[asterisk-users] Free IAX / SIP Softphone with attended transfer

Wed Jan 23 13:07:30 CST 2008

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] 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
> simon at simonellistonball.com
> 
> 
> 
> 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.
> >
> >
> >
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