similar to: UniMRCP and Asterisk 14

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

Hey all - In some previous conversations on the Asterisk mailing lists, we noticed that some users of Asterisk were using UniMRCP [1] with Asterisk, as well as some modules made and distributed by that project. Unfortunately, there were some licensing concerns with using UniMRCP with Asterisk. As such, we contacted the UniMRCP project regarding the licensing issues and, after discussing the issue

I'd like to announce the release of an open source connector bridge for Asterisk and UniMRCP. The connector bridge is an implementation of Asterisk's Generic Speech API using UniMRCP client stack. This module allows Asterisk to connect to MRCPv2 or MRCPv1 compliant servers for speech recognition. It also allows to offload Asterisk using client/server architecture MRCP provides. Moreover,

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

I'm getting cycles of repeated crashes which occur and then stop occurring. Looking at the dumps via gdb shows that something peculiar is happening that looks like memory corruption: Program terminated with signal 6, Aborted. #0 0x0000003686e30285 in raise () from /lib64/libc.so.6 (gdb) up #1 0x0000003686e31d30 in abort () from /lib64/libc.so.6 (gdb) up #2 0x0000003686e6971b in

All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. Asterisk version: Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux on 2014-08-11 13:55:25 UTC OS: Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 2014 x86_64 x86_64 x86_64 GNU/Linux When

Just a quick and dirty thought, try the MONITOR application. Pseudo-code: Anchor-point PLAYBACK ("press or say") MONITOR (use the split audio files mode, not the mixed - this way you can roughly separate which side did the "talking") READ (audio file "1 to 5", try to grab one digit) STOPMONITOR IF (READ variable timed-out, send the incoming half of the monitor file

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2:

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote: > Oh, what a good idea! That's exactly the kind of lateral thinking I > was hoping someone would come up with. > > I thought it was called MixMonitor, and tried to wrap my head around > it but couldn't. MixMonitor is related, but different (and as the name suggests, automatically mixes the two channels, so I

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

Hello Dovecot users, Unfortunately, Pigeonhole v0.4.4 had a rather big issue with error handling and the execution of the implicit keep. This problem presented itself for example when users exceeded their quota. This did not result in the proper rejection message. This is why I quickly release a new version. Changelog v0.4.5: + Added a Pigeonhole version banner to doveconf output. This way,

Hello Dovecot users, Unfortunately, Pigeonhole v0.4.4 had a rather big issue with error handling and the execution of the implicit keep. This problem presented itself for example when users exceeded their quota. This did not result in the proper rejection message. This is why I quickly release a new version. Changelog v0.4.5: + Added a Pigeonhole version banner to doveconf output. This way,

Hi, There is no change in the compiler flags. I'm using as it is from the original code. No change in the Makefile and I believe it is using the floating point only by default. We are using 8k samples and mono so the commands is as follows. [root at MEDIA opus-1.1]# ./opus_demo -d 8000 1 opus_encoded_crash.opus opus_encoded_crash.pcm *And segmentation is as below..*. ............ Calling

Folks, A new release candidate for 3.2.0 has just been checked into the xen-unstable tree. It''s available from staging and will be in the main tree when it has passed internal regression tests. Meanwhile, in preparation for 3.1.3, please let me know if there are any further patches from xen-unstable that should be backported into the 3.1 branch. You can pull the xen-3.1-testing.hg

Hi, We are able to reproduce the issue with the 1.1 opus_demo (sample file). We captured the frames in our server just before the opus_decode and fed the file to opus_demo (1.1) and it is crashing. Same file is tested with 1.1.1 and it is fine. So this is in line with our server testing observation and I think here we can conclude that the 1.1 library is crashing while handling a specific mode

Red Hat Enterprise Linux Server release 6.4 (Santiago) gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) We see the issue in all our Intel based Linux servers. Thanks Suresh On 21 April 2015 at 12:41, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Still can't reproduce. What OS and compiler version? > > Jean-Marc > > On 21/04/15 02:48 AM, Suresh Thiriveedi

Hi everybody! Now that RSYNC has RSYNC+ included a good usage would be to use RSYNC+ to gather update-date, then multicast that on your hosts and process it. So my question is: does anyone know of a product which does reliable multicasting? (source available would be preferred) Simple pointers are appreciated; if noone has one I'm thinking about writing one myself. Thanks for all help!

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to