[svn-commits] russell: trunk r61760 - /trunk/CHANGES
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Mon Apr 23 08:34:52 MST 2007
Author: russell
Date: Mon Apr 23 10:34:51 2007
New Revision: 61760
URL: http://svn.digium.com/view/asterisk?view=rev&rev=61760
Log:
Add OSP support for IAX2 to the changes file. Also, slightly reorganize some
of the content.
Modified:
trunk/CHANGES
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=61760&r1=61759&r2=61760
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Mon Apr 23 10:34:51 2007
@@ -1,6 +1,81 @@
-------------------------------------------------------------------------------
--- Functionality changes since Asterisk 1.4-beta was branched ----------------
-------------------------------------------------------------------------------
+
+AMI - The manager (TCP/TLS/HTTP)
+--------------------------------
+ * Added the URI redirect option for the built-in HTTP server
+ * The output of CallerID in Manager events is now more consistent.
+ CallerIDNum is used for number and CallerIDName for name.
+ * enable https support for builtin web server.
+ See configs/http.conf.sample for details.
+ * Added a new action, GetConfigJSON, which can return the contents of an
+ Asterisk configuration file in JSON format. This is intended to help
+ improve the performance of AJAX applications using the manager interface
+ over HTTP.
+ * SIP and IAX manager events now use "ChannelType" in all cases where we
+ indicate channel driver. Previously, we used a mixture of "Channel"
+ and "ChannelDriver" headers.
+ * Added a "Bridge" action which allows you to bridge any two channels that
+ are currently active on the system.
+
+Dialplan functions
+------------------
+ * Added the DEVSTATE() dialplan function which allows retrieving any device
+ 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
+
+CLI Changes
+-----------
+ * New CLI command "core show settings"
+ * Added 'core show channels count' CLI command.
+
+SIP changes
+-----------
+ * The default SIP useragent= identifier now includes the Asterisk version
+ * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
+ If set, and the incoming request carries authentication info,
+ the username to match in the users list is taken from the Digest header
+ rather than from the From: field. This feature is considered experimental.
+ * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
+ since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
+ * The "localmask" setting was removed in version 1.2 and the reminder about it
+ being removed is now also removed.
+ * A new option "busy-level" for setting a level of calls where asterisk reports
+ a device as busy, to separate it from call-limit
+ * A new realtime family called "sipregs" is now supported to store SIP registration
+ data. If this family is defined, "sippeers" will be used for configuration and
+ "sipregs" for registrations. If it's not defined, "sippeers" will be used for
+ registration data, as before.
+ * The SIPPEER function have new options for port address, call and pickup groups
+ * Added support for T.140 realtime text in SIP/RTP
+
+IAX2 changes
+------------
+ * Added the trunkmaxsize configuration option to chan_iax2.
+ * Added the srvlookup option to iax.conf
+ * Added support for OSP. The token is set and retrieved through the CHANNEL()
+ dialplan function.
+
+DUNDi changes
+-------------
+ * Added the ability to specify arguments to the Dial application when using
+ the DUNDi switch in the dialplan.
+ * Added the ability to set weights for responses dynamically. This can be
+ done using a global variable or a dialplan function. Using the SHELL()
+ function would allow you to have an external script set the weight for
+ each response.
+
+Voicemail Changes
+-----------------
+ * Added the ability to customize which sound files are used for some of the
+ prompts within the Voicemail application by changing them in voicemail.conf
+ * Added the ability for the "voicemail show users" CLI command to show users
+ configured by the dynamic realtime configuration method.
Miscellaneous
-------------
@@ -53,11 +128,9 @@
* 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.
- * Added the trunkmaxsize configuration option to chan_iax2.
* Added G729 passthrough support to chan_phone for Sigma Designs boards.
* Added the parkedcalltransfers option to features.conf
* Added 's' option to Page application.
- * Added the srvlookup option to iax.conf
* Added 'E' and 'V' commands to ExternalIVR.
* Added 'DBDel' and 'DBDelTree' manager commands.
* Added 'o' and 'X' options to Chanspy.
@@ -68,71 +141,3 @@
* Added a new realtime configuration module, res_config_sqlite
* Added a new dialplan application, Bridge, which allows you to bridge the
calling channel to any other active channel on the system.
-
-AMI - The manager (TCP/TLS/HTTP)
---------------------------------
- * Added the URI redirect option for the built-in HTTP server
- * The output of CallerID in Manager events is now more consistent.
- CallerIDNum is used for number and CallerIDName for name.
- * enable https support for builtin web server.
- See configs/http.conf.sample for details.
- * Added a new action, GetConfigJSON, which can return the contents of an
- Asterisk configuration file in JSON format. This is intended to help
- improve the performance of AJAX applications using the manager interface
- over HTTP.
- * SIP and IAX manager events now use "ChannelType" in all cases where we
- indicate channel driver. Previously, we used a mixture of "Channel"
- and "ChannelDriver" headers.
- * Added a "Bridge" action which allows you to bridge any two channels that
- are currently active on the system.
-
-Dialplan functions
-------------------
- * Added the DEVSTATE() dialplan function which allows retrieving any device
- 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
-
-CLI Changes
------------
- * New CLI command "core show settings"
- * Added 'core show channels count' CLI command.
-
-SIP changes
------------
- * The default SIP useragent= identifier now includes the Asterisk version
- * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
- If set, and the incoming request carries authentication info,
- the username to match in the users list is taken from the Digest header
- rather than from the From: field. This feature is considered experimental.
- * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
- since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
- * The "localmask" setting was removed in version 1.2 and the reminder about it
- being removed is now also removed.
- * A new option "busy-level" for setting a level of calls where asterisk reports
- a device as busy, to separate it from call-limit
- * A new realtime family called "sipregs" is now supported to store SIP registration
- data. If this family is defined, "sippeers" will be used for configuration and
- "sipregs" for registrations. If it's not defined, "sippeers" will be used for
- registration data, as before.
- * The SIPPEER function have new options for port address, call and pickup groups
- * Added support for T.140 realtime text in SIP/RTP
-
-DUNDi changes
--------------
- * Added the ability to specify arguments to the Dial application when using
- the DUNDi switch in the dialplan.
- * Added the ability to set weights for responses dynamically. This can be
- done using a global variable or a dialplan function. Using the SHELL()
- function would allow you to have an external script set the weight for
- each response.
-
-Voicemail Changes
------------------
- * Added the ability to customize which sound files are used for some of the
- prompts within the Voicemail application by changing them in voicemail.conf
- * Added the ability for the "voicemail show users" CLI command to show users
- configured by the dynamic realtime configuration method.
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