similar to: audio fading in and out?

Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from people. Soon I figured some network problem, I did a tcpdump and analyzed by wireshark ...the strange thing is this ... all packets that

I've had several users today inform me that whilst they were on a call, the volume kept fading in and out to such an extent that they thought the caller had hung up. I would dismiss this if it were a single person mentioning it, but it isn't .. Has anyone else seen anything like this ? We haven't, and we've been running for nearly a year ... CVS head as of 3 days ago, TE410p

well ... liveice has done this for the past 5 years.... but it's not an easy package to use ----- Original Message ----- From: "Geoff Shang" <gshang@uq.net.au> To: "icecast mailing list" <icecast@xiph.org> Sent: Monday, January 13, 2003 10:59 PM Subject: Re: [icecast] Fading between songs <p>> Hi: > > To do fading, you'll need to use a source

Hi: To do fading, you'll need to use a source that will: 1. Decode your songs 2. Do the necessary crossfading 3. Re-encode the stream and send it to icecast. right now, I know of no streamer that will do this. I'm aware of a very comprehensive package under development, but there's no public access to it yet that I know of. Note that there are strong feelings on both sides as

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

> Any idea or suggestion to avoid having two sound cards to do what I need to > do ? > > Or how to fade in / fade out without using the card' mixer, while playing > i.e. using mpg123 ? Depends on how much you want to program. Basically, the easiest would probably be to use OSS software mixing; if you buy the commercial OSS drivers, you can have several programs writing to

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

Hi all, Originally tried to post this without being subscribed, apologies if the list gets this twice. One of my users has a problem with many of his calls via my Asterisk? server. He describes the problem as having the sound slowly fade out and then fade back at a regular frequency. Has anyone experienced this before? Can you suggest a potential cause? The client is an IAX2 soft

Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. I've got a user who either can't be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me.

Hello all. I have the following setup: Fax machine | Sipura SPA-3120 | SIP 100BaseT | Asterisk 1.4 | IAX2 100BaseT | Asterisk 1.6 | ISDN PRI TE210P | Traditional Telco The fax lands on the Internal Asterisk 1.4 box, the sip config for this extension looks like: [35081] type=friend secret=************ qualify=yes port=5060 nat=no host=dynamic dtmfmode=rfc2833

Hi, Here is a simple patch that adds an option to the Fade plugin to enable/disable fading on minimize/open/close window events. This option is on by default but can be disabled to avoid conflicts with the Animation plugin. Fading on other stuff (fade during nonresponding app. desaturation, during switcher, and visual bell) shouldn't be affected (which is the main point of this patch).

Hello! I am encoding small snippets of audio (e.g. 100ms) that contain important audio from the first sample till the last one. (means they dont start/end silent). Doing so raised a couple of questions I couldnt solve by reading the docu/faq/internet search. 1. Does the "complexity" parameter influence only the speed of the encoder or also the speed of the decoder? (I need fast

HI all , i'm using icecast1.3.12/IceS0.23. Is it possible to "fade" songs ? i mean , begining the following song a few seconds before the previous ends ? (Tell me if it's necessary to upgrade to icecast2/ices2 ) Thanks in advance .... <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this

I have an account with a IAX service provider that I'm happy with but recently I started getting rather strange reports from users. They're saying that occasionally, they'll be on a call via the provider and the outbound audio appears to slowly fade out to nothing with a bit of static during the fade. Does this ring any bells with someone? I'm going to update * on the server

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

Hi There, I have just ported to Linux a program that I wrote years ago in Prolog under DOS to automate our Shortwave radio station. Back in 1990 the original program used ADPCM coded audio files and a special board. Now I can use MP3 files and a Soundblaster compatible card ;-) under Linux, and it runs from a console without any fancy interface. I absolutely need to fade-in and fade-out

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

Been googling trying to find a workaround, with no luck yet ... I hope y''all can help. My goal is to have a semi-transparent background screen on a DIV which will fade in and out over another DIV with a photo for a background. currently I have background-image: url(screen.png) which is a 32x32 PNG24 filled with white at 35% opacity. The problem is that as the div is fading-in, IE

This is off topic but I'm sure many subscribed here may be able to help. I'm writing an audio player, which includes vorbis of course, for the broadcasting industry and I need to prematurely fade out songs before they have finished. I've tried a linear and a logarithmic fade but neither sound natural to me. I was never good at logarithms in school and that was a long time ago. Can

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and