asterisk-java-users — Asterisk-Java User List

[Asterisk-java-users] Stable release of 3.x series?

[Steve S. <ss...@us...>]

Hello,

We're currently using the 2.0.x version of asterisk-java, and I've just 
determined we need to upgrade to something much newer that supports 
PJSIP.  I don't see much support for PJSIP in the 2.0.x branches.

It looks like there's a lot of recent activity, and I see what looks 
like support for PJSIP in the Master branch, with a lot of tags of 3.0, 
3.1, 3.5, 3.6, etc...

However, I don't see anything in github marked as an official release, 
and there doesn't seem to be a branch where 3.x is being maintained.

So that leads me to a couple questions.

Are the tags marked in the 3.x series compatible with 2.0.x? i.e. Can I 
largely drop something tagged as a 3.x release and expect code written 
against 2.0.x to be compatible?

Are the 3.x tags stable?  Is one recommended over another?  I'd like to 
use something newer than 2.0.4, which looks like the latest branch, but 
I'm not sure what to start with, and if things might break later if I 
then use some newer 3.x tag.


Thanks.

-- 

Re: [Asterisk-java-users] FastAGI call for custom commands

[Yves <yv...@gm...>]

Hi,

"streamFile" is not an customAGICommand. If you want to stream a
soundfile, just use

channel.streamFile("filename");

regards,
Yves

Am 17.06.2020 um 21:47 schrieb Wagner Feliziani via Asterisk-java-users:
> Hi,
>
> I am trying to call custom commands using FastAGI as stated in the tutorial : channel.sendCommand(AgiCommand)
>
> I tried :
>
> ...
> 	class CustomAgiCommand implements AgiCommand {
> 		String cmd;
> 		//public CustomAgiCommand(String cmd) {this.cmd = cmd;}
> 		public CustomAgiCommand(String cmd) { this.cmd = cmd; }
> 		@Override public String buildCommand() {
> 			return cmd;
> 		}
> 		@Override public void setAsteriskVersion(AsteriskVersion arg0) {}}
> ...
>
> channel.sendCommand( new CustomAgiCommand("streamFile(\"tt-monkeys\")")   );
>
> But it dont work :
>
> SEVERE: AgiException running AgiScript CallReceiveHandler on AJ DaemonPool-1.1
> org.asteriskjava.fastagi.InvalidOrUnknownCommandException: Invalid or unknown command: streamFile("tt-monkeys")
>          at org.asteriskjava.fastagi.internal.AgiChannelImpl.sendCommand(AgiChannelImpl.java:130)
>          at CallReceiveHandler.service(CallReceiveHandler.java:42)
>          at org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConnectionHandler.java:178)
>          at org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnectionHandler.java:149)
>          at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
>          at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
>          at java.lang.Thread.run(Thread.java:748)
>
> Am I doing this wrong ?
>
> Regards,
> wagfeliz
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] FastAGI call for custom commands

[Wagner F. <wag...@ya...>]

Hi, 

I am trying to call custom commands using FastAGI as stated in the tutorial : channel.sendCommand(AgiCommand)

I tried : 

...
	class CustomAgiCommand implements AgiCommand {
		String cmd;
		//public CustomAgiCommand(String cmd) {this.cmd = cmd;}
		public CustomAgiCommand(String cmd) { this.cmd = cmd; }
		@Override public String buildCommand() {			
			return cmd; 
		}
		@Override public void setAsteriskVersion(AsteriskVersion arg0) {}}
...

channel.sendCommand( new CustomAgiCommand("streamFile(\"tt-monkeys\")")   );

But it dont work : 

SEVERE: AgiException running AgiScript CallReceiveHandler on AJ DaemonPool-1.1
org.asteriskjava.fastagi.InvalidOrUnknownCommandException: Invalid or unknown command: streamFile("tt-monkeys")
        at org.asteriskjava.fastagi.internal.AgiChannelImpl.sendCommand(AgiChannelImpl.java:130)
        at CallReceiveHandler.service(CallReceiveHandler.java:42)
        at org.asteriskjava.fastagi.internal.AgiConnectionHandler.runScript(AgiConnectionHandler.java:178)
        at org.asteriskjava.fastagi.internal.AgiConnectionHandler.run(AgiConnectionHandler.java:149)
        at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
        at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
        at java.lang.Thread.run(Thread.java:748)

Am I doing this wrong ?

Regards,
wagfeliz

[Asterisk-java-users] Bridging two live AGI Channels

[Sreekumar J. <sr...@dh...>]

Dear team,

I'm stuck in the process of linking two live AgiChannels such that the
audio is shared either ways between the callees of each AgiChannel.

I've tried these methods -

AgiChannel-> dial() from one to another, either ways
AgiChannel->bridge() from one to another, either ways
OriginateAction to Application "ConfBridge" on both channels back to back

- of which none worked. They all seem to be waiting for the AgiChannel to
be hung up.

If any of you have gone through similar pain before and solved it, can you
please give general instruction on how to go ahead on this ???

Regards,

Sreekumar

[Asterisk-java-users] Availability for support for Asterisk 16

[Greg B. <gf...@gm...>]

We are attempting to upgrade to asterisk 16 and there appears to not be
support for it through this project.  If there is intent to support it we
would be happy to assist by Beta testing.
GregBiltz

Re: [Asterisk-java-users] Wrong PJSIP_HEADER read

[Salahuddin A. <tx...@gm...>]

Hello Yves,

Thanks for your response and I am very sorry for the late reply.

OS: Debian (Jessie)
Asterisk: 13.20.0
Asterisk-Java: 2.0.3

Actually, regenerate this issue is a little harder, but in our environment,
in sipp we send a cancel from caller side after few milliseconds (50-58ms)
then this issue has been occurring. We did not set any delay in our agi
script. I think if you use multiple getVariable in the invite agi and send
a cancel from caller side immediately after INVITE, this issue should have
occurred.

Please let us know if is it possible to do any help from our side to
regenerate this issue.

Thanks & Regards,
Salah Ahmed

On Sun, Dec 23, 2018 at 6:47 PM Yves <yv...@gm...> wrote:

> Hello Ahmed,
>
> thanks for you valuable findings. To find out, whats going on, I´d like to
> copy your setup; so, what versions are you using concerning
> - OperatingSystem
> - Asterisk
> - Asterisk-Java
>
> How do you "time" the Hangup so that it occurs between the two
> getHeader-Methods... or do you delay the execution in between
> to better simulate, whats going on, if this special case occurs?
>
> regards,
> Yves
>
> Am 21.12.2018 um 01:08 schrieb Salahuddin Ahmed:
>
> Hello,
>
> Let's explain the issue,
> In our agi script has those following header read,
>
> AGI Code:
> =========================================
> String fromHeader = getHeaderValue("FROM");
> String paidHeader = getHeaderValue("P-ASSERTED-IDENTITY");
> String privacyHeader = getHeaderValue("PRIVACY");
> String toHeader = getHeaderValue("TO");
> =========================================
>
> 4573 port trace
> =========================================
> T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
>  GET
> VARIABLE"PJSIP_HEADER(read,FROM)".
>
> #
> T 172.32.10.70:38630 -> 172.32.60.11:4573 [AP]
>  HANGUP.
>
> #
> T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
>   GET VARIABLE
> "PJSIP_HEADER(read,P-ASSERTED-IDENTITY)".
>
> #
> T 172.32.10.70:38630 -> 172.32.60.11:4573 [AP]
>   200 result=1 (<sip:12016615429@172.32.100.100>;tag=1).
>
> #
> T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
>   GET VARIABLE
> "PJSIP_HEADER(read,PRIVACY)".
>
> #
> ===========================================
>
> When a new call arrived those lines are executing, and during that
> execution, if caller canceled the call then a HANGUP message send from
> asterisk to asterisk-java. If asterisk-java waits for any response of one
> of those 4 requests then I think asteris-agi get confused and the all
> further response will be shifted. Now according to code if asterisk-java
> received a response without "200 result=1" format, it discards that
> response, but this response does not belong to that request. In the next
> read, the actual value will appear.
>
> Now in this scenario,  what would be the solution? could anyone please
> suggest this?
>
> Thanks,
> Salah Ahmed
>
> On Tue, Dec 18, 2018 at 1:31 PM Salahuddin Ahmed <tx...@gm...>
> wrote:
>
>> Hello,
>>
>> Recently I have noticed an error in my service. After some investigation,
>> we found that its due to some calls is terminated very early.
>>
>> The scenario is, If Caller-A sends an INVITE to Asterisk, This INVITE
>> processed on an agi method. On that method, we read several sip custom and
>> default sip header for our processing.
>>
>> But during this processing, if Caller-A sends CANCEL then some header
>> value read get malformed. Actually, it seems it gets shifted, suppose if we
>> read To header first and then cseq, TO header value set to cseq value and
>> TO return NULL. We have added some debug log in asterisk code and found,
>> asterisk return good value. The time between INVITE and CANCEL is 50 ms.
>>
>> We have implemented AGIHangupException in our agi method, but this
>> exception does not catch by this CANCEL request.
>>
>> Any idea/solution is appreciated.
>>
>> Thanks in advance,
>> Salah Ahmed
>>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

