search for: i810_audio

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

...o, version 1.01, 02:25:38 Dec 24 2004 Jun 19 08:41:34 b2 kernel: PCI: Found IRQ 10 for device 00:1f.5 Jun 19 08:41:34 b2 kernel: PCI: Sharing IRQ 10 with 00:1f.3 Jun 19 08:41:34 b2 kernel: i810: Intel ICH2 found at IO 0xdc40 and 0xd800, MEM 0x0000 and 0x0000, IRQ 10 Jun 19 08:41:34 b2 kernel: i810_audio: Audio Controller supports 6 channels. Jun 19 08:41:34 b2 kernel: i810_audio: Defaulting to base 2 channel mode. Jun 19 08:41:34 b2 kernel: i810_audio: Resetting connection 0 Jun 19 08:41:34 b2 kernel: ac97_codec: AC97 Audio codec, id: ADS96 (Analog Devices AD1885) Jun 19 08:41:34 b2 kernel:...

i installed CentOS (and WBEL too) on a notebook gateway m520 and all work fine, but when i try to use the sound the notebook freeze. the soundcard is an intel i810_audio. somebody can help me, pls? PD: my english is no good.... :)

...bspeex.so.1: cannot open shared object file: No such file or directory Jan 21 19:22:29 WARNING[4042]: loader.c:440 load_modules: Loading module codec_speex.so failed! why do u guys think this is happening... this is the output of lsmod lsmod Module Size Used by Not tainted i810_audio 27720 0 (autoclean) ac97_codec 13640 0 (autoclean) [i810_audio] soundcore 6404 2 (autoclean) [i810_audio] radeon 116132 1 agpgart 47776 3 ppp_synctty 7904 1 (autoclean) ppp_generic 24444 3...

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing happens when i use sjphone to connect direct...

...e [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) My sound card information: Vendor : Intel Corp. Model : 82801CA/CAM AC'97 Audio Controller Module : i810_audio After running 'dial' command under the asterisk prompt, I got the following message without any sound. *CLI> -- Executing Wait("OSS/dsp", "1") in new stack -- Executing Answer("OSS/dsp", "") in new stack << Console call has been an...

...or something along those lines. The strange thing is that it has been playing the sound fine for about 5 seconds before this. It always fails at exactly the same point. Can someone point me to a fix? Or is this something unrepairable with Humoungous Games? I us Slackware 9.2,X v4.4.x, and the i810_audio mod. Thanks, Creighton

...web (make config - make install - insmod...). When i check loaded modules: # lsmod Module Size Used by Not tainted capi 20864 0 capifs 5424 1 [capi] kernelcapi 34048 1 [capi] capiutil 24864 0 [kernelcapi] i810_audio 25244 1 (autoclean) ac97_codec 17236 0 (autoclean) [i810_audio] e100 55496 1 hisax_fcpcipnp 7968 0 hisax_isac 7800 0 [hisax_fcpcipnp] hisax 131820 0 [hisax_fcpcipnp hisax_isac] isdn 1397...

I got wine working! This is awesome. Now all I need is sound. I use wine-20010216 and sound is fine under linux (alsa driver). Anyway here is the error I am getting: err:wave:DSDB_MapPrimary (0x40360894): Could not map sound device for direct access (errno=5) Any suggestion? Aaron

Hi, I've been try to get sound in wine (20040121) working on my machine for while now. As you've probably guessed I've got a SiS 7012 (which uses the intel i8x0 modules) and alsa (either natively or OSS emulation.) google came up with a few hits about this not working in the past, namely: http://www.winehq.org/hypermail/wine-devel/2002/12/0281.html But I'm not sure if this is

Hi Everybody, I''m a using xen for my first time in a debian sarge using the testing repository of Xen. I have the following problem: When boot with kernel 2.6.11-ksxen0, after a day or some running, the X crash and then the machine reboots. Will it''s running i''m seeing this erros on dmesg: [drm:i830_wait_ring] *ERROR* space: 131056 wanted 131064 [drm:i830_wait_ring]

the problem below keeps recarrying even after i have cleared this error when i run asterisk -vvv or -c the error occurs again please help ..Warning, flexible rate not heavily tested! .................WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File

...g : no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid bogomips : 4784.12 [7.3.] Module information (from /proc/modules): ipv6 273856 25 - Live 0xf8d4e000 i810_audio 35476 0 - Live 0xf8cc9000 ac97_codec 19340 1 i810_audio, Live 0xf8cc3000 ide_scsi 15716 0 - Live 0xf8bf1000 parport_pc 37228 1 - Live 0xf8cf4000 lp 11844 0 - Live 0xf8ca9000 parport 45608 2 parport_pc,lp, Live 0xf8ce7000 ext2 75528 3 - Live 0xf8cd3000 dm_mod 42304 8 - Live 0xf8c66000 raid5 23392 1...

...comes in via the ISDN and somebody leaves a voicemail, the sound file recorded is very choppy. If I actually take the call, the sound is not choppy so it's obviously something to do with the Asterisk box itself having to do the recording. Perhaps the sound card drivers? I'm using the stock i810_audio (OSS) drivers on Fedora Core 1. If I call from a local VoIP client to the Asterisk box and leave a voicemail, the sound is NOT choppy. No errors on the Asterisk console. Thanks in advance. Regards, Gonzalo

...is a nameserver) 80 httpd (this is a web server) 616 unknown 617 unknown 845 unknown 865 unknown 901 samba-swat 1025 listen lsmod binfmt_misc sg ipchains loop cs4243 vfat ad1848 nls_iso8859-1 uart401 nls_cp437 sound msdos fat sr_mod i810_audio ac97_codec nfs ide_scsi smbclient -L server (with return when requested for passwd - running as root - root does not have smbpassword Anonymous login successful Anonymous login successful Domain [Chemistry] OS=[Unix] .. Domain [Geology] OS=[Unix].. tree connect failed: N...

I'm trying to set up the console as an extension (so I can set up overhead paging), but I can't seem to get it to work. When I call my paging extension, I get an error that it can't open the device: -- Executing Ringing("Zap/9-1", "") in new stack -- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack << Call

Hi, I am trying to stream two different sources with two soundcards using liveice and icecast, all from single Linux box. Can anyone tell me if this is possible? I tried creating two sets in the liveice config file where I specified different devices. No erros but the latter source was streamed on both mount points. Running two instances of liveice talking to one instance of icecast failed.

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin