[Asterisk-Users] Read error om sound device
Tzafrir Cohen
tzafrir at cohens.org.il
Sun Jul 17 05:58:01 MST 2005
On Sun, Jul 17, 2005 at 02:44:27PM +0200, Administrator TOOTAI wrote:
> Tzafrir Cohen a écrit :
>
> >On Sun, Jul 17, 2005 at 10:47:29AM +0200, Administrator TOOTAI wrote:
> >
> >
> >>Hi list,
> >>
> >>I have an asterisk box running on a via C3 motherboaard/Debian Sarge.
> >>Installed version was the Debian packages one 1.0.7-bristuff. I use this
> >>box with the console dial command and it was working fine. Cards info are:
Have you installed the version from Sarge, or built one on your own?
> >>
> >>cat /proc/assound/cards
> >>0 [V8235 ]: VIA 8233 - VIA 8235
> >> VIA 8235 at )xe400, irq 11
> >>
> >>
> >
> >you use alsa, right?
> >
> >
> Yes
>
> >
> >
> >>Now I installed the bristuff+asterisk 1.0.9 and always have in my logs
> >>
> >>[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
> >>Read error on sound device: Resource temporarily unavailable
> >>
> >>
> >
> >Why use OSS, then?
> >
> >
> OSS emulation of alsa. No need to build chan_alsa which needs alsa sources
>
> >
> >
> >>Trying a "dial 3" command in console (config file from scratch) give me
> >>BackGround("OSS/dsp","demo-congrats")
> >>[...]
> >>Timeout on OSS/dsp
> >><<console Hangup>>
> >>
> >>/dev/dsp is a symlink to /dev/dsp0 owned by root/audio and crw-rw----
> >>
> >>
> >
> >asterisk is in the group audio, right?
> >
> > groups asterisk
> >
> >If it is not, this is a bug.
> >
> >
> Yes, asterisk is in group audio.
>
> groups asterisk
> asterisk : users audio
That's strange. isn't asterisk in the group asterisk? is its primary
group 'users'?
>
> Running asterisk this way (under asterisk user)
>
> [chan_oss.so] => (OSS Console Channel Driver)
> Unable to open /dev/dsp: Permission denied
> == No sound card detected -- console channel will be unavailable
> == Turn off OSS support by adding 'no-load=chan_oss.so' in
> /etc/asterisk/modules.conf
>
> Anyone else is using 1.0.9 with dial command in CLI?
No. But http://www.pbxfreeware.com/app_changrab.c has one (originate).
Should probably build just fine with asterisk-dev installed, but let me
know if you need a skeleton deb.
As for me, I never bothered, as I use a script to generate a call file
and thus have basically the same functionality.
--
Tzafrir Cohen | tzafrir at jbr.cohens.org.il | VIM is
http://tzafrir.org.il | | a Mutt's
tzafrir at cohens.org.il | | best
ICQ# 16849755 | | friend
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