[asterisk-commits] russell: trunk r98656 - /trunk/CHANGES
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Jan 13 17:43:07 CST 2008
Author: russell
Date: Sun Jan 13 17:43:06 2008
New Revision: 98656
URL: http://svn.digium.com/view/asterisk?view=rev&rev=98656
Log:
- 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
Modified:
trunk/CHANGES
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=98656&r1=98655&r2=98656
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Sun Jan 13 17:43:06 2008
@@ -43,22 +43,22 @@
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 @@
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 @@
* 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 @@
----------------------------------------
* 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 @@
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.
-
-A new channel driver: Unistim
------------------------------
+ state for a channel; also ensured that the DNDState Manager event is
+ emitted no matter how the DND state is set or cleared.
+
+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 @@
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 @@
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 @@
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 @@
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)
+
More information about the asterisk-commits
mailing list