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Add OSP support for IAX2 to the changes file. Also, slightly reorganize some

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Russell Bryant 18 years ago
parent 82d5673c81
commit 5cf93c4ca0
1 changed files with 75 additions and 70 deletions
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@ -2,6 +2,81 @@
--- 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 @@ Miscellaneous
* 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 @@ Miscellaneous
* 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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