similar to: Re: Sound Card help -- Solution -- and a question

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

The make seems to go okay. [root@somebox asterisk-1.2.7.1]# uname -a Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux mkdir -p /var/lib/asterisk/sounds/digits mkdir -p /var/lib/asterisk/sounds/priv-callerintros for x in sounds/digits/*.gsm; do \ if grep -q "^%`basename $x`%" sounds.txt; then \ install -m 644 $x

On Thu, 2005-01-06 at 12:00 -0600, asterisk-users- request@lists.digium.com wrote: > Andy Burns wrote: > > Shoval Tomer wrote: > > > >> Can anyone comment why shouldn't we use FC 3 for an * production > system? > > > > > > when I tried the X100P drivers on FC3 I had problems with udev, the > > workaround didn't work for me, maybe

Hi all, on my debian box i configured chan_oss to work with /dev/audio device. CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice 2 problems: 1. audio is very low in volume, even if i set 100 the mixer volume (via cmd line setmixer utility) 2. the sound is very crappy: the voice is "vibrant", words sounds like 'ttthhhiiisss iiisss aaa ttteeessstt". Seems

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Does anybody know what would be causing the errors below? I get these errors continuously until asterisk finally quits. This happens when I make 20 simultaneous SIP calls with the Dial Command. chan_oss.c:291 sound_thread: Failed to write sound chan_oss.c:200 send_sound: Unable to read output space __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new

Hello. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Hi, I have a redhat 8 linux box with a an X100P and a soundblaster sound card. When I start asterisk I get the following message: WARNING[65544]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable And then if I try to call in none of the audio files I have set to play in extensions.conf actually play. When I'm outside of asterisk and try playing a

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

The subject says it all. After digging through latency and other issues with all kinds of linux softphones, I've found that only * works alright for me as a VoIP client. Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab the card once and won't release it until shutdown, while other clients are friendly enough to grab the card only on calls. So, before getting lost in

hi guys... im a new asterisk user. i dont understand why asterisk doesnt start. pls take a look at the output below. i could start it a couple of times .. #asterisk -gc Asterisk CVS-v1-0-01/21/05-14:58:31, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================= [ Booting....................Jan

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable This is my sound card: Multimedia audio controller: Fortemedia, Inc Xwave QS3000A I am not sure... what I am doing wrong. Please help. Thanks,

Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235 ]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 Now I installed the bristuff+asterisk 1.0.9 and always have in my logs

I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on

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\"

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, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Hello-- How do I twiddle the sound drivers? I'm not that experienced with kernel twiddling and driver loading. I have Redhat 9. My previous attempts to play with the kernel and load extra drivers always ended with a new kernel that wouldn't boot. I know asterisk doesn't like to play with sound interfaces that aren't full duplex. So, when I built the system, I saw the MSI