[asterisk-speech-rec] Sphinx and AGI integration; Digium vs VoIP gate

[johny jj2]

Thank you for your answer!

I watched video-tutorials of Lumenvox and contacted them.
Unfortunately their services are much too expensive. I guess similar
thing can be said about Invox.

In other words I need to create this system on my own. May you rather
answer my original questions, please?

Let me summarize those:
1. How to connect Asterisk with Automatic Speech Recognition? I
created formal grammars, algorithm in source code of java application
for CMU Sphinx4 ASR, acoustic model and so on. I found there are two
ways of integrating Sphinx with Asterisk:
http://www.voip-info.org/wiki/view/Sphinx and
http://scribblej.com/svn/ . But later I found that most of these
things are done with dial-plan. Do I need to use this Sphinx4 at all?
Or do I only need acoustic model for my language created with
SphinxTrain?
2. I'd like the user to be able to choose if he/she wants to use DTMF
or ASR in the given session. I thought that it should be like: a) user
chooses with DTMF what he/she wants to use, b) based on this decision
it switches to DTMF main algorithm or ASR main algorithm. How to do
this?

Regards!

2009/12/17 praveen kumar <pbx.kumar at gmail.com>:
> Hi -
>
> You have more than one possibility
>
> - you can use Lumenvox and buy their licenses and write a program to do that
>
> - The other option I can recommend is to try www.invox.com
> (Intelligent Voice) and build your phone system there. Since it
> integrates with REST, HTML - you can simply collect information from
> caller using dtmf and/or speech and then post these params to the REST
> page which will output the sum. You can collect the output and use TTS
> to play it back. It should be 5-10 mins to build this system. The
> system is hosted - so you don't have to worry about renting servers
> etc. You pay per call or per min depending on the plan.
>
>
> Thanks.
>
> On Thu, Dec 17, 2009 at 12:55 PM, johny jj2 <johnyjj2 at gmail.com> wrote:
>> Hello!
>>
>> I would be very grateful if you can answer my questions, at least with
>> one sentence :-). Or simply answer e.g. "1-3, 2-1, 3-2" (it is my
>> choice at this moment) and give short explanation :-).
>>
>> I'm familiar with using SphinxTrain and Sphinx4. I'd like to create
>> such an IVR-ASR system that:
>> a. user calls special number
>> b. he or she speaks twelve digits
>> c. server recognizes digits, calculates control sum and inform the
>> user about this sum
>> d. second and third steps are repeated many times until the user says 'finish'
>>
>> There are some things which I should consider:
>>
>> ---------------------------------------------------------------------------------------
>> 1. HOW TO ENABLE ACCESS TO ASTERISK FROM MOBILE PHONE (choice of
>> hardware and services)
>> keywords: server, Digium card, SIP/ITSP provider, PSTN/DID number
>> ---------------------------------------------------------------------------------------
>>
>> I've got server with access to internet. Unfortunately this server
>> runs on Windows (but I try my best to convince its admin to switch to
>> Linux and I may succeed). What should I buy for this server? I thought
>> about:
>>
>> 1-1. http://www.planet.com.tw/en/product/product_ov.php?id=4160 (price
>> about 230 euro)
>> 1-2. Digium card (I don't know approximate prices)
>> 1-3. buying service from SIP provider (what may be the prices of such
>> a service?)
>> 1-4. or should I rent server?
>>
>> Ad. 1-2:
>>
>> I asked companies from my country and only two providers answered me.
>>
>> First one (HaloNet) told me that in order to configure Asterisk for
>> HaloNet I need: 1. account (https://www.halonet.pl/rejestracja), 2.
>> password to account, 3. name for SIP server (sip.halonet.pl).
>> Additionally, to test incoming calls, I need PSTN number. They told me
>> to register for the service and then send mail to them with request to
>> add test number. They also provided examplary configuration for
>> Asterisk. How to create or obtain my name for SIP server?
