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Kulya
vmj-qt
Commits
ab8f73dc
Commit
ab8f73dc
authored
Aug 04, 2010
by
Adrian Georgescu
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Fixed indentation of changelog entries
parent
b119471c
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debian/changelog
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ab8f73dc
blink (0.1.1) unstable; urgency=low
* First public release for Debian and Ubuntu Linux
* 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 and MSRP relay
* 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
* Audio recording
* 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
* 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
* Accept partial offers when INVITE contains multiple streams
* Multiple simultaneous conferences
* SIP, DNS, MSRP protocol trace to file
* First public release for Debian and Ubuntu Linux
* 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 and MSRP relay
* 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
* Audio recording
* 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
* 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
* Accept partial offers when INVITE contains multiple streams
* Multiple simultaneous conferences
* SIP, DNS, MSRP protocol trace to file
-- Adrian Georgescu <ag@ag-projects.com> Wed, 04 Aug 2010 22:53:46 +0200
blink (0.1.0) unstable; urgency=low
* Initial private released
-- Saul Ibarra <saul@ag-projects.com> Fri, 16 Jul 2010 12:54:28 +0200
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