similar to: T1 or T1PRI? which sould i use?

Dear all! Although the Psycho-Toolbox for Matlab is free software, Matlab isn't. I'm planning to do an experiment where it's essentail to travel to the subjects, not let the subjects come to where the Matlab licences are :-( So I need to use a free software for my experiment if I don't want to by an extra Matlab licence (which I don't want to). Did anyone ever try to do

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou.

Hi Julien, Quote from : http://dspguru.com/dsp/faqs/multirate/resampling "The problem is that for resampling factors close to 1.0, the interpolation factor can be quite large. For example, in the case described above of changing from the sampling rate from 48 kHz to 44.1 kHz, the ratio is only 0.91875, yet the interpolation factor is 147!" My guess is that Opus would perform similar to

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have an AVI file (~900MB) that I want to convert to OGM format, with a ~ DivX 5 video and uncompressed AC3 (5.1ch) audio stream. Nandub doesn't want to compress the audio, so I was wondering if anyone could recommend a (Windows) tool to help. Thanks for your time! -Chris - -- "Everyone has to find their own reason." ~ -Cloud,

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

Whats the best way of intergrating del.icio.us into a RoR app? -- Posted via http://www.ruby-forum.com/.

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Thanks, Chris

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Has anyone done any installations using the Xorcom TS-1 and multiple T1's? I'm looking for a reliable box to put in a closet and route calls between T1's. No voicemail, IVR, etc. Any other suggestions? Mark Willis Cartama

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

Hello. New to the list. We're in the process of deploying Asterisk. Actually, we're going live tomorrow, and I just found out that my Zaptel cards have been mis-configured. I'll preface this by saying that I have looked in the wiki, read through the samples, and attempted to call Digium (they're closed.) So I'm praying that someone on the list can help me out! I have two

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

> Mark Farver wrote: > > > On Fri, 2004-12-17 at 16:26 -0600, Kristian Kielhofner wrote: > > > >> We are getting pricing and one provider is telling us > that they have > >>quality issues with the "Integrated Access" product. From > what they > >>say it sounds like you can have audio dropouts on the voice > channels >

I''m using the mongrel_cluster recipes for Capistrano as laid out by codahale.com and the new ed. of the Agile Development book, but I''m having problems on restart and stop: I can start: [root current]# mongrel_rails cluster::start Starting 3 Mongrel servers... And I can see 3 processes running: [root current]# ps aux | grep mongrel mongrel 1947 0.0 8.2 24684 10776 ?