similar to: problem with sound device

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

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

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

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

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

Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : "FATAL: Module zaptel not found." . The same uccurs when i type "modprobe wcfxo" May you help me. Thank you in advance Juanjo

I tried numerous but most of them just don't work. They fail on the quality of the microphone. I tried two that support alsa. Kphone does, but dtmf (the numbers) are not recognized by Asterisk :-( I tried X-lite, but that has really terrible voice quality (mic). I tried Sflphone (has alsa support) and that worked on FC4 (not on FC3) and it does not work on Centos44 either: [root at raaf

I took the latest installation of lame (both stable and beta) and installed those with the default install variables. The last one I compiled was lame3.87. I then have the ices-0.0.1.beta5 and config'd the sources with ./configure --with-perl --with-lame and installed. Yet, when I run ices -R, it complains with the following line when I run the -R option: Support for reencoding with

Hi, As I´ve understood things, the Ogg Vorbis format supports more that two channels (stereo). Is there any tools to encode x sourcefiles into one .ogg file? I am a musican and am thrilled with the ide of makeing music in surround (or in more than stereo). This leads to my next question: is there (developing) any decoders for multichannel oggs to, let us say, 4.1 or 5.1 surround? Wouldn´t it

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

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

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten => _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI> console dial 111 at default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [111 at default:1]

Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel