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Add OSP support for IAX2 to the changes file. Also, slightly reorganize some
of the content. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
145
CHANGES
145
CHANGES
@@ -2,6 +2,81 @@
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--- Functionality changes since Asterisk 1.4-beta was branched ----------------
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-------------------------------------------------------------------------------
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AMI - The manager (TCP/TLS/HTTP)
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--------------------------------
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* Added the URI redirect option for the built-in HTTP server
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* The output of CallerID in Manager events is now more consistent.
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CallerIDNum is used for number and CallerIDName for name.
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* enable https support for builtin web server.
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See configs/http.conf.sample for details.
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* Added a new action, GetConfigJSON, which can return the contents of an
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Asterisk configuration file in JSON format. This is intended to help
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improve the performance of AJAX applications using the manager interface
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over HTTP.
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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indicate channel driver. Previously, we used a mixture of "Channel"
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and "ChannelDriver" headers.
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* Added a "Bridge" action which allows you to bridge any two channels that
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are currently active on the system.
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Dialplan functions
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------------------
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* Added the DEVSTATE() 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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* 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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CLI Changes
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-----------
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* New CLI command "core show settings"
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* Added 'core show channels count' CLI command.
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SIP changes
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-----------
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* The default SIP useragent= identifier now includes the Asterisk version
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from the Digest header
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rather than from the From: field. This feature is considered experimental.
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* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
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since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
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* The "localmask" setting was removed in version 1.2 and the reminder about it
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being removed is now also removed.
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* A new option "busy-level" for setting a level of calls where asterisk reports
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a device as busy, to separate it from call-limit
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* A new realtime family called "sipregs" is now supported to store SIP registration
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data. If this family is defined, "sippeers" will be used for configuration and
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"sipregs" for registrations. If it's not defined, "sippeers" will be used for
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registration data, as before.
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* The SIPPEER function have new options for port address, call and pickup groups
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* Added support for T.140 realtime text in SIP/RTP
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IAX2 changes
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------------
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* Added the trunkmaxsize configuration option to chan_iax2.
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* Added the srvlookup option to iax.conf
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* Added support for OSP. The token is set and retrieved through the CHANNEL()
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dialplan function.
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DUNDi changes
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-------------
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* Added the ability to specify arguments to the Dial application when using
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the DUNDi switch in the dialplan.
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* Added the ability to set weights for responses dynamically. This can be
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done using a global variable or a dialplan function. Using the SHELL()
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function would allow you to have an external script set the weight for
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each response.
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Voicemail Changes
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-----------------
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* Added the ability to customize which sound files are used for some of the
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prompts within the Voicemail application by changing them in voicemail.conf
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* Added the ability for the "voicemail show users" CLI command to show users
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configured by the dynamic realtime configuration method.
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Miscellaneous
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-------------
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@@ -53,11 +128,9 @@ Miscellaneous
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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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* Added the trunkmaxsize configuration option to chan_iax2.
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* Added G729 passthrough support to chan_phone for Sigma Designs boards.
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* Added the parkedcalltransfers option to features.conf
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* Added 's' option to Page application.
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* Added the srvlookup option to iax.conf
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* Added 'E' and 'V' commands to ExternalIVR.
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* Added 'DBDel' and 'DBDelTree' manager commands.
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* Added 'o' and 'X' options to Chanspy.
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@@ -68,71 +141,3 @@ Miscellaneous
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* Added a new realtime configuration module, res_config_sqlite
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* Added a new dialplan application, Bridge, which allows you to bridge the
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calling channel to any other active channel on the system.
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AMI - The manager (TCP/TLS/HTTP)
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--------------------------------
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* Added the URI redirect option for the built-in HTTP server
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* The output of CallerID in Manager events is now more consistent.
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CallerIDNum is used for number and CallerIDName for name.
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* enable https support for builtin web server.
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See configs/http.conf.sample for details.
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* Added a new action, GetConfigJSON, which can return the contents of an
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Asterisk configuration file in JSON format. This is intended to help
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improve the performance of AJAX applications using the manager interface
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over HTTP.
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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indicate channel driver. Previously, we used a mixture of "Channel"
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and "ChannelDriver" headers.
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* Added a "Bridge" action which allows you to bridge any two channels that
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are currently active on the system.
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Dialplan functions
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------------------
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* Added the DEVSTATE() 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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* 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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CLI Changes
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-----------
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* New CLI command "core show settings"
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* Added 'core show channels count' CLI command.
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SIP changes
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-----------
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* The default SIP useragent= identifier now includes the Asterisk version
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from the Digest header
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rather than from the From: field. This feature is considered experimental.
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* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
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since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
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* The "localmask" setting was removed in version 1.2 and the reminder about it
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being removed is now also removed.
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* A new option "busy-level" for setting a level of calls where asterisk reports
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a device as busy, to separate it from call-limit
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* A new realtime family called "sipregs" is now supported to store SIP registration
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data. If this family is defined, "sippeers" will be used for configuration and
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"sipregs" for registrations. If it's not defined, "sippeers" will be used for
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registration data, as before.
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* The SIPPEER function have new options for port address, call and pickup groups
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* Added support for T.140 realtime text in SIP/RTP
|
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|
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DUNDi changes
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-------------
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* Added the ability to specify arguments to the Dial application when using
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the DUNDi switch in the dialplan.
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* Added the ability to set weights for responses dynamically. This can be
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done using a global variable or a dialplan function. Using the SHELL()
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function would allow you to have an external script set the weight for
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each response.
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Voicemail Changes
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-----------------
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* Added the ability to customize which sound files are used for some of the
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prompts within the Voicemail application by changing them in voicemail.conf
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* Added the ability for the "voicemail show users" CLI command to show users
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configured by the dynamic realtime configuration method.
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