Source: snapvoip.blogspot.com

Following an article on Asterisk blog by Russell, "Sneak peek at new features" and provided a link to SVN reository. Following is a part of it (85 lines, there are 235 lines of description). I was surprised that so many features had sneaked by and yet we are happy to use…!

UPDATE!
Seems like formatting makes this impossible to read;

So you have to go Asterisk developer site (SVN repository) see the complete document.Thanks

1 -------------------------------------------------------------------------------    2 --- Functionality changes since Asterisk 1.4-beta was branched ----------------    3 -------------------------------------------------------------------------------    4      5 AMI - The manager (TCP/TLS/HTTP)    6 --------------------------------    7   * Added the URI redirect option for the built-in HTTP server    8   * The output of CallerID in Manager events is now more consistent.    9      CallerIDNum is used for number and CallerIDName for name.   10   * enable https support for builtin web server.   11      See configs/http.conf.sample for details.   12   * Added a new action, GetConfigJSON, which can return the contents of an   13      Asterisk configuration file in JSON format.  This is intended to help   14      improve the performance of AJAX applications using the manager interface   15      over HTTP.   16   * SIP and IAX manager events now use "ChannelType" in all cases where we   17      indicate channel driver. Previously, we used a mixture of "Channel"   18      and "ChannelDriver" headers.   19   * Added a "Bridge" action which allows you to bridge any two channels that   20      are currently active on the system.   21   * Added a "ListAllVoicemailUsers" action that allows you to get a list of all   22      the voicemail users setup.   23     24 Dialplan functions   25 ------------------   26   * Added the DEVSTATE() dialplan function which allows retrieving any device   27     state in the dialplan, as well as creating custom device states that are   28     controllable from the dialplan.   29   * Extend CALLERID() function with "pres" and "ton" parameters to   30      fetch string representation of calling number presentation indicator   31      and numeric representation of type of calling number value.   32   * MailboxExists converted to dialplan function   33     34 CLI Changes   35 -----------   36   * New CLI command "core show settings"   37   * Added 'core show channels count' CLI command.   38     39 SIP changes   40 -----------   41   * The default SIP useragent= identifier now includes the Asterisk version   42   * A new option, match_auth_username in sip.conf changes the matching of incoming requests.   43      If set, and the incoming request carries authentication info,   44      the username to match in the users list is taken from the Digest header   45      rather than from the From: field. This feature is considered experimental.   46   * The "musiconhold" and "musicclass" settings in sip.conf are now removed,   47      since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4   48   * The "localmask" setting was removed in version 1.2 and the reminder about it   49      being removed is now also removed.   50   * A new option "busy-level" for setting a level of calls where asterisk reports   51      a device as busy, to separate it from call-limit   52   * A new realtime family called "sipregs" is now supported to store SIP registration   53      data. If this family is defined, "sippeers" will be used for configuration and   54      "sipregs" for registrations. If it's not defined, "sippeers" will be used for   55      registration data, as before.   56   * The SIPPEER function have new options for port address, call and pickup groups   57   * Added support for T.140 realtime text in SIP/RTP   58   * The "checkmwi" option has been removed from sip.conf, as it is no longer   59      required due to the restructuring of how MWI is handled.  See the descriptions   60      in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf   61      for more information.   62   * Added rtpdest option to CHANNEL() dialplan function.   63   * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.   64     65 IAX2 changes   66 ------------   67   * Added the trunkmaxsize configuration option to chan_iax2.   68   * Added the srvlookup option to iax.conf   69   * Added support for OSP.  The token is set and retrieved through the CHANNEL()   70      dialplan function.   71     72 DUNDi changes   73 -------------   74   * Added the ability to specify arguments to the Dial application when using   75      the DUNDi switch in the dialplan.   76   * Added the ability to set weights for responses dynamically.  This can be   77      done using a global variable or a dialplan function.  Using the SHELL()   78      function would allow you to have an external script set the weight for   79      each response.   80   * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These   81      functions will allow you to initiate a DUNDi query from the dialplan,   82      find out how many results there are, and access each one.   83     84 ENUM changes85....235.....So you have to go Asterisk developer site (SVN repository) see the rest of the lines."Sneak peek at new features"

Source: snapvoip.blogspot.com

Following an article on Asterisk blog by Russell, "Sneak peek at new features" and provided a link to SVN reository. Following is a part of it (85 lines, there are 235 lines of description). …