asterisk-java-users — Asterisk-Java User List

Re: [Asterisk-java-users] ManagerAMI Tutorial giving error after logging

[Rounak S. <rou...@gm...>]

Yeah.I changed it later but was still getting an error.I changed the
manager.conf.
The [manager] in user should be assigned the following

[manager]
read=all
write=all

Furthermore how many people can asterisk call at a time?

Thanks

Re: [Asterisk-java-users] ManagerAMI Tutorial giving error after logging

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

hi,

as far as I can see (you did not show all the code / logs...) you used 
the wrong context...
while the example uses "default", you have named the correspong context 
in your dialplan
"incoming-call"...
So change the line

originateAction.setContext("default");

to

originateAction.setContext("incoming-call");

and try again..
yves

Am 08.06.2013 09:53, schrieb Rounak Saha:
> telnet 127.0.0.1 5038 is working fine.
> *Manager.conf*
> [general]
> enabled = yes
> port = 5038
> bindaddr = 127.0.0.1
> webenabled=yes
> allowmultiplelogin=yes
>
> [admin]
> secret = s12345
> read = system,call,log,verbose,command,agent,config,read,write,originate
> write = system,call,log,verbose,command,agent,config,read,write,originate
>
> [manager]
> secret = squirrel
> deny = 0.0.0.0/0.0.0.0 <http://0.0.0.0/0.0.0.0>
> permit = 127.0.0.1/255.0.0.0 <http://127.0.0.1/255.0.0.0>
> read=system,call,log,verbose,agent,command,user
> write=system,call,log,verbose,agent,command,user
>
>
> On Sat, Jun 8, 2013 at 2:49 AM, Rounak Saha <rou...@gm... 
> <mailto:rou...@gm...>> wrote:
>
>     I tried the code given in tutorial HelloManager
>     http://www.asterisk-java.org/0.3.1/tutorial.html
>
>     I changed the sip according to my asterisk program.The sip is
>     working fine if I am calling extension from a softphone.
>     I am able to succesfully log in
>
>     Here are the necessary files
>
>     *Sip.conf*
>     [1000abc]
>     type=peer
>     allow=all
>     secret=secret
>     host=dynamic
>     context=incoming-call
>     nat=yes
>
>     *extensions.conf*
>     [incoming-call]
>     exten => 100,1,Answer()
>      same =>n,Agi(agi://localhost/hello.agi)
>
>     exten=>300,1,Answer()
>      same=>n,Playback(/home/user/Desktop/b)
>      same=>n,Hangup()
>
>     *HelloManager*
>             originateAction = new OriginateAction();
>     originateAction.setChannel("SIP/1000abc");
>             originateAction.setContext("default");
>             originateAction.setExten("300");
>             originateAction.setPriority(new Integer(1));
>             originateAction.setTimeout(new Integer(30000));
>     //setTimeout is showing as deprecated
>
>     *Output*
>     Jun 08, 2013 2:35:46 AM
>      org.asteriskjava.manager.internal.ManagerConnectionImpl connect
>     INFO: *Connecting to localhost:5038*
>     Jun 08, 2013 2:35:46 AM
>     org.asteriskjava.manager.internal.ManagerConnectionImpl
>     setProtocolIdentifier
>     INFO: *Connected via Asterisk Call Manager/1.1*
>     Jun 08, 2013 2:35:46 AM
>     org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
>     INFO:*Successfully logged in*
>     Jun 08, 2013 2:35:46 AM
>     org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
>     INFO: Determined Asterisk version: Asterisk 1.8
>     *Error*
>     Jun 08, 2013 2:35:46 AM
>     org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
>     INFO: *Closing socket.*
>     Jun 08, 2013 2:35:46 AM
>     org.asteriskjava.manager.internal.ManagerReaderImpl run
>     INFO: Terminating reader thread: No more lines available: null
>
>     *LogFile*
>
>     [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Challenge'
>     [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Login'
>     [Jun  8 02:35:46] VERBOSE[9743] manager.c: == Manager 'manager'
>     logged on from 127.0.0.1
>     [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
>     [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
>     [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Logoff'
>     [Jun  8 02:35:46] VERBOSE[9743] manager.c: == Manager 'manager'
>     logged off from 127.0.0.1
>     *
>     *
>     *
>     *
>
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] ManagerAMI Tutorial giving error after logging

[Rounak S. <rou...@gm...>]

telnet 127.0.0.1 5038 is working fine.
*Manager.conf*
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
webenabled=yes
allowmultiplelogin=yes

[admin]
secret = s12345
read = system,call,log,verbose,command,agent,config,read,write,originate
write = system,call,log,verbose,command,agent,config,read,write,originate

[manager]
secret = squirrel
deny = 0.0.0.0/0.0.0.0
permit = 127.0.0.1/255.0.0.0
read=system,call,log,verbose,agent,command,user
write=system,call,log,verbose,agent,command,user


On Sat, Jun 8, 2013 at 2:49 AM, Rounak Saha <rou...@gm...> wrote:

> I tried the code given in tutorial HelloManager
> http://www.asterisk-java.org/0.3.1/tutorial.html
>
> I changed the sip according to my asterisk program.The sip is working fine
> if I am calling extension from a softphone.
> I am able to succesfully log in
>
> Here are the necessary files
>
> *Sip.conf*
> [1000abc]
> type=peer
> allow=all
> secret=secret
> host=dynamic
> context=incoming-call
> nat=yes
>
> *extensions.conf*
> [incoming-call]
> exten => 100,1,Answer()
>  same =>n,Agi(agi://localhost/hello.agi)
>
> exten=>300,1,Answer()
>  same=>n,Playback(/home/user/Desktop/b)
>  same=>n,Hangup()
>
> *HelloManager*
>         originateAction = new OriginateAction();
>         originateAction.setChannel("SIP/1000abc");
>         originateAction.setContext("default");
>         originateAction.setExten("300");
>         originateAction.setPriority(new Integer(1));
>         originateAction.setTimeout(new Integer(30000)); //setTimeout is
> showing as deprecated
>
> *Output*
> Jun 08, 2013 2:35:46 AM
>
>  org.asteriskjava.manager.internal.ManagerConnectionImpl connect
> INFO: *Connecting to localhost:5038*
> Jun 08, 2013 2:35:46 AM
> org.asteriskjava.manager.internal.ManagerConnectionImpl
> setProtocolIdentifier
> INFO: *Connected via Asterisk Call Manager/1.1*
> Jun 08, 2013 2:35:46 AM
> org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
> INFO:* Successfully logged in*
> Jun 08, 2013 2:35:46 AM
> org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
> INFO: Determined Asterisk version: Asterisk 1.8
> *Error*
> Jun 08, 2013 2:35:46 AM
> org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
> INFO: *Closing socket.*
> Jun 08, 2013 2:35:46 AM
> org.asteriskjava.manager.internal.ManagerReaderImpl run
> INFO: Terminating reader thread: No more lines available: null
>
> *LogFile*
>
> [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Challenge'
> [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Login'
> [Jun  8 02:35:46] VERBOSE[9743] manager.c:   == Manager 'manager' logged
> on from 127.0.0.1
> [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
> [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
> [Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Logoff'
> [Jun  8 02:35:46] VERBOSE[9743] manager.c:   == Manager 'manager' logged
> off from 127.0.0.1
> *
> *
> *
> *
>

[Asterisk-java-users] ManagerAMI Tutorial giving error after logging

[Rounak S. <rou...@gm...>]

I tried the code given in tutorial HelloManager
http://www.asterisk-java.org/0.3.1/tutorial.html

I changed the sip according to my asterisk program.The sip is working fine
if I am calling extension from a softphone.
I am able to succesfully log in

Here are the necessary files

*Sip.conf*
[1000abc]
type=peer
allow=all
secret=secret
host=dynamic
context=incoming-call
nat=yes

*extensions.conf*
[incoming-call]
exten => 100,1,Answer()
 same =>n,Agi(agi://localhost/hello.agi)

exten=>300,1,Answer()
 same=>n,Playback(/home/user/Desktop/b)
 same=>n,Hangup()

*HelloManager*
        originateAction = new OriginateAction();
        originateAction.setChannel("SIP/1000abc");
        originateAction.setContext("default");
        originateAction.setExten("300");
        originateAction.setPriority(new Integer(1));
        originateAction.setTimeout(new Integer(30000)); //setTimeout is
showing as deprecated

*Output*
Jun 08, 2013 2:35:46 AM

 org.asteriskjava.manager.internal.ManagerConnectionImpl connect
INFO: *Connecting to localhost:5038*
Jun 08, 2013 2:35:46 AM
org.asteriskjava.manager.internal.ManagerConnectionImpl
setProtocolIdentifier
INFO: *Connected via Asterisk Call Manager/1.1*
Jun 08, 2013 2:35:46 AM
org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO:* Successfully logged in*
Jun 08, 2013 2:35:46 AM
org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
INFO: Determined Asterisk version: Asterisk 1.8
*Error*
Jun 08, 2013 2:35:46 AM
org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
INFO: *Closing socket.*
Jun 08, 2013 2:35:46 AM org.asteriskjava.manager.internal.ManagerReaderImpl
run
INFO: Terminating reader thread: No more lines available: null

*LogFile*

[Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Challenge'
[Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Login'
[Jun  8 02:35:46] VERBOSE[9743] manager.c:   == Manager 'manager' logged on
from 127.0.0.1
[Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
[Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Command'
[Jun  8 02:35:46] DEBUG[9743] manager.c: Running action 'Logoff'
[Jun  8 02:35:46] VERBOSE[9743] manager.c:   == Manager 'manager' logged
off from 127.0.0.1
*
*
*
*

Re: [Asterisk-java-users] Outbound call from fastAGI

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

as far as i remember there was an exec command... you could e.g. use 
that to execute a "DIAL" command...

yves

Am 06.06.2013 09:07, schrieb Rounak Saha:
> I meant by using AGI only(not using manager API)  can I make an 
> outbound call
>
>
> On Thu, Jun 6, 2013 at 11:42 AM, Rounak Saha <rou...@gm... 
> <mailto:rou...@gm...>> wrote:
>
>     Hello everyone,
>     I am using fastAGI protocol to integrate asterisk with JAVA.
>     I want to make an outbound call to one of help center from java
>     program.
>     How can I do it?
>     I cannot find any command in fastAGI command for this feature.
>
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

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

hi,

I would try different engines and use the possibility of evaluating 
them... most companies offer an evaluation period.
i used cepstral without issues, but the quality is sometimes robotic... 
i think at&t or nuance have better voices meanwhile..

