[asterisk-users] confbridge

[Tony Mountifield]

In article <5034D4B7.8000906 at pagestation.com>,
Jerry Geis <geisj at pagestation.com> wrote:
> 
> So I commented in the "full" in the logger and restarted. set verbose 
> and debug.
> the only thing I saw was below. dsp.c Setup Tone. See below.
> 
> [Aug 22 08:02:31] DEBUG[31329] channel.c: Didn't receive a media frame 
> from 
> Local/app_confbridge_call_out at smvoice-local-public-address-playfile-621a;2 
> within 500 ms of answering. Continuing anyway
> [Aug 22 08:02:31] DEBUG[31329] app_confbridge.c: Trying to find 
> conference bridge 'PA0001'
> [Aug 22 08:02:31] DEBUG[31329] bridging.c: Joining bridge channel 
> 0x7fb07c0032e8 to bridge 0x7fb07801e8f8
> [Aug 22 08:02:31] DEBUG[31329] bridging.c: Added channel 
> Local/app_confbridge_call_out at smvoice-local-public-address-playfile-621a;2(0x7fb07800f4a8) 
> to bridge array on 0x7fb07801e8f8, new count is 2
> [Aug 22 08:02:31] DEBUG[31329] bridging.c: Bridge 0x7fb07801e8f8 is 
> happy that channel 
> Local/app_confbridge_call_out at smvoice-local-public-address-playfile-621a;2 
> already has read format slin
> [Aug 22 08:02:31] DEBUG[31329] bridging.c: Bridge 0x7fb07801e8f8 is 
> happy that channel 
> Local/app_confbridge_call_out at smvoice-local-public-address-playfile-621a;2 
> already has write format slin
> [Aug 22 08:02:31] DEBUG[31329] bridging.c: Giving bridge technology 
> softmix notification that 0x7fb07c0032e8 is joining bridge 0x7fb07801e8f8
> [Aug 22 08:02:31] DEBUG[31329] dsp.c: Setup tone 1100 Hz, 500 ms, 
> block_size=160, hits_required=21
> [Aug 22 08:02:31] DEBUG[31329] dsp.c: Setup tone 2100 Hz, 2600 ms, 
> block_size=160, hits_required=116
> [Aug 22 08:02:31] DEBUG[31340] pbx.c: Launching 'AGI'

The dsp stuff doesn't matter. It is not generating tones, just setting up
a detector to listen for them. It does this any time a dsp is set up using
ast_dsp_new(), and there is probably one in confbridge for talk detection.

> Also I dont want to do any transcoding and its talking about slin 
> format. seems likes thats the
> native format for conference. Do I need to add slin to my formats for 
> the end locations. All I have right now are ulaw,alaw,gsm.

Conference mixing has to be done in slin for the sample addition to work.
You don't need slin as an endpoint format (most endpoints won't understand
it anyway) - Asterisk will efficiently transcode between ulaw and slin
as required.

> Rename the sounds directory (I just tried again) did nothing this time. 
> Not sure what I had
> done??? Anyway from above looks like the dsp.c tone is whats doing it.

No, I think it's something else.

Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org