On 08/22/2012 08:46 AM, Jerry Geis wrote:
Hi Jerry,
Firstly, in logging.conf, make sure you have a line as follows:
full => notice,warning,error,debug,verbose,dtmf,fax
If you made any changes, then in the asterisk CLI, do: reload logger
Then again in the CLI, do:
set verbose 5
set debug 5
Then try your scenario and look afterwards at /var/log/asterisk/full.
Tony
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@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@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@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@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'
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.
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.
What next?
Jerry
I finally found this - it was not asterisk, it was me. I had in the
dialplan two locations
that brought other asterisk boxes into conf. Its was being called twice.
So first call
into a box worked, then the second call was giving me a busy.
Thanks Tony! For all your help. Meetme must have been "slightly" smarter
to say that
device is already in the meetme so don't do a second call, where
confbridge did not do that.
Jerry
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