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

[praveen kumar]

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!
>
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