On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer <jsnee...@gmail.com> wrote: > On Mon, Mar 9, 2009 at 5:44 PM, nik600 <nik...@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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users