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andre.herrlich at gmx.de
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PostPosted: Tue Jan 22, 2008 7:41 am    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

Hello,

any one advise a good, strong and free softphone that can work with SIP
or/and IAX lines and supports attended transfer ?

Thanks for help.

Mit freundlichen Gr??en / best regards

Andr? Herrlich
IT-Operator / Developer

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02625 Bautzen
Tel.: + 49 - (0)3591 - 2722 - 1451
Fax: + 49 - (0)3591 - 2722 - 1459
E-Mail: aherrlich at letmerepair.com
Internet: www.letmerepair.com
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philipp.kempgen at amo...
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PostPosted: Tue Jan 22, 2008 12:23 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

Andre Herrlich wrote:

Quote:
any one advise a good, strong and free softphone that can work with SIP
or/and IAX lines and supports attended transfer ?

IMHO there are no good softphones - at least not for
Mac OS X and I think that is true for Linux as well.
They're either not stable or their interface is unusable.

Regards,
Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de

Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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zoachien at securax.org
Guest





PostPosted: Tue Jan 22, 2008 12:55 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

Hello,

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 ?

Greetings,

Joachim
Philipp Kempgen wrote:
Quote:
Andre Herrlich wrote:


Quote:
any one advise a good, strong and free softphone that can work with SIP
or/and IAX lines and supports attended transfer ?


IMHO there are no good softphones - at least not for
Mac OS X and I think that is true for Linux as well.
They're either not stable or their interface is unusable.

Regards,
Philipp Kempgen

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andres at paglayan.com
Guest





PostPosted: Tue Jan 22, 2008 11:46 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

On Jan 22, 2008, at 10:23 AM, Philipp Kempgen wrote:

Quote:
Andre Herrlich wrote:

Quote:
any one advise a good, strong and free softphone that can work
with SIP
or/and IAX lines and supports attended transfer ?

IMHO there are no good softphones - at least not for
Mac OS X and I think that is true for Linux as well.
They're either not stable or their interface is unusable.

I've been happy with both, zoiper and x-lite at a osx,
(...may be I have a low threshold for hapiness?)

Quote:

Regards,
Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de

Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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rentorbuy at yahoo.com
Guest





PostPosted: Wed Jan 23, 2008 2:27 am    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

--- zoa <zoachien at securax.org> wrote:

Quote:

Hello,

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 ?

Attended transfer is not available in the free version
and that's what the OP is looking for.

I've been using SJPhone (v. 1.60.289a , not the newer
one) and it has been working just fine (supports
attended transfers). It's free software but it's
development seems to have come to a stop.

I was hoping to move to Ekiga but haven't had time to
test it yet.

____________________________________________________________________________________
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zoachien at securax.org
Guest





PostPosted: Wed Jan 23, 2008 3:35 am    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

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:
Quote:

zoa wrote:
Quote:
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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simon at simonelliston...
Guest





PostPosted: Wed Jan 23, 2008 7:55 am    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

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:

Quote:

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:
Quote:

zoa wrote:
Quote:
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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gordon+asterisk at dro...
Guest





PostPosted: Wed Jan 23, 2008 12:49 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

On Wed, 23 Jan 2008, Zoa wrote:

Quote:

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.

I liked Zoiper when it was idefisk however I'm very irritated that they
changed the account limit to 2 in Zoiper after it was seemingly unlimited
in idefisk, so guess what I stick with...

Gordon
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chr.ejlertsen at has.dk
Guest





PostPosted: Wed Jan 23, 2008 2:07 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

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 Smile, but I hope this could the first really good sip phone with
switchboard properties out there.

Regards
Christian Ejlertsen



Quote:
-----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:

Quote:

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:
Quote:

zoa wrote:
Quote:
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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zoachien at securax.org
Guest





PostPosted: Wed Jan 23, 2008 5:51 pm    Post subject: [asterisk-users] Free IAX / SIP Softphone with attended tran Reply with quote

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:
Quote:
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 Smile, but I hope this could the first really good sip phone with
switchboard properties out there.

Regards
Christian Ejlertsen




Quote:
-----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:


Quote:
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:

Quote:
zoa wrote:

Quote:
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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