[asterisk-users] Fwd: add a new queue strategy: SBR

[nik600]

On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer <jsneerin at gmail.com> wrote:
> On Mon, Mar 9, 2009 at 5:44 PM, nik600 <nik600 at gmail.com> wrote:
>> Thanks, i've tested and it works (1.4.23.1).
>>
>> Just 2 questions:
>>
>> 1) this approach seems to be an "hack" and not the implementation of a
>> feature is it really used in corporate solutions?
>> 2) using queue show 001 i can't see the ringing status, is that
>> correct (In Use, Not in Use,Paused works now properly)?
>
> I've never really noticed the lack of a "ringing" status. Our queue
> setup has just worked, so I usually only have to use "queue show" when
> there's a problem. I do know that the AMI reports the ringing status.
>
> The Local/n solution has the added problem of not handling attended
> transfers correctly. When using a Local channel with the /n flag, if
> an agent performs an attended or SIP transfer, or does a 3-way call on
> their own phone and then hangs up, Queue() will still consider the
> agent "In Use" until the original transferred call is hung up.
>
>> Maybe polling the device state using the SIP channel would be better,
>> but as you told me this feature is available only on 1.6.x.
>
> It was backported to 1.4.19, but the patch no longer applies cleanly
> to newer versions. There were some locking changes just after that
> version. If you want to give it a try, I found it at:
>
> http://ftp.iq-labs.net/state_interface-1.4/
>
> Then there's this:
>
> http://reviewboard.digium.com/r/116/
>
> The corresponding func_devstate has also been backported, but it's pretty old:
>
> http://svncommunity.digium.com/view/russell/asterisk-1.4/func_devstate-1.4/
>
> I got the 1.4.19 backport to compile against a 1.4.20.1 codebase, but
> Asterisk would core as soon as app_queue.so loaded, so clearly I
> didn't quite get it right. I eventually punted and changed my dynamic
> queues to just use the actual SIP/xxxxx channel names. It's been
> working fine for over a year now.
>

thanks for these explanation, at this point i think that the better
thing is to use the SIP/xxxx channel and do something else on a third
party system to store an "additional information" about the agent
using that phone, it's more stable and clear on asterisk side.

Thanks

-- 
/*************/
nik600
http://www.kumbe.it