[asterisk-users] ANN: Asterisk-Java 1.0.0.M3 Released

[Stefan Reuter]

Hi,

We've just released milestone 3 of Asterisk-Java 1.0.0. Next to a few
bug fixes this new milestone makes Asterisk-Java OSGi compliant and adds
support for the modern SLF4J logging framework.
Have a look at
http://blogs.reucon.com/asterisk-java/2009/10/19/asterisk_java_1_0_0_m3_released.html

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50670 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  stefan.reu...@reucon.com
Jabber:  stefan.reu...@reucon.com
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760







signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fastagi

[Stefan Reuter]

Yes this is a well-known change. If you use the latest Snapshot of
Asterisk-Java 1.0.0 (http://asterisk-java.org) the issue should be resolved.

=Stefan

Alex Balashov wrote:
> Appearances suggest that some part(s) of the AGI protocol changed 
> between 1.4 and 1.6.
> 
> hh174 wrote:
> 
>> Hello,
>>
>> I have a problem with fastagi.
>> In fact I have a fastagi written in Java.
>> Communcation between asterisk 1.6 and the server works correctly, except
>> when a 'HANGUP' is sent by asterisk...
>> In this case, the java server doesn't read the message.
>> I have tride with PHP, same result.
>>
>> A ngrep show differences between the HANGUP messages and the others (
>> [AP] vs [AUP] )
>>
>> T XXX.73.102.179:53824 -> XXX.73.102.188:1687 [AP]
>>   200 result=1
>>
>> T XXX.73.102.179:53824 -> XXX.73.102.188:1687 [AUP]
>>   HANGUP.
>>
>>
>> Any idea?
>>
>> Kind regards
>>
>>
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 


-- 
reuter network consulting
Neusser Str. 110
50670 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  stefan.reu...@reucon.com
Jabber:  stefan.reu...@reucon.com
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk + Openfire

[Stefan Reuter]

jonas kellens wrote:
> [general]
> displaysystemname = yes
> enabled = yes
> webenabled = yes   (is this necessary for Openfire ???)

no you don't need it

> port = 5038
> bindaddr = 0.0.0.0
> 
> [openfire]
> secret=XX
> deny=0.0.0.0/0.0.0.0
> permit=192.168.2.5/255.255.255.0

I usually add
read = all
write = all

=Stefan

> Openfire takes forever to connect to Asterisk... My Asterisk-firewall is
> down.

Anthing in the logs of Openfire?

=Stefan



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parsing Asterisk's .conf files from Perl, Java or PHP file

[Stefan Reuter]

>> Then, my next question, is there widely available librairies to parse
>> Asterisk's config files-like files ?

Asterisk-Java has some support for this:

http://asterisk-java.org/development/apidocs/index.html?org/asteriskjava/config/package-summary.html

The basic things are pretty straight forward, however there are some
edge cases like templates that make parsing the config files non-trivial.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager API in PHP

[Stefan Reuter]

Olivier wrote:
>> Would it help if you could use Asterisk-Java's implementation of the
>> Manager API for your script? Similar to what we already did for FastAGI
>> at
>> 
>> http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html
>> 
>> If you could use that approach let me know and I'll see if we can extend
>> scripting support to cover the Manager API and AJ's live package.
> 
> I'll see if I can try it.
> It seems I should have to download latest 1.0.0-SNAPSHOT for that.
> 
> Thanks for this tip.

Yes 1.0.0-SNAPSHOT is quite stable.
But please note that the ScriptEngine support currently only includes
the FastAGI part, for the Manager API there would still be some work to
be done. It's not hard to do I am just waiting for someone to ask for it
and to test it :-)

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50670 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  stefan.reu...@reucon.com
Jabber:  stefan.reu...@reucon.com
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager API in PHP

[Stefan Reuter]

Olivier wrote:
> I need a hack to query current calls coming in and going out an Asterisk
> 1.6.1 system and send this list back as a custom UserEvent to other
> listening endpoints.
> For various reasons, I might need to write this hack in PHP though I'm
> more experienced with Asterisk Java.

Would it help if you could use Asterisk-Java's implementation of the
Manager API for your script? Similar to what we already did for FastAGI
at
http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html

If you could use that approach let me know and I'll see if we can extend
scripting support to cover the Manager API and AJ's live package.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Agent-Login/out in 1.6

[Stefan Reuter]

David Anthony O Reilly wrote:
> hehe What were the developers thinking by removing the old system! It
> worked perfect!! and by the looks of it nobody has ever recovered from
> the command removal unless they hack around with the voicemail system.

I think the best solution is to either use an AGI script if agents
should be able to login/logout through the phone, this allows you to
store the agents in a database or any other external system.
On the other hand many call centers use special applications for their
agents and allow them to manage their availability through a GUI. Such
applications can make use of the Manager API. They probably more
convenient as they also allow to display additional information about
the calls they handle and the queues they are subscribed to.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] AGI scripts in Groovy, JavaScript, JRuby or PHP running on the Java Virtual Machine

[Stefan Reuter]

Hi,

We've just finished adding support for writing AGI scripts in a variety
of popular scripting languages to Asterisk-Java.
The FastAGI server in Asterisk-Java allows you to move your AGI scripts
to a dedicated server and increases performance by eleminating the need
to start the language interpreter for each request.
Our current snapshot release includes an AGI demo in Groovy, JavaScript
and PHP to get you up and running quickly.
Check it out at
http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html
and provide feedback.
Will this also be useful for non-Java developers?

Best regards,

Stefan

-- 
reuter network consulting
Neusser Str. 110
50670 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  stefan.reu...@reucon.com
Jabber:  stefan.reu...@reucon.com
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Preferred language for Asterisk AGIs development ?

[Stefan Reuter]

Tobias Wolf wrote:
> This be true for AGI, but there is also FastAGI and with it the excellent
> asterisk-java package:
> 
> http://asterisk-java.org/
> 
> It supports writing AGI Scripts in JAVA, which communicates over TCP with 
> Asterisk. AMI is supported too ...
> 
> Last but not least it has a nice Abstraction Layer, the Live API, which gives 
> you some good insight and control of what is going on, inside the asterisk 
> server.

It should be added that using the Java platform with Asterisk-Java for
AGI (and AMI) development also offers an easy way to use a whole variety
of other languages that run on the JVM.
These include Scala, JavaScript, Groovy, Ruby (JRuby), Python (Jython),
PHP (through Quercus) and BeanShell to only name a few.

You may want to have a look at
http://blogs.reucon.com/asterisk-java/2008/02/15/agi_scripts_in_beanshell.html
http://blogs.reucon.com/asterisk-java/2007/11/20/agi_scripts_in_groovy.html
http://blogs.reucon.com/asterisk-java/2008/08/10/outbound_message_delivery_using_agi_and_ami_in_scala.html

=Stefan



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Originate Status Monitoring

[Stefan Reuter]

Hi,

The only reliable solution I've found for this is to set a custom
variable with the Originate action and query new channels for that
variable when they appear.
We've also used this strategy successfully when implementing
Asterisk-Java's live API.
Depending on which language you are going to use for your application
solutions similar to Asterisk-Java may be available that hide this and
similar obstacles.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Removing "Parsing /etc/asterisk/manager.conf" from CLI

[Stefan Reuter]

Adrian A wrote:
> Is there any way of removing this line from showing on the console? I
> have a script that logs in every few seconds to manager (...)

