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: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] 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 > > > > > > _______________________________________________ > > -- 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 > > > > > _______________________________________________ > -- 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 _______________________________________________ -- 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