Re: [Asterisk-java-users] Wrong PJSIP_HEADER read

[Yves <yv...@gm...>]

Hello Ahmed,

thanks for you valuable findings. To find out, whats going on, I´d like 
to copy your setup; so, what versions are you using concerning
- OperatingSystem
- Asterisk
- Asterisk-Java

How do you "time" the Hangup so that it occurs between the two 
getHeader-Methods... or do you delay the execution in between
to better simulate, whats going on, if this special case occurs?

regards,
Yves

Am 21.12.2018 um 01:08 schrieb Salahuddin Ahmed:
> Hello,
>
> Let's explain the issue,
> In our agi script has those following header read,
>
> AGI Code:
> =========================================
> String fromHeader = getHeaderValue("FROM");
> String paidHeader = getHeaderValue("P-ASSERTED-IDENTITY");
> String privacyHeader = getHeaderValue("PRIVACY");
> String toHeader = getHeaderValue("TO");
> =========================================
>
> 4573 port trace
> =========================================
> T 172.32.60.11:4573 <http://172.32.60.11:4573> -> 172.32.10.70:38630 
> <http://172.32.10.70:38630> [AP]
>  GET VARIABLE"PJSIP_HEADER(read,FROM)".
> #
> T 172.32.10.70:38630 <http://172.32.10.70:38630> -> 172.32.60.11:4573 
> <http://172.32.60.11:4573> [AP]
>  HANGUP.
> #
> T 172.32.60.11:4573 <http://172.32.60.11:4573> -> 172.32.10.70:38630 
> <http://172.32.10.70:38630> [AP]
>   GET VARIABLE "PJSIP_HEADER(read,P-ASSERTED-IDENTITY)".
> #
> T 172.32.10.70:38630 <http://172.32.10.70:38630> -> 172.32.60.11:4573 
> <http://172.32.60.11:4573> [AP]
>   200 result=1 (<sip:12016615429@172.32.100.100 
> <mailto:sip%3A12016615429@172.32.100.100>>;tag=1).
> #
> T 172.32.60.11:4573 <http://172.32.60.11:4573> -> 172.32.10.70:38630 
> <http://172.32.10.70:38630> [AP]
>   GET VARIABLE "PJSIP_HEADER(read,PRIVACY)".
> #
> ===========================================
>
> When a new call arrived those lines are executing, and during that 
> execution, if caller canceled the call then a HANGUP message send from 
> asterisk to asterisk-java. If asterisk-java waits for any response of 
> one of those 4 requests then I think asteris-agi get confused and the 
> all further response will be shifted. Now according to code if 
> asterisk-java received a response without "200 result=1" format, it 
> discards that response, but this response does not belong to that 
> request. In the next read, the actual value will appear.
>
> Now in this scenario,  what would be the solution? could anyone please 
> suggest this?
>
> Thanks,
> Salah Ahmed
>
> On Tue, Dec 18, 2018 at 1:31 PM Salahuddin Ahmed <tx...@gm... 
> <mailto:tx...@gm...>> wrote:
>
>     Hello,
>
>     Recently I have noticed an error in my service. After some
>     investigation, we found that its due to some calls is terminated
>     very early.
>
>     The scenario is, If Caller-A sends an INVITE to Asterisk, This
>     INVITE processed on an agi method. On that method, we read several
>     sip custom and default sip header for our processing.
>
>     But during this processing, if Caller-A sends CANCEL then some
>     header value read get malformed. Actually, it seems it gets
>     shifted, suppose if we read To header first and then cseq, TO
>     header value set to cseq value and TO return NULL. We have added
>     some debug log in asterisk code and found, asterisk return good
>     value. The time between INVITE and CANCEL is 50 ms.
>
>     We have implemented AGIHangupException in our agi method, but this
>     exception does not catch by this CANCEL request.
>
>     Any idea/solution is appreciated.
>
>     Thanks in advance,
>     Salah Ahmed
>
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Wrong PJSIP_HEADER read

[Salahuddin A. <tx...@gm...>]

Hello,

Let's explain the issue,
In our agi script has those following header read,

AGI Code:
=========================================
String fromHeader = getHeaderValue("FROM");
String paidHeader = getHeaderValue("P-ASSERTED-IDENTITY");
String privacyHeader = getHeaderValue("PRIVACY");
String toHeader = getHeaderValue("TO");
=========================================

4573 port trace
=========================================
T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
 GET
VARIABLE"PJSIP_HEADER(read,FROM)".

#
T 172.32.10.70:38630 -> 172.32.60.11:4573 [AP]
 HANGUP.

#
T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
  GET VARIABLE
"PJSIP_HEADER(read,P-ASSERTED-IDENTITY)".

#
T 172.32.10.70:38630 -> 172.32.60.11:4573 [AP]
  200 result=1 (<sip:12016615429@172.32.100.100>;tag=1).

#
T 172.32.60.11:4573 -> 172.32.10.70:38630 [AP]
  GET VARIABLE
"PJSIP_HEADER(read,PRIVACY)".

#
===========================================

When a new call arrived those lines are executing, and during that
execution, if caller canceled the call then a HANGUP message send from
asterisk to asterisk-java. If asterisk-java waits for any response of one
of those 4 requests then I think asteris-agi get confused and the all
further response will be shifted. Now according to code if asterisk-java
received a response without "200 result=1" format, it discards that
response, but this response does not belong to that request. In the next
read, the actual value will appear.

Now in this scenario,  what would be the solution? could anyone please
suggest this?

Thanks,
Salah Ahmed

On Tue, Dec 18, 2018 at 1:31 PM Salahuddin Ahmed <tx...@gm...> wrote:

> Hello,
>
> Recently I have noticed an error in my service. After some investigation,
> we found that its due to some calls is terminated very early.
>
> The scenario is, If Caller-A sends an INVITE to Asterisk, This INVITE
> processed on an agi method. On that method, we read several sip custom and
> default sip header for our processing.
>
> But during this processing, if Caller-A sends CANCEL then some header
> value read get malformed. Actually, it seems it gets shifted, suppose if we
> read To header first and then cseq, TO header value set to cseq value and
> TO return NULL. We have added some debug log in asterisk code and found,
> asterisk return good value. The time between INVITE and CANCEL is 50 ms.
>
> We have implemented AGIHangupException in our agi method, but this
> exception does not catch by this CANCEL request.
>
> Any idea/solution is appreciated.
>
> Thanks in advance,
> Salah Ahmed
>

[Asterisk-java-users] Wrong PJSIP_HEADER read

[Salahuddin A. <tx...@gm...>]

Hello,

Recently I have noticed an error in my service. After some investigation,
we found that its due to some calls is terminated very early.

The scenario is, If Caller-A sends an INVITE to Asterisk, This INVITE
processed on an agi method. On that method, we read several sip custom and
default sip header for our processing.

But during this processing, if Caller-A sends CANCEL then some header value
read get malformed. Actually, it seems it gets shifted, suppose if we read
To header first and then cseq, TO header value set to cseq value and TO
return NULL. We have added some debug log in asterisk code and found,
asterisk return good value. The time between INVITE and CANCEL is 50 ms.

We have implemented AGIHangupException in our agi method, but this
exception does not catch by this CANCEL request.

Any idea/solution is appreciated.

Thanks in advance,
Salah Ahmed

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[adriano s. <sle...@bs...>]

Hi  Brandon!
Some updates....
I am already able to dial through the management interface. Now all i have
to do is use AGI to send some useful info to the call.
If you have any other tips that you want to share I will be very grateful
to you!

