to audit # of calls, connetions and failures.

After some thinking, I was thinking about "originatingToExtension"
from a "virtual" channel and on answering, redirecting that to
the extension which is head of the message/service.

Like:
<extensions.conf>
...
[dialing]
exten => 0,1,Answer
exten => 0,n,Wait,60

and
originateToExtension("Local/0@dialing","dialout","<dest_no>",1,25000);

But this is not behaving as I expected. It returns before the call is
connected, failures are not correctly typed, etc.

What is the right way (or a working one) of doing this ?

-Carlos
-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] No such channel?

[<vaz...@zh...>]

to illustrate my problem, you'll find the log-output of a call here:
http://195.141.78.164/logfile.txt

here is the (new) context i use:
http://195.141.78.164/extensions_conf.txt

this is the sip-config of my voip-provider:
http://195.141.78.164/sip_conf.txt

right now I'm using the originate-function this way:

originateToExtension("SIP/XXX@CityTel", "kronos", "SIP/YYY", 1, 30000);

...where XXX = caller-number and YYY = callee-number. different syntax 
as in the mail i sent before, but the problem stays the same. :-)

thanks a lot for your kind help!

> > Couldn't find channel: org.asteriskjava.live.NoSuchChannelException:  
> > Channel 'SIP/CityTel/<caller-number>' is not available.
>   
can you show us what you see on the asterisk console (with set verbose
9) when this happens?

> > Does that mean, that this doesn't work with two EXTERNAL numbers or  
> > is there another problem I don't see?
>   
Nope connecting two external numbers is no problem.

> > Probably more an AGI then an Asterisk-Java-Question...
>   
You are not using AGI but the Manager API.

=Stefan

Re: [Asterisk-java-users] AsteriskQueue Interface

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

Following myself:
I kind of solved the out/in issue via dialplan programming using
automatic pausing of agents when receiving calls direct or generating
calls, but I don't like the idea of the library functionality being
dependent on certain dialplan logic.

Also, elaborating on AsteriskQueue information, it would be nice to have
counters by "buckets" like answered in less than 5, 5-15, 15-30, 30-60,
60-120, 120-300, 300+ or whatever scale.
Missing from before is abandon counters and bucket counters too.
Longest talk time and hold time with timestamps ?

Buckets are hard to do w/o some backing store, so I guess some of this
will have to go to a special implementation and not to standard library.

-Carlos

Carlos G Mendioroz @ 28/09/2006 08:56 -0300 dixit:
> Hi,
> I'm also looking into this...
> 
> A start could be defining the AsteriskAgent model data ?
> -ID
> -Associated extension (as reported by Loginchan)
> -Associated channel (as in SIP/XXX used for login)
> -Mode (call back/dedicated)
> -Status (inbound/outbound/free/<queueName>)
> (I'm a litle confused though because of the capacity of an agent to
> have more than one call at one time..., invalidating this approach,
> but I'm trying to have some way of mapping agent to logical extension)
> 
> -Since (i.e. time at which entered current state)
> 
> and an association from AsteriskQueue to AsteriskAgent,
> which needs also some data:
> -Membership
> -CallsTaken
> -Penalty
> -LastCall
> -Paused
> 
> AsteriskQueue could be extended to have some running averages like
> avgHold, avgConnTime.
> 
> Thoughts ?
> I hope my abrupt chime in is ok :)
> -Carlos
> 
> Chris Howard @ 27/09/2006 10:24 -0300 dixit:
>> Stefan Reuter wrote:
>>> Hey Chris,
>>>
>>>   
>>>> 1. Number of queue members logged into the queue.
>>>> 2. Number of queue members available to take calls
>>>> 3. Number of channels currently waiting in the queue (looks like 
>>>> AsteriskQueue will do this one)
>>>> 4. Average wait time.
>>>>     
>>> Yes these sound like a good fit for AsteriskQueue.
>>>
>>>   
>>>> It looks like the AsteriskQueue interface is a good start for this but 
>>>> I'm not sure if it's even possible to get the other information without 
>>>> parsing through a big list of queuemember responses.
>>>>     
>>> There already is an empty method in live.internal.QueueManager called
>>> handleQueueMemberEvent() that is called on initialization, where we
>>> could put some meat in.
>>> A new live object QueueMember (or AsteriskAgent) will be needed to
>>> represent agents.
>>> The interesting part starts after initialization, i.e. tracking agent
>>> events and updating the model. The relevant ones seem to be
>>> - AgentLoginEvent
>>> - AgentLogoffEvent
>>> - AgentCallbackLoginEvent
>>> - AgentCallbackLogoffEvent
>>> To associate agents with the callers they are serving (and to see
>>> whether they are busy or not) looking for AgentCalledEvents will be
>>> needed as well as observing the associated channel for hangup or
>>> redirection so we know when the agent is available again.
>>>
>>> Once that has been done some new methods could be added to AsteriskQueue
>>> and AsteriskAgent like support for easy login/logout, etc.
>>>   
>> We are doing something similar in our queue management application 
>> (external from Asterisk-Java).  Getting the initial state is fun as you 
>> probably already have callers in the queue and agents on the phone.  I'm 
>> willing to invest some resources in this as it would make my life much 
>> easier to keep this info in Aasterisk-Java and make my presentation 
>> piece more light weight.  Where do we start?
>>
>>>   
>>>> I'm thinking about 
>>>> adding an action to asterisk QueueStatSummaryRequest that will take as 
>>>> an argument the queue name and return the above info as a response (I 
>>>> really hate the way asterisk tries to send events as a response).  Any 
>>>> thoughts?
>>>>     
>>> I hope that this is not needed because the idea of the live package is
>>> to observe asterisk events and update an in memory model of asterisks
>>> state so you dont have to talk to asterisk each time you query the state.
>>>   
>> For our immediate needs we created a QueueSummary action that returns 
>> the info needed.  This patch has already been accepted in trunk.
>> http://bugs.digium.com/view.php?id=8035
>>
> 

