[asterisk-users] confbridge

[Jerry Geis]

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 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'
>
>
> 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