Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Ok good piece software easy on the eyes as they say and I have to say this before I start listing a lot of things that I would love to see, for it to be usable as a good high performance phone. Working with industrial pc switchboards and soft phones of various vendors for some years now, and it all boils down to. How much functionality you can boil into the keyboard. No mouse action should be needed to search a number add an F-key for it. No mouse action should be needed to dial or transfer a number. No mouse action should be needed unless absolutely unavoidable. A_PARTY = caller B_PARTY = operator / called person C_PARTY = number to transferred to STATES: Example to keep it within the numeric key-pad when you receive a call and transfer it. STEP 1 A call is presented. LINE_STATE: Ringing TRANSFER_STATE: inactive TALKING_TO_STATE: inactive STEP 2 Press numeric enter to pick up call. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY STEP 3 Transfer the call Scenario 1: Search out the number in the phonenbook by pressing ex: F10, while talking to the caller, the phone book appears search by name, number or whatever is available and mark the number with arrow keys and dial with NUM-enter. Scenario 2 Press enter a new dial box appears. Type in the number to call. Press enter. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CALLING_C_PARTY TALKING_TO_STATE: DIALBACKTONE STEP 4 The person transferring the call can now make a choice either to do a attended transfer or a blind transfer. Scenario Blind transfer: Simply pressing NUM-enter should do a blind transfer, and the call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: Attended transfer: The person transferring the call can talk to the C_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Should the operator wish for switching back do the previous call that currently placed on hold it could be done by pressing the NUM+ key placing the C_PARTY on hold and reconnecting the A_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: A_PARTY Switch back by NUM+ LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Connect the call by NUM-enter at any point talking to either the A_PARTY or C_PARTY. The call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: disconnect the party you are talking to Press NUM- If the states are as follows. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY The C_PARTY would be disconnected and the states would go to. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY And the here we go again with a new transfer or a goodbye and hang up with NUM-. Some side notes: The calling transfer functions are already in the phone alle that needs to be done is associate the functions to the states and numeric keys. The features could be activated by putting the phone in operator mode, if this was the case you could turn of the DTMF and just start typing the new number and hit NUM-enter twice to transfer the call fast. 1 enter to dial number the other to transfer. DTMF could be turned of since the operator rarely calls any ivr, that needs a DTMF response, if so you could leave dtmf open on the QWERTY number keys HEX 30 31 33 34 so on. A tcp port on the phone that allowed for picking up calls and hanging up calls, and perhaps being able to read the number status would make is possible for people write some very nice callcenter agent software for this phone, without having to worry about the functionality of a phone in their agent software. These things might be on the table already if so happy days and then I can't wait to see the product then. Shw that was a little longer than expected. Just my way to keep it simple :), but I hope this could the first really good sip phone with switchboard properties out there. Regards Christian Ejlertsen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Elliston Ball Sent: 23. januar 2008 13:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste
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
