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Updates and re-organization to make it easier to digest this information 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54465 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2007-02-14 20:31:10 +00:00
parent 18b268814d
commit c05d59a018
1 changed files with 22 additions and 10 deletions
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@ -4,10 +4,7 @@ Changes since Asterisk 1.4-beta was branched:
the DUNDi switch in the dialplan.
* 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
* enable https support for builtin web server.
See configs/http.conf.sample for details.
* Argument support for Gosub application
* MailboxExists converted to dialplan function
* Ability to set process limits without restarting Asterisk
* SS7 support in chan_zap (via libss7 library)
* Proper codec support in chan_skinny.
@ -27,8 +24,6 @@ Changes since Asterisk 1.4-beta was branched:
statistics during a reload.
* Added rotatetimestamp option to logger.conf which will use
the time to name the logger files instead of sequence number.
* The output of CallerID in Manager events is now more consistent.
CallerIDNum is used for number and CallerIDName for name.
* setinterfacevar option in queues.conf also now sets a variable
called MEMBERNAME which contains the member's name.
* Added Masquerade manager event for when a masquerade happens between
@ -43,9 +38,6 @@ Changes since Asterisk 1.4-beta was branched:
Read() - timeout now can be floating pt.
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* 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.
* Added 'C' option to Meetme which causes a caller to continue in the dialplan
when kicked out.
* Added option to run macro when a queue member is connected to a caller,
@ -59,7 +51,6 @@ Changes since Asterisk 1.4-beta was branched:
* 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 URI redirect option for the built-in HTTP server
* 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
@ -67,10 +58,29 @@ Changes since Asterisk 1.4-beta was branched:
* Added the srvlookup option to iax.conf
* Added 'E' and 'V' commands to ExternalIVR.
* Added 'DBDel' and 'DBDelTree' manager commands.
* Added 'core show channels count' CLI command.
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.
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
-----------
@ -83,3 +93,5 @@ SIP changes
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