similar to: echo / delay problem

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

I'm having a problem defining my channels Here is my zapata.conf ----------------------------------------------------------------------------------------------------- ; Zapata telephony interface ; ; Configuration file [channels] ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes

Dear all, I need to use echo cancellation in a VOIP application using speex narrowband codec.I am facing some problems with echo cancellation. It will be very helpfull if someone can put some light on the following issues. 1: =================== -- When call is started EC works fine for first few seconds. Bur after a certain period [usually 20 sec or more] it does not work or works very bad

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.

Cool, thanks for all the great comments. I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module 2) Is it best to use a TDM card with an TFX and FXO module or is the the FXO = X100p and FXS on TDM a good method? 3) How do I

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

> I am writing to gain a better understanding of the limitations of speex echo > cancellation, esp. with respect to the fixed point implementation. > If these limitations have been documented elsewhere already, please let me > know! Nothing officially documented, sorry. > I observe experimentally that when one or both of the echo or ref data for > speex_echo_cancel() have

Hi all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This

Dear helpers I constructed a data frame with this structure > str(dados1) `data.frame': 485 obs. of 16 variables: $ Emissor : int 1 1 1 1 1 1 1 1 1 1 ... $ Marisca.Rio : int 1 1 1 1 1 1 1 1 1 1 ... $ Per?odo : int 1 1 1 1 1 1 1 1 1 1 ... $ Reproducao : int 3 3 3 3 3 3 3 3 3 3 ... $ Estacao : int 2 2 2 2 2 2 2 2 2 2 ... $ X30cm : int

Dear Prof Ripley, First of all, unless you are an english professor, then I do not think you have any business policing language. I'm still very much a student, both in R, and regarding signal analysis. My competence on the subject as compared too your own level of expertise, or my spelling for that matter, may be a contension for you, but it would have been better had you kept that opinion

What I did was to plot your initial values, then plot the smoothed values and guess the constants. That is, I got an "eyeball" fit to the smoothed values. As I have described this as "gross cheating" in the past, you should either split your data, estimate on one subset and then test on another, or estimate on your data and test on a replication. If you get pretty much the same

Hi all,I had a question last week on asking for a function that will help me draw three different circles on x,y,z axis based on polar coordinates (Each X,Y,Z circle are coming from three independent measurements of 1-360 polar coordinates). It turned out that there ?is no such function in R and thus I am trying to write my own piece of code that hopefully I will be able to share. I have spent

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Please look at what I see in your code below (run-on code mush) to understand part of why it is important for you to send your email as plain text as the Posting Guide indicates. You might find [1] helpful. [1] https://wiki.openstack.org/wiki/MailingListEtiquette -- Sent from my phone. Please excuse my brevity. On June 25, 2017 2:42:26 PM EDT, Alaios via R-help <r-help at r-project.org>

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

Hi! I'm learning mgcv, and reading Simon Wood's book on GAMs, as recommended to me earlier by some folks on this list. I've run into a question to which I can't find the answer in his book, so I'm hoping somebody here knows. My outcome variable is binary, so I'm doing a binomial fit with gam(). I have five independent variables, all continuous, all uniformly