[asterisk-dev] New agents does not get noted for waiting callers in the queue

Kristijan Vrban vrban.lkml at googlemail.com
Wed Feb 15 09:11:26 CST 2012


my solution was to run the load_realtime_queue function in a detach thread:
see: https://reviewboard.asterisk.org/r/1744

And it's working as expected for me. No audio dropout anymore when
reload member from sql. Whould this approach be acceptable?

Kristijan


2012/2/14 Kristijan Vrban <vrban.lkml at googlemail.com>:
> many thanks for the enlightenment. even if it is not the answer, which
> I wanted to hear.
>
> Kristijan
>
> 2012/2/13 Kevin P. Fleming <kpfleming at digium.com>:
>> On 02/13/2012 10:33 AM, Kristijan Vrban wrote:
>>>
>>> Sure, the non polling solution would be the preferred. But the polling
>>> solution by adding the "load_realtime_queue" into "is_our_turn"
>>> function would be an
>>> acceptable quick fix solution. But the issues, that the audio get
>>> dropout while doing some functional queue processing (make a sql
>>> loopkup) irritated me.
>>> Why depend audio processing from what the queue do? Because that
>>> means, even if the queue does "normal" things, this can always be the
>>> source of audio dropout.
>>> e.g. add a sleep(1) into "is_our_turn" function, then you have 1sec
>>> audio dropout.
>>>
>>> Without know the hole asterisk structural, but shouldn't be the audio
>>> processing independent from any logical processing?
>>
>>
>> That is not how Asterisk works. The thread servicing a channel is
>> responsible for everything that channel needs; the code in Asterisk modules
>> does not sit 'outside' the channels themselves, just directing their
>> activities. The code you are seeing in app_queue is run *by* the channel
>> thread for that channel, so yes, if its processing takes too long, then
>> there will be audio disruptions.
>>
>> This is why the most popular high-volume queuing applications (Aheeva,
>> ViciDial, others) aren't written that way; they are external applications
>> that control channels in Asterisk via AMI/AGI and similar interfaces.
>>
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at www.digium.com & www.asterisk.org
>>
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