similar to: NUT and MGE Ellipse ASR 1500 under FreeBSD.

Hello, I'm facing problems as I'm installing NUT to monitor a MGE Ellipse ASR600USBS on a FreeBSD 6.1 box. UPS is recognized as : ugen0: MGE UPS SYSTEMS ELLIPSE, rev 1.10/42.41, addr 2 I'm temporarily using root profile (devfs/devd scripts not modified atm). newhidups starts & runs fine (unplugging mains makes newhidups outputs lots of data when started with -DDDD flags)

Hello everybody, Our supoprt team is facing a problem which froze the CPU. They suspect that this happens after an automatic battery test, which lets the battery goes down to 30% and then stop the test. At this level of charge, our kernel also triggers a shutdown (CPU and UPS). I saw (in the doc found here : tools.*mge*ops.net/download/intl/products/*ellipse*-asr/28-umeasr_08.pdf) that this

Hi all Arnaud, Some newly purchased Ellipse ASR-model UPS find their way to my office for a first evaluation test. It concerns the Ellipse 1500 and Ellipse 1000 models. As you know we are still working with hidups. Peripheral recognition, start of NUT, mains short interruptions management, all that work fine... as expected ;-) But changes occured on the delayed shutdown procedure after the

Hello guys, I have to manage an Eaton Ellipse ASR 750 VA UPS under a FreeBSD 8.2 server. I used nut from ports collection but I'm unable to detect the UPS by USB cable. Here is the output of 'usbconfig -u 4 -a 2 dump_device_desc' : ugen4.2: <ELLIPSE EATON> at usbus4, cfg=0 md=HOST spd=LOW (1.5Mbps) pwr=ON bLength = 0x0012 bDescriptorType = 0x0001 bcdUSB = 0x0110

Hi all, I am trying to communicate to this Eaton Ellipse (see above) using nut from a Debian Squeeze system on Armel architecture. But have not succeeded so far. Version: Linux b3 2.6.39.4-11 #1 Tue Apr 3 21:45:12 FET 2012 armv5tel GNU/Linux Nut version is 2.4.3-1.1squeeze2 this information if from dpkg. NUT installation method:from debian package for armel. The problem is I cannot

On Jun 20, 2013, at 10:35 AM, Louis Chaillet wrote: > With respect clarification of the documentation (and reference to your question Charles). The documentation does suggest that you can shutdown the ups by the upsdrvclt shutdown [ups] command. Specifically in "http://www.networkupstools.org/docs/man/upsdrvctl.html" in the section commands under warning. In the "Network UPS

Hi Charles, thank you for your answer. This was indeed the core of the problem. I have amended the configuration files. It means now the communication works as required. Alas I now have another problem. Just for the record here is the configuration after the changes. The configuration is as follows. #nut.conf MODE=standalone #ups.conf [eaton] driver = usbhid-ups port = auto

Hi, folks! There is FreeBSD 7.1, NUT 2.4.1, UPS APC Smart 2200 XL. NUT is compiled from ports with following options: --------------------------------------- _OPTIONS_READ=nut-2.4.1_5 WITH_SERIAL=true WITH_USB=true WITH_SNMP=true WITHOUT_NEON=true WITHOUT_PDU=true WITHOUT_HAL=true WITHOUT_IPV6=true WITHOUT_CGI=true --------------------------------------- UPS is successfully recognized by OS

Dear Sir, I am attempting to get a Broad Band Modem working on: sony# uname -a FreeBSD sony.family.hom 6.2-PRERELEASE FreeBSD 6.2-PRERELEASE #2: Tue Dec 19 16:55:50 EST 2006 root@sony.family.hom:/usr/obj/usr/src/sys/SONY01 i386 The device is a Sprint PC5740 pc card. When I perform a "man umodem" the card is listed (vendor = Curitel) UMODEM(4)

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

Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is

Hello, All of the Opus quality studies that I've seen focused on human-perceived quality. I'm interested to know of any experience with machined "perceived" quality, particularly related to speech recognition or biometrics. I'm also interested in folks thoughts on optimizing Opus for ASR. For example, removing certain classes of comfort noise, filtering non-speech bands,

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

Hi all, I have a little question I was not able to find answered on the Net:: How can I measure ASR (Answer-Seize Ratio) ? Which is its definition ? In other terms, it is said to be a ratio between "answered calls" and "total attempted calls". (cfr:http://en.wikipedia.org/wiki/Least_cost_routing) The question is: should I remove from the "total attempted calls" the

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

Hi Everyone, I am in search of a reliable open source tool for the real time monitoring of ASR & ACD, so any help or suggestions will be highly appreciated. Regards, Faisal Rehman -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111020/d283a5af/attachment-0001.htm>