SIPTAPI Changelog

v. 0.2.17
- Fix CRLF in log file
- Set Allow header on outgoing INVITE to suppress incoming UPDATEs
- Correctly handle incoming reINVITE (send 200 OK with fake SDP)
- Refuse all other requests (except reINVITE and NOTIFY) with 405

v. 0.2.16
- New option "Immediate BYE". If checked, after REFER, SIPTAPI will not wait 
  for a NOTIFY with sipfrag "200 OK" to send the BYE but instead the BYE will
  be sent immediately after receiving the final response on the REFER.

v. 0.2.15
- New option "Don't send BYE". If checked, SIPTAPI will not send BYE to terminate the outgoing call
  (eg. after timeout receiving NOTIFY or receiving final NOTIFY), except if the user hangs up call via TAPI.
  This feature is only useful, if your SIP proxy is actually not a real SIP proxy but some kind of application
  layer glue, that just uses SIP messages to interact with SIPTAPI.

v. 0.2.14
- better handling of NOTIFYs after REFER: wait with sending BYE until NOTIFY with 
  final response in sipfrag body or until timeout happens
- new auto-answer option for Siemens phones 
  http://wiki.siemens-enterprise.com/wiki/Asterisk_Feature_Example_Configuration#Auto_Answer
  Alert-Info: <http://example.com>;info=alert-autoanswer

v. 0.2.13
- added auto-answer: add header to indicate to phone that it should answer the call
  SNOM: http://asterisk.snom.com/index.php/Asterisk_1.4/Intercom
    Call-Info: <sip:127.0.0.1>;answer-after=0
  Polycom: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
    Alert-Info: Ring Answer

v. 0.2.12
- added authentication user

v. 0.2.11
- added codecs in initial SDP: G722 and G728

v. 0.2.10
- added workaround for alternative configuration with XLite/Bria
- enhanced documentation

v. 0.2.9
- added file which describes a workaround for TAPI problems in Outlook 2007/2010
- added "reverse-mode": normally SIPTAPI calls the user's phone and then refers 
  it to the real target. In reverse mode SIPTAPI will call directly the target, 
  and once the target answers, it gets refered to the user's phone.
  Note: In reverse mode you wont get any audible call progress indication (e.g. ringback, busy tone ...)
- location of SIPTAPI settings changed (should work better with security features of Vista and Windows 7)
  old location: HKEY_LOCAL_MACHINE\SOFTWARE\Software\enum.at\SipTapi
  new location: HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Telephony\SipTapiSF
- Sourceforge code repository migrated from CVS to SVN

v. 0.2.8
- outbound proxy is now always a loose-router, if ;lr parameter is not present it will be added internally
- fix bug: wait maximum 25seconds for NOTIFY, then send BYE anyway
- logfile renamed to c:\siptapi_0.2.log

v. 0.2.7 (not released)
- fix bug: proper handling of 401 response
- new (faster) handling of line close/drop
- successfully tested against Telinta hosted platform (http://www.telinta.com)

v. 0.2.6
- fix bug: works now also with Lotus Organizer
- fix bug: better error handling for SIP errors (still todo: async)

v. 0.2.5
- supports 64bit too

v. 0.2.2 is tested with:
- ser/openser and Cisco 7960 phones
- asterisk CVS_21_08_2005 (canreinvite=no)
