similar to: (possibly) new use for asterisk

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello all >> >> As of 5pm PST today (7/13), I pulled

Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead

> Well, I thought about it as well but up to now didn't try. Via which > way do you intend to have e.g. a line-in to the iPaq with reasonable > quality. Would be really interesting to figure out such things ... > and then use a WLAN-cradle to transport data to the internet *g* The main use would be covering events at sports fields, auditoriums, etc. that are away from the

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters:

Hi all, I get these errors while trying to run RealProducer 8.5 in WINE: err:heap:HEAP_ValidateInUseArena Heap 40310000: in-use arena 4036ae80 next block has PREV_FREE flag err:ntdll:RtlpWaitForCriticalSection Critical section 0x40310070 wait timed out, retrying (60 sec) fs=0297 err:ntdll:RtlpWaitForCriticalSection Critical section 0x40310070 wait timed out, retrying (60 sec) fs=0297

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

Hello Vorbis Developers. I'm in the proccess of writing an OggVorbis audio player using libao and libvorbisfile. I've greatly appreciated the simplicity of the API and good documentation with the souce of the libraries. I've descided that I really want a mixer on my player, and I'm unsure how to best go about it. LibAO (unless I'm absolutely blind) doesn't

Hi everyone. First, sorry for my bad english. I'm brazilian and I'm learning english yet. Well, i've installed wine 1.1.12 in my kubuntu 8.04, but I don't achieve to run GTA San Andreas. When I try to run the game, it returns this error: err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:mixer:ALSA_MixerInit No master control found on

Do es anyone know what the following means for sound under Wine? fixme:mixer:ALSA_MixerInit No master control found on MPU-401 UART, disabling mixer Severa of us are trying to get WOW internal voice chat and teamspeak to work (different uses - wow for groups, teamspeak for guild). Im wondering if the above output has a workaround until a fix is made or even if it's related to our partial

Package: logcheck-database Version: 1.2.32 Severity: wishlist Ignore rules to supress messages generated from pugging in, and then removing, a USB headset (one speaker). ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: drivers\/usb\/class\/audio\.c: v1.0.0:USB Audio Class driver$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usbaudio: assuming that a stereo channel connected directly to a mixer is

Just bringing up a Centos-6 (x86_64, FWIW) and I can't find the mixer app... on my C5 box(es), double click on the speaker icon in the top panel and I get a mixer app appears on the screen. My new C6 box doesn't do that. I've installed all the PULSEAUDIO RPMS I can find, but still nothing. what am I missing here? thanks! Fred

I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet {~~~~~~~~~} +---------------+ +-----------+ +---------------+ +---------------+ {

I have to run this windows application on my linuxbox. But when manycam run give an arror message "Can't load driver for the virtual Manycam capture device. Please reboot your computer and restar manycam" The console wine output is that: . at dknight-ubuntu:~/.wine/drive_c/Programmi/ManyCam 2.3$ wine ManyCam.exe fixme:mixer:ALSA_MixerInit No master control found on Trust WB-6250X

Dudes.... I'm running Debian sarge and i can't get the %$#%# sound to work. I installed Starcraft but it doesn't recognize any sound at all. My sound card is a SB extigy. HELP PLZ! Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20051208/bf4d5ccf/attachment.htm

On Sun, Dec 22, 2019 at 5:23 PM David G. Miller <dave at davenjudy.org> wrote: > I do something similar to what you're trying to do when I digitize my > vinyl LPs. I take output from my amp/turntable into my sound card and > use Audacity to digitize it and write an appropriate sound file. I also > monitor the sound using Audacity by telling it to also route the sound >

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

Hello everybody... I figure this is not exactly an icecast question, but maybe someone might be able to help. Icecast runs nicely as far as I can see, and so does liveice, apart from the fact I don't get any input signal fed into it. I think it's due to the fact it expects the signal on line in by default and the souncard only has a mic in. So I tried to use the built in mixer mode,

wow, Enrico ... it's a great help ;-) I comment: ----- Original Message ----- From: "Enrico Minack" <enrico.minack@informatik.tu-chemnitz.de> To: <icecast@xiph.org> Sent: Wednesday, February 11, 2004 8:00 PM Subject: Re: [icecast] Q: Is it possible? <p>> Hi Raúl, > > interesting project ;-) So this is what I would recommend. For a number of > remote

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a