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


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