Regards,
Adriano Santos

Em sáb, 24 de nov de 2018 às 19:54, adriano santos <sle...@bs...>
escreveu:

> Hello Brandon!
> Thank you for your suggestions. I will try to use the agi in the context
> to and start the call through the management interface.
> If I succeed, I'll leave the shared solution here.
>
> Regards,
> Adriano Santos
>
> Em sáb, 24 de nov de 2018 às 15:21, Brandon Haugen <bh...@ta...>
> escreveu:
>
>> Hey Adriano,
>>
>> I was looking for a way to send audio through the Manager Interface and I
>> found this in the FAQ for Asterisk Java.
>>
>> Currently, there is no support for sending media or files across the
>>> Manager interface or the Gateway interface. Having access to the media
>>> stream without participating in the call would require saving it to a file
>>> or modifications to Asterisk itself, and you can expose any files (media or
>>> otherwise) through some other means (HTTP, SSH, etc).
>>
>>
>> So you will need to play the audio through either regular dialplan code
>> in *extensions.conf* or by using Agi and Asterisk Java together. Since
>> you are specifying the context as "from-internal" you can call out to Agi
>> from within that context and use the code I shared earlier in this thread.
>>
>> Thanks,
>> Brandon
>>
>> On Fri, Nov 23, 2018 at 6:31 PM adriano santos <sle...@bs...>
>> wrote:
>>
>>> Hi!
>>>
>>> I'm trying to use the management interface to originate a call, but in
>>> the moment I can not do play an audio file. The code is something like this:
>>>
>>> @Slf4j
>>> public class OriginateCall {
>>> private static final int PORT = 5038;
>>> private static final String HOSTNAME = "*.*.*.*";
>>> private static final String USERNAME = "*";
>>> private static final String PASSWORD = "*";
>>> private static final String CONTEXT = "context";
>>> // Number to call, prefaced by 1+areacode (just like your home phone).
>>> public String call = "some_number";
>>> private ManagerConnection managerConnection;
>>>
>>> public OriginateCall() {
>>> ManagerConnectionFactory factory = new
>>> ManagerConnectionFactory(HOSTNAME, PORT, USERNAME, PASSWORD);
>>>
>>> this.managerConnection = factory.createManagerConnection();
>>> }
>>>
>>> public void call() throws IllegalStateException, IOException,
>>> AuthenticationFailedException, TimeoutException {
>>> OriginateAction originateAction;
>>> ManagerResponse originateResponse;
>>>
>>> originateAction = new OriginateAction();
>>> /*
>>> * Format the call for dialing Channel example:
>>> Local/12065551234@outgoing-42
>>> */
>>> Integer timeoutCall = 50000;
>>> originateAction.setChannel("SIP/" + call + "@" + CONTEXT);
>>>         originateAction.setContext("from-internal");
>>>         originateAction.setCallerId("55011100"); // what will be showed
>>> on the phone screen (in most cases your phone)
>>>         originateAction.setExten(""/*[targetExten]*/); //where to call..
>>> the target extension... internal extension or the outgoing number.. the
>>> 0[nomberToCall]
>>>         originateAction.setPriority(1);// priority of the call
>>>
>>> originateAction.setTimeout(timeoutCall ); // the time that a pickup
>>> event will be waited for
>>>         originateAction.setVariable("UUID",
>>> UUID.randomUUID().toString()); // asigning a unique ID in order to be able
>>> to hangup the call.
>>>
>>> managerConnection.login();
>>>
>>> /*
>>> * send the originate action
>>> */
>>> originateResponse = managerConnection.sendAction(originateAction, 50000);
>>>
>>> // print out whether the originate succeeded or not
>>> log.info("Call Response -> {}", originateResponse.getResponse());
>>>
>>> // and finally log off and disconnect
>>> managerConnection.logoff();
>>> }
>>> public static void main(String... args) throws IllegalStateException,
>>> IOException, AuthenticationFailedException, TimeoutException {
>>> OriginateCall originate = new OriginateCall();
>>>     originate.call();
>>> }
>>>
>>> }
>>>
>>> I'll view the information you sent me. Again thank you for your time and
>>> knowledge.
>>>
>>> Regards,
>>> Adriano Santos
>>>
>>> Em sex, 23 de nov de 2018 às 19:45, Brandon Haugen <
>>> bh...@ta...> escreveu:
>>>
>>>> Hey Adriano,
>>>>
>>>> Yes, the code I shared is showing how to receive a call and respond
>>>> with audio.
>>>>
>>>> I have a scenario where we allow users to initiate a call to themselves
>>>> (by pressing a button in a Web App) and when they pick up the call, we
>>>> stream a couple of audio files (one after the other) to give them
>>>> instructions for the system they are using. In our scenario though, we are
>>>> just initiating the call through the use of Call Files (
>>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files). When
>>>> the user picks up, they just drop into the context specified in the Call
>>>> File which directs them to our Asterisk Java AgiScript. In this case, I
>>>> still use `getData` on the `AgiChannel` to stream the files when I need
>>>> DTMF, otherwise I just use the `exec` method on the AgiChannel to use the
>>>> Playback application within Asterisk.
>>>>
>>>> If you are going to originate the Call in some other way, I won't be of
>>>> much help at the moment as I have not worked with the Asterisk Manager
>>>> Interface, (
>>>> https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API)
>>>> yet. Asterisk Java does also support the Manager Interface so you might
>>>> find something that more closely resembles what you are looking to do at
>>>> http://asterisk-java.org .
>>>>
>>>> On Fri, Nov 23, 2018 at 3:21 PM adriano santos <sle...@bs...>
>>>> wrote:
>>>>
>>>>> Hi  Brandon! Thank you for your response.
>>>>>
>>>>> Is the example you sent to receive a call and respond with correct
>>>>> audio?
>>>>> What I need to do is originate a call to cell phone and play audio.
>>>>> For now no need for DTMF.
>>>>>
>>>>> Regards,
>>>>> Adriano Santos
>>>>>
>>>>> Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <
>>>>> bh...@ta...> escreveu:
>>>>>
>>>>>> Hey Adriano,
>>>>>>
>>>>>> I have within the last few months started using Asterisk-Java for a
>>>>>> project and you can certainly playback an audio file. There are a couple of
>>>>>> things that you will have to think about to determine the best approach for
>>>>>> your situation, here is what I can tell you though
>>>>>>
>>>>>>    - If you do *not* need to listen for DTMF input (keypresses)
>>>>>>    while playing the audio file then you can play an audio file on an
>>>>>>    AgiChannel in Java by using something like `agiChannel.exec("playback",
>>>>>>    "<path to audio file>");`
>>>>>>       - This example in the Asterisk Java GitHub repository shows
>>>>>>       answering a call, playing a file back, and then hanging up (note that in
>>>>>>       the example they rely on methods defined in the AgiOperations class so they
>>>>>>       aren't operating directly on the AgiChannel like the line of code I am
>>>>>>       sharing is doing)
>>>>>>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>>>>>>    - If you *do* need to listen for DTMF input while playing the
>>>>>>    audio file then you will want to use the `getData` or `getOption` methods
>>>>>>    defined on AgiChannel. You will need to pick which method is appropriate
>>>>>>    depending on the length of DTMF you need to listen for (`getOption` returns
>>>>>>    a single char while `getData` will return a String)
>>>>>>
>>>>>> Thanks,
>>>>>> Brandon
>>>>>>
>>>>>> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
>>>>>> wrote:
>>>>>>
>>>>>>> Hello, guys!
>>>>>>>
>>>>>>> I would like to play an audio file on a initialized call via java. I
>>>>>>> haven't found an example for this.
>>>>>>> Is this possible?
>>>>>>>
>>>>>>> --
>>>>>>> Adriano P. Santos
>>>>>>>
>>>>>>> "O homem erudito é um descobridor de fatos que já existem - mas o
>>>>>>> homem sábio é um criador de valores que não existem e que ele faz existir."
>>>>>>> Albert Einstein
>>>>>>> _______________________________________________
>>>>>>> Asterisk-java-users mailing list
>>>>>>> Ast...@li...
>>>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>>>
>>>>>> _______________________________________________
>>>>>> Asterisk-java-users mailing list
>>>>>> Ast...@li...
>>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Adriano P. Santos
>>>>>
>>>>> "O homem erudito é um descobridor de fatos que já existem - mas o
>>>>> homem sábio é um criador de valores que não existem e que ele faz existir."
>>>>> Albert Einstein
>>>>> _______________________________________________
>>>>> Asterisk-java-users mailing list
>>>>> Ast...@li...
>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>
>>>> _______________________________________________
>>>> Asterisk-java-users mailing list
>>>> Ast...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>
>>>
>>>
>>> --
>>> Adriano P. Santos
>>>
>>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>>> sábio é um criador de valores que não existem e que ele faz existir."
>>> Albert Einstein
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>
>
> --
> Adriano P. Santos
>
> "O homem erudito é um descobridor de fatos que já existem - mas o homem
> sábio é um criador de valores que não existem e que ele faz existir."
> Albert Einstein
>


-- 
Adriano P. Santos

"O homem erudito é um descobridor de fatos que já existem - mas o homem
sábio é um criador de valores que não existem e que ele faz existir."
Albert Einstein

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[adriano s. <sle...@bs...>]

Hello Brandon!
Thank you for your suggestions. I will try to use the agi in the context to
and start the call through the management interface.
If I succeed, I'll leave the shared solution here.