Maybe a better solution is to rethink your architecture. The Manager API
is well suited for long running connections, so there is no need to
reconnect every few seconds.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Higher level API on top of AMI and AGI (was Re: "Real" API for Perl?)

[Stefan Reuter]

Lee Jenkins wrote:
> I thought that the OP was asking for something to perl what Asterisk-Java 
> does 
> for java coders.  I would definitely consider Asterisk-Java to be a 
> framework, 
> though not so much with PasAGI which is more of an class object wrapper 
> around 
> AGI functions that I wrote a while back because I'm lazy that way ;)

Indeed and I think such a higher level API could be implemented in
different languages. There is/was a port of the Asterisk-Java API to
.Net at least. I think especially the "live" API of Asterisk-Java is
worth having a look at. It provides an object view on top of AMI with
rich objects like Channel and methods like hangup() and redirect().
So it makes the developer focus on his tasks rather than thinking in
terms of actions and responses.

Asterisk 1.6 includes a new feature that allows using AMI as a transport
for AGI commands, there abstraction becomes even more important.
For Asterisk-Java I am currently adding support for that in a way that
allows the developer to run the same "AGI" code either through FastAGI
or AMI without knowing about the underlying details.

I think this kind of abstraction is key to the success in
Asterisk-enabling other applications.

If someone is interested in defining a language-neutral general higher
level API that can be implemented in a variety of languages I am happy
to support this effort.

> I'm not sure what your point is, but I'll say that I'm a definite proponent 
> of 
> abstraction layers provided they don't bar access to lower level logic when I 
> need it.  I think you'll agree that good abstractions lend themselves to 
> reuse 
> and reduced development time (easy of use, less runtime logic errors, easier 
> to 
> extend, etc).

And don't miss the additional benefit of supporting multiple versions of
Asterisk that you get almost for free. Asterisk-Java will run with
Asterisk from 1.0 to 1.6 without changing your code even if the Asterisk
guys decide to rename properties and the like.
Just have a look at doc/manager_1_1.txt in the betas of Asterisk 1.6 and
decide what your efforts would be to support Asterisk 1.4 and 1.6 if you
stick to low level APIs.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3.1 released

[Stefan Reuter]

Asterisk-Java 0.3.1, a free Java library for Asterisk PBX integration,
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 for this scenario: The FastAGI protocol and the Manager API.

Asterisk-Java 0.3.1 is a maintenance release that solves the
following issues:
* [AJ-81] - executeCliCommand() always executes "show voicemail users"
* [AJ-86] - getChannelByName doesn't return the latest channel
* [AJ-80] - getMeetMeRooms() should only return active rooms
* [AJ-68] - Support for Bridge Action
* [AJ-74] - Support Strategy property in QueueParamsEvent

Asterisk-Java takes advantage of the features of Java 5.0 and therfore
requires a Java Virtual Machine of at least version 1.5.0.

Asterisk-Java is used in several commercial environments and by
the following Open Source projects:
* Asterisk-JTAPI
  JTAPI implementation for Asterisk.
  http://asterisk-jtapi.sf.net/
* Asterisk-IM
  A plugin for the Openfire XMPP (jabber) server. It provides
  integrated presence between your IM client and phone, notification
  of incoming calls by IM and originate calls from supported IM
  clients.
  http://www.igniterealtime.org/projects/openfire/
* Asterisk Desktop Manager (ADM)
  A desktop application that will allow for automatic on-call volume
  reduction, one click dial from clipboard, integrated phonebook
  and more.
  http://adm.hamnett.org/

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




signature.asc
Description: OpenPGP digital signature
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP and ODBC voicemail storage

[Stefan Reuter]

Olivier wrote:
> 1. With IMAP, is it necessary to save a copy of voicemails in /var/log
> files so that a user can still listen to his (or her) own voicemails
> with his own hardphone ?

no listening your voicemails are only stored in the IMAP folder and
accessed from both the email client and the phones. The cool thing is
that they are only marked read/deleted/... once so you listen to a
message on the phone and your corresponding "email" is instantly marked
read.

> 2. How then, can you make sure to skip non-voice mails stored in the
> same email repository ?

I usually put voicemail in a separate imap folder but I am sure it also
works with only one inbox. Whether it's a voice mail or a regular email
can easily be detected by looking at the message headers.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Queues monitoring software

[Stefan Reuter]

You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/

I am not sure if it supports all features you are looking for but it
should be a good start.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Monitor events?

[Stefan Reuter]

Anthony Francis wrote:
> I guess I should clarify. My name is Anthony, I was the one that said I 
> had written a patch, if Daniel also said he had done so and I missed 
> that email I apologize.

Well then disclaim it and post it to the asterisk bug tracker or post
its issue id if you already did.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Monitor events?

[Stefan Reuter]

Hey Daniel,

I think adding the events would be a good idea.
Just open an issue on http://bugs.digium.com/ and attach your patch
there. Be sure to send a disclaimer to digium so your patch can be
included in the distribution (see
http://asterisk.org/developers/bug-guidelines for details).

> actually you probably know i am using your java-asterisk :)

and of course if you already have patches for Asterisk-Java that support
your new events post it to http://jira.reucon.org referencing the digium
issue.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Monitor events?

[Stefan Reuter]

Anthony Francis wrote:
> There are no events generated when the monitor stops and starts, but 
> since you are implicitly recording in your dialplan one way or another 
> you can just add a userevent step before recording and after.

You can also start monitoring through the Manager API in which case you
could also generate corresponding user events. It is also possible to
map monitoring to dtmf digits in features.conf. In that case generating
user events would be hard.
So a better solution is probably to add events directly to res_monitor.c
so that they fire automatically.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3 released

[Stefan Reuter]

Asterisk-Java 0.3, a Java library for Asterisk PBX integration,
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 for this scenario: The FastAGI protocol and the Manager API.

Asterisk-Java 0.3 is the new stable release with full support for
Asterisk 1.4 and the new Live API (org.asteriskjava.live).
The Live API takes care of the lowlevel action and event handling
of the Manager API and offers an intuitive API for Java developers.
Asterisk-Java takes advantage of the features of Java 5.0 and therfore
requires a Java Virtual Machine of at least version 1.5.0.

Asterisk-Java is used in several commercial environments and by
the following Open Source projects:
* Asterisk-JTAPI
  JTAPI implementation for Asterisk.
  http://asterisk-jtapi.sf.net/
* Asterisk-IM
  A plugin for the Openfire XMPP (jabber) server. It provides
  integrated presence between your IM client and phone, notification
  of incoming calls by IM and originate calls from supported IM
  clients.
  http://www.igniterealtime.org/projects/openfire/
* Asterisk Desktop Manager (ADM)
  A desktop application that will allow for automatic on-call volume
  reduction, one click dial from clipboard, integrated phonebook
  and more.
  http://adm.hamnett.org/

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




signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk and SAP

[Stefan Reuter]

nik600 wrote:
> has everyone interfaced Asterisk in a SAP production enviroment?

we have integrated SAP systems using JCo and Asterisk-Java. A FastAGI
application accesses to data in R/3 and provides it to the caller.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] agi with java?

[Stefan Reuter]

Matthew Pease wrote:
> Hi all -
>  Searching for java agi in the mailing list archives turns up ancient
> posts.