>>
>> Second one (Ipfon) told me to 1. create an account
>> (https://rejestrator.ipfon.pl/index.php?version=ipfon_starter&scenario=telefon),
>> 2. configure trunk for Asterisk
>> (http://forum.ipfon.pl/index.php?topic=64).
>>
>> I also asked on Ekiga mailing list (it is not form my country;
>> http://mail.gnome.org/archives/ekiga-list/2009-December/msg00046.html).
>> They told that they cannot provide what I need. They told about ITSP
>> (not SIP) providers and DID (not PSTN) number. I thought I understand
>> that I need PSTN number from SIP provider. They told I need DID number
>> from ITSP provider and I'm really confused. So what do I need exactly?
>>
>> After all I guess it would work like this: user -> mobile phone ->
>> call -> servers of providers -> network cloud -> my server ->
>> Asterisk. Am I right?
>>
>> Ad 1-4:
>>
>> At first I thought about using server which they can provide me.
>> Access to physical, proprietary device would be necessary for 1-1 and
>> 1-2. However for 1-3 I can consider both options (to have my own
>> server or to rent server from somebody else). It is popular thing to
>> buy some space on server to upload webpage. Are there similar services
>> for what I'd like to do? In other words I need Linux server with
>> Asterisk and probably Sphinx. The disadvantage of my server is that
>> I've got Windows and perhaps I will have to use Asterisk in Windows
>> (however it is not a sure thing, there is possiblity that I would be
>> able to convince administrator to switch to Linux).
>>
>> --------------------------------------------------------------------------------
>> 2. HOW TO ENABLE SPEECH RECOGNITION ON SERVER WITH ASTERISK (choice of software)
>> keywords: AGI scripts, Sphinx4, ScribbleJ plugin, PocketSphinx
>> --------------------------------------------------------------------------------
>>
>> 2-1. I found this: http://www.voip-info.org/wiki/view/Sphinx . It is
>> AGI script to be called from Asterisk. Am I right that the only what I
>> need is Asterisk and Sphinx4?
>> 2-2. I found this: http://scribblej.com/svn/ . What kind of advantage
>> does it have if it looks like the same can be done much easier with
>> 2-1? For this solution it would look like: Asterisk <-> ScribbleJ
>> plugin <-> Sphinx4 (if it is possible to integrate it with Sphinx4, it
>> was tested only for PocketSphinx).
>> 2-3. Are there any other ways possible?
>>
>> ------------------------------------------------------------------------------------
>> 3. WHERE TO SPECIFY ALGORITHM? (Asterisk + Sphinx or Asterisk +
>> AGI/AEL/LUA scripts)
>> ------------------------------------------------------------------------------------
>>
>> I am also curious about the way how to specify the algorithm of the talk.
>>
>> 3-1. Formal grammars and source code for Sphinx4 application
>>
>> At first I thought about writing application for Sphinx4. The
>> application is written in java, normally executed as "java -mx256m
>> -jar bin/ApplicationName.jar". I create: a) acoustic model (it is not
>> English and it cannot be downloaded from VoxForge so I had to create
>> it myself in SphinxTrain), b) language model (created with lmtoolkit
>> online), c) formal grammars (it is crucial for the algorithm), d) list
>> of words, list of phonemes, e) main application (java source code). I
>> create (a) from (b) and (d) and then I use (c) and (a) for (e).
>>
>> 3-2. Dialplan with AEL/LUA script
>>
>> But later I talked a little bit on #asterisk at Freenode. (I installed
>> Pidgin in order to contact ScribbleJ, author of the plugin, but I
>> couldn't contact him after all). They told me "Implement the logic in
>> the dialplan. Or if you choose to use an embedded language like AEL or
>> LUA". So don't I need java source code from Sphinx4 at all? Do I need
>> to have installed Sphinx4 at all :-)? May you give me link to some
>> kind of tutorial about creating these dialplans? Do I still need
>> formal grammars from 3-1?
>>
>> Thanks very much for help in advance :-)!
>> Greetings!