similar to: oss Errors

Hi, I hope someone could help me, or at least give me a hint about this error. ERROR => Aug 18 10:48:51 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to display condition 16 on OSS/dsp -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Aug 18 10:48:53 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to

Hi, When I start asterisk with asterisk -vvvvc it works preaty well, but I get the following warnings: Warning, flexibel rate not heavily tested! WARNING[1074494176]: File res_musiconhold.c, Line 462 (moh_register ): Unable to open pseudo channel for timing... Sound may be choppy. WARNING[1074494176]: File res_musiconhold.c, Line 462 (moh_register ): Unable to open pseudo channel for timing...

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

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

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

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 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

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

Has anybody had any problems with their soundcards like this: Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on sound device: Resource temporarily unavailable This shows on the console and I have no clue what it is.. voice prompts sound good.... Any clues? __________________________________________________________________ Anton Krall

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

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try

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, 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

Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

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've been running an asterisk server for several months on my RedHat 9 box. Today I decided to update some of the glibc packages on my box and upgrade it to the latest asterisk cvs as well. I can start the asterisk server without incident. I can place a call in from the outside and it takes the call fine; however as soon as I hangup, the server appears to just crash. I have no idea

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