yves

Am 06.06.2013 07:58, schrieb Rounak Saha:
> Hii yves,
>   Thanks for clearing my doubt. Idea of caching and building a md5 for 
> the text is brilliant.
> what kind of TTS engine are you using? an online service or a 
> serverside engine?
>  Actually my organisation has not yet decided which TTS engine they 
> are going to use but it will be mostly serverside engine.
>  Any suggestion that should be taken in mind while using serverside 
> engine?
>
>
> Cheers
> Rounak
>
>
>
>
>
>
>
>
> On Tue, Jun 4, 2013 at 11:59 AM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     see inline...
>
>     yves
>
>
>     Am 03.06.2013 17:25, schrieb Rounak Saha:
>>     Hii,
>>     Is it necessary for the soundFile to be in
>>     /var/lib/asterisk/sounds/?What do you exactly mean by "soundfile
>>     has to be in the right folder..!"?If yes,how can I do it?
>     I would recommend using subfolders beneath
>     /var/lib/asterisk/sounds, e.g. /var/lib/asterisk/sounds/TTS etc.
>     you can configure another sounddirectory in the asterisk conf
>     files, but it is valid for ALL soundfiles (prompts etc.).
>
>>
>>     that means I have to store the voice file return by TTS software
>>     to the /var/lib/asterisk/sounds/ and then use play it, all when
>>     the caller is online!!!I think its a time taking process?
>     yes, thats right... but time taking process...? no, not really...
>     what kind of TTS engine are you using? an online service or a
>     serverside engine? I wrote an IVR using an online TTS and all the
>     process of generating and retrieving the
>     file, converting it and storing it in the right place was (of
>     course depending on length and bandwith) a matter of less than one
>     second. caching generated files (if the text is
>     the same, it has not to be generated over and over again) and
>     producing them in advance or while playing a prompt does "speed
>     up" the process even more...
>     I have developed a simple cashing mechanism... build a md5 of the
>     text that has to be generated and see, if a file with that name
>     exists... if yes, its there and can be played,
>     if not, generate it, store it under the md5 filename and than it
>     is there for now and for future use...
>
>>
>>     What is the best way to integrate TTS with my java program?
>     I have done with cepstral for linux as well as with google and
>     with AT&T (both online...) AT&T has the best voice quality and
>     natural sounding voices. Depending on
>     Bandwith an offline solution is probably preferrable.
>
>>
>>     And what if my Java server and asterisk server are not in same
>>     physical system.
>>     Do I have to copy the soundFile from java server to asterisk
>>     server may be using nfs or gufer?
>     you could mount the TTS-folder on the other machine via nfs and
>     set the mount point beneath the /var/lib/asterisk/sounds folder...
>     That should work.
>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>     On Mon, Jun 3, 2013 at 8:36 PM, Yves A. <yv...@gm...
>>     <mailto:yv...@gm...>> wrote:
>>
>>         asterisk can´t handle mp3 by default.... you have to compile
>>         support for mp3 (licence issues...)...
>>         or you convert mp3 to wav (att.: wav != wav...) or gsm after
>>         tts generation.... sox can help you there....
>>
>>         take a look in the log / verbose cli output to see, if any
>>         error occur...
>>
>>         soundfile has to be in the right folder..! normally:
>>         /var/lib/asterisk/sounds/ and is addressed relatively
>>         to that path...
>>
>>         use streamFile(file) in your AGI to stream the file to the
>>         caller...
>>
>>         yves
>>
>>
>>
>>         Am 03.06.2013 16:47, schrieb Rounak Saha:
>>>         Hello everyone,
>>>
>>>         I have integrated my JAVA program(using asterisk-java) with
>>>         a text-to-speech software where I send a string  to the TTS
>>>         software and it return me a .mp3 file (which can be
>>>         converted to any desire format).
>>>
>>>         Now I want this mp3 to be played to the caller.
>>>
>>>         Which fastAGI command should I use to play the mp3?
>>>
>>>         Do I have to first transfer my .mp3 file to Asterisk
>>>         database and then ask it to run it?If yes,How should I transfer?
>>>
>>>
>>>
>>>
>>>
>>>         ------------------------------------------------------------------------------
>>>         Get 100% visibility into Java/.NET code with AppDynamics Lite
>>>         It's a free troubleshooting tool designed for production
>>>         Get down to code-level detail for bottlenecks, with <2% overhead.
>>>         Download for free and get started troubleshooting in minutes.
>>>         http://p.sf.net/sfu/appdyn_d2d_ap2
>>>
>>>
>>>         _______________________________________________
>>>         Asterisk-java-users mailing list
>>>         Ast...@li...  <mailto:Ast...@li...>
>>>         https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>         ------------------------------------------------------------------------------
>>         Get 100% visibility into Java/.NET code with AppDynamics Lite
>>         It's a free troubleshooting tool designed for production
>>         Get down to code-level detail for bottlenecks, with <2% overhead.
>>         Download for free and get started troubleshooting in minutes.
>>         http://p.sf.net/sfu/appdyn_d2d_ap2
>>         _______________________________________________
>>         Asterisk-java-users mailing list
>>         Ast...@li...
>>         <mailto:Ast...@li...>
>>         https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>>     ------------------------------------------------------------------------------
>>     Get 100% visibility into Java/.NET code with AppDynamics Lite
>>     It's a free troubleshooting tool designed for production
>>     Get down to code-level detail for bottlenecks, with <2% overhead.
>>     Download for free and get started troubleshooting in minutes.
>>     http://p.sf.net/sfu/appdyn_d2d_ap2
>>
>>
>>     _______________________________________________
>>     Asterisk-java-users mailing list
>>     Ast...@li...  <mailto:Ast...@li...>
>>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>     ------------------------------------------------------------------------------
>     How ServiceNow helps IT people transform IT departments:
>     1. A cloud service to automate IT design, transition and operations
>     2. Dashboards that offer high-level views of enterprise services
>     3. A single system of record for all IT processes
>     http://p.sf.net/sfu/servicenow-d2d-j
>     _______________________________________________
>     Asterisk-java-users mailing list
>     Ast...@li...
>     <mailto:Ast...@li...>
>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] getOptionCommand Issue

[Miguel S. <m.s...@gm...>]

Is better to use a char. Sometimes you'll need to include a special
character in your options string, like '#', '*' etc.
 El 06/06/2013 11:48, "Atul Agrawal" <adi...@gm...> escribió:

> Hello everyone,
> This is my asterisk extension conf file as soon as caller calls handle
> call to Java.
> I want to know if the following will work.
>
> Asterisk extension.conf
>      exten=>100,Answer()
>      same=>n,AGI(agi://127.0.0.1/hello.agi)
>
> Java file
>
>
>  public void service(AgiRequest request, AgiChannel channel)           throws AgiException
>
>     {
>
>  streamFile("welcome");
> */**Now I will play a main menu and it will ask user to choose.I then
> *want to save it and perform action based on that.I*s the use correct?
>  *   variable option should be string or int?
> /
>                        int option=getOption("mainmenu","12349");
>                        switch(option){
>
>                                  //will perform function based on what
> user pressed
>                        }
>
>                        hangup();
>           }
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] getOptionCommand Issue

[Miguel S. <m.s...@gm...>]

Hi,
getOption returns a primitive java char.
You can provide a third parameter : Timeout (milliseconds)

Then you can implement your switch statement using the char returned value.

char option=getOption("mainmenu","12349",10000);

        switch(option){
            case '1': ... break;
            case '2' : ... break;
            .
            .
            .
            else ...;
        }
El 06/06/2013 11:48, "Atul Agrawal" <adi...@gm...> escribió:

> Hello everyone,
> This is my asterisk extension conf file as soon as caller calls handle
> call to Java.
> I want to know if the following will work.
>
> Asterisk extension.conf
>      exten=>100,Answer()
>      same=>n,AGI(agi://127.0.0.1/hello.agi)
>
> Java file
>
>
>  public void service(AgiRequest request, AgiChannel channel)           throws AgiException
>
>     {
>
>  streamFile("welcome");
> */**Now I will play a main menu and it will ask user to choose.I then
> *want to save it and perform action based on that.I*s the use correct?
>  *   variable option should be string or int?
> /
>                        int option=getOption("mainmenu","12349");
>                        switch(option){
>
>                                  //will perform function based on what
> user pressed
>                        }
>
>                        hangup();
>           }
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] getOptionCommand Issue

[Rounak S. <rou...@gm...>]

Please Ignore.
The above works fine.. :-)
Sorry for the inconveninece


On Thu, Jun 6, 2013 at 3:16 PM, Atul Agrawal <adi...@gm...> wrote:

> Hello everyone,
> This is my asterisk extension conf file as soon as caller calls handle
> call to Java.
> I want to know if the following will work.
>
> Asterisk extension.conf
>      exten=>100,Answer()
>      same=>n,AGI(agi://127.0.0.1/hello.agi)
>
> Java file
>
>
>  public void service(AgiRequest request, AgiChannel channel)           throws AgiException
>
>     {
>
>  streamFile("welcome");
> */**Now I will play a main menu and it will ask user to choose.I then
> *want to save it and perform action based on that.I*s the use correct?
>  *   variable option should be string or int?
> /
>                        int option=getOption("mainmenu","12349");
>                        switch(option){
>
>                                  //will perform function based on what
> user pressed
>                        }
>
>                        hangup();
>           }
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

[Asterisk-java-users] getOptionCommand Issue

[Atul A. <adi...@gm...>]

Hello everyone,
This is my asterisk extension conf file as soon as caller calls handle call
to Java.
I want to know if the following will work.

Asterisk extension.conf
     exten=>100,Answer()
     same=>n,AGI(agi://127.0.0.1/hello.agi)

Java file

 public void service(AgiRequest request, AgiChannel channel)
throws AgiException
    {

 streamFile("welcome");
*/**Now I will play a main menu and it will ask user to choose.I then *want
to save it and perform action based on that.I*s the use correct?    *
variable option should be string or int?
/
                       int option=getOption("mainmenu","12349");
                       switch(option){

                                 //will perform function based on what user
pressed
                       }

                       hangup();
          }

Re: [Asterisk-java-users] Outbound call from fastAGI

[Chris M. <ch...@mr...>]

Write a .call file to /var/spool/asterisk/outgoing from your agi script.
This is an alternative to manager api

Google for asterisk call file
On Jun 6, 2013 5:10 PM, "Rounak Saha" <rou...@gm...> wrote:

> I meant by using AGI only(not using manager API)  can I make an outbound
> call
>
>
> On Thu, Jun 6, 2013 at 11:42 AM, Rounak Saha <rou...@gm...>wrote:
>
>> Hello everyone,
>> I am using fastAGI protocol to integrate asterisk with JAVA.
>> I want to make an outbound call to one of help center from java program.
>> How can I do it?
>> I cannot find any command in fastAGI command for this feature.
>>
>>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] Outbound call from fastAGI

[Rounak S. <rou...@gm...>]

I meant by using AGI only(not using manager API)  can I make an outbound
call


On Thu, Jun 6, 2013 at 11:42 AM, Rounak Saha <rou...@gm...> wrote:

> Hello everyone,
> I am using fastAGI protocol to integrate asterisk with JAVA.
> I want to make an outbound call to one of help center from java program.
> How can I do it?
> I cannot find any command in fastAGI command for this feature.
>
>

[Asterisk-java-users] Outbound call from fastAGI

[Rounak S. <rou...@gm...>]

Hello everyone,
I am using fastAGI protocol to integrate asterisk with JAVA.
I want to make an outbound call to one of help center from java program.
How can I do it?
I cannot find any command in fastAGI command for this feature.

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

[Rounak S. <rou...@gm...>]

Hii yves,
  Thanks for clearing my doubt. Idea of caching and building a md5 for the
text is brilliant.
       what kind of TTS engine are you using? an online service or a
serverside engine?

 Actually my organisation has not yet decided which TTS engine they are
going to use but it will be mostly serverside engine.
 Any suggestion that should be taken in mind while using serverside engine?


Cheers
Rounak








On Tue, Jun 4, 2013 at 11:59 AM, Yves A. <yv...@gm...> wrote:

>  see inline...
>
> yves
>
>
> Am 03.06.2013 17:25, schrieb Rounak Saha:
>
> Hii,
> Is it necessary for the soundFile to be in /var/lib/asterisk/sounds/?What
> do you exactly mean by "soundfile has to be in the right folder..!"?If
> yes,how can I do it?
>
> I would recommend using subfolders beneath /var/lib/asterisk/sounds, e.g.
> /var/lib/asterisk/sounds/TTS etc.
> you can configure another sounddirectory in the asterisk conf files, but
> it is valid for ALL soundfiles (prompts etc.).
>
>
>  that means I have to store the voice file return by TTS software to the /var/lib/asterisk/sounds/
> and then use play it, all when the caller is online!!!I think its a time
> taking process?
>
> yes, thats right... but time taking process...? no, not really...
> what kind of TTS engine are you using? an online service or a serverside
> engine? I wrote an IVR using an online TTS and all the process of
> generating and retrieving the
> file, converting it and storing it in the right place was (of course
> depending on length and bandwith) a matter of less than one second. caching
> generated files (if the text is
> the same, it has not to be generated over and over again) and producing
> them in advance or while playing a prompt does "speed up" the process even
> more...
> I have developed a simple cashing mechanism... build a md5 of the text
> that has to be generated and see, if a file with that name exists... if
> yes, its there and can be played,
> if not, generate it, store it under the md5 filename and than it is there
> for now and for future use...
>
>
> What is the best way to integrate TTS with my java program?
>
> I have done with cepstral for linux as well as with google and with AT&T
> (both online...) AT&T has the best voice quality and natural sounding
> voices. Depending on
> Bandwith an offline solution is probably preferrable.
>
>
>  And what if my Java server and asterisk server are not in same physical
> system.
> Do I have to copy the soundFile from java server to asterisk server may be
> using nfs or gufer?
>
> you could mount the TTS-folder on the other machine via nfs and set the
> mount point beneath the /var/lib/asterisk/sounds folder... That should work.
>
>
>
>
>
>
>
>
>
>
>
> On Mon, Jun 3, 2013 at 8:36 PM, Yves A. <yv...@gm...> wrote:
>
>>  asterisk can´t handle mp3 by default.... you have to compile support
>> for mp3 (licence issues...)...
>> or you convert mp3 to wav (att.: wav != wav...) or gsm after tts
>> generation.... sox can help you there....
>>
>> take a look in the log / verbose cli output to see, if any error occur...
>>
>> soundfile has to be in the right folder..! normally:
>> /var/lib/asterisk/sounds/ and is addressed relatively
>> to that path...
>>
>> use streamFile(file) in your AGI to stream the file to the caller...
>>
>> yves
>>
>>
>>
>> Am 03.06.2013 16:47, schrieb Rounak Saha:
>>
>>  Hello everyone,
>>
>>  I have integrated my JAVA program(using asterisk-java) with a
>> text-to-speech software where I send a string  to the TTS software and it
>> return me a .mp3 file (which can be converted to any desire format).
>>
>>  Now I want this mp3 to be played to the caller.
>>
>>  Which fastAGI command should I use to play the mp3?
>>
>>  Do I have to first transfer my .mp3 file to Asterisk database and then
>> ask it to run it?If yes,How should I transfer?
>>
>>
>>
>>
>>
>>  ------------------------------------------------------------------------------
>> Get 100% visibility into Java/.NET code with AppDynamics Lite
>> It's a free troubleshooting tool designed for production
>> Get down to code-level detail for bottlenecks, with <2% overhead.
>> Download for free and get started troubleshooting in minutes.http://p.sf.net/sfu/appdyn_d2d_ap2
>>
>>
>>
>> _______________________________________________
>> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Get 100% visibility into Java/.NET code with AppDynamics Lite
>> It's a free troubleshooting tool designed for production
>> Get down to code-level detail for bottlenecks, with <2% overhead.
>> Download for free and get started troubleshooting in minutes.
>> http://p.sf.net/sfu/appdyn_d2d_ap2
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

