Added changelog for initial release

a80207aa · Adrian Georgescu · 5ef773ad · a80207aa · a80207aa · a80207aa
Commit a80207aa authored Aug 04, 2010 by Adrian Georgescu
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__init__.py blink/__init__.py +2 -2

changelog debian/changelog +41 -1

changelog.html doc/changelog.html +40 -5

No files found.
--- a/blink/__init__.py
+++ b/blink/__init__.py
@@ -4,8 +4,8 @@
 __all__ = ['Blink']


-__version__ = '0.1.0'
-__date__    = 'July 5, 2010'
+__version__ = '0.1.1'
+__date__    = 'August 16th, 2010'


 import os

--- a/debian/changelog
+++ b/debian/changelog
+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
+
+ -- Adrian Georgescu <ag@ag-projects.com>  Wed, 04 Aug 2010 22:53:46 +0200
+
 blink (0.1.0) unstable; urgency=low

-  * Initial release.

 -- Saul Ibarra <saul@ag-projects.com>  Fri, 16 Jul 2010 12:54:28 +0200
--- a/doc/changelog.html
+++ b/doc/changelog.html
 <html>
 <head>
-<title>Blink Changelog</title>
+<title>Blink QTY Changelog</title>
 <link rel='stylesheet' type='text/css' href='style.css'>
 </head>

-<body>
+<h2>Version 0.1.1</h2>

-<h2>Version 0.1.0</h2>
-<p>Date: TBD  
+<p>August 16th, 2010

 <ul>
-<li>Initial release
+<li>First public release for Debian and Ubuntu Linux
+<li>Multiple SIP accounts
+<li>Easy to setup accounts, only the SIP address and password are required
+<li>Bonjour discovery mechanism
+<li>Automatic detection of IP address changes
+<li>TLS Security for both signaling and media
+<li>NAT traversal using ICE and MSRP relay
+<li>Built-in DNS resolver to by-pass broken implementations in NAT routers
+<li>Re-INVITE support for adding and removing media streams
+<li>One-click SIP account sign-up at http://sip2sip.info
+<li>Integration with AG Projects Multimedia Service Platform
+<li>Integration with third-party SIP service providers
+<li>Wideband Audio (G722 & speex)
+<li>Multiple parallel calls
+<li>Play hold tone and disconnect tone
+<li>In-band DTMF support for legacy devices
+<li>Per account ringtones
+<li>Silent mode (do not ring on incoming call)
+<li>Mute microphone
+<li>Audio recording
+<li>Displays packet loss and round trip time
+<li>Displays selected audio codec and sampling rate
+<li>Control for input, output and alert audio devices
+<li>Automatic DTMF mapping between letters and digits
+<li>Support for entering PSTN numbers and SIP addresses
+<li>Strip unwanted characters from telephone numbers
+<li>Redial last call
+<li>Multi-party conferencing with unlimited number of participants
+<li>Drag and Drop contacts to conferences
+<li>Mute individual participants
+<li>Audio recording
+<li>Display the caller icon and name retrieved from Address Book
+<li>Reject calls with 486 Busy  or 603 Decline
+<li>Accept partial offers when INVITE contains multiple streams
+<li>Multiple simultaneous conferences
+<li>SIP, DNS, MSRP protocol trace to file
+
 </ul>

 </body>