similar to: Sphinx

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do?

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How

Hi all I install sphinx2 successfully but when i execute sphinx2-server, i have the error below: ad_oss.c(105): Failed to open audio device(/dev/dsp): No such device FATAL_ERROR: "server.c", line 476: ad_open() failed What's the matter? I also want to know how i can do in Asterisk to use the sphinx server.Must i write entirely an eagi script? And when it's wrote, which

Greetings Friends, I would be grateful if you can help me undestand how to make my R code more efficiently. I have read in R intoductory tutorial that a for loop is not used so ofter (and is not maybe not that efficient) compared to other languages. So I am trying to build understanding how to get the equivalent of a for loop using more R-oriented thinking. If I got it right one way to do that

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

Does the System() dialplan application have a limit on how long it can run? Either a time limit, or server load limit? I'm trying to pipe the output of Sphinx2 into Text2Wave, but Asterisk just runs by it to the next extension priority, with no errors. If I run the same command via the system shell, all is good, though it does take a few seconds, probably about 5 seconds to run. Yes,

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Hi Matt, Any decent quad, quad-core Opteron system should be able to handle it with ease! :) -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark AudioVisual http://www.quarkav.net http://www.threatcode.com/

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

Hi Here is another patch regarding time in ogg123 which is more controvercial than the other one I send some days ago. (see attachment) When working with Daisy/SMIL [1] files it would be helpfull if the user could stop play at a specific time. Currently .ogg files are not allowed in the Daisy format but that will hopefully happen one day if I work hard at it. A clip in a Daisy file could look

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file