[Asterisk-Users] Problems with asterisk and gnophone on Gentoo
box (long)
John Baker
johnb at listbrokers.com
Fri Mar 19 15:05:05 MST 2004
Kevin -
Guess what, I've got the same chip! It's on my DFI LanParty board. Now
we're getting somewhere.
Try the intel8x0 driver for alsa. Installation instructions are here:
http://www.alsa-project.org/alsa-doc/doc-php/template.php3?module=intel8x0
Also, are you getting sound anywhere else at all besides Asterisk?
Sorry if you said so already, it's kind of unclear.
The reason I ask is that if you aren't getting any sound and you think
everything's working right (cat /proc/interrupts shows the chip
receiving and responding to interrupts, etc) then check your
motherboard.
I had removed a couple of jumpers in order to use the audio out header
off the motherboard for a jack in ther front. I changed my mind on it
and didn't put the jumpers back - they were originally side to side, not
across, so I figured it didn't matter. It did and it stopped my sound
out the back jack.
When I put the jumpers in the right place, voila! Audio out.
John
On Fri, 2004-03-19 at 08:49, Kevin wrote:
> Thanks for your reply, John.
>
> On Thursday 18 March 2004 18:30, John Baker wrote:
> > What sound chip are you using? I thought I had the via82xx and spent
> > a couple days jacking with it before I figured out I was wrong.
>
> I suppose that could be my problem, but I'm pretty sure it's not:
>
> On Thursday 18 March 2004 10:40, Kevin wrote:
> > My hardware and driver setup is as follows:
> >
> > ================================================
> > => M/B: ASUS A7N8X with onboard nForce2 audio system
> >
> > => bash-2.05b# lspci|grep audio
> > 00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97
> > Audio Controler (MCP) (rev a1)
> >
> > => bash-2.05b# cat /proc/pci
> > Bus 0, device 6, function 0:
> > Multimedia audio controller: nVidia Corporation nForce2 AC97
> > Audio Controler (MCP) (rev 161).
> > IRQ 21.
> > Master Capable. No bursts. Min Gnt=2.Max Lat=5.
> > I/O at 0xd400 [0xd4ff].
> > I/O at 0xd800 [0xd87f].
> > Non-prefetchable 32 bit memory at 0xe0001000 [0xe0001fff].
> >
> > => bash-2.05b# lsmod|grep audio
> > nvaudio 36180 0
> > ac97_codec 13076 0 [nvaudio]
> > soundcore 4196 4 [snd nvaudio]
> > ================================================
>
> If the nvaudio module from nVidia is not the right module for this sound
> device, does anyone here know which one is?
>
> Or I wonder if the problem lies with that module, since I don't think it
> comes from the alsa development community. Does the underlying sound
> card module (nvaudio in my case) need to have OSS emulation support
> coded into it? Or does that happen at a different level? Maybe nVidia
> just didn't include OSS emulation support in that module---maybe they
> just planned on having it used with alsa and not oss? I'm reaching
> here...
>
> >
> > Here's my alsa setup in modules.conf:
> >
> > # --- ALSACONF verion 1.0.0pre1 ---
> > alias char-major-116 snd
> > alias char-major-14 soundcore
> > alias char-major-15 off
> > alias sound-service-0-0 snd-mixer-oss
> > alias sound-service-0-1 snd-seq-oss
> > alias sound-service-0-3 snd-pcm-oss
> > alias sound-service-0-8 snd-seq-oss
> > alias sound-service-0-12 snd-pcm-oss
>
> > alias snd-card-0 snd-intel8x0
> > alias sound-slot-0 snd-intel8x0
>
> > # --- END: Generated by ALSACONF, do not edit. --
>
> Here's mine (pretty close):
>
> ==========================================
> bash-2.05b$ cat /etc/modules.d/alsa|grep -v ^\#
>
> alias char-major-116 snd
> alias char-major-14 soundcore
>
> alias snd-card-0 nvaudio # testing
> alias snd-slot-0 snd-card-0 # testing
>
> alias sound-service-0-0 snd-mixer-oss
> alias sound-service-0-1 snd-seq-oss
> alias sound-service-0-3 snd-pcm-oss
> alias sound-service-0-8 snd-seq-oss
> alias sound-service-0-12 snd-pcm-oss
>
> alias /dev/mixer snd-mixer-oss
> alias /dev/dsp snd-pcm-oss
> alias /dev/midi snd-seq-oss
>
> options snd cards_limit=1
> ==========================================
>
> I note with particular interest your lines:
> > alias snd-card-0 snd-intel8x0
> > alias sound-slot-0 snd-intel8x0
>
> ...whereas in mine, I have:
> alias snd-card-0 nvaudio
> alias snd-slot-0 snd-card-0
>
> Do you know which is correct?
