- Break up the Misc. section a bit with a new section for Misc. New Modules
- Change spacing a bit in some places for consistent indentation git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98656 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
russell
parent
02a222b4cc
commit
0c6b474f15
140
CHANGES
140
CHANGES
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@ -43,22 +43,22 @@ AMI - The manager (TCP/TLS/HTTP)
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Dialplan functions
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------------------
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* Added the DEVICE_STATE() dialplan function which allows retrieving any device
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state in the dialplan, as well as creating custom device states that are
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controllable from the dialplan.
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state in the dialplan, as well as creating custom device states that are
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controllable from the dialplan.
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* Extend CALLERID() function with "pres" and "ton" parameters to
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fetch string representation of calling number presentation indicator
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and numeric representation of type of calling number value.
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* MailboxExists converted to dialplan function
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* A new option to Dial() for telling IP phones not to count the call
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as "missed" when dial times out and cancels.
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as "missed" when dial times out and cancels.
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* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
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mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
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held for any given channel. Also, locks are automatically freed when a
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channel is hung up.
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mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
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held for any given channel. Also, locks are automatically freed when a
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channel is hung up.
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* Added HINT() dialplan function that allows retrieving hint information.
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Hints are mappings between extensions and devices for the sake of
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determining the state of an extension. This function can retrieve the list
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of devices or the name associated with a hint.
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Hints are mappings between extensions and devices for the sake of
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determining the state of an extension. This function can retrieve the list
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of devices or the name associated with a hint.
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* Added EXTENSION_STATE() dialplan function which allows retrieving the state
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of any extension.
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* Added SYSINFO() dialplan function which allows retrieval of system information
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@ -114,13 +114,13 @@ SIP changes
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states it is not needed. For phones, however, that do require it the "registertrying" option
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has been added so it can be enabled.
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* A new option called "callcounter" (global/peer/user level) enables call counters needed
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for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
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used to enable this functionality).
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for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
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used to enable this functionality).
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* New settings for timer T1 and timer B on a global level or per device. This makes it
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possible to force timeout faster on non-responsive SIP servers. These settings are
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considered advanced, so don't use them unless you have a problem.
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possible to force timeout faster on non-responsive SIP servers. These settings are
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considered advanced, so don't use them unless you have a problem.
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* Added a dial string option to be able to set the To: header in an INVITE to any
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SIP uri.
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SIP uri.
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* Added a new global and per-peer option, qualifyfreq, which allows you to configure
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the qualify frequency.
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@ -150,11 +150,6 @@ Console Channel Driver changes
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* Added experimental support for video send & receive to chan_oss.
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This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
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a video source.
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* Added a new channel driver, chan_console, which uses portaudio as a cross
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platform audio interface. It was written as a channel driver that would
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work with Mac CoreAudio, but portaudio supports a number of other audio
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interfaces, as well. Note that this channel driver requires v19 or higher
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of portaudio; older versions have a different API.
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Phone channel changes (chan_phone)
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----------------------------------
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@ -181,8 +176,8 @@ Zaptel channel driver (chan_zap) Changes
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----------------------------------------
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* SS7 support in chan_zap (via libss7 library)
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* In India, some carriers transmit CID via dtmf. Some code has been added
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that will handle some situations. The cidstart=polarity_IN choice has been added for
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those carriers that transmit CID via dtmf after a polarity change.
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that will handle some situations. The cidstart=polarity_IN choice has been added for
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those carriers that transmit CID via dtmf after a polarity change.
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* CID matching information is now shown when doing 'dialplan show'.
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* Added zap show version CLI command to chan_zap.
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* Added setvar support to zapata.conf channel entries.
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@ -192,21 +187,26 @@ Zaptel channel driver (chan_zap) Changes
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event indicating the new state of the mailbox is also generated, so that
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the normal MWI facilities in Asterisk work as usual.
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* Added signalling type 'auto', which attempts to use the same signalling type
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for a channel as configured in Zaptel. This is primarily designed for analog
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ports, but will also work for digital ports that are configured for FXS or FXO
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signalling types. This mode is also the default now, so if your zapata.conf
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does not specify signalling for a channel (which is unlikely as the sample
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configuration file has always recommended specifying it for every channel) then
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the 'auto' mode will be used for that channel if possible.
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for a channel as configured in Zaptel. This is primarily designed for analog
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ports, but will also work for digital ports that are configured for FXS or FXO
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signalling types. This mode is also the default now, so if your zapata.conf
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does not specify signalling for a channel (which is unlikely as the sample
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configuration file has always recommended specifying it for every channel) then
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the 'auto' mode will be used for that channel if possible.
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* Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
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state for a channel; also ensured that the DNDState Manager event is
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emitted no matter how the DND state is set or cleared.
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state for a channel; also ensured that the DNDState Manager event is
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emitted no matter how the DND state is set or cleared.