Regards,
Adriano Santos

Em sáb, 24 de nov de 2018 às 15:21, Brandon Haugen <bh...@ta...>
escreveu:

> Hey Adriano,
>
> I was looking for a way to send audio through the Manager Interface and I
> found this in the FAQ for Asterisk Java.
>
> Currently, there is no support for sending media or files across the
>> Manager interface or the Gateway interface. Having access to the media
>> stream without participating in the call would require saving it to a file
>> or modifications to Asterisk itself, and you can expose any files (media or
>> otherwise) through some other means (HTTP, SSH, etc).
>
>
> So you will need to play the audio through either regular dialplan code in
> *extensions.conf* or by using Agi and Asterisk Java together. Since you
> are specifying the context as "from-internal" you can call out to Agi from
> within that context and use the code I shared earlier in this thread.
>
> Thanks,
> Brandon
>
> On Fri, Nov 23, 2018 at 6:31 PM adriano santos <sle...@bs...>
> wrote:
>
>> Hi!
>>
>> I'm trying to use the management interface to originate a call, but in
>> the moment I can not do play an audio file. The code is something like this:
>>
>> @Slf4j
>> public class OriginateCall {
>> private static final int PORT = 5038;
>> private static final String HOSTNAME = "*.*.*.*";
>> private static final String USERNAME = "*";
>> private static final String PASSWORD = "*";
>> private static final String CONTEXT = "context";
>> // Number to call, prefaced by 1+areacode (just like your home phone).
>> public String call = "some_number";
>> private ManagerConnection managerConnection;
>>
>> public OriginateCall() {
>> ManagerConnectionFactory factory = new ManagerConnectionFactory(HOSTNAME,
>> PORT, USERNAME, PASSWORD);
>>
>> this.managerConnection = factory.createManagerConnection();
>> }
>>
>> public void call() throws IllegalStateException, IOException,
>> AuthenticationFailedException, TimeoutException {
>> OriginateAction originateAction;
>> ManagerResponse originateResponse;
>>
>> originateAction = new OriginateAction();
>> /*
>> * Format the call for dialing Channel example:
>> Local/12065551234@outgoing-42
>> */
>> Integer timeoutCall = 50000;
>> originateAction.setChannel("SIP/" + call + "@" + CONTEXT);
>>         originateAction.setContext("from-internal");
>>         originateAction.setCallerId("55011100"); // what will be showed
>> on the phone screen (in most cases your phone)
>>         originateAction.setExten(""/*[targetExten]*/); //where to call..
>> the target extension... internal extension or the outgoing number.. the
>> 0[nomberToCall]
>>         originateAction.setPriority(1);// priority of the call
>>
>> originateAction.setTimeout(timeoutCall ); // the time that a pickup event
>> will be waited for
>>         originateAction.setVariable("UUID",
>> UUID.randomUUID().toString()); // asigning a unique ID in order to be able
>> to hangup the call.
>>
>> managerConnection.login();
>>
>> /*
>> * send the originate action
>> */
>> originateResponse = managerConnection.sendAction(originateAction, 50000);
>>
>> // print out whether the originate succeeded or not
>> log.info("Call Response -> {}", originateResponse.getResponse());
>>
>> // and finally log off and disconnect
>> managerConnection.logoff();
>> }
>> public static void main(String... args) throws IllegalStateException,
>> IOException, AuthenticationFailedException, TimeoutException {
>> OriginateCall originate = new OriginateCall();
>>     originate.call();
>> }
>>
>> }
>>
>> I'll view the information you sent me. Again thank you for your time and
>> knowledge.
>>
>> Regards,
>> Adriano Santos
>>
>> Em sex, 23 de nov de 2018 às 19:45, Brandon Haugen <bh...@ta...>
>> escreveu:
>>
>>> Hey Adriano,
>>>
>>> Yes, the code I shared is showing how to receive a call and respond with
>>> audio.
>>>
>>> I have a scenario where we allow users to initiate a call to themselves
>>> (by pressing a button in a Web App) and when they pick up the call, we
>>> stream a couple of audio files (one after the other) to give them
>>> instructions for the system they are using. In our scenario though, we are
>>> just initiating the call through the use of Call Files (
>>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files). When
>>> the user picks up, they just drop into the context specified in the Call
>>> File which directs them to our Asterisk Java AgiScript. In this case, I
>>> still use `getData` on the `AgiChannel` to stream the files when I need
>>> DTMF, otherwise I just use the `exec` method on the AgiChannel to use the
>>> Playback application within Asterisk.
>>>
>>> If you are going to originate the Call in some other way, I won't be of
>>> much help at the moment as I have not worked with the Asterisk Manager
>>> Interface, (
>>> https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API)
>>> yet. Asterisk Java does also support the Manager Interface so you might
>>> find something that more closely resembles what you are looking to do at
>>> http://asterisk-java.org .
>>>
>>> On Fri, Nov 23, 2018 at 3:21 PM adriano santos <sle...@bs...>
>>> wrote:
>>>
>>>> Hi  Brandon! Thank you for your response.
>>>>
>>>> Is the example you sent to receive a call and respond with correct
>>>> audio?
>>>> What I need to do is originate a call to cell phone and play audio. For
>>>> now no need for DTMF.
>>>>
>>>> Regards,
>>>> Adriano Santos
>>>>
>>>> Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <
>>>> bh...@ta...> escreveu:
>>>>
>>>>> Hey Adriano,
>>>>>
>>>>> I have within the last few months started using Asterisk-Java for a
>>>>> project and you can certainly playback an audio file. There are a couple of
>>>>> things that you will have to think about to determine the best approach for
>>>>> your situation, here is what I can tell you though
>>>>>
>>>>>    - If you do *not* need to listen for DTMF input (keypresses) while
>>>>>    playing the audio file then you can play an audio file on an AgiChannel in
>>>>>    Java by using something like `agiChannel.exec("playback", "<path to audio
>>>>>    file>");`
>>>>>       - This example in the Asterisk Java GitHub repository shows
>>>>>       answering a call, playing a file back, and then hanging up (note that in
>>>>>       the example they rely on methods defined in the AgiOperations class so they
>>>>>       aren't operating directly on the AgiChannel like the line of code I am
>>>>>       sharing is doing)
>>>>>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>>>>>    - If you *do* need to listen for DTMF input while playing the
>>>>>    audio file then you will want to use the `getData` or `getOption` methods
>>>>>    defined on AgiChannel. You will need to pick which method is appropriate
>>>>>    depending on the length of DTMF you need to listen for (`getOption` returns
>>>>>    a single char while `getData` will return a String)
>>>>>
>>>>> Thanks,
>>>>> Brandon
>>>>>
>>>>> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
>>>>> wrote:
>>>>>
>>>>>> Hello, guys!
>>>>>>
>>>>>> I would like to play an audio file on a initialized call via java. I
>>>>>> haven't found an example for this.
>>>>>> Is this possible?
>>>>>>
>>>>>> --
>>>>>> Adriano P. Santos
>>>>>>
>>>>>> "O homem erudito é um descobridor de fatos que já existem - mas o
>>>>>> homem sábio é um criador de valores que não existem e que ele faz existir."
>>>>>> Albert Einstein
>>>>>> _______________________________________________
>>>>>> Asterisk-java-users mailing list
>>>>>> Ast...@li...
>>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>>
>>>>> _______________________________________________
>>>>> Asterisk-java-users mailing list
>>>>> Ast...@li...
>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>
>>>>
>>>>
>>>> --
>>>> Adriano P. Santos
>>>>
>>>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>>>> sábio é um criador de valores que não existem e que ele faz existir."
>>>> Albert Einstein
>>>> _______________________________________________
>>>> Asterisk-java-users mailing list
>>>> Ast...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>>
>>
>> --
>> Adriano P. Santos
>>
>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>> sábio é um criador de valores que não existem e que ele faz existir."
>> Albert Einstein
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>


-- 
Adriano P. Santos

"O homem erudito é um descobridor de fatos que já existem - mas o homem
sábio é um criador de valores que não existem e que ele faz existir."
Albert Einstein

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[Brandon H. <bh...@ta...>]

Hey Adriano,

I was looking for a way to send audio through the Manager Interface and I
found this in the FAQ for Asterisk Java.

Currently, there is no support for sending media or files across the
> Manager interface or the Gateway interface. Having access to the media
> stream without participating in the call would require saving it to a file
> or modifications to Asterisk itself, and you can expose any files (media or
> otherwise) through some other means (HTTP, SSH, etc).


So you will need to play the audio through either regular dialplan code in
*extensions.conf* or by using Agi and Asterisk Java together. Since you are
specifying the context as "from-internal" you can call out to Agi from
within that context and use the code I shared earlier in this thread.