> (sound-slot-0 or snd-slot-0)
> ^^^^^ ^^^
>
> I checked the alsa docs and a sample configuration file and it seems
> that sound-slot-0 is correct, not snd-slot-0. I got snd-slot-0 from
> Alastair:
>
> On Thursday 18 March 2004 14:57, Alastair Maw wrote:
> > But I suspect that your real problem is that in addition to the lines
> > you specified in modules.d/alsa, you must have the following:
> >
> > alias snd-card-0 snd-via82xx <-- replace with your ALSA driver
> > alias snd-slot-0 snd-card-0 <-- required for OSS support under
>
> The problem is though, I can't get functional sound with asterisk using
> _either_ configuration! After re-reading the alsa docs on this, I
> tried changing it back to what I had previously (sound-slot-0) and I
> still get the same error from asterisk (see below).
>
> >
> > I'm thinking maybe soundcore is what you're missing, since on mine
> > it's definitely used.
>
> Well, as you can see above, I do have soundcore loaded in the kernel and
> it is being used by snd and nvaudio. Although I'm not sure, I'm
> guessing I wouldn't have any sound at all without it.
>
> >
> > As proof, here's the pertinent readoff from lsmod:
> >
> > snd-mixer-oss 13456 0 (autoclean) [snd-pcm-oss]
> > snd-intel8x0 20612 1
> > snd-ac97-codec 50176 0 [snd-intel8x0]
> > snd-pcm 78464 0 [snd-pcm-oss snd-intel8x0]
> > snd-page-alloc 8876 0 [snd-intel8x0 snd-pcm]
> > snd-timer 19204 0 [snd-pcm]
> > snd-mpu401-uart 4856 0 [snd-intel8x0]
> > snd-rawmidi 17728 0 [snd-mpu401-uart]
> > snd-seq-device 5644 0 [snd-rawmidi]
> > snd 42468 0 [snd-pcm-oss snd-mixer-oss
> > snd-intel8x0 snd-ac97-codec snd-pcm snd-timer snd-mpu401-uart
> > snd-rawmidi snd-seq-device]
> > soundcore 6244 4 [snd]
> >
>
> Well, I didn't have some of these modules loaded. Not sure why they
> weren't auto-loaded... should they have been? Or are they associated
> with midi stuff---which I haven't used and maybe that's why they
> weren't auto-loaded?
>
> After reading this, I used modprobe to manually load snd-ac97-codec
> (though I did already have the module, "ac97_codec" loaded into the
> kernel---what is that? old module?). In my filesystem, I have the
> following ac97 modules:
> /lib/modules/2.4.22-gentoo-r7/kernel/drivers/sound/ac97_codec.o
> /lib/modules/2.4.22-gentoo-r7/kernel/sound/pci/ac97/snd-ac97-codec.o
>
> Right now, both are loaded:
> bash-2.05b# lsmod|grep ac97
> snd-ac97-codec 48428 0 (unused)
> snd 33892 0 [snd-mpu401-uart snd-rawmidi
> snd-ac97-codec snd-pcm-oss snd-pcm snd-mixer-oss snd-seq-oss
> snd-seq-midi-event snd-seq snd-timer snd-seq-device]
> ac97_codec 13076 0 [nvaudio]
> bash-2.05b#
>
> I then loaded snd-mpu401-uart (which seems to have required and thus
> autoloaded snd-rawmidi).
>
> Now I have:
> bash-2.05b# lsmod|grep snd
> snd-mpu401-uart 3904 0 (unused)
> snd-rawmidi 14688 0 [snd-mpu401-uart]
> snd-ac97-codec 48428 0 (unused)
> snd-pcm-oss 39140 0 (unused)
> snd-pcm 65828 0 [snd-pcm-oss]
> snd-page-alloc 6452 0 [snd-pcm]
> snd-mixer-oss 13392 0 [snd-pcm-oss]
> snd-seq-oss 27456 0 (unused)
> snd-seq-midi-event 3840 0 [snd-seq-oss]
> snd-seq 40528 2 [snd-seq-oss snd-seq-midi-event]
> snd-timer 15556 0 [snd-pcm snd-seq]
> snd-seq-device 4176 0 [snd-rawmidi snd-seq-oss snd-seq]
> snd 33892 0 [snd-mpu401-uart snd-rawmidi
> snd-ac97-codec snd-pcm-oss snd-pcm snd-mixer-oss snd-seq-oss
> snd-seq-midi-event snd-seq snd-timer snd-seq-device]
> soundcore 4196 4 [snd nvaudio]
> bash-2.05b# lsmod|grep audio
> nvaudio 36180 0
> ac97_codec 13076 0 [nvaudio]
> soundcore 4196 4 [snd nvaudio]
> bash-2.05b#
>
> I don't think these are required by asterisk, though, and I still have
> the same error from asterisk as before (see below).
