search for: chirps

I'm trying to solve a homework problem using R. The problem gives a list of cricket chirps per second and corresponding temperature, and asks to give the equation for the linear model and then predict the temperature to produce 18 chirps per second. So far, I have: > # Homework 11.2.1 and 11.3.3 > chirps <- scan() 1: 20 2: 16 3: 19.8 4: 18.4 5: 17.1 6: 15.5 7: 14.7 8: 17.1 9: 1...

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

RF trains fine with X, but fails on prediction > library(randomForest) > chirps <- c(20,16.0,19.8,18.4,17.1,15.5,14.7,17.1,15.4,16.2,15,17.2,16,17,14.1) > temp <- c(88.6,71.6,93.3,84.3,80.6,75.2,69.7,82,69.4,83.3,78.6,82.6,80.6,83.5,76 .3) > X <- cbind(1,chirps) > rf <- randomForest(X, temp) > yp <- predict(rf, X) Error in predict.randomForest(rf, X)...

Has anyone had an issue with the polycom's not discontinuing the mwi chirp even after the message has been acknowledged? -- James M. Milne Nuvio Corporation CCNA - CCNP - CIPTSS - CCIE milne.james@gmail.com

..." for the "Battery disconnected" and "Battery reattached" log entries? The rebooting came as a result of me turning off modems, router, external drives, monitor, cordless phone and finally, the computer, trying to locate a quiet but annoying "chirp". The "chirps" stopped when I tilted the UPS to look at the front panel. As you can see, log entries stopped at as suddenly as they started. [root at madeleine ~]# grep apcupsd /var/log/messages | grep Aug Aug 1 22:56:20 madeleine apcupsd[3342]: Battery disconnected. Aug 1 23:11:36 madelei...

I was wondering if anyone has had a problem with an initial chirp from asterisk after dialing an outside line through a TDM400? I am using 3 polycom 500's and it happens regardless of what phone I am at. So I was thinking maybe it was an issue with asterisk or the uLaw codec? Thanks. Jared Armstrong -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear helpers, When I use the function cgt of library Rwave, the time axis of the plot is always on the [0,1] scale regardless of the original time. This happedn when I tried to reproduce some pictures of "Practical Time-Frequency Analysis", e.g. Figure 3.5 and 3.6 using the codes provided in the book. But the figures of the book display the real sampling time using the same codes.

...xs_ks - no difference. Various things were tried to no avail, until I made a long distance call over a different carrier to our usual carrier (we use Optus, I made the call over Telstra). When the remote end answered, my end disconnected. What was happening was, when the call is answered, 5 quick chirps are sent down the line. However, because of the bug in the Cisco 7960 causing the first 1/2 a second or so of a conversation to be cut off - I didn't hear these chirps and as such I didn't think of the next bit: Basically, because I had busycount set to 3 and busydetect set to yes, these c...

Hi Jean-Marc, I use the resampler to convert various sampling frequencies to 48 kHz on my Blackfin platform (fixed-point) 48K -> 16K speex -> 48K chain does not sound very good compared to plain 16K. But the main issue is when processing loud signals, I have truncation (and not clipping/saturation) I could hear it and see it with various music and speech messages. See example.png. I also

Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible "Message waiting" indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks!

Hello, i try to send out a sms, but with no success. The trunk is a E100P, and the sms should go out to the Telekom SM-SC. What i want to to at the first run is, sending out a sms when a certain number is dialed. I tried: In extensions.conf: exten => 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,"Hi there") exten => 35953,2,SMS(${TRUNK}/9350193010) exten => 35953,3,Hangup

Hello, I faithfully followed the instructions from: http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk but still the message waiting indicator doesn't flash when a message is waiting. There is a brief intermittent chirp but nothing more. Using latest firmware 1.4.1 Thanks for your suggestions,

I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Using the following MATLAB snippet: fnames={'chirp','perfect','block','filter'} for k=1:length(fnames) fn=fnames{k}; myfile=fopen([fn, '.fl'], 'r', 'ieee-le'); x=fread(myfile, Inf, 'float32', 0, 'ieee-le'); fclose(myfile); X=specgram(x,2048,1,kaiser(2048,14)); spectrogram_floor=-96;

...figured this must be a change on the Shoutcast side and probably required that I upgrade to 1.3.10 and move into the new millenium with the adoption of IceS (instead of Shout). I think I have everything now running on a secondary test server, but MTD (Mean Time between client Disconnects and other chirps, skips, etc...) is now every few minutes. I'm also a little lost on the whole "XML config" setup, but I'm still plugging through the docs... Shoutcast directory listings are still not showing up... I'm hoping I don't need to run their servers to make this work... ;-( My...

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

I just upgraded a box to 1.2.3 this morning after encountering the issues noted earlier on the list. Everything is great. In fact, a LOT better. In the past few weeks, I've been battling with BV to address dropped outgoing voice packets (the flipside is that I haven't experienced this with other providers during tests), and an annoying mechnical 'chirp' at the start of a call.

I read through the "Writing R Extensions" and the "Debugging in R" website (http://www.stats.uwo.ca/faculty/murdoch/software/debuggingR/), looking for some hints about how to solve the issue of debugging problems encountered during the "R CMD check" process, but nothing seems to be mentioned about addressing issues encountered. Specifically, I am working with the R

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc