twinkle 1:1.4.2-2 (i386 binary) in ubuntu karmic

 Soft-phone for making telephone calls using SIP over an IP network.
 .
 Twinkle supports direct IP phone to IP phone communication or a network
 using a SIP proxy to route your calls.
 .
 In addition to making basic voice calls Twinkle provides you the
 following features regardless of the services that your VoIP service
 provider might offer.
 .
  2 call appearances (lines)
  Multiple active call identities
  Custom ring tones
  Call Waiting
  Call Hold
  3-way conference calling
  Mute
  Call redirection on demand
  Call redirection unconditional
  Call redirection when busy
  Call redirection no answer
  Reject call redirection request
  Blind call transfer
  Call transfer with consultation (attended call transfer) (new)
  Reject call transfer request
  Call reject
  Repeat last call
  Do not disturb
  Auto answer
  Message Waiting Indication
  Voice mail speed dial
  User definable scripts triggered on call events
   E.g. to implement selective call reject or distinctive ringing
  RFC 2833 DTMF events
  In-band DTMF
  Out-of-band DTMF (SIP INFO)
  STUN support for NAT traversal
  Send NAT keep alive packets when using STUN
  NAT traversal through static provisioning
  Missed call indication
  History of call detail records for incoming, outgoing, successful and missed
  DNS SRV support
  Automatic fail-over to an alternate server if a server is unavailable
  Other programs can originate a call via Twinkle, e.g. call from address book
  System tray icon
  System tray menu to originate and answer calls while Twinkle stays hidden
  User definable number conversion rules
  Simple address book
  Support for UDP and TCP (new) as transport for SIP
  Presence
  Instant messaging
  Simple file transfer with instant message
  Instant message composition indication
  Command line interface (CLI)
 .
 VoIP security
  Secure voice communication by ZRTP/SRTP
  MD5 digest authentication support for all SIP requests
  AKAv1-MD5 digest authentication support for all SIP requests (new)
  Identity hiding
 .
 Audio codecs
  G.711 A-law (64 kbps payload, 8 kHz sampling rate)
  G.711 u-law (64 kbps payload, 8 kHz sampling rate)
  GSM (13 kbps payload, 8 kHz sampling rate)
  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)
  Speex wide band (28 kbps payload, 16 kHz sampling rate)
  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)
  G.726 (16, 24, 32 or 40 kbps payload, 8 kHz sampling rate)
 .
 For all codecs the following preprocessing options are available to improve
 quality at the far end of a call.
  Automatic gain control (AGC) (new)
  Noise reduction (new)
  Voice activity detection (VAD) (new)
  Acoustic echo control (AEC) [experimental] (new)