Thanks,
Brandon

On Fri, Nov 23, 2018 at 6:31 PM adriano santos <sle...@bs...> wrote:

> Hi!
>
> I'm trying to use the management interface to originate a call, but in the
> moment I can not do play an audio file. The code is something like this:
>
> @Slf4j
> public class OriginateCall {
> private static final int PORT = 5038;
> private static final String HOSTNAME = "*.*.*.*";
> private static final String USERNAME = "*";
> private static final String PASSWORD = "*";
> private static final String CONTEXT = "context";
> // Number to call, prefaced by 1+areacode (just like your home phone).
> public String call = "some_number";
> private ManagerConnection managerConnection;
>
> public OriginateCall() {
> ManagerConnectionFactory factory = new ManagerConnectionFactory(HOSTNAME,
> PORT, USERNAME, PASSWORD);
>
> this.managerConnection = factory.createManagerConnection();
> }
>
> public void call() throws IllegalStateException, IOException,
> AuthenticationFailedException, TimeoutException {
> OriginateAction originateAction;
> ManagerResponse originateResponse;
>
> originateAction = new OriginateAction();
> /*
> * Format the call for dialing Channel example:
> Local/12065551234@outgoing-42
> */
> Integer timeoutCall = 50000;
> originateAction.setChannel("SIP/" + call + "@" + CONTEXT);
>         originateAction.setContext("from-internal");
>         originateAction.setCallerId("55011100"); // what will be showed on
> the phone screen (in most cases your phone)
>         originateAction.setExten(""/*[targetExten]*/); //where to call..
> the target extension... internal extension or the outgoing number.. the
> 0[nomberToCall]
>         originateAction.setPriority(1);// priority of the call
>
> originateAction.setTimeout(timeoutCall ); // the time that a pickup event
> will be waited for
>         originateAction.setVariable("UUID", UUID.randomUUID().toString());
> // asigning a unique ID in order to be able to hangup the call.
>
> managerConnection.login();
>
> /*
> * send the originate action
> */
> originateResponse = managerConnection.sendAction(originateAction, 50000);
>
> // print out whether the originate succeeded or not
> log.info("Call Response -> {}", originateResponse.getResponse());
>
> // and finally log off and disconnect
> managerConnection.logoff();
> }
> public static void main(String... args) throws IllegalStateException,
> IOException, AuthenticationFailedException, TimeoutException {
> OriginateCall originate = new OriginateCall();
>     originate.call();
> }
>
> }
>
> I'll view the information you sent me. Again thank you for your time and
> knowledge.
>
> Regards,
> Adriano Santos
>
> Em sex, 23 de nov de 2018 às 19:45, Brandon Haugen <bh...@ta...>
> escreveu:
>
>> Hey Adriano,
>>
>> Yes, the code I shared is showing how to receive a call and respond with
>> audio.
>>
>> I have a scenario where we allow users to initiate a call to themselves
>> (by pressing a button in a Web App) and when they pick up the call, we
>> stream a couple of audio files (one after the other) to give them
>> instructions for the system they are using. In our scenario though, we are
>> just initiating the call through the use of Call Files (
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files). When
>> the user picks up, they just drop into the context specified in the Call
>> File which directs them to our Asterisk Java AgiScript. In this case, I
>> still use `getData` on the `AgiChannel` to stream the files when I need
>> DTMF, otherwise I just use the `exec` method on the AgiChannel to use the
>> Playback application within Asterisk.
>>
>> If you are going to originate the Call in some other way, I won't be of
>> much help at the moment as I have not worked with the Asterisk Manager
>> Interface, (
>> https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API)
>> yet. Asterisk Java does also support the Manager Interface so you might
>> find something that more closely resembles what you are looking to do at
>> http://asterisk-java.org .
>>
>> On Fri, Nov 23, 2018 at 3:21 PM adriano santos <sle...@bs...>
>> wrote:
>>
>>> Hi  Brandon! Thank you for your response.
>>>
>>> Is the example you sent to receive a call and respond with correct
>>> audio?
>>> What I need to do is originate a call to cell phone and play audio. For
>>> now no need for DTMF.
>>>
>>> Regards,
>>> Adriano Santos
>>>
>>> Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <
>>> bh...@ta...> escreveu:
>>>
>>>> Hey Adriano,
>>>>
>>>> I have within the last few months started using Asterisk-Java for a
>>>> project and you can certainly playback an audio file. There are a couple of
>>>> things that you will have to think about to determine the best approach for
>>>> your situation, here is what I can tell you though
>>>>
>>>>    - If you do *not* need to listen for DTMF input (keypresses) while
>>>>    playing the audio file then you can play an audio file on an AgiChannel in
>>>>    Java by using something like `agiChannel.exec("playback", "<path to audio
>>>>    file>");`
>>>>       - This example in the Asterisk Java GitHub repository shows
>>>>       answering a call, playing a file back, and then hanging up (note that in
>>>>       the example they rely on methods defined in the AgiOperations class so they
>>>>       aren't operating directly on the AgiChannel like the line of code I am
>>>>       sharing is doing)
>>>>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>>>>    - If you *do* need to listen for DTMF input while playing the audio
>>>>    file then you will want to use the `getData` or `getOption` methods defined
>>>>    on AgiChannel. You will need to pick which method is appropriate depending
>>>>    on the length of DTMF you need to listen for (`getOption` returns a single
>>>>    char while `getData` will return a String)
>>>>
>>>> Thanks,
>>>> Brandon
>>>>
>>>> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
>>>> wrote:
>>>>
>>>>> Hello, guys!
>>>>>
>>>>> I would like to play an audio file on a initialized call via java. I
>>>>> haven't found an example for this.
>>>>> Is this possible?
>>>>>
>>>>> --
>>>>> Adriano P. Santos
>>>>>
>>>>> "O homem erudito é um descobridor de fatos que já existem - mas o
>>>>> homem sábio é um criador de valores que não existem e que ele faz existir."
>>>>> Albert Einstein
>>>>> _______________________________________________
>>>>> Asterisk-java-users mailing list
>>>>> Ast...@li...
>>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>>
>>>> _______________________________________________
>>>> Asterisk-java-users mailing list
>>>> Ast...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>
>>>
>>>
>>> --
>>> Adriano P. Santos
>>>
>>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>>> sábio é um criador de valores que não existem e que ele faz existir."
>>> Albert Einstein
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>
>
> --
> Adriano P. Santos
>
> "O homem erudito é um descobridor de fatos que já existem - mas o homem
> sábio é um criador de valores que não existem e que ele faz existir."
> Albert Einstein
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[adriano s. <sle...@bs...>]

Hi!

I'm trying to use the management interface to originate a call, but in the
moment I can not do play an audio file. The code is something like this:

@Slf4j
public class OriginateCall {
private static final int PORT = 5038;
private static final String HOSTNAME = "*.*.*.*";
private static final String USERNAME = "*";
private static final String PASSWORD = "*";
private static final String CONTEXT = "context";
// Number to call, prefaced by 1+areacode (just like your home phone).
public String call = "some_number";
private ManagerConnection managerConnection;

public OriginateCall() {
ManagerConnectionFactory factory = new ManagerConnectionFactory(HOSTNAME,
PORT, USERNAME, PASSWORD);

this.managerConnection = factory.createManagerConnection();
}

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

originateAction = new OriginateAction();
/*
* Format the call for dialing Channel example: Local/12065551234@outgoing-42
*/
Integer timeoutCall = 50000;
originateAction.setChannel("SIP/" + call + "@" + CONTEXT);
        originateAction.setContext("from-internal");
        originateAction.setCallerId("55011100"); // what will be showed on
the phone screen (in most cases your phone)
        originateAction.setExten(""/*[targetExten]*/); //where to call..
the target extension... internal extension or the outgoing number.. the
0[nomberToCall]
        originateAction.setPriority(1);// priority of the call

originateAction.setTimeout(timeoutCall ); // the time that a pickup event
will be waited for
        originateAction.setVariable("UUID", UUID.randomUUID().toString());
// asigning a unique ID in order to be able to hangup the call.

managerConnection.login();

/*
* send the originate action
*/
originateResponse = managerConnection.sendAction(originateAction, 50000);

// print out whether the originate succeeded or not
log.info("Call Response -> {}", originateResponse.getResponse());

// and finally log off and disconnect
managerConnection.logoff();
}
public static void main(String... args) throws IllegalStateException,
IOException, AuthenticationFailedException, TimeoutException {
OriginateCall originate = new OriginateCall();
    originate.call();
}

}

I'll view the information you sent me. Again thank you for your time and
knowledge.

