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asterisk / CHANGES

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Asterisk 0.1.8
 -- Keep track of version information
 -- Add -f to cause Asterisk not to fork
 -- Keep important information in voicemail .txt file
 -- Adtran Voice over Frame Relay updates
 -- Implement option setting/querying of channel drivers
 -- IAX performance improvements and protocol fixes
 -- Substantial enhancement of console channel driver
 -- Add IAX registration.  Now IAX can dynamically register
 -- Add flash-hook transfer on tormenta channels
 -- Added Three Way Calling on tormenta channels
 -- Start on concept of zombie channel
 -- Add Call Waiting CallerID
 -- Keep track of who registeres contexts, includes, and extensions
 -- Added Call Waiting(tm), *67, *70, and *82 codes
 -- Move parked calls into "parkedcalls" context by default
 -- Allow dialplan to be displayed
 -- Allow "=>" instead of just "=" to make instantiation clearer
 -- Asterisk forks if called with no arguments
 -- Add remote control by running asterisk -vvvc
 -- Adjust verboseness with "set verbose" now
 -- No longer requires libaudiofile
 -- Install beep
 -- Make PBX Config module reload extensions on SIGHUP
 -- Allow modules to be reloaded when SIGHUP is received
 -- Variables now contain line numbers
 -- Make dialer send in band signalling
 -- Add record application
 -- Added PRI signalling to Tormenta driver
 -- Allow use of BYEXTENSION in "Goto"
 -- Allow adjustment of gains on tormenta channels
 -- Added raw PCM file format support
 -- Add U-law translator
 -- Fix DTMF handling in bridge code
 -- Fix access control with IAX
* Asterisk 0.1.7
 -- Update configuration files and add some missing sounds
 -- Added ability to include one context in another
 -- Rewrite of PBX switching
 -- Major mods to dialler application
 -- Added Caller*ID spill reception
 -- Added Dialogic VOX file format support
 -- Added ADPCM Codec
 -- Add Tormenta driver (RBS signalling)
 -- Add Caller*ID spill creation
 -- Rewrite of translation layer entirely
 -- Add ability to run PBX without additional thread
* Asterisk 0.1.6
 -- Make app_dial handle a lack of translators smoothly
 -- Add ISDN4Linux support -- dtmf is weird...
 -- Minor bug fixes
* Asterisk 0.1.5
 -- Fix a small mistake in IAX
 -- Fix the QuickNet driver to work with newer cards
* Asterisk 0.1.4
 -- Update VoFR some more
 -- Fix the QuickNet driver to work with LineJack
 -- Add ability to pass images for IAX.
* Asterisk 0.1.3
 -- Update VoFR for latest sangoma code
 -- Update QuickNet Driver
 -- Add text message handling
 -- Fix transfers to use "default" if not in current context
 -- Add call parking
 -- Improve format/content negotiation
 -- Added support for multiple languages
 -- Bug fixes, as always...
* Asterisk 0.1.2
 -- Updated README file with a "Getting Started" section
 -- Added sample sounds and configuration files.
 -- Added LPC10 very low bandwidth (low quality) compression
 -- Enhanced translation selection mechanism.
 -- Enhanced IAX jitter buffer, improved reliability
 -- Support echo cancelation on PhoneJack
 -- Updated PhoneJack driver to std. Telephony interface
 -- Added app_echo for evaluating VoIP latency
 -- Added app_system to execute arbitrary programs
 -- Updated sample configuration files
 -- Added OSS channel driver (full duplex only)
 -- Added IAX implementation
 -- Fixed some deadlocks.
 -- A whole bunch of bug fixes
* Asterisk 0.1.1
 -- Revised translator, fixed some general race conditions throughout *
 -- Made dialer somewhat more aware of incompatible voice channels
 -- Added Voice Modem driver and A/Open Modem Driver stub
 -- Added MP3 decoder channel
 -- Added Microsoft WAV49 support
 -- Revised License -- Pure GPL, nothing else
 -- Modified Copyright statement since code is still currently owned by author
 -- Added RAW GSM headerless data format
 -- Innumerable bug fixes
* Asterisk 0.1.0
 -- Initial Release