Have a look at http://asterisk-java.org and the tutorial at
http://asterisk-java.org/development/tutorial.html - it include a hello
world AGI script in Java.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Blindside Web Conferencing

[Stefan Reuter]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hey Brad,

I am not sure if you know about the Asterisk-IM plugin for Openfire.
Basically it supports dialing contacts and arbitrary numbers through
Spark and updates presence based on being on call or not.
One of our next steps would be to integrate conferencing so you could
setup (and control) a voice conference much the same way you can do with
Jabber groupchat.
We also have a web conferencing app in a pre beta state sitting around
for some time now (based on Asterisk-Java, DWR and Tomcat) with the
original intend to use it for a commercial service which never got
really started though.
I am not sure if we could come together in some way but if you are
interested feel free to contact me off-list.

=Stefan
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGW1V9cVCZDfrn+pMRAt0TAJ4n0BPLDu1EBqqZg5RtIy4tEsLsJgCeJQFW
yePaEzQ9FX65+SoTGxs8B6M=
=TUw5
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Blindside Web Conferencing

[Stefan Reuter]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

> Yes, we are looking for that. Do you know of any projects that provides
> those? I know one written in TCL/TK.

You might also want to have a look at
http://www.version2software.com/v2whiteboard.html - its a plugin for the
Java based Jabber client Spark (from igniterealtime.org)

=Stefan
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv
pQOw6ugERcCUKy7pjDHf/qs=
=JI7J
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: [Asterisk-java-users] asterisk-java.org up again with bonus article on Local/ channels

[Stefan Reuter]

robert home wrote:
> does any one know what happened to www.asterisk-java.org
> or when it'll be back

We had problems with the IN NS records at PSI. The problem is fixed now
though it might still take a few hours for the changes to propagate.

I am sorry for any inconvinience this outage may have caused and have
provided a bonus article on Local/ channels to say sorry.
The article include a nice diagram on how using Local channels and
Originate relates to the events you see on the Manager API.

=Stefan

P.S. If you still encounter problems please contact me off-list and I'll
have a look if I still missed anything.

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk-Java website

[Stefan Reuter]

Doug Garstang wrote:
> Well, it _was_ up again Friday, and now it's down again Monday! :(

sorry, there seem to be problem with the nameservers.
I'll hava a look at it asap.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] managers

[Stefan Reuter]

Todd H wrote:
>  Am I allowed to have multiple managers logged in with the same manager
> username at the same time?   I'm referring to the id names in
> manager.conf.  I expect so, but just want to check to help in
> troubleshooting a problem.

Yes you are.

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: Asterisk Java w/ Threads

[Stefan Reuter]

Eric "ManxPower" Wieling wrote:
> In the past, the Asterisk Manager Interface was prone to crashes if it
> had more than 1 client connected to it.  The proxy solved that issue.  I
> think this issue was resolved in 1.2.

Yes, this was indeed a problem with 1.0. I didn't encounter any problems
regarding this for 1.2 and 1.4.
Connecting from one application application to multiple Asterisk servers
(which was the question) has never been a problem though.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Asterisk Java w/ Threads

[Stefan Reuter]

With Asterisk-Java the proposed solution to connect to multiple Asterisk
servers is to create multiple AsteriskManagerConnection obeject.
Each ManagerConnection handles its own thread so there is no need for
custom thread handing code.
All you have to do is to make sure is the EventListener objects you pass
to these connections synchronize access to shared data (if there are
such accesses).
I think this approach is rather simple for the user and don't see a
benefit in adding a proxy to that picture.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Java w/ Threads

[Stefan Reuter]

Jesus Mogollon wrote:
> The best option would be to use AstManProxy and connect your event
> manager to it.

why would adding a new system in between be better than directly
connecting to multiple Asterisk servers?

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Java w/ Threads

[Stefan Reuter]

Doug Garstang wrote:
> Can the Asterisk Java API be written with threads?
sure.

> Ie, I need to connect
> to multiple Asterisk systems from the one java application. I tried to
> make my  class which implements ManagerEventListener, also implement
> Runnable, but got errors because the Runnable interface doesn't throw
> InterruptedException.
It would certainly help if you provided some example of what you tried.

> Anywho...
A better place to ask questions regarding Asterisk-Java is the
asterisk-java-users list:
http://asterisk-java.org/development/mail-lists.html

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How can I use "Asterisk Manager API" to hold and retrive an active call?

[Stefan Reuter]

James Zhang wrote:
> These are common functions. Why "Asterisk Manager"
> doesn't  provide commands to hold and retrive an active channel?
> If it must be implemented by AGI, could anyone give a direction or steps?

Sure the Manager API provides all thing to do that.
Maybe you are just using the wrong library on top of the Manager API ;)

Asterisk-Java as an example lets you retrieve active channels, iterate
over them, hangup, redirect, ... whatever.

Example to hangup all active channels:

for (AsteriskChannel channel : server.getChannels())
{
channel.hangup();
}

http://asterisk-java.org

I am sure other libraries provide similar abstraction.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk-Java 0.3 Milestone 2

[Stefan Reuter]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

we've just released Asterisk-Java 0.3-m2 at http://asterisk-java.org.

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
for this scenario: The FastAGI protocol and the Manager API.

Asterisk-Java is free software distributed under the terms of the Apache
License 2.0.

Here is the Changelog:

Bug

* [AJ-47] - AGI does not support multi line data
* [AJ-51] - Problems with non-english locales
* [AJ-52] - Fix shutdown when using the live api

Improvement

* [AJ-41] - Add ability to get ManagerConnection from AsteriskServer
* [AJ-49] - Support socket read timeout

New Feature

* [AJ-35] - Support timestamp property on manager events
* [AJ-42] - Add support QueueSummary action to Queue manager
interface
* [AJ-44] - Support PauseMonitor and UnpauseMonitor actions
* [AJ-45] - Support ZapRestart action

Task

* [AJ-53] - Refactor BaseAgiScript to extend AgiOperations and
implement AgiScript so users can extend AgiOperation to provide their
own add-on features


Have fun,

Stefan
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF0D1pcVCZDfrn+pMRAv9GAKCB1Ozij3l5eLGOu1pOx7kq+PHG4wCePsjR
Jm+tu6reyyrDtVDGgATnTwI=
=My0x
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How accurate is show translation?

[Stefan Reuter]

James Harper wrote:
> Ah... I use debian, and they tend to have pretty strict policies on
> anything that isn't free (as in speech). Without having looked into it
> further, that would probably explain it.
> 
> If you have never seen a debian compiled asterisk that can't transcode
> ilbc then maybe I should be looking elsewhere...

You might want to have a look at
deb http://pkg-voip.buildserver.net/debian sarge main
for non-crippled Asterisk on Debian and Ubuntu.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SOS building fastagi C

[Stefan Reuter]

Yelson Vivas wrote:
> Ps: java is not an option i develop a fast agi which worked fine up to
> 30 call but later got frozen, i have to do it in C

This problem wont be solved by just switching to C ;)
Crappy and non-performing stuff can be written in any language. Maybe
its better to fix your design than switching the language.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk AGI question

[Stefan Reuter]

David R. wrote:
> Can AGI be used to have a web application talk back and forth between
> Asterisk and itself?  What if the web application is on a separate box?

Yes, if its on another box you should use FastAGI (you should do this
anyway for performance reasons ;), see
http://www.voip-info.org/wiki-Asterisk+FastAGI

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ANNOUNCEMENT: Asterisk-Java 0.3-m1 released

[Stefan Reuter]

Asterisk-Java 0.3-m1, a Java library for Asterisk PBX integration,
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 for this scenario: The FastAGI protocol and the Manager API.

