[Asterisk-Users] Problems with asterisk and gnophone on Gentoo
box
John Baker
johnb at listbrokers.com
Thu Mar 18 16:30:34 MST 2004
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.
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. --
I'm thinking maybe soundcore is what you're missing, since on mine it's
definitely used.
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]
What motherboard are you using? Again, make sure you've got the right
chip selected for alsa.
John
On Thu, 2004-03-18 at 16:11, Kevin wrote:
> On Thursday 18 March 2004 14:57, Alastair Maw wrote:
> > On 18/03/04 18:46, Kevin wrote:
> > > Thanks for your reply, Alastair. I did use that guide in getting
> > > myself set-up with sound, and do have alsa-oss installed:
> >
> > You need to have it all insmod'ed as well (which I guess it will be):
> >
> > root at glide almaw # lsmod | grep oss
> > snd-seq-oss 29216 0
> > snd-seq-midi-event 3584 0 [snd-seq-oss]
> > snd-seq 37584 2 [snd-seq-oss snd-seq-midi-event]
> > snd-seq-device 4304 0 [snd-rawmidi snd-seq-oss
> > snd-seq]
> > snd-pcm-oss 38436 0
> > snd-pcm 60960 0 [snd-via82xx snd-pcm-oss]
> > snd-mixer-oss 13680 0 [snd-pcm-oss]
> > snd 33636 1 [<<<...snip...>>>]
> >
>
> Yep. I have this, or something very close to it anyway:
>
> bash-2.05b# lsmod | grep oss
> snd-pcm-oss 39140 0 (unused)
> snd-pcm 65828 0 [snd-pcm-oss]
> 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-seq-device 4176 0 [snd-seq-oss snd-seq]
> snd 33892 0 [snd-pcm-oss snd-pcm snd-mixer-oss
> snd-seq-oss snd-seq-midi-event snd-seq snd-timer snd-seq-device]
>
> I wonder if the (unused) messages are telling me something important
> here...
>
> I see that your output does not have them, apparently indicating that
> something is using them.
>
> In addition to the sound apps I have that use alsa, I also use xmms with
> a libOSS.so plugin for accessing the oss system. xmms does work for me
> with this plugin (and doesn't when I use the libALSA.so plugin). Is it
> safe to conclude therefore, that xmms _is_ properly accessing the OSS
> emulation support in the alsa system with this libOSS.so plugin? If
> so, is it safe to conclude that my OSS emulation is working properly?
>
> >
> >
> > Also make sure your dsp device is accessible for the user running
> > OSS:
> >
> > root at glide almaw # ls -l /dev/dsp
> > lr-xr-xr-x 1 root root 9 Mar 9 10:02 /dev/dsp ->
> > sound/dsp
> >
> > root at glide almaw # ls -l /dev/sound/dsp
> > crw-rw---- 1 almaw audio 14,3 Jan 1 1970 /dev/sound/dsp
> >
>
> Yup, have that too. And just to make sure, I'm running asterisk and
> gnophone as root. I've had success running asterisk v1-0_stable from
> CVS as root on other linux distributions like SuSE 9.0, but I think it
> doesn't use alsa---I think it uses straight OSS---not sure though.
>
> >
> >
> > 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
>
> Ah! Though I missed it with my grep in my original reply, I followed up
> with another that showed them being present (about 10 minutes before
> you posted---probably not on the list yet). But your post here shows
> me that I had a syntax problem in my config file.
>
> Whereas I had:
> alias sound-slot-0 snd-card-0
> ^^
>
> I obviously should have had:
> alias snd-slot-0 snd-card-0
>
> That certainly helps (or I think it should anyway).
>
> Unfortunately, after fixing this flaw in the config file and rebooting
> in order to reload all the modules, I still have the same problems
> (with both gnophone and asterisk). I ran modules-update,
> checked /etc/modules.conf for proper carryover of these changed
> settings and rebooted again, but still no joy.
>
> Asterisk startup output:
>
> *****************************************************
> [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 18 16:39:40 WARNING[16384]: chan_oss.c:352 setformat: Requested 8000
> Hz, got 7866 Hz -- sound may be choppy
> Mar 18 16:39:40 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 18 16:39:40 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>
> *CLI>
> *****************************************************
>
> Where's that comment about 8000 Hz vs. 7866 Hz coming from? And I'm
> pretty sure this sound card _does_ support full duplex operation (it
> does in KDE/ARTS anyway), so I don't understand why the asterisk
> comment about duplex.
>
> Does the asterisk config file oss.conf need any special tweaking to run
> with an OSS emulation system under ALSA (as opposed to a real OSS
> system)? I'm reaching here for straws here, I know.
>
> Alastair, you've really been terrific with your specific answers. Thank
> you so much. But is there anything else I'm doing wrong here? If you
> don't see anything then I maybe someone else on the list does. Does
> anyone else on the list have Asterisk working with ALSA emulated OSS on
> a Gentoo box---not that it being Gentoo probably has much or anything
> to do with this, but... If someone else has a similar configuration
> that's working, then maybe I can compare and figure out what the
> problem is. Alastair, are you running asterisk on a Gentoo box with
> ALSA emulated OSS support? What else could it be, I wonder?
>
> I'm running asterisk 0.7.2 from the portage tree. Should I upgrade to
> v1-0_stable from CVS or is that unlikely to be the issue here?
>
> Thanks again for all your help.
>
> -Kevin
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