similar to: Noisy/Clicky hangup

I''m using Prototype/Scriptaculous on a large web app. Everything runs briskly in IE, and everything runs briskly in Firefox except for one page. On this page, only in Firefox, any code that touches Prototype runs very, very slowly. In particular: I use Effect.BlindDown/BlindUp for dropdown menus. Same html and css on all pages. Flies along on all the other pages; here BlindDown crawls,

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr

I was really looking forward to Asterisk 1.0 et al, but it is a major disappointment. I have never experienced any Asterisk release that was interacting with Digium hardware so unreliably. Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as the call is being picked up at the other end. I have tried various X100P (original Digium) cards, various phone lines and just about every

Why, in R, does (0.1 + 0.05) > 0.15 evaluate to True? What am I missing here? How can I ensure this (ostensibly incorrect) behavior doesn't introduce bugs into my code? Thanks for your time. Dave Mitchell [[alternative HTML version deleted]]

I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxxxxxx number but I intend to add the ability to get a free United States DID

Howdy, We've been trying to get Asterisk to work properly for us and we aren't having much luck when it comes to simply speaking on the phone. We are experiencing hangups don't occur at any specific time. The equipment we use are two X101P's and one T100P. We originally tried with with the USB S100U as well (so channel bank isn't an issue). Sometimes the caller will be

(another in an ongoing list of things i just want to clarify for the sake of future courses taught on centos.) from this RHEL doc page: http://docs.redhat.com/docs/en-US/Red_Hat_Enterprise_Linux/5/html/Deployment_Guide/s1-openssh-server-config.html the reader is advised to, for the sake of security, remove/disable vsftpd, ostensibly in favour of sftp/sftp-server. really? i can obviously

Hi: I'm trying to plot dates on the x-axis of a code, but the legend is not being displayed. I receive the following error: Error in match.arg(x, c("bottomright", "bottom", "bottomleft", "left", : 'arg' should be one of bottomright, bottom, bottomleft, left, topleft, top, topright, right, center In addition: Warning message: longer

Greetings, I use S+, Stata, SAS and R depending on the weather... The script file editors in S+, SAS and Stata all allow me to submit a portion of a developing script file to the running process. In R I have to use an external editor and either source the entire file or do lots of copy-and-pasting. Are there any editors out there which interface to R more closely and let me submit a section

Hello. I am trying to invert a matrix, and I am finding that I can get different answers depending on whether I set LAPACK true or false using "qr". I had understood that LAPACK is, in general more robust and faster than LINPACK, so I am confused as to why I am getting what seems to be invalid answers. The matrix is ostensibly the Hessian for a function I am optimizing. I want to get

I tried to install Wine, but I get this message in console. /home/user> cd wine-1.0 /home/user/wine-1.0> .tools/wineinstall Wine Installer v1.0 Running configure... checking build system type ... i686-pc-linux-gnuoldld checking host system type ... i686-pc-linux-gnuoldld checking whether make sets $(MAKE) ... no checking for gcc ... no checking for cc ... no checking for cl.exe ... no

R help.start() opens a web browser as expected, but the ?Manual? html links are dead (although the ?Reference? and ?Miscellaneous Materials? html links do work). I get a similar problem from within RStudio. I?d like to get pointy-clicky html manuals working, to help migrate users from SAS to R. I do see R manuals on this system from the EPEL R rpm's in info and pdf form, for example at

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

> This wiki should cover most of the basic stuff that gets asked over and >over again just to help reduce the amount of repetition that most of you >have probably noticed takes place here. Problem is, Wikis in general suck and voip-info.org in particular is quite useless except as a random clicky-clicky exercise. You ever use the search on voip-info.org? It's almost like someone

This was asked many years ago but I thought I would check to see if things have changed. Is it possible to take over a call in progress - using a replacement Asterisk server? In other words, if 2 user agents are connected through an Asterisk PBX, and I tracked the call ID, IP of each UA (and anything else needed), could I remove the PBX and put a new one in its place (at the same IP

Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message.

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. ----------the dial and the problem----------- -- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new stack --

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

Hi, I am plotting a policy function (result from a dynamic stochastic optimization problem, discretized approximation). The policy function maps from an 2 x 2 x 2 x 3 x B x F state space to a B x F state space (B and F are usually between 4-6, and represent domestic and foreign savings. The other variables are income (Y), inflation (Pi), domestic and foreign interest rates (R and Z)). I