[asterisk-users] Attended transfers manager or phone

[Christian Ejlertsen]

Thank you very much, that was a new angle I hadn't thought of time to
investigate a little more :). The joys of learning new things :)

- Christian

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Mojo with Horan & Company, LLC
> Sent: 16. januar 2008 01:06
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Attended transfers manager or phone
> 
> 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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