[asterisk-speech-rec] asterisk-speech-rec Digest, Vol 38, Issue 6

Mon Dec 21 12:14:13 CST 2009

I have had some very good successes with continuous speech recognition
and understanding working with the $50 LumenVOX software package.  The
time savings was well worth the $50.  Installation took less than an
hour and the grammar language is very easy to work with.

I selected the bottom end LumenVOX package simply for evaluation and
found it sufficient for my needs.  My only complaints are that the
speech engine must be run on a 32 bit box, the engine is very dependent
upon using a Fedora platform, and the bundled tools must be run on a
Windows platform.

On 12/21/2009 12:00 PM, asterisk-speech-rec-request at lists.digium.com wrote:
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> Today's Topics:
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>    1. Re: Sphinx and AGI integration;	Digium vs VoIP 	gate
>       (praveen kumar)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 21 Dec 2009 01:42:55 -0800
> From: praveen kumar <pbx.kumar at gmail.com>
> Subject: Re: [asterisk-speech-rec] Sphinx and AGI integration;	Digium
> 	vs VoIP 	gate
> To: Use of speech recognition in Asterisk
> 	<asterisk-speech-rec at lists.digium.com>
> Message-ID:
> 	<da73877d0912210142r5de6cc73mceaf9544de93ac3b at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Sphinx was never written keeping telephony applications in mind. Last
> time I checked, they had WSJ models for 8k but they did a pretty bad
> job.
>
> Each port from Nuance costs $500-2000 + maintenance depending on the
> grammar size.
>
> So, expect speech to be expensive.
>
> Good luck.
>
> On Sun, Dec 20, 2009 at 6:05 AM, johny jj2 <johnyjj2 at gmail.com> wrote:
>   
>> 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!
>>>>         
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