similar to: Integration with external ASR engines

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello: I found that Amazon cannot find a copy of "Statistical Models in S"? I am about to embark on some tree-based and perhaps ANOVA models and have following options: (*) Find another book/online doc that covers this material (perhaps one recommended on the R FAQ page) (*) Use R documentation (*) Try even harder to land the white book. I have a decent conceptual and

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

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

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

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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.

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

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,

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

Hello, I'm in the early stages of designing an Emergency calling service IVR application. The IVR application asks simple one or two questions like "which is the postal code of the area you are currently calling from ?" "Is the correct ?". The expected values are a 5-digits number like "twenty-five-thousand-two-hundreds-twelve" or

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

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

The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT & T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once.

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------