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
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Maybe it's been replaced by the Monitor app?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
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.
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
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
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
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
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/
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Peter Svensson wrote:
I think the complaint is that asterisk does not use the destination
address for the incoming request packet as the source address for the
outgoing packet holding the reply. This will prevent the requestor from
matching the quadtuple (src addr, dst addr, src port, dst port)
hi,
bristuff comes with these two applications - and too little info to
understand what they are for. Anyone has a clue and is willing to share
it?
i don't really know what they are good for but i had a look at the
source and think i know what they do.
imagine you have two extension 8767
ideas?
stefan
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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|MAC).cnf as it contains the
actual sip firmware.
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
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
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
hi,
Is there a way to route incoming ISDN calls to different contexts based on
the MSN dailled?
i am sending all calls to a context called capi-in where i use GotoIf based
on DNID to route them.
capi.conf:
[interfaces]
msn=123
incomingmsn=123,456,789
context=capi-in
extensions.conf:
Hi,
i am looking for a tool to merge the two wav files of a monitored call
into one. soxmix does that well but actually merges the two channels.
I would prefer a solution that creates a stereo wav file of the two mono
files so you have the called party on one (e.g. left) channel and the
calling
Um, sorry, but if SMS is not for sending to mobile phones, then what is it
for (if it matters, I'm not in the US) ?
i am in germany and use app_sms to send sms messgaes to mobile phones.
app_sms does not talk directly to mobile phones but to the sms message
center that in turn sends the sms
I realize that the most interesting aspect for some is to talk back
and forth between phone and asterisk on the local server which is
nice too.
yes i almost forgot about that one. app_sms can also act as a message
center itself: so if you connect some sms capable fixed line phone to
asterisk
is MY MSN ? and b: ?
If the message center number was 01234567890 what would that line be using
ZAP ?
then you dont need a msn...
when using ZAP I suppose its something like Channel: Zap/r1/01234567890
the normal syntax for dialing out
stefan
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On Tue, 2004-12-21 at 17:05 +0100, Gutzke Klaus wrote:
Is it possible to use the sms_app over zap without the .call file?
in newer versions of asterisk there is smsq - a tool that sends sms from
the command line.
see the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms
I tried the
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
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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
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.
to
reconnect every few seconds.
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to use for your application
solutions similar to Asterisk-Java may be available that hide this and
similar obstacles.
=Stefan
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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
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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
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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
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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
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
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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
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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
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
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).
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
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.
=Stefan
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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
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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
();
}
http://asterisk-java.org
I am sure other libraries provide similar abstraction.
=Stefan
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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
--
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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
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Telefon
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
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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
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E-Mail
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.
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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
developers?
Best regards,
Stefan
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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
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
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
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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
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
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
/wiki-Asterisk+FastAGI
=Stefan
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than switching the language.
=Stefan
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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
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
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
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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
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
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
...
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
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
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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
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
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:
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
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
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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
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Description:
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
a list of all active channels.
=Stefan
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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:
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
/Communications/MobicentsAsteriskRA
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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
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this.
no it is not designed to handle this.
Have a look at http://www.voip-info.org/wiki-Asterisk+Manager+Proxy
=Stefan
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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
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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
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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
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/index.php?page=Asterisk+Manager+API+Action+DBPut
=Stefan
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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
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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
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.
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
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
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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
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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
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
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
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
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
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
The problem I'm having with understanding this is for incoming calls
from broadvoice. If I remove the context=from-broadvoice from the
above, incoming calls from broadvoice are dropped into the bogon-calls
context (no service available message).
just add the context = from-broadvoice to the
hi,
On Fri, 2004-08-13 at 10:32, [EMAIL PROTECTED] wrote:
Aug 13 12:11:00 ERROR[1225231280]: Queue name too long
thats actually just the cause ;)
dont use the calleridnum (thats actually the sms text you are about to
send!) as a queue name, better use a fixed string like 'default'
[smsdial]
hi daniel,
I'd guess because there are several enqueued with the same
name and when they aren't send they'll be overwritten in queue?!
no they won't be overwritten, a queue can hold many sms to send.
How can I fix this?
no need to fix
regards,
stefan
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:
;
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
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?
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
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.
allows you to easily implement AGI
scripts in Java: http://asterisk-java.org
=Stefan
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Germany
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