Regards,
Adriano Santos

Em sex, 23 de nov de 2018 às 19:45, Brandon Haugen <bh...@ta...>
escreveu:

> Hey Adriano,
>
> Yes, the code I shared is showing how to receive a call and respond with
> audio.
>
> I have a scenario where we allow users to initiate a call to themselves
> (by pressing a button in a Web App) and when they pick up the call, we
> stream a couple of audio files (one after the other) to give them
> instructions for the system they are using. In our scenario though, we are
> just initiating the call through the use of Call Files (
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files). When the
> user picks up, they just drop into the context specified in the Call File
> which directs them to our Asterisk Java AgiScript. In this case, I still
> use `getData` on the `AgiChannel` to stream the files when I need DTMF,
> otherwise I just use the `exec` method on the AgiChannel to use the
> Playback application within Asterisk.
>
> If you are going to originate the Call in some other way, I won't be of
> much help at the moment as I have not worked with the Asterisk Manager
> Interface, (
> https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API)
> yet. Asterisk Java does also support the Manager Interface so you might
> find something that more closely resembles what you are looking to do at
> http://asterisk-java.org .
>
> On Fri, Nov 23, 2018 at 3:21 PM adriano santos <sle...@bs...>
> wrote:
>
>> Hi  Brandon! Thank you for your response.
>>
>> Is the example you sent to receive a call and respond with correct audio?
>> What I need to do is originate a call to cell phone and play audio. For
>> now no need for DTMF.
>>
>> Regards,
>> Adriano Santos
>>
>> Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <bh...@ta...>
>> escreveu:
>>
>>> Hey Adriano,
>>>
>>> I have within the last few months started using Asterisk-Java for a
>>> project and you can certainly playback an audio file. There are a couple of
>>> things that you will have to think about to determine the best approach for
>>> your situation, here is what I can tell you though
>>>
>>>    - If you do *not* need to listen for DTMF input (keypresses) while
>>>    playing the audio file then you can play an audio file on an AgiChannel in
>>>    Java by using something like `agiChannel.exec("playback", "<path to audio
>>>    file>");`
>>>       - This example in the Asterisk Java GitHub repository shows
>>>       answering a call, playing a file back, and then hanging up (note that in
>>>       the example they rely on methods defined in the AgiOperations class so they
>>>       aren't operating directly on the AgiChannel like the line of code I am
>>>       sharing is doing)
>>>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>>>    - If you *do* need to listen for DTMF input while playing the audio
>>>    file then you will want to use the `getData` or `getOption` methods defined
>>>    on AgiChannel. You will need to pick which method is appropriate depending
>>>    on the length of DTMF you need to listen for (`getOption` returns a single
>>>    char while `getData` will return a String)
>>>
>>> Thanks,
>>> Brandon
>>>
>>> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
>>> wrote:
>>>
>>>> Hello, guys!
>>>>
>>>> I would like to play an audio file on a initialized call via java. I
>>>> haven't found an example for this.
>>>> Is this possible?
>>>>
>>>> --
>>>> Adriano P. Santos
>>>>
>>>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>>>> sábio é um criador de valores que não existem e que ele faz existir."
>>>> Albert Einstein
>>>> _______________________________________________
>>>> Asterisk-java-users mailing list
>>>> Ast...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>>
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>>
>>
>> --
>> Adriano P. Santos
>>
>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>> sábio é um criador de valores que não existem e que ele faz existir."
>> Albert Einstein
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>


-- 
Adriano P. Santos

"O homem erudito é um descobridor de fatos que já existem - mas o homem
sábio é um criador de valores que não existem e que ele faz existir."
Albert Einstein

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[Brandon H. <bh...@ta...>]

Hey Adriano,

Yes, the code I shared is showing how to receive a call and respond with
audio.

I have a scenario where we allow users to initiate a call to themselves (by
pressing a button in a Web App) and when they pick up the call, we stream a
couple of audio files (one after the other) to give them instructions for
the system they are using. In our scenario though, we are just initiating
the call through the use of Call Files (
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files). When the
user picks up, they just drop into the context specified in the Call File
which directs them to our Asterisk Java AgiScript. In this case, I still
use `getData` on the `AgiChannel` to stream the files when I need DTMF,
otherwise I just use the `exec` method on the AgiChannel to use the
Playback application within Asterisk.

If you are going to originate the Call in some other way, I won't be of
much help at the moment as I have not worked with the Asterisk Manager
Interface, (
https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Manager+TCP+IP+API)
yet. Asterisk Java does also support the Manager Interface so you might
find something that more closely resembles what you are looking to do at
http://asterisk-java.org .

On Fri, Nov 23, 2018 at 3:21 PM adriano santos <sle...@bs...> wrote:

> Hi  Brandon! Thank you for your response.
>
> Is the example you sent to receive a call and respond with correct audio?
> What I need to do is originate a call to cell phone and play audio. For
> now no need for DTMF.
>
> Regards,
> Adriano Santos
>
> Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <bh...@ta...>
> escreveu:
>
>> Hey Adriano,
>>
>> I have within the last few months started using Asterisk-Java for a
>> project and you can certainly playback an audio file. There are a couple of
>> things that you will have to think about to determine the best approach for
>> your situation, here is what I can tell you though
>>
>>    - If you do *not* need to listen for DTMF input (keypresses) while
>>    playing the audio file then you can play an audio file on an AgiChannel in
>>    Java by using something like `agiChannel.exec("playback", "<path to audio
>>    file>");`
>>       - This example in the Asterisk Java GitHub repository shows
>>       answering a call, playing a file back, and then hanging up (note that in
>>       the example they rely on methods defined in the AgiOperations class so they
>>       aren't operating directly on the AgiChannel like the line of code I am
>>       sharing is doing)
>>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>>    - If you *do* need to listen for DTMF input while playing the audio
>>    file then you will want to use the `getData` or `getOption` methods defined
>>    on AgiChannel. You will need to pick which method is appropriate depending
>>    on the length of DTMF you need to listen for (`getOption` returns a single
>>    char while `getData` will return a String)
>>
>> Thanks,
>> Brandon
>>
>> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
>> wrote:
>>
>>> Hello, guys!
>>>
>>> I would like to play an audio file on a initialized call via java. I
>>> haven't found an example for this.
>>> Is this possible?
>>>
>>> --
>>> Adriano P. Santos
>>>
>>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>>> sábio é um criador de valores que não existem e que ele faz existir."
>>> Albert Einstein
>>> _______________________________________________
>>> Asterisk-java-users mailing list
>>> Ast...@li...
>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>>
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>
>
> --
> Adriano P. Santos
>
> "O homem erudito é um descobridor de fatos que já existem - mas o homem
> sábio é um criador de valores que não existem e que ele faz existir."
> Albert Einstein
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[adriano s. <sle...@bs...>]

Hi  Brandon! Thank you for your response.

Is the example you sent to receive a call and respond with correct audio?
What I need to do is originate a call to cell phone and play audio. For now
no need for DTMF.

Regards,
Adriano Santos

Em sex, 23 de nov de 2018 às 18:34, Brandon Haugen <bh...@ta...>
escreveu:

> Hey Adriano,
>
> I have within the last few months started using Asterisk-Java for a
> project and you can certainly playback an audio file. There are a couple of
> things that you will have to think about to determine the best approach for
> your situation, here is what I can tell you though
>
>    - If you do *not* need to listen for DTMF input (keypresses) while
>    playing the audio file then you can play an audio file on an AgiChannel in
>    Java by using something like `agiChannel.exec("playback", "<path to audio
>    file>");`
>       - This example in the Asterisk Java GitHub repository shows
>       answering a call, playing a file back, and then hanging up (note that in
>       the example they rely on methods defined in the AgiOperations class so they
>       aren't operating directly on the AgiChannel like the line of code I am
>       sharing is doing)
>       https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
>    - If you *do* need to listen for DTMF input while playing the audio
>    file then you will want to use the `getData` or `getOption` methods defined
>    on AgiChannel. You will need to pick which method is appropriate depending
>    on the length of DTMF you need to listen for (`getOption` returns a single
>    char while `getData` will return a String)
>
> Thanks,
> Brandon
>
> On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...>
> wrote:
>
>> Hello, guys!
>>
>> I would like to play an audio file on a initialized call via java. I
>> haven't found an example for this.
>> Is this possible?
>>
>> --
>> Adriano P. Santos
>>
>> "O homem erudito é um descobridor de fatos que já existem - mas o homem
>> sábio é um criador de valores que não existem e que ele faz existir."
>> Albert Einstein
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>


-- 
Adriano P. Santos

"O homem erudito é um descobridor de fatos que já existem - mas o homem
sábio é um criador de valores que não existem e que ele faz existir."
Albert Einstein