The 0.3-m1 milestone release focuses on ease of use and provides the
new org.asteriskjava.live package that takes care of the lowlevel action
and event handling of the Manager API and offers an intuitive API for
Java developers. Asterisk-Java has been updated to take advantage of the
new features of Java 5.0 and therfore requires a Java Virtual Machine
of at least version 1.5.0.

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

Asterisk-Java is available under Apache 2.0 license at
http://asteriskjava.org



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event

[Stefan Reuter]

Matt Florell wrote:
> If you are originating a call with a Local/ channel you cannot use the
> uniqueID alone to track it. The only field that will follow all legs
> of a Local/ channel originated call is the CallerID, and that is only
> if you add the "o" flag to your Dial string.

As I pointed out earlier you can also use any other channel variable for
this. The callerId (if you dont need for other things) has the advantage
though that you dont have to send another action to query for it as its
available in most events anyway.

> It's a very messy prospect to track calls through the Manager API. In
> fact two years ago I drafted this Whitepaper that would help
> tremendously, nothing ever came of it:
> http://www.freedomphones.net/Manager_API_modification_whitepaper.txt

The Manager API could be enhanced, sure. But until now everybody works
around it happily and fears the things that will break once there are
real changes applied :)

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager Interface API's

[Stefan Reuter]

John Novack wrote:
> I, for one, didn't take his comment as anything other than constructive
> Lack of documentation is an issue, open source or not.

To make this thread even more constructive:
What kind of documentation do you expect from a Manager API package?
What features do you expect?
- A plain wrapper for the Actions, Responses and Events?
- An abstracted view on Asterisk's concept like channels, extensions,
queues and so on?

And last not least: Would a language independant specification help?
Something like: There is a channel concept (object) with the properties
id, name, caller id, ... and the operations hangup, redirect, ...

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager Interface API's

[Stefan Reuter]

Douglas Garstang wrote:
> 
> Can anyone recommend the best Manager Interface API, putting language
> preferences aside?

Asterisk-Java of course ;)
http://asterisk-java.org/latest for the stable release
and
http://asterisk-java.org/0.3-SNAPSHOT for the dev snapshot.

Includes a short tutorial and javadoc for everything else.


> The python and perl ones have bupkiss documentation. I can't understand
> why anyone would even write an api and make it publically available
> without documenting it.
>  
> Doug.
>  
> 
> 
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Associating an Originate Request to a Channel before the call is answered

[Stefan Reuter]

Janahan Vivekanandan wrote:
> I know that I can use the ActionID to accomplish this once I receive
> the OriginateSuccess event, but I need to be able to cancel the call
> before it is answered(I'm pretty sure OriginateSuccess is only sent
> after the  call is answered, correct me if I'm wrong...:-)

You are right and the canonical answer seems to be you can't.
I had the same problem a few weeks ago for the development of
Asterisk-Java and solved it by setting a channel variable in the
Originate action which I check for with every NewChannel event.
Not nice, but it a working hack.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CSTA support for asterisk

[Stefan Reuter]

Hans-Jürgen Brand wrote:
> just have a look at http://sourceforge.net/projects/oscsta

looks rather empty...



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager interface

[Stefan Reuter]

[EMAIL PROTECTED] wrote:
> You could try out Snap -- www.snapanumber.com
> , it has the features you need. We also do
> custom developement, so this may help get your project moving along faster.

The problem with Snap and any other solution that directly opens the
Manager API to client workstations is that it opens a rather huge
security issue. Using the Manager API you are able to completely control
the Asterisk server (up to the UNIX level, even root access if Asterisk
is running under root). This may be ok for smaller closed groups with
fully trusted users, but it certainly is not a solution for the paranoid.
A better way to handle this is a server that connects to the Manager API
and only passes the relevant information to the clients.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] bugs.digium.com

[Stefan Reuter]

> On Thu, 2006-07-27 at 08:32 -0600, Douglas Garstang wrote:
>> Someone with the id of
>> 'russell' in his infinite wisdom has deemed that this isn't a bug,
>> closed it, and given me -2 karma points. 

There may be developers/bug marshalls who close bugs without considering
the user's issues in depth but russel certainly is not one of them.
Providing patches for bugs or enhancements handled by him has always
been a please.
So please keep up the good work, I feel your passion for Asterisk!

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Associate manager events to a previous Originate action

[Stefan Reuter]

Johannes Zweng wrote:
> Although I can associate every incoming event to a specific channel on
> Asterisk (because of the Uniqueid field) I see no possibility to identify
> without doubts which channels were created as a result of my Originate
> action. 

add an ActionId property to your Originate action and you will receive
the same ActionId as part of the OriginateSuccess or OriginateFailure
event. The OriginateSuccess event also contains a link to the channel so
you can relate them after you received the OriginateSuccess event.
For the OriginateSuccess/-Failure events to be sent you must also set
Async to true when sending the Originate action.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Simple But important question (for me)

[Stefan Reuter]

Camilo Echeverry wrote:
> 1- receive the call (obvious)
> 2- get the Caller ID
> 3- Send the CID to another application and get some info from a Database
> example: Your address is "some address"
> 4- Get that info and convert it into voice (by mixing various audio files)
> 5- return it to the Caller (as audio)

Yes, Asterisk can do all of these. You might want to look at AGI/FastAGI
for implementing it.
# http://www.voip-info.org/wiki/view/Asterisk+AGI
# http://www.voip-info.org/wiki-Asterisk+FastAGI

If your primary development language is Java you might also be
interested in Asterisk-Java which allows you to easily implement AGI
scripts in Java: http://asterisk-java.org

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Running a poll server with asterisk

[Stefan Reuter]

Time Bandit wrote:
> AGI is your answer.

Or you stick to the dialplan and use Asterisk's internal DB.
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] remote setting - AGI or what?

[Stefan Reuter]

Christophorus Laube wrote:
> But I want to set the number remotely and client initiated. Is AGI able
> to do such things or what can I use else?

Have a look at the Manager API and the DbPut action for this.

http://www.voip-info.org/wiki/view/Asterisk+manager+API
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+DBPut

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Simple windows / web Asterisk user software?

[Stefan Reuter]

Rod Bacon wrote:

> ADM (Asterisk Desktop Manager) is close to what I'm after, but is still a 
> little BETA for my liking.

If you have any issues regarding ADM please let us know, its the only
way we can improve things and make it fit our users' needs!

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Recent debian packages?

[Stefan Reuter]

Jean-Michel Hiver wrote:
> I'd like to use the convenience of apt packaging, but debian sarge's
> default asterisk is something like 1.0.7.
> 
> Are there any apt repositories which provide newer versions of the
> software?

sure: http://pkg-voip.buildserver.net/debian

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] View Agent Status on the Web

[Stefan Reuter]

Pimjai Wesnarat wrote:
> I want to be able to see the status of my Agents on a web interface. I
> have no idea how to do so.
> I have found a few sample script to communicate with queues manager to
> view queues.But I couldn't find any on viewing the agent status. Could
> anybody give me a clue?