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A new channel driver: Unistim
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-----------------------------
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New Channel Drivers
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-------------------
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* Added a new channel driver, chan_unistim. See doc/unistim.txt and
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configs/unistim.conf.sample for details. This new channel driver allows
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you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
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* Added a new channel driver, chan_console, which uses portaudio as a cross
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platform audio interface. It was written as a channel driver that would
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work with Mac CoreAudio, but portaudio supports a number of other audio
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interfaces, as well. Note that this channel driver requires v19 or higher
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of portaudio; older versions have a different API.
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DUNDi changes
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-------------
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@ -350,8 +350,8 @@ Music On Hold Changes
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to this music on hold class.
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* Support for realtime music on hold has been added.
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* In conjunction with the realtime music on hold, a general section has
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been added to musiconhold.conf, its sole variable is cachertclasses. If this
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is set, then music on hold classes found in realtime will be cached in memory.
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been added to musiconhold.conf, its sole variable is cachertclasses. If this
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is set, then music on hold classes found in realtime will be cached in memory.
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AEL Changes
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-----------
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@ -373,11 +373,11 @@ AEL Changes
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fashion: Set(LOCAL(myvar)=someval); ("local" is now
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an AEL keyword).
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* utils/conf2ael introduced. Will convert an extensions.conf
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file into extensions.ael. Very crude and unfinished, but
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will be improved as time goes by. Should be useful for a
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first pass at conversion.
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file into extensions.ael. Very crude and unfinished, but
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will be improved as time goes by. Should be useful for a
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first pass at conversion.
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* aelparse will now read extensions.conf to see if a referenced
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macro or context is there before issueing a warning.
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macro or context is there before issueing a warning.
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Call Features (res_features) Changes
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------------------------------------
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@ -417,39 +417,10 @@ Logger changes
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and to ensure that the oldest log file gets deleted.
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* Added realtime support for the queue log
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Miscellaneous
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-------------
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* Ability to use libcap to set high ToS bits when non-root
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on Linux. If configure is unable to find libcap then you
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can use --with-cap to specify the path.
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* Added maxfiles option to options section of asterisk.conf which allows you to specify
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what Asterisk should set as the maximum number of open files when it loads.
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* Added the jittertargetextra configuration option.
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Miscellaneous New Modules
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-------------------------
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* Added a new CDR module, cdr_sqlite3_custom.
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* The cdr_manager module has a [mappings] feature, like cdr_custom,
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to add fields to the manager event from the CDR variables.
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* Added a new realtime configuration module, res_config_sqlite
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* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
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configuration files for the IP channel drivers. The new option is "cos".
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This information is also documented in doc/qos.tex, or the IP Quality of Service
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section of asterisk.pdf.
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* When originating a call using AMI or pbx_spool that fails the reason for failure
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will now be available in the failed extension using the REASON dialplan variable.
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* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
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It allows you to configure a prefix for auto-monitor recordings.
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* Added support for writing and running your dialplan in lua. See
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configs/extensions.lua.sample for examples of how to do this.
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* A new extension pattern matching algorithm, based on a trie, is introduced
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here, that could noticeably speed up mid-sized to large dialplans.
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It is NOT used by default, as duplicating the behaviour of the old pattern
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matcher is still under development. A config file option, in extensions.conf,
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in the [general] section, called "extenpatternmatchingnew", is by default
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set to false; setting that to true will force the use of the new algorithm.
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Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
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be used to switch the algorithms at run time.
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* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
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specifying which socket to use to connect to the running Asterisk daemon
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(-s)
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* Added a new codec translation module, codec_resample, which re-samples
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signed linear audio between 8 kHz and 16 kHz to help support wideband
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codecs.
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@ -473,3 +444,36 @@ Miscellaneous
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on as the channel's audio. This is very useful for building custom
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vocoders or doing recording or analysis of the channel's audio in another
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application.
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Miscellaneous
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-------------
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* Ability to use libcap to set high ToS bits when non-root
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on Linux. If configure is unable to find libcap then you
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can use --with-cap to specify the path.
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* Added maxfiles option to options section of asterisk.conf which allows you to specify
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what Asterisk should set as the maximum number of open files when it loads.
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* Added the jittertargetextra configuration option.
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* The cdr_manager module has a [mappings] feature, like cdr_custom,
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to add fields to the manager event from the CDR variables.
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* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
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configuration files for the IP channel drivers. The new option is "cos".
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This information is also documented in doc/qos.tex, or the IP Quality of Service
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section of asterisk.pdf.
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* When originating a call using AMI or pbx_spool that fails the reason for failure
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will now be available in the failed extension using the REASON dialplan variable.
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* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
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It allows you to configure a prefix for auto-monitor recordings.
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* Added support for writing and running your dialplan in lua. See
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configs/extensions.lua.sample for examples of how to do this.
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* A new extension pattern matching algorithm, based on a trie, is introduced
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here, that could noticeably speed up mid-sized to large dialplans.
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It is NOT used by default, as duplicating the behaviour of the old pattern
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matcher is still under development. A config file option, in extensions.conf,
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in the [general] section, called "extenpatternmatchingnew", is by default
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set to false; setting that to true will force the use of the new algorithm.
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Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
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be used to switch the algorithms at run time.
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* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
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specifying which socket to use to connect to the running Asterisk daemon
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(-s)
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