Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real "simple" answer to.
I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken off-hook and then being able to dial the number without having to answer the dial-back first. 1. One solution, though an ugly one, would be using Originate, but use a phone that has some sort tcp/ip interface that allows for taking the phone off-hook. 2. A Better solution would be using a phone that allows dialling and taking the phone off-hook on-hook etc. via some tcp/ip interface. 3. Yet another solution, though I do not favour this one since I really don't want to maintain the sip phone code, would be programming a soft sip phone with all the bells and whistles and adding the switchboard functionality to that (name searching, status email so on and so forth. In the end all I need is just a software or hardware phone, sip/iax, which can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps status requests. If such a phone exists that would do the trick, the rest is manageable via the Asterisk Manager console. I'm guessing some people have messed with this problem before so I hope that someone has some information about this kind of thing :) Thank you in advance Christian _______________________________________________ -- 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