You can use the Manager API and issue an "Agent" action. This results in
a series of "Agents" events. The properties of this event are described
here:

http://www.asteriskjava.org/0.3-SNAPSHOT/apidocs/org/asteriskjava/manager/event/AgentsEvent.html

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk Manager interface

[Stefan Reuter]

Devraj Mukherjee wrote:
> Part of the issue is number of socket connections that the
> client opens back to the manager itnerface. Most of these connections
> are short lived.
> 
> Is this is a problem from a design perspective? Or is the management
> interface designed to handle this.

no it is not designed to handle this.
Have a look at http://www.voip-info.org/wiki-Asterisk+Manager+Proxy

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] [patch] fix for redirect manager action with BRIstuffed Asterisk

[Stefan Reuter]

Hi,

BRIstuff contains two bugs in its implementation of the Redirect manager
 action:
1. If the property ExtraUnqiueId is used, the Priority property is used
to redirect the extra channel (instead of ExtraPriority)
2. If the property ExtraChannel is used, 0 is used to redirect the extra
channel regardless of the Priority and ExtraPriority properties.

A patch for manager.c is available at
http://www.reucon.net/~srt/bristuff_redirect.patch as a result to a bug
filed against Asterisk-Java at http://jira.reucon.org/browse/AJ-34

I've sent a notice to kpj.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk + mobicents

[Stefan Reuter]

Hi Harry,

> I look at the mobicents project.
> Somebody has experience within both projects ? 
I dont have any real experience with mobicents but I now that some guys
from mobicents built a resource adapter for asterisk about a year ago.
Here are some notes about it:
http://wiki.java.net/bin/view/Communications/MobicentsAsteriskRA

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Instant Message?

[Stefan Reuter]

Kyle Sexton wrote:
> I've also had horrible experiences with the Asterisk plugin.  The second
> I enable it, no one can log into their IM client anymore.

did you report that on their forum?
I installed it some time ago and it worked quite well besides some
issues with staying "on the phone" when using Asterisk queues.
Andrew is usually very helpful in resolving any issues regarding the
Jive (now Wildfire) plugin.

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] AGI hangup problem

[Stefan Reuter]

Branko Samardzic wrote:
> I have problem with Asterisk.
> [sendCommand]=EXEC "DIAL" "IAX2/somehost/somenumber|10"
> [readReply]=200 result=-1
> [sendCommand]=GET VARIABLE "ANSWEREDTIME"
> 1086422 [Thread-7] ERROR - establishConnection: exec encountered exception
> net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up.
> This means that AGI is not capable of extracting data about call performed.
> Is there any workaround?

just catch the exception and go on...



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] transfer calls via Manager Api

[Stefan Reuter]

nik600 wrote:
> nik600 wrote:
> but...how can i know the channel used in a specific period from the called?
> 
> for examle...i know SIP/200 but how can i know that the channel is
> SIP/200sfhj3e ?

either by following the events (NewChannel, Rename, ...) or by issuing a
StatusAction that will return a list of all active channels.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Köln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Programming the Manager API

[Stefan Reuter]

Michael Collins wrote:
> Just curious if someone out there might have already solved this problem
> and created a Python module that you could borrow...

A python package is available from
http://py-asterisk.berlios.de/py-asterisk.php. It doesn't seem to be
activly maintained but it might serve as a starting point.

=Stefan


signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] transfer calls via Manager Api

[Stefan Reuter]

nik600 wrote:
> Is it possible to transfer an existing call from the extension ...
> SIP/xxx to another extension in a specific context?

you can do this with the redirect action:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect

=Stefan



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sip channel status - how?

[Stefan Reuter]

Peter Hoppe wrote:
> Thank you very much for that hint! I am using asterisk-java at the
> moment to retrieve the channel information and I now have a way of
> retrieving such channel information sending a "sip show channels"
> command via the manager interface. I then parse the answer from the
> server. But it seems that "show channels consise" is an even better way.
> I have tried it and it gives me even better information about active
> channels. I'll investigate the documentation of that command.

why dont u just use the Status action ?

=Stefan


signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How to record a call

[Stefan Reuter]

On Wed, 2005-12-21 at 19:34 -0800, Asterisk Mail wrote:
> I am relatively new to this area. I want to record a 2 party/
> conference call in some sound file format (maybe as a .wav file). If
> anyone can point me towards some documentation or some sample code it
> will be great.

http://www.voip-info.org/wiki-Asterisk+cmd+Monitor



signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Running asterisk within screen

[Stefan Reuter]

On Thu, 2005-12-01 at 22:29 -0800, Luki wrote:
> > Does anybody know, why it is not possible, to run asterisk within
> > screen?
> 
> Yes, it is possible but you can't scroll up so you only see the last
> ~40 lines. At least I didn't work for me but I didn't research this
> further.

in screen you can enter copy/view scrollback mode by pressing Ctrl-a Esc
then you can use page up/down. To leave scrollback mode press Esc again.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Recording Calls

[Stefan Reuter]

Felix Amaral schrieb:
>  Hi, I´ve recently installed my first Asterisk and it´s working. I can only
> make outbound calls trough internet. I was willing to record the phone calls
> in files maybe with wav or gsm extension. Can someboy help me a little with
> this?

http://www.voip-info.org/wiki-Asterisk+cmd+monitor



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cdr_manager.conf

[Stefan Reuter]

> Would anybody please tell me,
> If I keep enabled=yes, cdr_manager would be enable, I know
> but an 'enabled' cdr_manager would help me?
> How I can be benifited from this in terms of cdr management?
> What exactly it does if I keep enabled=yes?

As I said: If you set enabled to yes you receive CDR Events via the
Manager API (http://www.voip-info.org/wiki-Asterisk+manager+API).
If you do not enable it, you dont receive them via the Manager API but
only written to the CDR file, database or whatever you configured.

> or, what are the next step(s)?

Next steps? Have a look at the wiki and read about the Manager API, if
you come to the conclusion that you don't need it forget about cdr_manager.

=Stefan


signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cdr_manager.conf

[Stefan Reuter]

On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote:
> What is the purpose of cdr_manager.conf?
cdr_manager.conf allows you to configure asterisk to send call detail
records (cdr) via the Manager API.

> How I can configure it?
to enable CDR via Manager API a cdr_manager.conf looks like this:

;
; Asterisk Call Management CDR
;
[general]
enabled = yes

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ANNOUNCEMENT: Asterisk-Java 0.2 released

[Stefan Reuter]

Asterisk-Java 0.2, 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 for this scenario: The FastAGI protocol and the Manager API.

The 0.2 release focuses on the new features of the Asterisk 1.2 series 
though it still supports Asterisk 1.0.x.
Since 0.2-rc2 some minor bugs have been fixed and support for several 
last minute additions to Asterisk 1.2 has been added.

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 
  of incoming calls by IM and originate calls from supported IM 
  clients.
* Asterisk Desktop Manager (ADM)
  A desktop application that will allow for automatic on-call volume 
  reduction, one click dial from clipboard, integrated phonebook
  and more.

Asterisk-Java is available under Apache 2.0 license at
http://www.asteriskjava.org


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Looking for Info on Asterisk scripting

[Stefan Reuter]

Obelix schrieb:
> Is there a source of Asterisk programming techniques in various languages - ie
> Asterisk scripting in general, not the main Asterisk program itself?

What you are looking for is probably AGI (the Asterisk Gateway
Interface) that is to Asterisk what CGI is to a Webserver.

Have a look at the Wiki at
http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
it tells you all you need to know to get started.

=Stefan


signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAX clients

[Stefan Reuter]

On Wed, 2005-11-23 at 16:29 -0700, Jason Becker wrote:
> http://www.hem.za.org/jiaxclient/

Thanks for the pointer.
I should have been more clear with my request: What I am looking for is
a pure Java implementation. JIAXClient is a solution that is ok for many
use cases but is unacceptable in others as it merely wraps the native
library via JNI.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAX clients

[Stefan Reuter]

Yes it would be really interesting if there are any IAX libraries for
Java that are available under an open source license and that we might
improve further.
There is a growing demand for such a thing (for example see
http://forums.digium.com/viewtopic.php?t=2431)
Would be cool if we can create kind of a defacto standard, i.e.
something that everybody uses.

