similar to: Soundcard problems?

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

Hi, I'm a student working on an implementation of asterisk en linux (red hat 9.0). I'm trying to register a sip softphone (XLite) en asterisk en local. I have 2 problems : - When I execute asterisk there is a warning and a noise in the headphones : May 16 22:12:22 WARNING[3817]: chan_oss.c:269 sound_thread: Read error on sound device: Resource temporarily unavailable Does it come from

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

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

I am trying to get a console/dsp application going with 1.4.18 and not hearing any audio. In the CLI I see the call coming in, I see the Dial(Console/dsp) I see <auto answered> I see ALSA default but I hear no audio. What can I do to tell what is happening here. I have in modules.conf: noload chan_oss.so load chan_alsa.so For kicks I tried it the other way to noload chan_alsa.so and load

Hi My motto is to connect two computers on the same network with Voip without using any special hardware,i have downloaded Asterisk, I was suggested to use LinPhone as a soft phone as it is very easy to install I have installed Asterisk on my computer and iam using it as a server. And whe i DAIL 1234 at CLI i get the following errors repeatedly Apr 21 17:29:13 WARNING[1167272128]:

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

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 have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Hi, I'm a new asterisk user developing an AGI application. As part of my application I'd like to play sounds on the server's speakers, but it seems that I can't do this while asterisk is running. When I try to play sounds using the play or aplay command, it blocks until I stop asterisk. My guess is that asterisk is using the sound device and this means that other programs

Hi, First at alll, I beg your pardon because maybe I explained bad my questions (because my low level english) I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC has an audio card ac97 chipset intel i810 in its motherboard. I want to use asterisk in this way: |PC linux | --ethernet--

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

I have a Digium TDM400 in my system and I'm using my main system as my asterisk box at home (very light load). When I start up *, though, it grabs my sound card and I cannot play other music through it (e.g. x/ XMMS). I have moved the alsa.conf and oss.conf files so that there is no configuration for them (though those files seemed to do little), but still the sound card is grabbed. How can

Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_zap.so] => (Zapata Telephony w/PRI) Jul 28 15:05:49 WARNING[8249]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Jul 28 15:05:49 ERROR[8249]: chan_zap.c:6460 mkintf: Unable to open

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

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

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

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Zitat von Tzafrir Cohen <tzafrir.cohen at xorcom.com>: Hi, > Off-topic: any reason you don't use chan_alsa? This was the "Armbian installation", I didn't configured it extra... > Are you sure you quote the error message right? Copy+Paste... ;) But I searched a little bit and I really don't think, I need this module... As I undestand, I just need it, if I

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA