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 placed to 'dsp0' on console >>
<< Auto-answered >>
-- Called dsp0
-- OSS/dsp answered Zap/9-1
Dec 6 10:08:29 WARNING[901141]: chan_oss.c:413 soundcard_setinput: --
Remote UNIX connection disconnected
Unable to re-open DSP device: Device or resource busy
Dec 6 10:08:29 WARNING[901141]: chan_oss.c:572 oss_write: Unable to set device
to input mode
Dec 6 10:08:29 WARNING[901141]: file.c:550 ast_readaudio_callback: Failed to
write frame
-- Playing 'new/whistle' (language 'en')
<< Hangup on console >>
== Spawn extension (cmcm-internal, 6789, 2) exited non-zero on
'Zap/9-1'
-- Hungup 'Zap/9-1'
To make sure it wasn't a permissions issue, I set the dsp0 device to 666:
crw-rw-rw- 1 root audio 14, 3 Oct 14 15:24 /dev/dsp0
The audio device works fine with sox, mpg123, etc. I'm mystified about what
could be wrong with this.
Here's my oss.conf:
[general]
autoanswer=yes
context=paging
extension=s
The paging context in extensions.conf:
[paging]
; Using server sound card
exten => 6789,1,Ringing
exten => 6789,2,Dial(Console/dsp0,18,A(new/whistle))
exten => 6789,3,Hangup
; using paging module
;exten => 6789,1,Dial(Zap/17,18)
;exten => 6789,2,Hangup
System configuration:
Debian Sarge w/ Asterisk 1.0.1-1
T100P card going to Adtran TA750 channel bank with 5 FXS and 1 FXO card
(analog extensions on FXS ports, FXO ports currently unused)
Trunks via IAX2 to Nufone
Any insights into this would be appreciated.
Thanks,
David
--
David Carter ** dcarter@sigfs.org ** dcarter@visi.com
PGP Key 581CBE61: E07EE199C767C752 8A8B1A9F015BF2EA
Key available at www.keyserver.net
On Mon, Dec 06, 2004 at 10:18:48AM -0600, David Carter wrote:> 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 placed to 'dsp0' on console >> > << Auto-answered >> > -- Called dsp0 > -- OSS/dsp answered Zap/9-1 > Dec 6 10:08:29 WARNING[901141]: chan_oss.c:413 soundcard_setinput: -- Remote UNIX connection disconnected > Unable to re-open DSP device: Device or resource busy > Dec 6 10:08:29 WARNING[901141]: chan_oss.c:572 oss_write: Unable to set device to input mode > Dec 6 10:08:29 WARNING[901141]: file.c:550 ast_readaudio_callback: Failed to write frame > -- Playing 'new/whistle' (language 'en') > << Hangup on console >> > == Spawn extension (cmcm-internal, 6789, 2) exited non-zero on 'Zap/9-1' > -- Hungup 'Zap/9-1'At this point, I'm guessing that the problem lies with my sound chip (nForce2 audio) which only supports stereo, not mono. (Thus, the 'Unable to set device to input mode' above.) I'm in the process of compiling the ALSA driver to see if it will do the conversion itself, as the OSS-compatible i810_audio driver will not. (Since AC97-compatible audio devices only do 2-channel 48KHz audio, it doesn't surprise me that this is failing -- I doubt Asterisk does this conversion internally right now.) What tipped me off to this was playing a mono sound file with sox: playing /usr/share/asterisk/sounds/new/whistle.wav sox: Sound card appears to only support 2 channels. Overriding format If the ALSA driver doesn't work, I'll have to find an old, cheap sound card to put in my server, but one which supports mono audio natively. If anyone else has had experience using an AC97 audio chip as the Asterisk console, I'd love to hear about it. Otherwise, I'll have to find another card (and use up a PCI slot) to get this working. -- David Carter ** dcarter@sigfs.org ** dcarter@visi.com PGP Key 581CBE61: E07EE199C767C752 8A8B1A9F015BF2EA Key available at www.keyserver.net