similar to: signal amplified by asterisk

Hello there, recently installed CentOS6 on a (quite old) 64-bit system, and from the GNOME's sound preference app, I can see that the Output volume range is said "amplified" from 63% to 100%. Below 63% it's unamplified. No idea what's implied behind this amplification (I don't see this on other desktops), but I notice that when "amplified" the sound is pretty

I need decide on if our site will go with Java or Ruby on Rails. The major factor is that does Farret support Lucene''s ChineseAnalyzer or CJKAnalyzer or not. Can anyboby shine some lights on Farret''s Chinese search support? Really appreciate. -- Posted via http://www.ruby-forum.com/.

Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ?Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy

Hello list... 1) Is there an IAX softphone that supports any kind of voicemail indicator? 2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of

hi, dear all, does anyboby have an simple example of classification by using randomForest? Thanks a lot! best assicma

Hello, can anyboby tell me, how can I disable the echo cancellation for sip? thx a lot... best regards, Felix -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110216/4e54620f/attachment.htm>

Hi, I am using Samba as a file server only; with no whatsoever printing service. Therefore I would like to get rid of the printer-icon when the users access the samba-server. How can I do that ? I have tried to take off everything related to printing in the configuration file (/etc/samba/smb.conf) but the printer-icon is still there. Has anyboby got this problem ? Does anyone know how to

Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there

Hi user, I try to make plot with two axes y, it is posibility?. Can anyboby help me.... thank for all fernando [[alternative HTML version deleted]]

Hello evry one, I'm a frensh consulting Engineer in statistics and i work under R times to times. I would like to build an Graphic User Interface as we can do under MS Excel but i don't know how to do it. The aim is to have a GUI betwen a user and the console or the function (for exemple toto(a,b,c) ) In the GUI , i would like to have 3 boxes empty where i can put the values of a, b, and

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

Hi, I've had a small case with noise suppression and AGC. I have a fairly noisy environment here, and with the default parameters, noise suppression works fairly well while I talk. However, when I shut up, AGC starts slowly increasing the gain until it has amplified whatever noise is left to levels about equal to having no filtering at all. As soon as I talk, AGC backs down fairly quick

Hi all, this is my first post in this mailing group. I hope that anyboby could help me parsing a xml file. I found this website http://www.omegahat.org/RSXML/gettingStarted.html but unfortunately my XML file is not as easy as the one in the example. Example: <?xml version="1.0" encoding="UTF-8"?> <?xml-stylesheet

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

On Tue, Feb 28, 2017 at 10:49 AM, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > On Tue, February 28, 2017 9:22 am, Rob DeSanno wrote: > > Last time I saw it, I had just upgraded my CentOS 7 box with the > > 3.10.0-514 kernel and it rebooted already configured into debug mode. > Not > > sure if this is a ???feature?? of the newer kernels or not but glad

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

Chris Lattner <clattner at apple.com> writes: > I completely agree. The "branch" I most care about is mainline, and > losing the ability to say "fixed in r1234" (with some sort of > monotonically increasing number) would be a tragic loss. The monotonically increasing number is useful for giving you a fuzzy idea about how that change relates to other changes

While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the following. I limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty soon Asterisk ran out of RTP ports. Traced the problem back to how * is handling SUBSCRIBE. A sip structure is allocated as soon as a request is received, which also allocates RTP ports. Normally, this is not a problem as the structure

Hi, I'm a new Speex list member, and so far I'm very impressed with Speex. BUT automatic gain control seems to me to be virtually unusable. If a recorded utterance begins with even a tiny noise, that noise is amplified to the maximum volume. (See, e.g. the initial utterance in the TIMIT corpus, "She kept your dark suit...") This seems like a bug to me. If list members