[Greg H. <gre...@gm...>]

Hi Yves,

I am sorry to be confusing.

My application has a GUI interface that includes a dialer module.  The user
first would dial the conference room number (1024).  The lower layer A-J
interface would use Originate() to create a call to that conference room.
 The Meetme app would run in the dial plan and play out the request for a
PIN.  The user would then enter that PIN ("123") and the A-J layer would
take that input and use PlayDtmfAction to drive it to Meetme.  So it's up
to the user to enter the correct PIN to be sent in PlayDtmfAction, to be
allowed to enter room 1024.

Like I said, if the user dialed another SIP extension like 1000, then once
the connection is made further dialer input will get successfully passed
via PlayDtmfAction to that other SIP endpoint.

As far as bridging the extension, I think that might not be done in the
dialplan until the PIN is accepted.  If I request channel info for the
active call, the destination channel ID first is 1024@from-internal:1.  In
cases where the PIN gets accepted, the destination will be different when
the channel info gets requested again (something like STARTMEETME/1024).
 Since I am hearing the announcement to request PIN, I think I should have
a bridge in place but then why doesn't app_read.c collect the digits I
enter?

As far as OriginateToApplication, I understand it will not alter the dial
plan.  I just thought that since I specify meetme as an app there, then
further calls to the channelID associated with this session may be able to
direct things like DTMF to meetme.  Just a guess though!

Thanks again,

Greg


On Wed, Jun 5, 2013 at 3:25 PM, Yves A. <yv...@gm...> wrote:

>  Hi,
>
> now I am totally confused... I thought, your application has to play the
> PIN DTMF... now you say, that the user still has to enter the pin...
> So what is the problem to bridge the extension of your user with the
> conference-room extension? After connecting the two partys you´re
> done, the user hears the announcement and enters the pin...? If you can´t
> go this way I think I still did not catch your real objective...
>
> OriginateToApplication will not alter any part of the dialplan... its just
> a method to dial directly "into" asterisk apps... they do even not have
> to occur anywhere in the dialplan...
>
> As I mentioned earlier, you can use the originate(originateAction) method
> to jump to a certain priority in the dialplan... therefor you use
> the method originateAction.setPriority(int priority)...
>
> If the things still do not work the way you need, I think we should chat
> or phone... in that case eMail me directly and I´ll send you my
> skype ID.
>
> have fun,
> yves
>
>
>

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

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

Hi,

now I am totally confused... I thought, your application has to play the 
PIN DTMF... now you say, that the user still has to enter the pin...
So what is the problem to bridge the extension of your user with the 
conference-room extension? After connecting the two partys you´re
done, the user hears the announcement and enters the pin...? If you 
can´t go this way I think I still did not catch your real objective...

OriginateToApplication will not alter any part of the dialplan... its 
just a method to dial directly "into" asterisk apps... they do even not have
to occur anywhere in the dialplan...

As I mentioned earlier, you can use the originate(originateAction) 
method to jump to a certain priority in the dialplan... therefor you use
the method originateAction.setPriority(int priority)...

If the things still do not work the way you need, I think we should chat 
or phone... in that case eMail me directly and I´ll send you my
skype ID.

have fun,
yves



Am 05.06.2013 18:14, schrieb Greg Horton:
> Hi Yves,
>
> It is a definite requirement that I not be able to alter a customer 
> Asterisk configuration.  Possibly something like tagging on a .conf 
> file in the future but that's about it.
>
> I do know I am dialing a meeting room but it is still up to the user 
> to enter a proper PIN number, so I guess I could do something like 
> collect all entered info from a user then submit all at once to meetme 
> using OriginateToApplication().  Hopefully that will alter the 
> Asterisk dial plan so it will know it already has the PIN and it will 
> not have to play the recording to ask for it?
>
> It's great to learn about OriginateToApplication().  I use Originate() 
> today for all dialed calls, and maybe that is an issue.  I wonder if I 
> try using OriginateToApplication(), specifying meetme and without a 
> PIN, maybe internally a route will be setup in Asterisk to know to 
> send my PIN digits to the meetme app.  I can play with that a bit.
>
> Thanks!
>
> Greg
>
>
> On Wed, Jun 5, 2013 at 8:21 AM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     OK,
>
>     if you can play DTMF on "normal" calls, but not on meetme
>     conferences... have you tried to change the config parameter I
>     recommended in my last eMail or
>     can´t you change this, because you want to avoid ANY changes in
>     the config files on the target asterisk?
>
>     If you know that you are dialling into a meetme room (and I think
>     you know, otherwise I would be totally confused..), you can dial
>     directly into the
>     asterisk application meetme and pass the required parameters like
>     roomnumber and pin... no need to play around with DTMF...
>
>     The following lines given as an example would establish a
>     connection from the (SIP-)extension 567 to the meetme room 1024
>     that is protected with
>     the PIN 123456.
>
>     DefaultAsteriskServer asteriskServer = new
>     DefaultAsteriskServer("###your asterisk server ip or dns-name###",
>     "###username with appropriate rights###", "###password###);
>     asteriskServer.getManagerConnection().login();
>     asteriskServer.originateToApplication("SIP/567", "meetme",
>     "1024,,123456", 15000l);
>
>     regards,
>     yves
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

[Greg H. <gre...@gm...>]

Hi Yves,

It is a definite requirement that I not be able to alter a customer
Asterisk configuration.  Possibly something like tagging on a .conf file in
the future but that's about it.

