[Kde-announce-apps] Twinkle 1.0

[Michel de Boer]

Name: Twinkle
Version: 1.0
Type: Telephony
Depend: Qt 3.x
License: GPL
Homepage: http://www.twinklephone.com
More Info:
http://www.kde-apps.org/content/show.php?content=26926

Description:
 Twinkle is a soft phone for your voice over IP
communcations using the SIP protocol. You can use
it for direct IP phone to IP phone communication
or in a network using a SIP proxy to route your
calls.
Some of the features offered are call waiting,
call hold, 3-way conference call, call transfer,
and call reject. It supports STUN or a statically
configured public IP address for NAT traversal.
When using STUN, it will send keep-alive packets
to keep NAT bindings alive. It supports ZRTP for
secure voice communication.

Changelog:
 22 jan 2007 - 1.0
=================
- Local address book
- Message waiting indication (MWI)
  * Sollicted MWI as specified by RFC 3842
  * Unsollicited MWI as implemented by Asterisk
- Voice mail speed dial
- Call transfer with consultation
  * This is a combination of a consultation call
on the other line
    followed by a blind transfer.
- Attended call transfer
  * This is a combination of a consultation call
on the other line
    followed by a replacement from B to C of the
call on the first line.
    This is only possible if the C-party supports
"replaces".
    If "replaces" is not supported, then twinkle
automatically falls
    back to "transfer with consultation".
- User identity hiding
- Multi language support
  This version contains Dutch and German
translations
- Send BYE when a CANCEL/2XX INVITE glare occurs.
- When call release was not immediate due to
network problems or protocol errors,
  the line would be locked for some time. Now
Twinkle releases a call in the
  background immediately freeing the line for new
calls.
- Escape reserved symbols in a URI by their
hex-notation (%hex).
- Changed key binding for Bye from F7 to ESC
- When a lock file exists at startup, Twinkle asks
if you want to override it
- New command line options: --force, --sip-port,
--rtp-port
- Ring tone and speaker device list now also shows
playback only devices
- Microphone device list now also shows capture
only devices
- Validate audio device settings on startup,
before making a call, before
  answering a call.
- SIP_FROM_USER, SIP_FROM_HOST, SIP_TO_USER,
SIP_TO_HOST variables for call scripts.
- display_msg parameter added to incoming call
script
- User profile options to indicate which codec
preference to follow
- Twinkle now asks permission for an incoming
REFER asynchronously. This
  prevents blocking of the transaction layer.
- Highlight missed calls in call history
- Support for G.726 ATM AAL2 codeword packing
- replaces SIP extension (RFC 3891)
- norefesub SIP extension (RFC 4488)
- SIP parser supports IPv6 addresses in SIP URI's
and Via headers
  (Note: Twinkle does not support transport over
IPv6)
- Support mid-call change of SSRC
- Handling of SIGCHLD, SIGTERM and SIGINT on
platforms implementing
  LinuxThreads instead of NPTL threading (e.g.
sparc)