=Stefan



signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Speech recognition or TTS with Asterisk?

[Stefan Reuter]

On Thu, 2005-11-17 at 12:10 -0700, John Brookes wrote:
> Can this be implemented in Java?

sure that can be implemented in Java. Have a look at Asterisk-Java at
http://asteriskjava.org.
Asterisk-Java is to Java what phpagi is to PHP.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] AGI Dial command return status

[Stefan Reuter]

Hi Derek,

I don't think AGI-only is the best approach for billing.
You can easily use the Manager API for that (there you get Link and
Unlink or CDR events that you can process much better).
Using Asterisk-Java you can quite easily combine AGI and the Manager
API.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How uniqueids are formed - possible race conditions for linked channels?

[Stefan Reuter]

> I want to track the ringing event of the outgoing channel.
> Unfortunatelly the link event is fired not before connect.

I suppose you are still running Asterisk 1.0.x.
For Asterisk 1.0.x i know of no clean solution for that problem.
Asterisk 1.2 introduced the Dial event that is triggered before the Link
event (at dial time) and provides names and uniqueids of the source and
destination channel.
So having a look at Asterisk 1.2-beta2 is probably the way to go.

=Stefan

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How uniqueids are formed - possible race conditions for linked channels?

[Stefan Reuter]

On Sat, 2005-11-05 at 13:42 +0100, Roger Schreiter wrote:
> Now I wonder, whether I can rely on that scheme.
> I assume, the timestamp part can be different, e.g.
> if between the creation of the incoming channel and
> the creation of the outgoing channel the system clock
> switches to the next second. (Or even more, if an AGI-
> script or anything else has consumed more time in between.)

in general you can't rely on it. It really depends on how the call is
processed.
Usually there are more reliable ways to determine which channels are
linked, for example through the Manager API that generates Link and
Unlink events.
To propose the best solution we must know more about your actual use
case.

=Stefan

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ANNOUNCEMENT: Asterisk-Java 0.2-rc2 released

[Stefan Reuter]

Asterisk-Java 0.2-rc2, 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 for this scenario: The FastAGI protocol and the Manager API.

The 0.2-rc2 release candidate focuses on the new features of 
the Asterisk 1.2 series though it still supports Asterisk 1.0.x.
The changes include
* Bug fix for variables in OriginateAction (AJ-15)
* Support for FaxReceived event from spandsp (AJ-20)
* Better integration with Spring Framework via
  SimpleMappingStrategy and AGIServerThread

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 
  of incoming calls by IM and originate calls from supported IM 
  clients.
* Asterisk Desktop Manager (ADM)
  A desktop application that will allow for automatic on-call volume 
  reduction, one click dial from clipboard, integrated phonebook
  and more.

Asterisk-Java is available under Apache 2.0 license at
http://www.asteriskjava.org



signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ANNOUNCEMENT: Asterisk-Java 0.2-rc1 released

[Stefan Reuter]

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 
for this scenario: The FastAGI protocol and the Manager API.

The 0.2-rc1 release candidate focuses on the new features of 
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 
  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 
  of incoming calls by IM and originate calls from supported IM 
  clients.
* Asterisk Desktop Manager (ADM)
  A desktop application that will allow for automatic on-call volume 
  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



signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] DIALSTATUS for Originate

[Stefan Reuter]

On Sun, 2005-08-28 at 12:45 -0700, Geoff Karl wrote:
> If you are using Async and the action ID for some reason the Event:
> Newstate doesn't respond with the ActionID, but only a automatically
> generated "Uniqueid".

When using Async you receive an OriginateSuccess or OriginateFailure
event.
These events contain the proper ActionID (i.e. the one you set with the
Originate action) and they contain an integer field reason, that
indicates the reason for the failure.

=Stefan

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint

[Stefan Reuter]

> No, I need an endpoint I can put on a webpage

if you are looking for a web based sip user agent there is sip
communicator which can be loaded using java webstart.
look at https://sip-communicator.dev.java.net/

the jnlp is here:
http://clarinet.u-strasbg.fr/sip-communicator/download/webstart/sip-communicator.jnlp

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

[Stefan Reuter]

On Thu, 2005-08-18 at 13:01 -0700, Asterisk wrote:
> I'm looking to develop some custom AGI that will be MySQL intensive.  It
> appears Asterisk supports many different development environments.  Which
> would be best suited for Asterisk and MySQL?

First you should decide if you want to run short lived AGI script
processes on the same box along with Asterisk or want Asterisk to
connect to one long lived process that serves multiple calls via TCP/IP
(FastAGI). Usually you will prefer FastAGI as it allows AGI scripts to
be processed on a different machine if needed and eliminates the cost of
spawning a process (and additional setup like opening a db connection)
for each call.
Next you want to decide on you favorite programming language, there is
support for almost everything. Have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20AGI
If you want go with Java have a look at Asterisk-Java's support for
FastAGI at http://asterisk-java.sourceforge.net/tutorial.html

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Identify call flow from manager events

[Stefan Reuter]

On Sat, 2005-08-13 at 11:59 +0200, Christian Peter wrote:
> I know I have the action id to identify events which belong together.
> But if I have a call going inside asterisk and asterisk rings a phone
> these are two channels with different action ids. How can I know that
> these channels belong together?
> 
> I know there are link events but what if the phone doesn't answer? Then
> I have two separate channels.

With Asterisk stable all you get are the link events when the channels
are linked. With Asterisk CVS-HEAD there is a dial event that informs
your Manager application at dial time about the caller ids and the ids
of the channels.

> Can I rely on the number in the action id after the dot? Eg.
> 111.0 is the incoming channel and
> 111.1 is the next channel belonging to the incoming call
> action

not if other channels are created around that time.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk and .NET

[Stefan Reuter]

> Are there any Asterisk interfaces with .NET?

There is a port of the Manager API implementation of Asterisk-Java
available for .NET from Chad Kitching.
You can download it from http://www3.mb.sympatico.ca/~chadk/

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIPPeersAction class file not foundintheAsterisk-java.jar file

[Stefan Reuter]

You dont run applications using the Manager API as AGI scripts but as
standalone Java applications.
So in you case proably via
java -cp asterisk-java-0.1.jar:. ManagerAPI

