similar to: Red Hat patch kills streaming

Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs "Soundblaster PCI 128-Voice" soundcard ($13). This card is on the "approved" RedHat list. However, the documentation inside the package says "VIBRA 128". Anyhow, kudzu doesn't see then

I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Anyone else seen this eth-device and what brings it up? [root at oh1mrr etc]# ifconfig __tmp1192348344 Link encap:Ethernet HWaddr 52:54:AB:DD:2E:D1 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:2094689 errors:0 dropped:0 overruns:0 frame:0 TX packets:1645734 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX

The problem was because it is missing the "loop" module. the hotplug loads it automatically, while udev does not. The error xenoprof: Initialization failed. Intel processor model 15 for P6 class family is not supported" does not interference the winxp load.. so its ok. Does anyone know how to load osx86? It hangs at "booting from cd-rom". do I need kqemu/qemu xen? I

I noticed that my previous message is not very complete so I send here an "enhanced version". Please disregard the old one an reply to this one only. ( you can delete the ATTENTION word from subject ) Wilson (defiler@null.net) wrote : > There are two ways to decode multi-channel audio. In hardware, or in > software. > Hardware: A receiver or processor takes a Dolby Digital

Ok, is this possible: I want to have a 128k and a 24k stream of a particualr audio program, plugged into the line in of my Ensoniq AudioPCI 128 (es1370 chipset) Is there any way to do this with just one soundcard, or do I need two? Thanks Scott W --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a

I've set up a new * system as a test and development system. When I start asterisk, I get a fast clicking on all FXS ports (about 4 or 5 clicks per second.) Otherwise the system works great! When I make a call to another extension or to the console the clicking stops and the sound quality is great from then on. However, when I restart * the clicking resumes. I'm using an Adtran

>On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: >> > Hello, >> > >> > I have a strange situation: >> > >> > I can talk to other SIP phones and via ISDN to the outside, but I >don't hear >> > playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played,

Works just fine - though it might not be documented (seems like this info was dropped from the sample liveice.cfg file in the later distributions). Basically, at the end of liveice.cfg add the line: ENCODER_STREAM_SET 1 then define the values that would be different on the 2nd stream, ie: NAME Scott's 24kb/s stream SERVER xxxx.yyy.com PORT 80xx PASSWORD (if different than 128k) BITRATE

I have a small system based on one TDM400 card with - 2 FXO modules for incoming lines - 1 FXS module for one phone The system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS

Hi gang! Running 5.1 with Gnome as the default desktop, but have installed KDE too so I can run kde apps. Kmidi is the default midi player, so if I click on a midi file in firefox it brings up KMIDI. KMIDI goes merrily along playing the file, but I get NO SOUND. My motherboard has built in audio but it's disabled and I"m using an old Audiopci 128 card which seems to work fine for

Hi, I have 2 sound cards on my system a SIS controller with an i_810 module ( generic rockwell itu chipset ) and another recognized as cirrus log ( cs466xx module ) ..this is creative soundblaster 128 pci ... i used sndconfig to configure them and the sound plays on both...cirrus logic is the primary device presently ( red hat fedora core ).. The other apps using sound work well but no

https://bugs.freedesktop.org/show_bug.cgi?id=63101 Priority: medium Bug ID: 63101 Assignee: nouveau at lists.freedesktop.org Summary: [Ubuntu 10.04.4 LTS 32-bit] NVIDIA GeForce 7300 GT AGP graphics card will not display any text characters on VIA Technologies Apollo MVP3-based mainboard QA Contact:

I have Red Hat Linux RHEL 4 with Samba version 3.0.10-1.4E that mounts a local Windows 2003 Server disc with the following command: /sbin/mount.cifs //img1/docs /var/www/documents -o dir_mode=0777,file_mode=0444, gid=100,uid=501,username={xxxxxxx},password={pppppp} The mount succeeds and the mount appears on the output of 'df' and the documents are accessible and life is as grand

Hi, today I wanted to start Monkey Island 4 with wine. I'am using cvs version of wine from today (04.Feb.2001), I also patched the transgaming directx patch into it. My system: SuSE 7.0 with KDE2.1beta2, GeForce 2MX with 0.96 NVidia drivers, Soundblaster PCI 128 (ES1371) with Kernel drivers. I use wine with installed Windows. My startup command is: wine "c:\Monkey4\Monkey.exe"

After over a month (well, ok, no more than an hour a day :) of planning, getting hardware, tinkering and testing, I'm about to my Ultimate Home Phone System (tm) online. Connectivity to the outside world is provided by: A. 1 POTS phone line connected through an X100P ($11/month, needed to carry DSL) B. 1 Vonage ATA186 connected through an X100P (needed for the rate center :( ) C. 3 Broadvoice

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

Hello All, I'm trying to establish a MASTER SLAVE configuration of UPS using NUT Driver. I'm able to shutdown the master after UPS is on BATTERY ( POWER BLACKOUT) .However I'm trying to shutdown the other SLAVE MACHINES BEFORE MASTER. But at the remote side ( Slaves ) shutdown after UPS blackout does not happen. My configuration in Master are /usr/etc/upsd.users

I am, as this question will make very obvious, not exactly a guru when it comes to audio under Linux. I'd like to record some prompts using a nice microphone and my Ensoniq sound card, and then convert them to gsm/wav for use with asterisk. What Linux command would I use to a) do the recordings, and then b) convert them to the appropriate formats? I know there are