asterisk users - Jan 2008 - Attended transfers manager or phone

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

Some phones have the auto-answer ability.  So your phone could have two 
extensions, one for normal use and one for auto-answer use.  Redirect or 
Originate, as you were, to the auto-answer extension on the phone.  So 
the phone would already put itself offhook, and asterisk would continue 
and build up the other end of the bridge.

Polycom soundpoint phones, for example, but many others have this ability.

an example extension setup might be

exten => 110,1,Dial(SIP/110)

exten => #110,1,SipAddHeader(.......whatever your phone needs to make it 
autoanswer)
exten => #110,2,Dial(SIP/110)

Don't know about phones that allow ip control of their state, though.

Moj

Christian Ejlertsen wrote:
> 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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