Thank you very much for the feedback, i definately like the suggestions and i will do my best to get this on the roadmap. (which should be pretty easy as i actually kind of make the roadmap :p), so expect in done in one of the following releases. The things to turn it into a callcenter application are already there, not with a TCP port, but you could use it with command line options (even if the phone is already running) or through a com object. Documentation can be found here: http://www.zoiper.com/downloads/Zoiper_API_Documentation.pdf Examples can be found here : http://www.zoiper.com/biz3.php
I have an example for jscript somewhere tool, contact me offlist if you want it. Let me know offlist if you need any biz licenses to try it out, i;d be happy to provide you with them. Zoa. Christian Ejlertsen wrote: > 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. > > Sheeeew 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. >> >> It would be nice to have a keyboard hang-up, something like ESC, ditto >> for things like cancel buttons around the app. >> >> I really like the fact it does both SIP and IAX. >> >> Onto sillier issues: the icon is nice, but it would be great to have >> proper gamma anti-aliasing on the mac one. >> >> >> Just my .02 on the free mac os version, I might have to check out the >> biz edition too. It's all looking good. Good luck with the next release! >> >> Simon >> >> Simon Elliston Ball >> [EMAIL PROTECTED] >> >> >> >> On 23 Jan 2008, at 08:35, Zoa wrote: >> >> >>> You can find it here: >>> >>> http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz >>> >>> Note that the linux version does not support TLS and SRTP yet. >>> >>> * Instructions: * >>> >>> 1) Download zoiper201-linux.tar.gz >>> 2) Extract Zoiper. If you don't use a GUI application for archive >>> processing, here is the command line: >>> >>> tar zxf zoiper201-linux.tar.gz >>> ./zoiper >>> >>> 3) Start Zoiper. >>> >>> *ZoIPer depends on ALSA library, so it* **must** *be installed! >>> >>> * >>> >>> Zoa >>> >>> Robert Moskowitz wrote: >>> >>>> zoa wrote: >>>> >>>>> Have you tried our Zoiper softphone yet (www.zoiper.com) - new >>>>> version scheduled for in a couple of days ? If so, can you send me >>>>> any remarks of list so that we can keep those things in mind for >>>>> future versions ? >>>>> >>>> Do you know where I can get it as an rpm to install on Centos 5 with >>>> Gnome? >>>> >>>> I do not have the time resources to do compiles. >>>> >>>> I am really a security protocol researcher and would be very >>>> interested in seeing what you have done for SIP TLS and SRTP. But for >>>> the later, I am all Linux. The one XP system is a corp box that I >>>> cannot add any software too. >>>> >>> >>> _______________________________________________ >>> -- 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 > _______________________________________________ -- 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