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] AsteriskQueue Interface

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

Following myself:
I kind of solved the out/in issue via dialplan programming using
automatic pausing of agents when receiving calls direct or generating
calls, but I don't like the idea of the library functionality being
dependent on certain dialplan logic.

Also, elaborating on AsteriskQueue information, it would be nice to have
counters by "buckets" like answered in less than 5, 5-15, 15-30, 30-60,
60-120, 120-300, 300+ or whatever scale.
Missing from before is abandon counters and bucket counters too.
Longest talk time and hold time with timestamps ?

Buckets are hard to do w/o some backing store, so I guess some of this
will have to go to a special implementation and not to standard library.

-Carlos

Carlos G Mendioroz @ 28/09/2006 08:56 -0300 dixit:
> Hi,
> I'm also looking into this...
> 
> A start could be defining the AsteriskAgent model data ?
> -ID
> -Associated extension (as reported by Loginchan)
> -Associated channel (as in SIP/XXX used for login)
> -Mode (call back/dedicated)
> -Status (inbound/outbound/free/<queueName>)
> (I'm a litle confused though because of the capacity of an agent to
> have more than one call at one time..., invalidating this approach,
> but I'm trying to have some way of mapping agent to logical extension)
> 
> -Since (i.e. time at which entered current state)
> 
> and an association from AsteriskQueue to AsteriskAgent,
> which needs also some data:
> -Membership
> -CallsTaken
> -Penalty
> -LastCall
> -Paused
> 
> AsteriskQueue could be extended to have some running averages like
> avgHold, avgConnTime.
> 
> Thoughts ?
> I hope my abrupt chime in is ok :)
> -Carlos
> 
> Chris Howard @ 27/09/2006 10:24 -0300 dixit:
>> Stefan Reuter wrote:
>>> Hey Chris,
>>>
>>>   
>>>> 1. Number of queue members logged into the queue.
>>>> 2. Number of queue members available to take calls
>>>> 3. Number of channels currently waiting in the queue (looks like 
>>>> AsteriskQueue will do this one)
>>>> 4. Average wait time.
>>>>     
>>> Yes these sound like a good fit for AsteriskQueue.
>>>
>>>   
>>>> It looks like the AsteriskQueue interface is a good start for this but 
>>>> I'm not sure if it's even possible to get the other information without 
>>>> parsing through a big list of queuemember responses.
>>>>     
>>> There already is an empty method in live.internal.QueueManager called
>>> handleQueueMemberEvent() that is called on initialization, where we
>>> could put some meat in.
>>> A new live object QueueMember (or AsteriskAgent) will be needed to
>>> represent agents.
>>> The interesting part starts after initialization, i.e. tracking agent
>>> events and updating the model. The relevant ones seem to be
>>> - AgentLoginEvent
>>> - AgentLogoffEvent
>>> - AgentCallbackLoginEvent
>>> - AgentCallbackLogoffEvent
>>> To associate agents with the callers they are serving (and to see
>>> whether they are busy or not) looking for AgentCalledEvents will be
>>> needed as well as observing the associated channel for hangup or
>>> redirection so we know when the agent is available again.
>>>
>>> Once that has been done some new methods could be added to AsteriskQueue
>>> and AsteriskAgent like support for easy login/logout, etc.
>>>   
>> We are doing something similar in our queue management application 
>> (external from Asterisk-Java).  Getting the initial state is fun as you 
>> probably already have callers in the queue and agents on the phone.  I'm 
>> willing to invest some resources in this as it would make my life much 
>> easier to keep this info in Aasterisk-Java and make my presentation 
>> piece more light weight.  Where do we start?
>>
>>>   
>>>> I'm thinking about 
>>>> adding an action to asterisk QueueStatSummaryRequest that will take as 
>>>> an argument the queue name and return the above info as a response (I 
>>>> really hate the way asterisk tries to send events as a response).  Any 
>>>> thoughts?
>>>>     
>>> I hope that this is not needed because the idea of the live package is
>>> to observe asterisk events and update an in memory model of asterisks
>>> state so you dont have to talk to asterisk each time you query the state.
>>>   
>> For our immediate needs we created a QueueSummary action that returns 
>> the info needed.  This patch has already been accepted in trunk.
>> http://bugs.digium.com/view.php?id=8035
>>
> 