=Stefan

On Mon, 2005-08-08 at 16:31 +0530, Bharat M. Sarvan wrote:
> Ok Mr. Stefan,
>   The contents of the file fastagi-mapping.properties are as
> follows
> 
> Hello.agi = ManagerAPI
> 
> Where ManagerAPI is my class filename ManagerAPI.class.
> 
> And the directory structure is as given below:
> 
> /home/Bharat/AGISERVER/BharatJava/
> 
> Where the BharatJava directory holds the files
> 
> fastagi-mapping.properties
> asterisk-java-0.1.jar
> and the .java files
> 
> 
>
> And the Asterisk server is running on Fedora Core 1. I start the asterisk
> server using the command "asterisk -" at the command
> prompt.
> 
> So I was having doubt about the execution of the part of the code you have
> sent me where we login into the Asterisk server using the class
> ManagerConnection.
> 
> Is there any way out so that I can issue the command other than using the
> sendAction Method of the ManagerConnection to which I pass the object of the
> class CommandAction.
> 
> My asterisk server is already up and running. I just need to issue the
> command using the CommandAction. But even if I run the sample code
> "ManagerAPI" given on the link
> http://asterisk-java.sourceforge.net/tutorial.html. I am getting the error
> for "No script configured for agi://"
> 
> Please do reply as to how I get through this problem...
>  
> 
> 
> 
> 
> Regards,
> Bharat M. Sarvan
> Software Engineer - VoIP
> EZZI BPO Pvt Ltd.,
> PUNE.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
> Sent: Saturday, August 06, 2005 7:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] SIPPeersAction class file not
> foundintheAsterisk-java.jar file
> 
> Hi,
> 
> >   I have all the necessary files for the code to be executed. The
> > fastagi-mapping.properties file is also correct. But still I am getting
> the
> > error for 
> > 
> > The IP address is correct and as well as the agi file name. Does it make a
> > difference giving a Tab or a space when giving the mapping of agi file
> name
> > and class file name in the fastagi-mapping.properties file.
> 
> no that makes no difference
> 
> > Is there any other reason for getting this error
> 
> no
> please post the contents of fastagi-mapping.properties, your directory
> structure and the command you use to run java.
> and please note: THERE IS NO WAY TO GET SIP PEERS VIA FASTAGI ANYWAY!
> 
> =Stefan
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIPPeersAction class file not found intheAsterisk-java.jar file

[Stefan Reuter]

Hi,

>   I have all the necessary files for the code to be executed. The
> fastagi-mapping.properties file is also correct. But still I am getting the
> error for 
> 
> The IP address is correct and as well as the agi file name. Does it make a
> difference giving a Tab or a space when giving the mapping of agi file name
> and class file name in the fastagi-mapping.properties file.

no that makes no difference

> Is there any other reason for getting this error

no
please post the contents of fastagi-mapping.properties, your directory
structure and the command you use to run java.
and please note: THERE IS NO WAY TO GET SIP PEERS VIA FASTAGI ANYWAY!

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIPPeersAction class file not found in theAsterisk-java.jar file

[Stefan Reuter]

> Well thanks Stefan, for the help but when I am executing the AGI script I
> am getting the errors as below:

If you want to retrieve sip peers from Asterisk you won't do this via an
AGI as I explained. You will just run the main() method of the Manager
class I sent you in my last mail as an example, like:
$ java -cp asterisk-java-0.1.jar:. Manager

> SEVERE: No script configured for agi://65.125.224.207/bharat.agi
>
> What does it mean by "No Script configured for agi://" and can you please
> tell me how do I come up with this error?

That means you when you use FastAGI (which you should NOT in this case)
you failed to provide a correct fastagi-mapping.properties file on the
CLASSPATH. You find more information on how to set it up correctly at
http://asterisk-java.sourceforge.net/tutorial.html

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIPPeersAction class file not found in the Asterisk-java.jar file

[Stefan Reuter]

On Thu, 2005-08-04 at 18:25 +0530, Bharat M. Sarvan wrote:
> I am working on Fastagi and I am making use of
> Asterisk-java. But I don’t find the class file for SIPPeersAction.

The SIPPeersAction is not part of Asterisk-Java 0.1, it is available in
CVS-HEAD only.
Besides that the action classes in net.sf.asterisk.manager.action can
only be used with the Manager API and not with FastAGI.
So if you want to retrieve a list of sip peers you need to do that via
the Manager API. With Asterisk 1.0.x and Asterisk-Java 0.1 you can do
this via the CommandAction. Only if you are already using Asterisk
CVS-HEAD and Asterisk-Java CVS-HEAD you can use the new SipPeerAction.

Example with CommandAction:

import java.util.Iterator;

import net.sf.asterisk.manager.ManagerConnection;
import net.sf.asterisk.manager.ManagerConnectionFactory;
import net.sf.asterisk.manager.action.CommandAction;
import net.sf.asterisk.manager.response.CommandResponse;

public class Manager
{
private ManagerConnection c;

public Manager() throws Exception
{
c = new ManagerConnectionFactory().getManagerConnection("host", 
"user", "pass");
}

public void run() throws Exception
{
c.login();

CommandAction action;
CommandResponse response;
Iterator lineIterator;

action = new CommandAction();
action.setCommand("sip show peers");

response = (CommandResponse) c.sendAction(action);
lineIterator = response.getResult().iterator();

while (lineIterator.hasNext())
{
System.out.println(lineIterator.next());
}

c.logoff();
}

public static void main(String[] args) throws Exception
{
new Manager().run();
}
}

This produces something like:

Name/usernameHostDyn Nat ACL Mask Port
Status
1313/131310.13.0.61   D   A  255.255.255.255  5061
Unmonitored
1312/131210.13.0.61   D   A  255.255.255.255  5061
Unmonitored
1311/131110.13.0.61   D   A  255.255.255.255  5061
Unmonitored
1310/1310(Unspecified)D   A  255.255.255.255  0
Unmonitored
1303/1303(Unspecified)D   N  255.255.255.255  0
Unmonitored
1302/1302(Unspecified)D   A  255.255.255.255  0
Unmonitored
1301/1301(Unspecified)D   A  255.255.255.255  0
Unmonitored

=Stefan


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] TAPI Interface

[Stefan Reuter]

On Thu, 2005-07-28 at 12:48 -0300, Isamp wrote:
> Hi All !!!
> 
> Somebody can inform me where to get more information about the TAPI 
> (M$)  interface of the Asterisk ?

google? voip-info?

http://www.voip-info.org/wiki-Asterisk+TAPI
http://www.omniis.com/ntsgr/cms/page.asp?688


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Unattended Agent Login

[Stefan Reuter]

> So what I basically need is a way to log in an agent using
> AgentCallbackLogin
> at an extension without them having to answer / pickup a phone to do so. I
> looked at the Manager API but did not find any command related to agent
> logins.

Yes even with latest CVS there are no Manager Actions for those commands.
Probably the best way would be to just add
- AgentCallbackLogin
- AgentLogoff
to the Manager API.

> I then thought up the possibility to login agents using predefined
> extensions,
> to which I will add the AgentCallbackLogin command and later use Originate
> using the Manager API to set up the calls. Has anyone got experience doing
> the agent login in that way or some tips on how to do the actual
> implementation?

You can certainly hack this using local channels... but umm thats not nice.

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Manager API commands QueueStatus and Queues

[Stefan Reuter]

> Does anyone know what the descriptions are for the data that 
> "QueueStatus" and "Queues" manager API commands return? Any information 
> would be helpful. Thanks in advance.

anything that I know about the events is in the javadocs of
Asterisk-Java. Have a look at

http://asterisk-java.sourceforge.net/apidocs/net/sf/asterisk/manager/event/package-summary.html

If you get any further information about the attributes please let me
know so I can include it.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Stefan Reuter]

> It doesn't have to be IAX. Do you know how to
> configure it with another protocol?

have a look at
http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ANNOUNCEMENT: Asterisk-java 0.1 released

[Stefan Reuter]

Asterisk-java 0.1 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 for this scenario: The FastAGI  protocol and the Manager API.

The FastAGI implementation supports all commands currently available
from Asterisk.

The Manager API implementation supports receiving events from the
Asterisk server (e.g. call progess, registered peers, channel state)
and sending actions to Asterisk (e.g. originate call,
agent login/logoff, start/stop voice recording).

