Re: [asterisk-users] Attended transfers manager or phone
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
[asterisk-users] 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 ___ -- 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
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