>
> > What motherboard are you using? Again, make sure you've got the
> > right chip selected for alsa.
>
> M/B: ASUS A7N8X with onboard nForce2 audio system
>
> Again, thanks very kindly for your reply, John, but nothing seems to be
> working here. Any other ideas on this folks?
>
> Could this have something to do with the fact that I'm using the devfs
> filesystem on this Gentoo box?
>
> bash-2.05b# mount
> /dev/hdb4 on / type reiserfs (rw)
> none on /dev type devfs (rw)
> none on /proc type proc (rw)
> none on /dev/shm type tmpfs (rw)
>
> bash-2.05b# ls -l /dev/dsp
> lr-xr-xr-x 1 root root 9 Mar 19 03:38 /dev/dsp ->
> sound/dsp
> bash-2.05b# ls -l /dev/sound/
> total 0
> crw------- 1 adam audio 14, 3 Dec 31 1969 dsp
> crw------- 1 adam audio 14, 0 Dec 31 1969 mixer
> crw------- 1 adam audio 14, 1 Dec 31 1969 sequencer
> crw------- 1 adam audio 14, 8 Dec 31 1969 sequencer2
>
> On my SuSE 9.0 box (where I am getting asterisk to run and make
> sound---though it's not getting input from the microphone which is
> another problem...), I'm not using the devfs filesystem.
>
> I'll close with a summary of the error messages I'm getting from
> gnophone and asterisk.
>
> For anyone who's picking up this thread recently, I'll reiterate that I
> _do_ seem to have every other aspect of sound working on this box.
>
> gnophone output:
> ===================================================
> bash-2.05b# gnophone
> Card /dev/dsp is no good because Device does not support mono PCM data
> Loaded and activated '/usr/lib/gnophone/modules/audio-oss.so'
> Loaded and activated '/usr/lib/gnophone/modules/audio-phone.so'
> iax.c line 654 in iax_init: Started on port 5036
> Listening on port 5036
> Initialized phone core
> No audio devices found
> bash-2.05b#
> ===================================================
>
>
> Asterisk output:
> ===================================================
> [format_vox.so] => (Dialogic VOX (ADPCM) File Format)
> == Registered file format vox, extension(s) vox
> [app_waitforring.so] => (Waits until first ring after time)
> == Registered application 'WaitForRing'
> [app_setcidnum.so] => (Set CallerID Number)
> == Registered application 'SetCIDNum'
> [chan_oss.so] => (OSS Console Channel Driver)
> Mar 19 08:40:41 WARNING[16384]: chan_oss.c:352 setformat: Requested 8000
> Hz, got 7866 Hz -- sound may be choppy
> Mar 19 08:40:41 WARNING[16384]: 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 '/etc/asterisk/oss.conf': Found
> Mar 19 08:40:41 WARNING[229391]: chan_oss.c:238 sound_thread: Read error
> on sound device: Resource temporarily unavailable
> [app_db.so] => (Database access functions for Asterisk extension logic)
> == Registered application 'DBget'
> == Registered application 'DBput'
> == Registered application 'DBdel'
> == Registered application 'DBdeltree'
> [chan_sip.so] => (Session Initiation Protocol (SIP))
> == Parsing '/etc/asterisk/sip.conf': Found
> == SIP Listening on 0.0.0.0:5060
> == Using TOS bits 0
> == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
> == Registered application 'SIPDtmfMode'
> == Parsing '/etc/asterisk/enum.conf': Found
> Asterisk Ready.
> *CLI> dial
> *CLI> -- Executing Wait("OSS/dsp", "1") in new stack
> -- Executing Answer("OSS/dsp", "") in new stack
> << Console call has been answered >>
> -- Executing DigitTimeout("OSS/dsp", "5") in new stack
> -- Set Digit Timeout to 5
> -- Executing ResponseTimeout("OSS/dsp", "10") in new stack
> -- Set Response Timeout to 10
> -- Executing BackGround("OSS/dsp", "demo-congrats") in new stack
> Mar 19 08:41:15 WARNING[262161]: chan_oss.c:408 soundcard_setinput:
> Unable to re-open DSP device: Device or resource busy
> Mar 19 08:41:15 WARNING[262161]: chan_oss.c:567 oss_write: Unable to set
> device to input mode
> Mar 19 08:41:15 WARNING[262161]: file.c:521 ast_readaudio_callback:
> Failed to write frame
> -- Playing 'demo-congrats' (language 'en')
> == Spawn extension (local, s, 5) exited non-zero on 'OSS/dsp'
> << Hangup on console >>
>
> *CLI>
> ===================================================
>
> Many thanks to John and Alastair for replies, and I would be most
> grateful for any other replies.
>
> TIA.
>
> -Kevin
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