I do know I am dialing a meeting room but it is still up to the user to
enter a proper PIN number, so I guess I could do something like collect all
entered info from a user then submit all at once to meetme using
OriginateToApplication().  Hopefully that will alter the Asterisk dial plan
so it will know it already has the PIN and it will not have to play the
recording to ask for it?

It's great to learn about OriginateToApplication().  I use Originate()
today for all dialed calls, and maybe that is an issue.  I wonder if I try
using OriginateToApplication(), specifying meetme and without a PIN, maybe
internally a route will be setup in Asterisk to know to send my PIN digits
to the meetme app.  I can play with that a bit.

Thanks!

Greg


On Wed, Jun 5, 2013 at 8:21 AM, Yves A. <yv...@gm...> wrote:

>  OK,
>
> if you can play DTMF on "normal" calls, but not on meetme conferences...
> have you tried to change the config parameter I recommended in my last
> eMail or
> can´t you change this, because you want to avoid ANY changes in the config
> files on the target asterisk?
>
> If you know that you are dialling into a meetme room (and I think you
> know, otherwise I would be totally confused..), you can dial directly into
> the
> asterisk application meetme and pass the required parameters like
> roomnumber and pin... no need to play around with DTMF...
>
> The following lines given as an example would establish a connection from
> the (SIP-)extension 567 to the meetme room 1024 that is protected with
> the PIN 123456.
>
> DefaultAsteriskServer asteriskServer = new DefaultAsteriskServer("###your
> asterisk server ip or dns-name###", "###username with appropriate
> rights###", "###password###);
> asteriskServer.getManagerConnection().login();
> asteriskServer.originateToApplication("SIP/567", "meetme", "1024,,123456",
> 15000l);
>
> regards,
> yves
>
>

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

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

OK,

if you can play DTMF on "normal" calls, but not on meetme conferences... 
have you tried to change the config parameter I recommended in my last 
eMail or
can´t you change this, because you want to avoid ANY changes in the 
config files on the target asterisk?

If you know that you are dialling into a meetme room (and I think you 
know, otherwise I would be totally confused..), you can dial directly 
into the
asterisk application meetme and pass the required parameters like 
roomnumber and pin... no need to play around with DTMF...

The following lines given as an example would establish a connection 
from the (SIP-)extension 567 to the meetme room 1024 that is protected with
the PIN 123456.

DefaultAsteriskServer asteriskServer = new 
DefaultAsteriskServer("###your asterisk server ip or dns-name###", 
"###username with appropriate rights###", "###password###);
asteriskServer.getManagerConnection().login();
asteriskServer.originateToApplication("SIP/567", "meetme", 
"1024,,123456", 15000l);

regards,
yves


Am 04.06.2013 14:39, schrieb Greg Horton:
> Hi Yves,
>
> The example I give is from a test system for my app, thus only 2 
> participant max.  I understand that is not a useful scenario.  In 
> short, our customer will have an existing Asterisk PBX which may have 
> been setup by FreePBX, TrixBox, manually, or anything else.  We are an 
> app that accesses that Asterisk system and cannot alter its 
> configuration.  In this case the customer will have configured 
> Asterisk to use Meetme and to require a PIN to enter the room.  Our 
> app can drive calls using Asterisk and A-J, but has no RTP/SIP (yet). 
>  The dialer in the GUI is a way to initiate calls from a known 
> Asterisk extension.  If I initiate a call to a known endpoint that has 
> an Asterisk channelID (other than Meetme) then I can use the dialer 
> send digits to Asterisk via PlayDtmfAction, and it works.  This could 
> be useful for something like prepaid cards, automated help desks, etc. 
>  When I call into the MeetMe conference room I just cannot get the 
> DTMF passed towards that channelID.  The Dialplan is waiting for a PIN 
> and I can't hook up with where it wants to read the PIN from (in 
> app_read.c).
>
> Maybe Asterisk's app_read can only get the PIN if it comes in an RTP 
> media stream (in-band or 2833) or via SIP Info method (or whatever IAX 
> does).  If that is the case there is nothing I can do at this time, 
> but I am just not sure it works that way.
>
> Thanks,
>
> Greg
>
>
> On Tue, Jun 4, 2013 at 2:44 AM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     hi,
>
>     as far as I remember, there is an option where you can set if dtmf
>     tones should pass the conference call... maybe this does the trick
>     for you.
>     if not..:
>     sorry, I still did not get it... dialer, gui, meetme..?? how does
>     this stick together? what is the sense of the conferenceroom with
>     a limit of two users?
>     if it is a dialerapplication, why not use direct bridging? (or, if
>     you have more than one agent and use some kind of predictive
>     dialing, you would normally
>     use queues...)
>     so... to be fully with you, I need to understand the whole
>     dialprocess as detailed as possible...
>     yves
>
>
>
>
> ------------------------------------------------------------------------------
> How ServiceNow helps IT people transform IT departments:
> 1. A cloud service to automate IT design, transition and operations
> 2. Dashboards that offer high-level views of enterprise services
> 3. A single system of record for all IT processes
> http://p.sf.net/sfu/servicenow-d2d-j
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

[Greg H. <gre...@gm...>]

Hi Yves,

The example I give is from a test system for my app, thus only 2
participant max.  I understand that is not a useful scenario.  In short,
our customer will have an existing Asterisk PBX which may have been setup
by FreePBX, TrixBox, manually, or anything else.  We are an app that
accesses that Asterisk system and cannot alter its configuration.  In this
case the customer will have configured Asterisk to use Meetme and to
require a PIN to enter the room.  Our app can drive calls using Asterisk
and A-J, but has no RTP/SIP (yet).  The dialer in the GUI is a way to
initiate calls from a known Asterisk extension.  If I initiate a call to a
known endpoint that has an Asterisk channelID (other than Meetme) then I
can use the dialer send digits to Asterisk via PlayDtmfAction, and it
works.  This could be useful for something like prepaid cards, automated
help desks, etc.  When I call into the MeetMe conference room I just cannot
get the DTMF passed towards that channelID.  The Dialplan is waiting for a
PIN and I can't hook up with where it wants to read the PIN from (in
app_read.c).

Maybe Asterisk's app_read can only get the PIN if it comes in an RTP media
stream (in-band or 2833) or via SIP Info method (or whatever IAX does).  If
that is the case there is nothing I can do at this time, but I am just not
sure it works that way.

Thanks,

Greg


On Tue, Jun 4, 2013 at 2:44 AM, Yves A. <yv...@gm...> wrote:

>  hi,
>
> as far as I remember, there is an option where you can set if dtmf tones
> should pass the conference call... maybe this does the trick for you.
> if not..:
> sorry, I still did not get it... dialer, gui, meetme..?? how does this
> stick together? what is the sense of the conferenceroom with a limit of two
> users?
> if it is a dialerapplication, why not use direct bridging? (or, if you
> have more than one agent and use some kind of predictive dialing, you would
> normally
> use queues...)
> so... to be fully with you, I need to understand the whole dialprocess as
> detailed as possible...
> yves
>
>

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

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

hi,

as far as I remember, there is an option where you can set if dtmf tones 
should pass the conference call... maybe this does the trick for you.
if not..:
sorry, I still did not get it... dialer, gui, meetme..?? how does this 
stick together? what is the sense of the conferenceroom with a limit of 
two users?
if it is a dialerapplication, why not use direct bridging? (or, if you 
have more than one agent and use some kind of predictive dialing, you 
would normally
use queues...)
so... to be fully with you, I need to understand the whole dialprocess 
as detailed as possible...
yves


Am 03.06.2013 17:34, schrieb Greg Horton:
> Hi Yves,
>
> I have an Asterisk Manager app where I have a dialer in the GUI.  I 
> dial out to a number from there, on behalf of a true Asterisk 
> extension as defined in sip.conf.  So I can dial out to MeetMe 
> conference room 1024 on behalf of my Asterisk extension 1410.  For the 
> 1410 end I will get a channelID like SIP/1410-00034bc1.  When I dial 
> 1024, the dialplan I showed moves through entry 10 
> (Read(PIN,enter-conf-pin-number,,,,)) and waits on a timer.  I see 
> that in Asterisk CLI.  At that time I can query the 1410 channel and I 
> see that the destination is 1024@from-internal:1.  That apparently is 
> not a valid destination address to use for PlayDtmfAction as when I 
> try that I see in Asterisk CLI that the user sent no data (or 
> something like that).
>
> Now I know that :1 is not relevant for my needs.  Thanks for that.
>
> Note that if I called another extension from 1410 like 1000, I could 
> use PlayDtmfAction to generate a DTMF digit to the destination 
> channelID like SIP/1000-00003453bc.
>
> I hope that explains my goal??
>
> Can you give me an idea of what to do using AGI?
>
> Thanks,
>
> Greg
>
>
> On Mon, Jun 3, 2013 at 11:00 AM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     hi,
>
>     just very short... i am in a hurry...
>
>     :1 means first leg of the call....
>
>     what you mean is jumping to a priority?? (11) you should be able
>     to do so... there is a parameter where you the specify the priority...
>     if thats not working... try to go with agi... let me know.... and
>     explain your objective... dialplan is not complete / fully clear 2
>     me...
>
>     yves
>
>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

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

see inline...

yves


Am 03.06.2013 17:25, schrieb Rounak Saha:
> Hii,
> Is it necessary for the soundFile to be in 
> /var/lib/asterisk/sounds/?What do you exactly mean by "soundfile has 
> to be in the right folder..!"?If yes,how can I do it?
I would recommend using subfolders beneath /var/lib/asterisk/sounds, 
e.g. /var/lib/asterisk/sounds/TTS etc.
you can configure another sounddirectory in the asterisk conf files, but 
it is valid for ALL soundfiles (prompts etc.).
>
> that means I have to store the voice file return by TTS software to 
> the /var/lib/asterisk/sounds/ and then use play it, all when the 
> caller is online!!!I think its a time taking process?
yes, thats right... but time taking process...? no, not really...
what kind of TTS engine are you using? an online service or a serverside 
engine? I wrote an IVR using an online TTS and all the process of 
generating and retrieving the
file, converting it and storing it in the right place was (of course 
depending on length and bandwith) a matter of less than one second. 
caching generated files (if the text is
the same, it has not to be generated over and over again) and producing 
them in advance or while playing a prompt does "speed up" the process 
even more...
I have developed a simple cashing mechanism... build a md5 of the text 
that has to be generated and see, if a file with that name exists... if 
yes, its there and can be played,
if not, generate it, store it under the md5 filename and than it is 
there for now and for future use...
>
> What is the best way to integrate TTS with my java program?
I have done with cepstral for linux as well as with google and with AT&T 
(both online...) AT&T has the best voice quality and natural sounding 
voices. Depending on
Bandwith an offline solution is probably preferrable.
>
> And what if my Java server and asterisk server are not in same 
> physical system.
> Do I have to copy the soundFile from java server to asterisk server 
> may be using nfs or gufer?
you could mount the TTS-folder on the other machine via nfs and set the 
mount point beneath the /var/lib/asterisk/sounds folder... That should work.
>
>
>
>
>
>
>
>
>
>
> On Mon, Jun 3, 2013 at 8:36 PM, Yves A. <yv...@gm... 
> <mailto:yv...@gm...>> wrote:
>
>     asterisk can´t handle mp3 by default.... you have to compile
>     support for mp3 (licence issues...)...
>     or you convert mp3 to wav (att.: wav != wav...) or gsm after tts
>     generation.... sox can help you there....
>
>     take a look in the log / verbose cli output to see, if any error
>     occur...
>
>     soundfile has to be in the right folder..! normally:
>     /var/lib/asterisk/sounds/ and is addressed relatively
>     to that path...
>
>     use streamFile(file) in your AGI to stream the file to the caller...
>
>     yves
>
>
>
>     Am 03.06.2013 16:47, schrieb Rounak Saha:
>>     Hello everyone,
>>
>>     I have integrated my JAVA program(using asterisk-java) with a
>>     text-to-speech software where I send a string  to the TTS
>>     software and it return me a .mp3 file (which can be converted to
>>     any desire format).
>>
>>     Now I want this mp3 to be played to the caller.
>>
>>     Which fastAGI command should I use to play the mp3?
>>
>>     Do I have to first transfer my .mp3 file to Asterisk database and
>>     then ask it to run it?If yes,How should I transfer?
>>
>>
>>
>>
>>
>>     ------------------------------------------------------------------------------
>>     Get 100% visibility into Java/.NET code with AppDynamics Lite
>>     It's a free troubleshooting tool designed for production
>>     Get down to code-level detail for bottlenecks, with <2% overhead.
>>     Download for free and get started troubleshooting in minutes.
>>     http://p.sf.net/sfu/appdyn_d2d_ap2
>>
>>
>>     _______________________________________________
>>     Asterisk-java-users mailing list
>>     Ast...@li...  <mailto:Ast...@li...>
>>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>     ------------------------------------------------------------------------------
>     Get 100% visibility into Java/.NET code with AppDynamics Lite
>     It's a free troubleshooting tool designed for production
>     Get down to code-level detail for bottlenecks, with <2% overhead.
>     Download for free and get started troubleshooting in minutes.
>     http://p.sf.net/sfu/appdyn_d2d_ap2
>     _______________________________________________
>     Asterisk-java-users mailing list
>     Ast...@li...
>     <mailto:Ast...@li...>
>     https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

[Usman R. <usm...@gm...>]

you need the file to be on the asterisk server in order for the agi to play
the file.Nfs could be one possible solution, if you are generating the tts
file on another server.

--Usman.

Re: [Asterisk-java-users] How to use A-J to inject input to Dialplan?

[Greg H. <gre...@gm...>]

Hi Yves,

I have an Asterisk Manager app where I have a dialer in the GUI.  I dial
out to a number from there, on behalf of a true Asterisk extension as
defined in sip.conf.  So I can dial out to MeetMe conference room 1024 on
behalf of my Asterisk extension 1410.  For the 1410 end I will get a
channelID like SIP/1410-00034bc1.  When I dial 1024, the dialplan I showed
moves through entry 10 (Read(PIN,enter-conf-pin-number,,,,)) and waits on a
timer.  I see that in Asterisk CLI.  At that time I can query the 1410
channel and I see that the destination is 1024@from-internal:1.  That
apparently is not a valid destination address to use for PlayDtmfAction as
when I try that I see in Asterisk CLI that the user sent no data (or
something like that).

Now I know that :1 is not relevant for my needs.  Thanks for that.

Note that if I called another extension from 1410 like 1000, I could use
PlayDtmfAction to generate a DTMF digit to the destination channelID like
SIP/1000-00003453bc.

I hope that explains my goal??

Can you give me an idea of what to do using AGI?

Thanks,

Greg


On Mon, Jun 3, 2013 at 11:00 AM, Yves A. <yv...@gm...> wrote:

>  hi,
>
> just very short... i am in a hurry...
>
> :1 means first leg of the call....
>
> what you mean is jumping to a priority?? (11) you should be able to do
> so... there is a parameter where you the specify the priority...
> if thats not working... try to go with agi... let me know.... and explain
> your objective... dialplan is not complete / fully clear 2 me...
>
> yves
>

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

[Rounak S. <rou...@gm...>]

Hii,
Is it necessary for the soundFile to be in /var/lib/asterisk/sounds/?What
do you exactly mean by "soundfile has to be in the right folder..!"?If
yes,how can I do it?

that means I have to store the voice file return by TTS software to
the /var/lib/asterisk/sounds/
and then use play it, all when the caller is online!!!I think its a time
taking process?

What is the best way to integrate TTS with my java program?

And what if my Java server and asterisk server are not in same physical
system.
Do I have to copy the soundFile from java server to asterisk server may be
using nfs or gufer?










On Mon, Jun 3, 2013 at 8:36 PM, Yves A. <yv...@gm...> wrote:

>  asterisk can´t handle mp3 by default.... you have to compile support for
> mp3 (licence issues...)...
> or you convert mp3 to wav (att.: wav != wav...) or gsm after tts
> generation.... sox can help you there....
>
> take a look in the log / verbose cli output to see, if any error occur...
>
> soundfile has to be in the right folder..! normally:
> /var/lib/asterisk/sounds/ and is addressed relatively
> to that path...
>
> use streamFile(file) in your AGI to stream the file to the caller...
>
> yves
>
>
>
> Am 03.06.2013 16:47, schrieb Rounak Saha:
>
> Hello everyone,
>
>  I have integrated my JAVA program(using asterisk-java) with a
> text-to-speech software where I send a string  to the TTS software and it
> return me a .mp3 file (which can be converted to any desire format).
>
>  Now I want this mp3 to be played to the caller.
>
>  Which fastAGI command should I use to play the mp3?
>
>  Do I have to first transfer my .mp3 file to Asterisk database and then
> ask it to run it?If yes,How should I transfer?
>
>
>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>

Re: [Asterisk-java-users] Playing the voice file returned by TTS Software?

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

asterisk can´t handle mp3 by default.... you have to compile support for 
mp3 (licence issues...)...
or you convert mp3 to wav (att.: wav != wav...) or gsm after tts 
generation.... sox can help you there....

take a look in the log / verbose cli output to see, if any error occur...

soundfile has to be in the right folder..! normally: 
/var/lib/asterisk/sounds/ and is addressed relatively
to that path...

use streamFile(file) in your AGI to stream the file to the caller...

yves



Am 03.06.2013 16:47, schrieb Rounak Saha:
> Hello everyone,
>
> I have integrated my JAVA program(using asterisk-java) with a 
> text-to-speech software where I send a string  to the TTS software and 
> it return me a .mp3 file (which can be converted to any desire format).
>
> Now I want this mp3 to be played to the caller.
>
> Which fastAGI command should I use to play the mp3?
>
> Do I have to first transfer my .mp3 file to Asterisk database and then 
> ask it to run it?If yes,How should I transfer?
>
>
>
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users