Re: [Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[Brandon H. <bh...@ta...>]

Hey Adriano,

I have within the last few months started using Asterisk-Java for a project
and you can certainly playback an audio file. There are a couple of things
that you will have to think about to determine the best approach for your
situation, here is what I can tell you though

   - If you do *not* need to listen for DTMF input (keypresses) while
   playing the audio file then you can play an audio file on an AgiChannel in
   Java by using something like `agiChannel.exec("playback", "<path to audio
   file>");`
      - This example in the Asterisk Java GitHub repository shows answering
      a call, playing a file back, and then hanging up (note that in
the example
      they rely on methods defined in the AgiOperations class so they aren't
      operating directly on the AgiChannel like the line of code I am
sharing is
      doing)
      https://github.com/asterisk-java/asterisk-java/blob/master/src/main/java/org/asteriskjava/examples/fastagi/ExampleCallIn.java
   - If you *do* need to listen for DTMF input while playing the audio file
   then you will want to use the `getData` or `getOption` methods defined on
   AgiChannel. You will need to pick which method is appropriate depending on
   the length of DTMF you need to listen for (`getOption` returns a single
   char while `getData` will return a String)

Thanks,
Brandon

On Thu, Nov 22, 2018 at 12:09 PM adriano santos <sle...@bs...> wrote:

> Hello, guys!
>
> I would like to play an audio file on a initialized call via java. I
> haven't found an example for this.
> Is this possible?
>
> --
> Adriano P. Santos
>
> "O homem erudito é um descobridor de fatos que já existem - mas o homem
> sábio é um criador de valores que não existem e que ele faz existir."
> Albert Einstein
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>

[Asterisk-java-users] How can i use this library to make a call and run an audio file on it

[adriano s. <sle...@bs...>]

Hello, guys!

I would like to play an audio file on a initialized call via java. I
haven't found an example for this.
Is this possible?

-- 
Adriano P. Santos

"O homem erudito é um descobridor de fatos que já existem - mas o homem
sábio é um criador de valores que não existem e que ele faz existir."
Albert Einstein

Re: [Asterisk-java-users] Asterisk AMI Java API

[MT <moh...@gm...>]

Hi Robert,

this warining is not important:
*Unsupported protocol version 'Asterisk Call Manager/2.9.0'. Use at your
own risk.*

if you need any property of any event you should implement it by yourself
otherwise the warning (*Unable to set property*) is not important.

for example if you need eventlist on Agents Complete Event you should edit
AgentCompleteEvent.java and add eventlist as a property like this:


public class AgentCompleteEvent extends AbstractAgentEvent
{
    /**
     * Serial version identifier.
     */
    private static final long serialVersionUID = 2108033737226142194L;

    private Long holdTime;
    private Long talkTime;
    private String reason;

private String eventlist;

public String getEventlist() {
return eventlist;
}

public void setEventlist(String eventlist) {
this.eventlist = eventlist;
}


you could create asterisk.java with maven after updating source files

On Fri, Sep 8, 2017 at 4:11 PM, Robert Samuel via Asterisk-java-users <
ast...@li...> wrote:

> Hello
>
>
> I am new to Asterisk.
>
> In our project we will use Asterisk 13.13.
>
>
> I want to use the Java API for the AMI (https://github.com/asterisk-
> java/asterisk-java)
> I am testing with the latest final version available in Maven Central:
> org.asteriskjava:asterisk-java:1.0.0-final
>
> When following the tutorial http://asterisk-java.org/tutorial/ I get the
> following warnings:
>
>
> * Unsupported protocol version 'Asterisk Call Manager/2.9.0'. Use at your
> own risk.
>
> * Unable to set property 'eventlist' to 'Complete' on
> org.asteriskjava.manager.event.AgentsCompleteEvent: no setter. Please
> report at http://jira.reucon.org/browse/AJ
>
> (and other similar logs)
>
>
>
> The project does not seem to be very active. The version 2.0.0 is
> mentioned in the documentation but not available in Maven Central
> repository.
>
>
> Is there a new library/project for the recent versions of Asterisk?
>
>
> Thanks
>
> Samuel Robert
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

[Asterisk-java-users] Asterisk AMI Java API

[Robert S. <Sam...@ak...>]

<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
<style type="text/css" style="display:none;"><!-- P {margin-top:0;margin-bottom:0;} --></style>
</head>
<body dir="ltr">
<div id="divtagdefaultwrapper" style="font-size:12pt;color:#000000;font-family:Calibri,Arial,Helvetica,sans-serif;">
<p>Hello</p>
<p><br>
</p>
<p>I am new to Asterisk.&nbsp;</p>
<p>In our project we will use Asterisk 13.13.</p>
<p><br>
</p>
<p>I want to use the&nbsp;Java API for the AMI (<a href="https://github.com/asterisk-java/asterisk-java" class="OWAAutoLink" id="LPlnk461249" previewremoved="true">https://github.com/asterisk-java/asterisk-java</a>)</p>
I am testing with the latest final&nbsp;version available in Maven Central:&nbsp;<span>org.asteriskjava:asterisk-java:1.0.0-final</span>
<div><br>
</div>
<div><span></span>When following the tutorial&nbsp;<a href="http://asterisk-java.org/tutorial/" class="OWAAutoLink" id="LPlnk479289" previewremoved="true" style="font-size: 12pt;">http://asterisk-java.org/tutorial/</a>&nbsp;I get the following warnings:
<p><br>
<span>*&nbsp;Unsupported protocol version 'Asterisk Call Manager/2.9.0'. Use at your own risk.</span></p>
<p><span>*&nbsp;Unable to set property 'eventlist' to 'Complete' on org.asteriskjava.manager.event.AgentsCompleteEvent: no setter. Please report at
<a href="http://jira.reucon.org/browse/AJ" class="OWAAutoLink" id="LPlnk239302" previewremoved="true">
http://jira.reucon.org/browse/AJ</a></span></p>
<p><span></span>(and other similar logs)</p>
<p><br>
</p>
<p><br>
The project does not seem to be very active. The version 2.0.0 is mentioned in the documentation but not available in Maven Central repository.</p>
<p><br>
</p>
<p>Is there a new library/project for the recent versions of Asterisk?</p>
<p><br>
</p>
<p>Thanks</p>
<p>Samuel Robert<a name="_MailAutoSig" id="LPNoLP" style="font-size: 12pt; font-family: Calibri, Arial, Helvetica, sans-serif;">&nbsp;</a></p>
<div id="Signature">
<div id="divtagdefaultwrapper" style="font-size:12pt; color:#000000; font-family:Calibri,Arial,Helvetica,sans-serif">
<div>
<p></p>
</div>
<p></p>
</div>
</div>
</div>
</div>
</body>
</html>

[Asterisk-java-users] Asterisk call controll

[Arne H. <arn...@de...>]

Hello everyone,

At work we are creating a setup/framework to test pbx's from several vendor
(one of them being Asterisk).

Let me just sketch what we have now and what issues we are having at the
moment.
We started with programming everything for Cisco (CUCM) using jtapi which
works.
We can tell the pbx to set up calls, put calls on hond, extend calls,...
everything works perfectly so far..

After that we started working on the same for Asterisk. Since we read that
Astrisk works with jtapi we were hoping for a smooth development. But
apparently the version of Asterisk we need to test doesn't work with jtapi.

After some searching we started using Asterisk-Java. But this gives some
problems also(or we are doing it wrong...).
We used the Dail-example and after some corrections we were able to do some
stuff.
We have a set-up with 2 Zoiper phones and try to make the pbx setup a call
between these two using the following line of code.

DialActivity dial = pbx.dial(from, fromCallerID, to, toCallerID);

At first we thought we finally did it, we saw on one of the phones that we
were being called. But problem was that this phone is the phone that should
make the call, not get called. The call it receives is from the 'to' number
but on the second phone nothing happens there is no call to be seen there...
We should be able to control who hangs up the phone, in some cases it is
the called party and others the calling party

If anyone has any idea where we went wrong or how we are supposed to do it,
could you please help us out?


Arne Herbots
IT Consultant
+32 4 75 38 24 53 <+32+4+75+38+24+53>
arn...@de...
[image: pattern devoteam]
[image: Devoteam sur Linkedin] <https://www.linkedin.com/company/devoteam>
[image: Devoteam sur Google Plus] <https://plus.google.com/+Devoteam-group>
[image: Devoteam sur Twitter] <https://twitter.com/devoteam>
[image: Devoteam] <http://www.devoteam.com/>[image: Innovative technology
consulting for business]

Re: [Asterisk-java-users] ManagerConnection for simultaneous multiple calls

[Amol V. <amo...@ea...>]

 Hello Yves,
Thanks for your reply...
  