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



signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Xten-lite for linux

[Stefan Reuter]

> > Where can i get that version?
> > Not found any link on xten site...
> 
> Sign up for their forums and then email them ([EMAIL PROTECTED] I
> think) with a request to join the Linux beta.

or have a look at http://xten.com/apps/xprolinuxbeta/



signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Bristuff

[Stefan Reuter]

On Wed, 2005-03-30 at 21:22 +0300, [EMAIL PROTECTED] wrote:
> Is bristuff tarball only needed for isdn cards with NT mode or do i need it 
> also
> to connect external ISDN phone line to my non-HFC ISDN card?

bristuff is only needed for HFC based cards, so if you only use an AVM
card for example there is no need for bristuff.

> other question is, what is bristuffs content? Is there for example CAPI driver
> in it? In other words,do I need CAPI in addition with bristuff and what else?

You need chan_capi if you want to use CAPI. chan_capi is not included in
bristuff but available from the same author as bristuff:
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz

If you use a HFC and non-HFC in the same box you need both - bristuff
and chan_capi.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Using HFC-S card

[Stefan Reuter]

On Wed, 2005-03-30 at 19:10 +0300, [EMAIL PROTECTED] wrote:
> How about if I am connecting ISDN card to the external ISDN phone line (to 
> local
> telephone companys s-bus) when card must be in TE mode, do I still have to 
> have
> HFC-s card that I could forward incoming calls from pbx to phone(s) or could
> that be any ISDN card?


You don't need a HFC card in that case.
Have a look at 
http://www.voip-info.org/tiki-index.php?page=Asterisk+CAPI+Channels 

=Stefan


signature.asc
Description: This is a digitally signed message part
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Error in placing call file in directory

[Stefan Reuter]

On Wed, 2005-03-16 at 14:20 +, Razza wrote:
> Chris Blake wrote :
> 
> -%<-
> >If anyone can help I`ll send the call file to you, or is it ok to
> clutter the list with it ?
> -%<-
> 
> 'Clutter' the list I'd be interested and at least it is pertinent to *
> ;o)

I am almost sure it has nothing to do with the file contents.
The warning "Unable to open %s: %s, deleting" is only generated at one
place in pbx/pbx_spool.c:

f = fopen(fn, "r+");
if (f) {
...
} else {
  ...
  ast_log(LOG_WARNING, "Unable to open %s: %s, deleting\n", fn,
strerror(errno));
  ...
}

So please double check that the user running asterisk has access to the
file. Just checking the file is not sufficent, also check the directory
permissions above.

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Calls from web interface

[Stefan Reuter]

> There was a thread some time back about making calls via * from a web
> interface...ie user clicks number on web page and call is made...

There are basically two ways to implement this.

The first one assumes that your webserver is running on the same machine
as Asterisk. Then your web application will have to create a .call file
in /var/spool/asterisk/outgoing.
Examples on how do this are available at
http://www.voip-info.org/wiki-Asterisk+auto-dial+out

The second option is to use the Manager API which allows you to trigger
a call via TCP/IP.
For more information see
http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+dialout
http://www.voip-info.org/wiki-Asterisk+manager+api

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] AGI kill

[Stefan Reuter]

On Wed, 2005-03-16 at 13:08 +0200, Nir Simionovich wrote:
> You are correct, FastAGI is a valid option. However, if he's basing his
> application on Asterisk Stable, FastAGI is not available in the stable
> version.

My version of Asterisk 1.0.6 includes FastAGI support and works pretty
well.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] live monitoring of SIP calls chan_spy

[Stefan Reuter]

On Wed, 2005-03-16 at 11:38 +0100, Martijn van Oosterhout wrote:
> Maybe it's been replaced by the Monitor app?
> Or does it do something else?

The Monitor application records calls and writes wav files it does not
allow real time spying.

ChanSpy seems to have disappeared. The bug 2379 that formerly contained
the patch is no longer available in the bug tracker.
See http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy

=Stefan


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] AGI kill

[Stefan Reuter]

>   You can't! As far as I can tell, once Asterisk eliminates an AGI upon
> hangup, it doesn't send any signal information to the AGI script. If you
> need to run some clean ups, the proper way to do so would be to execute
> an AGI upon hangup, utilizing DeadAGI.

You can also use FastAGI instead of AGI over stdin/stdout. When using
FastAGI hangup only caused the network connection to be closed but after
that you can do any clean up you want.

=Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SAY DIGITS problem

[Stefan Reuter]

On Sat, 2005-03-05 at 17:25 +0700, Nattapong Mongkolnavin wrote:
> I have a problem using AGI cmd "SAY DIGITS". For  some reason I cannot 
> here any thing when the script got executed. However if I use the cmd 
> "SAY NUMBER" I can here * reading the number fine.
> 
> fputs($stdout, "SAY DIGITS 1234");

SAY DIGITS takes two mandatory parameters:
the first contains the digits to say (1234 in your case)
the second contains the digits that end the command if pressed by the
user

Example
SAY DIGITS 1234 1
says 1234 and stops as soon as the user presses 1

SAY DIGITS 1234 1#
says 1234 and stops as soon as the user presses 1 or #

If you don't want the user to interrupt you can pass an empty string as
second parameter:

SAY DIGITS 1234 "" (of corse the quotes must be escaped in php)

Cheers

Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IVR stats

[Stefan Reuter]

On Wed, 2005-02-23 at 14:50 -0600, Anton Krall wrote:
> Not a bad choice.. Ive seen software like XT or XC something that does this
> but for call queues... So.. Maybe a simple logger command here and there :) 

You also get the events as NewExtenEvents via the Manager API. But you
will have to write your own little script that logs them.

For general information on the Manager API see
http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+API

For the NewExtenEvent see
http://www.voip-info.org/tiki-index.php?page=asterisk%20manager%20events
http://asterisk-java.sourceforge.net/api/net/sf/asterisk/manager/event/NewExtenEvent.html

Stefan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can't run AGI for outbound call

[Stefan Reuter]

On Tue, 2005-02-15 at 00:07 +, Ívar Ragnarsson wrote:
> The problem is Asterisk does not seem to know the AGI application.  I
> create a file test.call and place it in the outbound spool directory:
> 
> the test.call file looks like this: 
> #Simple test call script. 
> #call my NetMeeting client 
> Channel: h323/[EMAIL PROTECTED] 
> MaxRetries: 2 
> RetryTime: 60 
> WaitTime: 30 
> Application: AGI(agi-test.agi) 
> Data: 1234
[SNIP]
> Feb 14 23:53:25 WARNING[7958]: pbx.c:4164 ast_pbx_run_app: No
> such application 'AGI(agi-test.agi)' 

Asterisk is looking for an application called 'AGI(agi-test.agi)' and
that one obviously does not exist.
The Application property must only contain the name of the application,
i.e. AGI. Any parameters are given via the Data property.
What you probably want is:

Channel: h323/[EMAIL PROTECTED] 
MaxRetries: 2 
RetryTime: 60 
WaitTime: 30 
Application: AGI 
Data: agi-test.agi

stefan


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse

[Stefan Reuter]

Doug Lytle wrote:
Keep in mind, you need to include both the  
P003-07-3-00  and P0S3-07-3-00 in the SIPDefault.cnf and OS79XX.txt
You need the P003-07-3-00 in OS79XX.TXT as it contains the application 
loader and P0S3-07-3-00 in the SIP(Default|).cnf as it contains the 
actual sip firmware.

Stefan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users