[Asterisk-Users] Does * machine need a sound board for MOH?

[Rich Adamson]

Does anyone know whether the linux machine running * have to have a 
sound card on it in order for musiconhold to work for sip phones?

I've tried about everything (including tons of google searching) to get
it to work, and nothing.

When a call is placed on hold between two C7960's, the CLI indicates:
    -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack
    -- Called 3000
    -- SIP/3000-56a0 is ringing
    -- SIP/3000-56a0 answered SIP/3002-c418
    -- Attempting native bridge of SIP/3002-c418 and SIP/3000-56a0
    -- Started music on hold, class 'default', on SIP/3002-c418
    -- Stopped music on hold on SIP/3002-c418
  == Spawn extension (from-sip, 3000, 1) exited non-zero on 'SIP/3002-c418'

If I do a "ps ax" to see what's running, I see:
 16243 pts/1    S      0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 For-You.

If I go to the * console and listen to "For-You", it plays through the speakers
just fine.

However, after starting * with -vvvvvc, I see:

 [chan_oss.so] => (OSS Console Channel Driver)
WARNING[1074504864]: File chan_oss.c, Line 974 (load_module): XXX I don't work r
ight with non-full duplex sound cards XXX
  == Registered channel type 'Console' (OSS Console Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
WARNING[1142127920]: File chan_oss.c, Line 232 (sound_thread): Read error on sou
nd device: Resource temporarily unavailable
<snip>
 [app_mp3.so] => (Silly MP3 Application)
  == Registered application 'MP3Player'
<snip>


I can't seem to find any other reason for MOH to not work. If someone has
MOH working, would it be possible for you to send me a "sip debug" listing
of what's going on so I can compare it to mine? (offlist to radamson at routers.com)

If anyone can think of anything more I can try it would be greatly appreciated.

This system does have two X100P cards, everything else is working fine, its
just the MOH that won't work.

Thanks
Rich