similar to: Fading voice problem

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'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

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.

Hi I am trying to find a possible cause of 'audio fading in and out' which effects about 0.1% to 0.15% of calls placed to a voip provider for termination. (I am using SIP & alaw throughout in this case) I don't believe this to be something that is network / jitter etc related (?) and looking at historical posts this may be a telco issue (voltage?) at the voip provider gateway.

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

> 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

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

Evening folks: Not sure where this fits into the picture so bear with me for a few moments while I think out loud to lay out the particulars of my problem. -would love to be employed using R for a living but so far I have just worked with it on my home machine to develop a basic level of skill with it. -I run Ubuntu linux 8.04 LTS on a machine with a 2.5GHz Celeron processor and 1.2GB of memory.

On Tue, 2003-01-14 at 12:29, gtgbr@gmx.net wrote: > Scott Manley wrote: > [crossfading] > > well ... liveice has done this for the past 5 years.... but it's not an easy > > package to use > > One could add Winamp based streaming to this list - by using a > crossfading plugin and using a feature like Oddcast DSP's "Adv. Rec". thankyou ... but i run only

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, 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).

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

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

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO