Version 0.2.7
May 25th, 2011
- Improved interoperability with OnSIP service
- Fixed duplicate account detection
- No longer decode display_name as it's unicode now
- Adapted to the latest API changes in middleware
- Fixed string representation of SIP URIs with special characters (SIP Simple)
- Fixed SDP negotiation on bogus answers (SIP Simple)
- Reduced SDP size when streams are disabled (SIP Simple)
Version 0.2.6
March 22nd, 2011
- Fixed exception when NAT type detection is attempted without connectivity (SIP Simple)
- Fixed exceptions when contact URI can't be built for the desired route (SIP Simple)
- Fixed crashes and increased resilience when connectivity is lost (SIP Simple)
- Relax check on SDP origin to increase interoperability (SIP Simple)
Version 0.2.5
February 16th, 2011
- Fixed saving TLS options (SIP Simple)
Version 0.2.4
February 15th, 2011
- Added support for unicode device names
- Added menu entry and dialog for joining a conference
- Restructured main menu
- Improved DNS resolver capabilities (SIP Simple)
- Only handle records in the local. domain for bonjour (SIP Simple)
- Send 500 response if we fail to create incoming invitation (SIP Simple)
- Fixed race conditions in subscription handlers (SIP Simple)
- Fixed exception when the session is ended on error conditions (SIP Simple)
Version 0.2.3
December 14th, 2010
- Detect change of IP address
- Added web server tools activity indicator
- Fixed compatibility with older python-qt
- Made changes to Preferences thread safe (SIP Simple)
- Fixed TLS transport initialization (SIP Simple)
- Added DNS resolver autodetection capabilities (SIP Simple)
- Fixed matching of media codecs on incoming calls (SIP Simple)
Version 0.2.2
November 29th, 2010
- Fixed detection of audio codecs without a rtpmap line in SDP
- Fixed exception for MWI NOTIFY without a Message-Account body
Version 0.2.1
November 26th, 2010
- Allow name and group attributes to be missing when updating a contact
- Handle bonjour neighbour record updates
- Updated debian dependency on python-sipsimple
- Honor the account.sip.always_use_my_proxy setting
- Fixed opening the create account dialog on first run
Version 0.2.0
November 11h, 2010
- First Blink QT official release for MS Windows
- Added the preferences panel
- Enable inband DTMF by default
- Disable ICE by default
- Simplified MWI code and improved its user interface
- Improve handling of Google contacts
- Open the dialog for adding the initial account after the main window
- Switch to new plugged-in device automatically if we have active calls
- Added transparency for contact icons
- Added conference contact on first start
- Many bug fixes in the middleware
- Adapted to the latest changes in SIP SIMPLE client SDK
Version 0.1.4
September 6th, 2010
- Save preferred media when creating a contact
- Fixed broken dependency to python-aplication for non-Debian systems
- Display 'no new messages' text before getting MWI NOTIFY
Version 0.1.3
September 3rd, 2010
- Added support for inband DTMF dialing
- Improved logic for matching contacts to incoming sessions
- Added pstn prefix setting
- Fixed enabling Bonjour account item in the menu
- Added initial MWI support
Version 0.1.2
August 19th, 2010
- First beta release for Microsoft Windows
- Switch automatically to the plugged audio device
- Release notes available at http://icanblink.com/blink-qt-windows-beta.phtml
Version 0.1.1
August 13th, 2010
- First public release for Debian and Ubuntu Linux
- Release notes available at http://icanblink.com/blink-qt-beta.phtml
- Multiple SIP accounts
- Easy to setup accounts, only the SIP address and password are required
- Bonjour discovery mechanism
- Automatic detection of IP address changes
- TLS Security for both signaling and media
- NAT traversal using ICE
- Built-in DNS resolver to by-pass broken implementations in NAT routers
- Re-INVITE support for adding and removing media streams
- One-click SIP account sign-up at http://sip2sip.info
- Integration with AG Projects Multimedia Service Platform
- Integration with third-party SIP service providers
- Wideband Audio (G722 & speex)
- Multiple parallel calls
- Play hold tone and disconnect tone
- In-band DTMF support for legacy devices
- Per account ringtones
- Silent mode (do not ring on incoming call)
- Mute microphone
- Displays packet loss and round trip time
- Displays selected audio codec and sampling rate
- Control for input, output and alert audio devices
- Automatic DTMF mapping between letters and digits
- Support for entering PSTN numbers and SIP addresses
- Strip unwanted characters from telephone numbers
- Redial last call
- Multi-party conferencing with unlimited number of participants
- Multiple simultaneous conferences
- Drag and Drop contacts to conferences
- Mute individual participants
- Audio recording
- Display the caller icon and name retrieved from Address Book
- Reject calls with 486 Busy or 603 Decline
- SIP, DNS, MSRP protocol trace to file