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Commit 17ed33fc authored by Russell Bryant's avatar Russell Bryant
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- 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: https://origsvn.digium.com/svn/asterisk/trunk@98656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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......@@ -43,22 +43,22 @@ AMI - The manager (TCP/TLS/HTTP)
Dialplan functions
------------------
* Added the DEVICE_STATE() dialplan function which allows retrieving any device
state in the dialplan, as well as creating custom device states that are
controllable from the dialplan.
state in the dialplan, as well as creating custom device states that are
controllable from the dialplan.
* Extend CALLERID() function with "pres" and "ton" parameters to
fetch string representation of calling number presentation indicator
and numeric representation of type of calling number value.
* MailboxExists converted to dialplan function
* A new option to Dial() for telling IP phones not to count the call
as "missed" when dial times out and cancels.
as "missed" when dial times out and cancels.
* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
held for any given channel. Also, locks are automatically freed when a
channel is hung up.
mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
held for any given channel. Also, locks are automatically freed when a
channel is hung up.
* Added HINT() dialplan function that allows retrieving hint information.
Hints are mappings between extensions and devices for the sake of
determining the state of an extension. This function can retrieve the list
of devices or the name associated with a hint.
Hints are mappings between extensions and devices for the sake of
determining the state of an extension. This function can retrieve the list
of devices or the name associated with a hint.
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
of any extension.
* Added SYSINFO() dialplan function which allows retrieval of system information
......@@ -114,13 +114,13 @@ SIP changes
states it is not needed. For phones, however, that do require it the "registertrying" option
has been added so it can be enabled.
* A new option called "callcounter" (global/peer/user level) enables call counters needed
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
used to enable this functionality).
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
used to enable this functionality).
* New settings for timer T1 and timer B on a global level or per device. This makes it
possible to force timeout faster on non-responsive SIP servers. These settings are
considered advanced, so don't use them unless you have a problem.
possible to force timeout faster on non-responsive SIP servers. These settings are
considered advanced, so don't use them unless you have a problem.
* Added a dial string option to be able to set the To: header in an INVITE to any
SIP uri.
SIP uri.
* Added a new global and per-peer option, qualifyfreq, which allows you to configure
the qualify frequency.
......@@ -150,11 +150,6 @@ Console Channel Driver changes
* Added experimental support for video send & receive to chan_oss.
This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
a video source.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
work with Mac CoreAudio, but portaudio supports a number of other audio
interfaces, as well. Note that this channel driver requires v19 or higher
of portaudio; older versions have a different API.
Phone channel changes (chan_phone)
----------------------------------
......@@ -181,8 +176,8 @@ Zaptel channel driver (chan_zap) Changes
----------------------------------------
* SS7 support in chan_zap (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added zap show version CLI command to chan_zap.
* Added setvar support to zapata.conf channel entries.
......@@ -192,21 +187,26 @@ Zaptel channel driver (chan_zap) Changes
event indicating the new state of the mailbox is also generated, so that
the normal MWI facilities in Asterisk work as usual.
* Added signalling type 'auto', which attempts to use the same signalling type
for a channel as configured in Zaptel. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your zapata.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
for a channel as configured in Zaptel. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your zapata.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
* Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.
A new channel driver: Unistim
-----------------------------
New Channel Drivers
-------------------
* Added a new channel driver, chan_unistim. See doc/unistim.txt and
configs/unistim.conf.sample for details. This new channel driver allows
you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
work with Mac CoreAudio, but portaudio supports a number of other audio
interfaces, as well. Note that this channel driver requires v19 or higher
of portaudio; older versions have a different API.
DUNDi changes
-------------
......@@ -350,8 +350,8 @@ Music On Hold Changes
to this music on hold class.
* Support for realtime music on hold has been added.
* In conjunction with the realtime music on hold, a general section has
been added to musiconhold.conf, its sole variable is cachertclasses. If this
is set, then music on hold classes found in realtime will be cached in memory.
been added to musiconhold.conf, its sole variable is cachertclasses. If this
is set, then music on hold classes found in realtime will be cached in memory.
AEL Changes
-----------
......@@ -373,11 +373,11 @@ AEL Changes
fashion: Set(LOCAL(myvar)=someval); ("local" is now
an AEL keyword).
* utils/conf2ael introduced. Will convert an extensions.conf
file into extensions.ael. Very crude and unfinished, but
will be improved as time goes by. Should be useful for a
first pass at conversion.
file into extensions.ael. Very crude and unfinished, but
will be improved as time goes by. Should be useful for a
first pass at conversion.
* aelparse will now read extensions.conf to see if a referenced
macro or context is there before issueing a warning.
macro or context is there before issueing a warning.
Call Features (res_features) Changes
------------------------------------
......@@ -417,39 +417,10 @@ Logger changes
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
Miscellaneous
-------------
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
Miscellaneous New Modules
-------------------------
* Added a new CDR module, cdr_sqlite3_custom.
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
* Added a new realtime configuration module, res_config_sqlite
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented in doc/qos.tex, or the IP Quality of Service
section of asterisk.pdf.
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
It allows you to configure a prefix for auto-monitor recordings.
* Added support for writing and running your dialplan in lua. See
configs/extensions.lua.sample for examples of how to do this.
* A new extension pattern matching algorithm, based on a trie, is introduced
here, that could noticeably speed up mid-sized to large dialplans.
It is NOT used by default, as duplicating the behaviour of the old pattern
matcher is still under development. A config file option, in extensions.conf,
in the [general] section, called "extenpatternmatchingnew", is by default
set to false; setting that to true will force the use of the new algorithm.
Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
specifying which socket to use to connect to the running Asterisk daemon
(-s)
* Added a new codec translation module, codec_resample, which re-samples
signed linear audio between 8 kHz and 16 kHz to help support wideband
codecs.
......@@ -473,3 +444,36 @@ Miscellaneous
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
Miscellaneous
-------------
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented in doc/qos.tex, or the IP Quality of Service
section of asterisk.pdf.
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
It allows you to configure a prefix for auto-monitor recordings.
* Added support for writing and running your dialplan in lua. See
configs/extensions.lua.sample for examples of how to do this.
* A new extension pattern matching algorithm, based on a trie, is introduced
here, that could noticeably speed up mid-sized to large dialplans.
It is NOT used by default, as duplicating the behaviour of the old pattern
matcher is still under development. A config file option, in extensions.conf,
in the [general] section, called "extenpatternmatchingnew", is by default
set to false; setting that to true will force the use of the new algorithm.
Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
specifying which socket to use to connect to the running Asterisk daemon
(-s)
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