[Asterisk-Users] Problems with asterisk and gnophone on Gentoo box

Kevin asterisk at gnosys.biz
Thu Mar 18 15:11:48 MST 2004


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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