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

Re: [Asterisk-java-users] AsteriskQueue Interface

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

Hi,
I'm also looking into this...

A start could be defining the AsteriskAgent model data ?
-ID
-Associated extension (as reported by Loginchan)
-Associated channel (as in SIP/XXX used for login)
-Mode (call back/dedicated)
-Status (inbound/outbound/free/<queueName>)
(I'm a litle confused though because of the capacity of an agent to
have more than one call at one time..., invalidating this approach,
but I'm trying to have some way of mapping agent to logical extension)

-Since (i.e. time at which entered current state)

and an association from AsteriskQueue to AsteriskAgent,
which needs also some data:
-Membership
-CallsTaken
-Penalty
-LastCall
-Paused

AsteriskQueue could be extended to have some running averages like
avgHold, avgConnTime.

Thoughts ?
I hope my abrupt chime in is ok :)
-Carlos

Chris Howard @ 27/09/2006 10:24 -0300 dixit:
> Stefan Reuter wrote:
>> Hey Chris,
>>
>>   
>>> 1. Number of queue members logged into the queue.
>>> 2. Number of queue members available to take calls
>>> 3. Number of channels currently waiting in the queue (looks like 
>>> AsteriskQueue will do this one)
>>> 4. Average wait time.
>>>     
>> Yes these sound like a good fit for AsteriskQueue.
>>
>>   
>>> It looks like the AsteriskQueue interface is a good start for this but 
>>> I'm not sure if it's even possible to get the other information without 
>>> parsing through a big list of queuemember responses.
>>>     
>> There already is an empty method in live.internal.QueueManager called
>> handleQueueMemberEvent() that is called on initialization, where we
>> could put some meat in.
>> A new live object QueueMember (or AsteriskAgent) will be needed to
>> represent agents.
>> The interesting part starts after initialization, i.e. tracking agent
>> events and updating the model. The relevant ones seem to be
>> - AgentLoginEvent
>> - AgentLogoffEvent
>> - AgentCallbackLoginEvent
>> - AgentCallbackLogoffEvent
>> To associate agents with the callers they are serving (and to see
>> whether they are busy or not) looking for AgentCalledEvents will be
>> needed as well as observing the associated channel for hangup or
>> redirection so we know when the agent is available again.
>>
>> Once that has been done some new methods could be added to AsteriskQueue
>> and AsteriskAgent like support for easy login/logout, etc.
>>   
> We are doing something similar in our queue management application 
> (external from Asterisk-Java).  Getting the initial state is fun as you 
> probably already have callers in the queue and agents on the phone.  I'm 
> willing to invest some resources in this as it would make my life much 
> easier to keep this info in Aasterisk-Java and make my presentation 
> piece more light weight.  Where do we start?
> 
>>   
>>> I'm thinking about 
>>> adding an action to asterisk QueueStatSummaryRequest that will take as 
>>> an argument the queue name and return the above info as a response (I 
>>> really hate the way asterisk tries to send events as a response).  Any 
>>> thoughts?
>>>     
>> I hope that this is not needed because the idea of the live package is
>> to observe asterisk events and update an in memory model of asterisks
>> state so you dont have to talk to asterisk each time you query the state.
>>   
> For our immediate needs we created a QueueSummary action that returns 
> the info needed.  This patch has already been accepted in trunk.
> http://bugs.digium.com/view.php?id=8035
> 