 Wondering why this group is not super active. I found, asterisk-Java to be 
super useful and a powerful tool.
  
 By the way, a little bit of digging into API reveled 
OriginatingAction.setAsync method. Which did the trick for me. Now without 
changing my code too much I am able to make simultaneous calls with single 
ManagerConnection object. The same connection object acts as event listens. 
I have made it singletone wrapper around the ManagerConnection.
  
 Thanks again... and enjoy your weekend.
  
 Cheers,
Amol
  

  
  
  
  

----------------------------------------
 From: "Yves" <yv...@gm...>
Sent: Friday, March 24, 2017 11:44 PM
To: ast...@li...
Subject: Re: [Asterisk-java-users] ManagerConnection for simultaneous 
multiple calls   
 Hi Amol and welcome to the list.

not much traffic here, but it is still alive... ;-)

basically, you are doing everything right.
Your second calls waits for the first call to be set up, because it is 
intended to... sendAction the way you use it waits for the manager 
response.
If you want to return immediately, you should use sendaction in conjunction 
with a per call callback handler... take a look at the api doc, it is quite 
easy.
another approach to become even more "parallel" is using one thread per 
call.

cheers,
yves

Am 23.03.2017 um 17:01 schrieb Amol Vedak:

Hello Friends,  

   

I am new to Asterisk Java and I suppose my question may be very basic. 
Request you to kindly bare with me.  

My code connects to Asterisk using ManagerConnection instance. It logs in 
for one time and I use the same object multiple times in different threads. 
 

I hope that is OK.  

I initiate two simultaneous calls on the same ManagerConnection object 
managerConnection.sendAction  

What I notice is, calls do not happen simultaneously. Second call happens 
after first call.  

   

Is there a way to run simultaneous calls through same ManagerConnection 
object? Or do I have to create object every time and login / logoff 
everytime?  

I hit a search on Google without much luck.  

   

Any help/pointers in this direction are highly appreciated.  

   

Thanks & Regards,  

Amol  

   

  
        

 
----------------------------------------------------------------------------
-- Check out the vibrant tech community on one of the world's most engaging 
tech sites, Slashdot.org! http://sdm.link/slashdot         

 _______________________________________________ Asterisk-java-users 
mailing list Ast...@li... 
https://lists.sourceforge.net/lists/listinfo/asterisk-java-users    

  

Re: [Asterisk-java-users] ManagerConnection for simultaneous multiple calls

[Yves <yv...@gm...>]

Hi Amol and welcome to the list.

not much traffic here, but it is still alive... ;-)

basically, you are doing everything right.
Your second calls waits for the first call to be set up, because it is 
intended to... sendAction the way you use it waits for the manager response.
If you want to return immediately, you should use sendaction in 
conjunction with a per call callback handler... take a look at the api 
doc, it is quite easy.
another approach to become even more "parallel" is using one thread per 
call.

cheers,
yves

Am 23.03.2017 um 17:01 schrieb Amol Vedak:
>
> Hello Friends,
>
> I am new to Asterisk Java and I suppose my question may be very basic. 
> Request you to kindly bare with me.
>
> My code connects to Asterisk using ManagerConnection instance. It logs 
> in for one time and I use the same object multiple times in different 
> threads.
>
> I hope that is OK.
>
> I initiate two simultaneous calls on the same ManagerConnection object 
> /managerConnection/.sendAction
>
> What I notice is, calls do not happen simultaneously. Second call 
> happens after first call.
>
> Is there a way to run simultaneous calls through same 
> ManagerConnection object? Or do I have to create object every time and 
> login / logoff everytime?
>
> I hit a search on Google without much luck.
>
> Any help/pointers in this direction are highly appreciated.
>
> Thanks & Regards,
>
> Amol
>
>
>
> ------------------------------------------------------------------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

[Asterisk-java-users] ManagerConnection for simultaneous multiple calls

[Amol V. <amo...@ea...>]

Hello Friends,

 

I am new to Asterisk Java and I suppose my question may be very basic.
Request you to kindly bare with me.

My code connects to Asterisk using ManagerConnection instance. It logs in
for one time and I use the same object multiple times in different threads.

I hope that is OK.

I initiate two simultaneous calls on the same ManagerConnection object
managerConnection.sendAction

What I notice is, calls do not happen simultaneously. Second call happens
after first call.

 

Is there a way to run simultaneous calls through same ManagerConnection
object? Or do I have to create object every time and login / logoff
everytime?

I hit a search on Google without much luck.

 

Any help/pointers in this direction are highly appreciated.

 

Thanks & Regards,

Amol

 

 

Re: [Asterisk-java-users] Asterisk version compatibility

[Fabio M. <fmo...@te...>]

<html>
  <head>
    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    I've tried it with asterisk 11, sending calls and reading events
    works.<br>
    <br>
    Maybe something on the upper layer (asterisk live) doesn't work, but
    I believe internal data structures haven't changed a lot.<br>
    <br>
    You can peek in the source code to see if they recognize asterisk 13
    as supported version.<br>
    <br>
    bye.<br>
    <br>
    <div class="moz-cite-prefix">Il 05/08/2016 07:15, Wayne Merricks ha
      scritto:<br>
    </div>
    <blockquote
      cite="mid:b92...@th..."
      type="cite">
      <meta content="text/html; charset=windows-1252"
        http-equiv="Content-Type">
      Hi,<br>
      <br>
      I'm not sure if asterisk-java.org is maintained any more (the news
      certainly isn't) but there does seem to be a github project that
      is active:<br>
      <br>
      <a moz-do-not-send="true" class="moz-txt-link-freetext"
        href="https://github.com/asterisk-java/asterisk-java">https://github.com/asterisk-java/asterisk-java</a><br>
      <br>
      You might have more joy getting a response there, they have a
      "formal" 1.0 release and a beta 1.1 on the go as of February.<br>
      <br>
      Regards,<br>
      <br>
      Wayne<br>
      <br>
      <div class="moz-cite-prefix">On 23/05/16 10:09, Grant Bagdasarian
        wrote:<br>
      </div>
      <blockquote
        cite="mid:342...@CM....local"
        type="cite">
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          <p class="MsoNormal"><span lang="EN-US">Hello,<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">Does the asterisk-java
              library work with the latest LTS version of Asterisk (13
              LTS)?<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">I couldn’t find
              information about the supported asterisk versions.<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">We’re currently using
              the asterisk-java.1.0.0.m3 version on asterisk 1.6 and are
              planning to update to the latest version of asterisk, but
              I believe this does require asterisk-java to be updated as
              well.<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">If the library does
              not work with the latest version of asterisk, are there
              any other actively maintained java libraries for asterisk?<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">Regards,<o:p></o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
          <p class="MsoNormal"><span lang="EN-US">Grant <o:p></o:p></span></p>
          <p class="MsoNormal"><o:p> </o:p></p>
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Mobile security can be enabling, not merely restricting. Employees who
bring their own devices (BYOD) to work are irked by the imposition of MDM
restrictions. Mobile Device Manager Plus allows you to control only the
apps on BYO-devices by containerizing them, leaving personal data untouched!
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://ad.doubleclick.net/ddm/clk/304595813;131938128;j">https://ad.doubleclick.net/ddm/clk/304595813;131938128;j</a></pre>
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Re: [Asterisk-java-users] Asterisk version compatibility

[Wayne M. <way...@th...>]

Hi,

I'm not sure if asterisk-java.org is maintained any more (the news 
certainly isn't) but there does seem to be a github project that is active:

https://github.com/asterisk-java/asterisk-java

You might have more joy getting a response there, they have a "formal" 
1.0 release and a beta 1.1 on the go as of February.

Regards,

Wayne

On 23/05/16 10:09, Grant Bagdasarian wrote:
>
> Hello,
>
> Does the asterisk-java library work with the latest LTS version of 
> Asterisk (13 LTS)?
>
> I couldn’t find information about the supported asterisk versions.
>
> We’re currently using the asterisk-java.1.0.0.m3 version on asterisk 
> 1.6 and are planning to update to the latest version of asterisk, but 
> I believe this does require asterisk-java to be updated as well.
>
> If the library does not work with the latest version of asterisk, are 
> there any other actively maintained java libraries for asterisk?
>
> Regards,
>
> Grant
>
>
>
> ------------------------------------------------------------------------------
> Mobile security can be enabling, not merely restricting. Employees who
> bring their own devices (BYOD) to work are irked by the imposition of MDM
> restrictions. Mobile Device Manager Plus allows you to control only the
> apps on BYO-devices by containerizing them, leaving personal data untouched!
> https://ad.doubleclick.net/ddm/clk/304595813;131938128;j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users