similar to: Getting realtime ASR and ACD from Asterisk

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>

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

Hi, There is a long call setup time untill the call connects. How can I play a beep tone say every 4 seconds to the caller untill the call connects? Tx. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/38e17d3e/attachment.htm

Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/85bd0069/attachment.htm

Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100410/0d4e92e9/attachment.htm

Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? Asterisk Version 1.4.21.2 Executing [0872200189 at In:2] VoiceMail("SIP/voip-1fd034e0", "910|u") in new stack -- <SIP/voip-1fd034e0> Playing 'vm-theperson' (language

Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.....! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b2691a9 at 411.2.139.106'. Giving up. Tx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have any suggestions (existing scripts, etc) for figuring out ASR? We wanted to write something to be a little more in depth then what we have now. Wasn't sure if anyone had any suggestions for things that might be pre-existing, or what you've done to accomplish this. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Say anyone know howto debug this: Failed to initialize G.729 copy protection! X64 CentOS system. Running Asterisk as Non-root. Downloaded latest G729 driver and registered it sucessfully. Restarted Asterisk. But still get this error! Asterisk 1.4.21.2 built by shaunw @ xxx.xxx.biz on a x86_64 running Linux on 2008-08-06 19:11:02 UTC Intel(R) Xeon(R) CPU E5420 @ 2.50GHz show g729 No

Hi, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

Hello, I have Ellipse ASR 1500 UPS and now I'm trying to monutor it with NUT from FreeBSD. It seems that there are some problems with this UPS: =================================== ugen0: MGE UPS SYSTEMS ELLIPSE, rev 1.10/42.41, addr 2 > usbdevs -v Controller /dev/usb0: addr 1: full speed, self powered, config 1, OHCI root hub(0x0000), NEC(0x0000), rev 1.00 port 1 powered port 2

Hi ! i'm a new user, and we are planning to use asterisk mostly as a basic ACD. What we would like is basically getting the list of all waiting calls in asterisk, and then display it on our staff homebrew app, for them to choose which call to take. Basically the list should display the caller's number, the time he was waiting and such. Is there any way to grab the waiting list and

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,

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 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 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