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125 lines
5.0 KiB
125 lines
5.0 KiB
Asterisk 0.1.9
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-- Implement IAX quelching
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-- Allow Caller*ID to be overridden and suggested
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-- Configure defaults to use IAXTEL
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-- Allow remote dialplan polling via IAX
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-- Eliminate ast_longest_extension
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-- Implement dialplan request/reply
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-- Let peers have allow/disallow for codecs
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-- Change allow/deny to permit/deny in IAX
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-- Allow dialplan entries to match Caller*ID as well
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-- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
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-- Added chan_zap for zapata telephony kernel interface, removed chan_tor
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-- Add convenience functions
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-- Fix race condition in channel hangup
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-- Fix memory leaks in both asterisk and iax frame allocations
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-- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
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-- Add DISA application (Thanks to Jim Dixon)
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-- Add IAX transfer support
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-- Add URL and HTML transmission
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-- Add application for sending images
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-- Add RedHat RPM spec file and build capability
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-- Fix GSM WAV file format bug
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-- Move ignorepat to main dialplan
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-- Add ability to specificy TOS bits in IAX
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-- Allow username:password in IAX strings
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-- Updates to PhoneJack interface
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-- Allow "servermail" in voicemail.conf to override e-mail in "from" line
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-- Add 'skip' option to app_playback
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-- Reject IAX calls on unknown extensions
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-- Fix version stuff
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Asterisk 0.1.8
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-- Keep track of version information
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-- Add -f to cause Asterisk not to fork
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-- Keep important information in voicemail .txt file
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-- Adtran Voice over Frame Relay updates
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-- Implement option setting/querying of channel drivers
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-- IAX performance improvements and protocol fixes
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-- Substantial enhancement of console channel driver
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-- Add IAX registration. Now IAX can dynamically register
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-- Add flash-hook transfer on tormenta channels
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-- Added Three Way Calling on tormenta channels
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-- Start on concept of zombie channel
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-- Add Call Waiting CallerID
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-- Keep track of who registeres contexts, includes, and extensions
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-- Added Call Waiting(tm), *67, *70, and *82 codes
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-- Move parked calls into "parkedcalls" context by default
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-- Allow dialplan to be displayed
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-- Allow "=>" instead of just "=" to make instantiation clearer
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-- Asterisk forks if called with no arguments
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-- Add remote control by running asterisk -vvvc
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-- Adjust verboseness with "set verbose" now
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-- No longer requires libaudiofile
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-- Install beep
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-- Make PBX Config module reload extensions on SIGHUP
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-- Allow modules to be reloaded when SIGHUP is received
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-- Variables now contain line numbers
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-- Make dialer send in band signalling
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-- Add record application
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-- Added PRI signalling to Tormenta driver
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-- Allow use of BYEXTENSION in "Goto"
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-- Allow adjustment of gains on tormenta channels
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-- Added raw PCM file format support
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-- Add U-law translator
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-- Fix DTMF handling in bridge code
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-- Fix access control with IAX
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* Asterisk 0.1.7
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-- Update configuration files and add some missing sounds
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-- Added ability to include one context in another
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-- Rewrite of PBX switching
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-- Major mods to dialler application
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-- Added Caller*ID spill reception
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-- Added Dialogic VOX file format support
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-- Added ADPCM Codec
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-- Add Tormenta driver (RBS signalling)
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-- Add Caller*ID spill creation
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-- Rewrite of translation layer entirely
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-- Add ability to run PBX without additional thread
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* Asterisk 0.1.6
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-- Make app_dial handle a lack of translators smoothly
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-- Add ISDN4Linux support -- dtmf is weird...
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-- Minor bug fixes
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* Asterisk 0.1.5
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-- Fix a small mistake in IAX
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-- Fix the QuickNet driver to work with newer cards
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* Asterisk 0.1.4
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-- Update VoFR some more
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-- Fix the QuickNet driver to work with LineJack
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-- Add ability to pass images for IAX.
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* Asterisk 0.1.3
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-- Update VoFR for latest sangoma code
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-- Update QuickNet Driver
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-- Add text message handling
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-- Fix transfers to use "default" if not in current context
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-- Add call parking
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-- Improve format/content negotiation
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-- Added support for multiple languages
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-- Bug fixes, as always...
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* Asterisk 0.1.2
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-- Updated README file with a "Getting Started" section
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-- Added sample sounds and configuration files.
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-- Added LPC10 very low bandwidth (low quality) compression
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-- Enhanced translation selection mechanism.
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-- Enhanced IAX jitter buffer, improved reliability
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-- Support echo cancelation on PhoneJack
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-- Updated PhoneJack driver to std. Telephony interface
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-- Added app_echo for evaluating VoIP latency
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-- Added app_system to execute arbitrary programs
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-- Updated sample configuration files
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-- Added OSS channel driver (full duplex only)
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-- Added IAX implementation
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-- Fixed some deadlocks.
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-- A whole bunch of bug fixes
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* Asterisk 0.1.1
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-- Revised translator, fixed some general race conditions throughout *
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-- Made dialer somewhat more aware of incompatible voice channels
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-- Added Voice Modem driver and A/Open Modem Driver stub
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-- Added MP3 decoder channel
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-- Added Microsoft WAV49 support
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-- Revised License -- Pure GPL, nothing else
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-- Modified Copyright statement since code is still currently owned by author
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-- Added RAW GSM headerless data format
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-- Innumerable bug fixes
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* Asterisk 0.1.0
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-- Initial Release
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