-- 
Carlos G Mendioroz  <tr...@ac...>

Re: [Asterisk-java-users] AsteriskQueue Interface

[Chris H. <ch...@as...>]

Stefan Reuter wrote:
> Hey Chris,
>
>   
>> 1. Number of queue members logged into the queue.
>> 2. Number of queue members available to take calls
>> 3. Number of channels currently waiting in the queue (looks like 
>> AsteriskQueue will do this one)
>> 4. Average wait time.
>>     
>
> Yes these sound like a good fit for AsteriskQueue.
>
>   
>> It looks like the AsteriskQueue interface is a good start for this but 
>> I'm not sure if it's even possible to get the other information without 
>> parsing through a big list of queuemember responses.
>>     
>
> There already is an empty method in live.internal.QueueManager called
> handleQueueMemberEvent() that is called on initialization, where we
> could put some meat in.
> A new live object QueueMember (or AsteriskAgent) will be needed to
> represent agents.
> The interesting part starts after initialization, i.e. tracking agent
> events and updating the model. The relevant ones seem to be
> - AgentLoginEvent
> - AgentLogoffEvent
> - AgentCallbackLoginEvent
> - AgentCallbackLogoffEvent
> To associate agents with the callers they are serving (and to see
> whether they are busy or not) looking for AgentCalledEvents will be
> needed as well as observing the associated channel for hangup or
> redirection so we know when the agent is available again.
>
> Once that has been done some new methods could be added to AsteriskQueue
> and AsteriskAgent like support for easy login/logout, etc.
>   
We are doing something similar in our queue management application 
(external from Asterisk-Java).  Getting the initial state is fun as you 
probably already have callers in the queue and agents on the phone.  I'm 
willing to invest some resources in this as it would make my life much 
easier to keep this info in Aasterisk-Java and make my presentation 
piece more light weight.  Where do we start?

>   
>> I'm thinking about 
>> adding an action to asterisk QueueStatSummaryRequest that will take as 
>> an argument the queue name and return the above info as a response (I 
>> really hate the way asterisk tries to send events as a response).  Any 
>> thoughts?
>>     
> I hope that this is not needed because the idea of the live package is
> to observe asterisk events and update an in memory model of asterisks
> state so you dont have to talk to asterisk each time you query the state.
>   
For our immediate needs we created a QueueSummary action that returns 
the info needed.  This patch has already been accepted in trunk.
http://bugs.digium.com/view.php?id=8035

-- 
Chris Howard                            Email: ch...@as...
Director Software Development                  Direct: 256.705.0262
Asteria Solutions Group, Inc.                    Main: 256.705.0277
2904 WestCorp BLVD, Suite 203                     Fax: 256.705.0280
Huntsville, AL, 35805                        Fax2Mail: 256.705.0262       
http://www.asteriasgi.com                  Toll-Free: 877.ASGI.4.ME      

Re: [Asterisk-java-users] No such channel?

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

Hi Javier,

> Couldn't find channel: org.asteriskjava.live.NoSuchChannelException: =20
> Channel 'SIP/CityTel/<caller-number>' is not available.
can you show us what you see on the asterisk console (with set verbose
9) when this happens?

> Does that mean, that this doesn't work with two EXTERNAL numbers or =20
> is there another problem I don't see?
Nope connecting two external numbers is no problem.

> Probably more an AGI then an Asterisk-Java-Question...
You are not using AGI but the Manager API.

=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...