Nope.. sure doesn't.. You call the AgentLoginCallback extension from any phone.. Enter you agent ID.. and Password... then enter the extension your calls should go to and its done.
bkw On Wed, 13 Aug 2003, Devon Henderson wrote: > Being a relative Asterisk newbie, I may be wrong.. but as far as I can tell, > it doesn't. The standard queue/agent logic requires that you assign an > extension to a phone. > > Someone correct me if I'm wrong, please. :) > > - Devon > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Brian West > Sent: Wednesday, August 13, 2003 10:10 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Extension and phone management best > practices?? > > > AgentLoginCallback does this doesn't it? > > On Wed, 13 Aug 2003, Steven Critchfield wrote: > > > On Wed, 2003-08-13 at 11:28, Devon Henderson wrote: > > > We're still in the planning stages of our Asterisk implementation, but > we > > > have a requirement that the extension be mapped to a user, with the > phone as > > > a variable (we have hot seats in our contact center, and we also have > agents > > > that work both from remote locations and our contact center). > > > > > > So, I am also very interested to see what everyone has to say about > this. > > > > All of this can be done in a very simple dbapp or agi app. BAsically all > > you need is a way to identify the user and the channel they are on. Then > > you just consult your data store for the mapping. > > > > So your basic app should need a login script, a logout script, and a > > translation extension logic. > > > > login script should look to see if another user had logged in at that > > phone and disable that login. logout script can make the mapping invalid > > so dial will jump to voicmail. Your extension logic would just need to > > be a pattern match on your extensions, and a lookup of the extension in > > the data store, retrieving the current mapping and attempt to dial the > > phone listed. Might even want to include the technology in your mapping > > as it will let you go SIP, H323, IAX, and Zap channels. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users