Paging issues using console/dsp


I’m currently having issues getting paging working with TB2.2.1

I had this working using the beta for 2.2, but for some reason it does not want to work anymore with the production version and I"m not sure what I"m doing wrong.

I’ve enabled the and it is loaded and working.

I’ve created an custom device called 7070 which dials console/dsp

Both the Master volume and PCM channel are set just above mid way on alsamixer.

I’ve verified that the output on the sound card works by using another persons suggestion to record a wave file then use aplay to play it (which works fine, but I can’t seem to add this to the paging group and it records it before playing where I would prefer to just have it output to the Line out on the sound card at the same time it pages the handsets (aastra 480i’s).

This is what I get when I’m watching the debug:

dialparties.agi: priority is 1
dialparties.agi: Caller ID name is ‘Douglas Browning’ number is '218’
dialparties.agi: Methodology of ring is ‘none’

dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ‘’
– dialparties.agi: Added extension 7070 to extension map
– dialparties.agi: Extension 7070 cf is disabled
– dialparties.agi: Extension 7070 do not disturb is disabled
dialparties.agi: extnum: 7070
dialparties.agi: exthascw: 1
dialparties.agi: exthascfb: 0
dialparties.agi: extcfb:
dialparties.agi: exthascfu: 0
dialparties.agi: extcfu:
– dialparties.agi: dbset CALLTRACE/7070 to 218
– AGI Script dialparties.agi completed, returning 0
– Executing Dial(“SIP/218-08f982e0”, “console/dsp||tr”) in new stack
<< Call to ‘dsp’ on console from <(null)><218> >>
<< Auto-answered >>
– Called dsp
– OSS/dsp answered SIP/218-08f982e0

Any suggestions would be greatly appreciated!




I’ve been playing around with this to try to get it working.

I found that if I load alsa as opposed to oss then I can get the console/dsp to work.

But this seems to break the line-in music on hold.

Any suggestions?