Review Board 1.7.16


Support text messages outside of a call

Review Request #1042 - Created Dec. 1, 2010 and submitted

Russell Bryant
trunk
Reviewers
asterisk-dev
Asterisk
This branch contains a proposal for adding protocol independent support for processing text messages into and out of the dialplan, outside of a call.  The file doc/asterisk-messaging.txt contains more details on the proposal.  The introduction of the document is quoted here:

"    Asterisk has some limited support today for messaging.  The support that
exists primarily includes passing text messages in the context of a call.  The
SIP and IAX2 protocols have support for this, but that's it.

    There are a couple of other messaging protocols that are supported: Skype
and XMPP (Jabber).  The support of these is very minimal and not very integrated
into the architecture of Asterisk since these messages are not in the context of
a phone call.  They provide a combination of dialplan and manager interface
interfaces that are specific to each protocol.  There just is no current
architectural concept of dealing with text messages.

    The purpose of this proposal is to introduce text messaging into the
architecture of Asterisk.  For messaging support to exist in the true spirit of
Asterisk architecture, the design needs to achieve the following two goals:

    a) Protocol Independence
    b) Scriptable message routing

    The rest of this document goes through some details about how these goals
will be achieved in a way that is both architecturally compatible with Asterisk
as well as practical to implement."

----------

In addition to the documented proposal, I have made some good progress on implementation.  While the document includes some ideas for future enhancements, what is there so far should be usable.

 - core modifications to allow sending incoming messages through the dialplan
 - core modifications to allow outbound messages from the dialplan
 - modifications to res_jabber to allow inbound and outbound messages in the new architecture
 - changes to chan_sip to support inbound and outbound MESSAGE outside of a call
svn/testsuite/asterisk/team/russell/messaging:
  - This branch of the testsuite contains my tests for this branch, which include:
    - tests/sip/message_disabled
      - Ensure MESSAGE outside of a call is rejected when disabled.
    - tests/sip/message_unauth
      - When enabled, test sending a MESSAGE to Asterisk and send another back out from the dialplan.
    - tests/sip/message_auth
      - Same as the last test, but authenticate MESSAGE both inbound and outbound.
    - tests/sip/message_from_call
      - Set up a normal SIP call and send an out of call MESSAGE from the dialplan processing the call

I have also written some simple apps using the pjsua Python module from pjsip that can send and receive messages sent through Asterisk.

Lastly, I have done some manual testing of XMPP messages in and out of Asterisk using this code.
/trunk/CHANGES
Revision 321522 New Change
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==============================================================================
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==============================================================================
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===
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===
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=== This file documents the new and/or enhanced functionality added in
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=== This file documents the new and/or enhanced functionality added in
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=== the Asterisk versions listed below. This file does NOT include
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=== the Asterisk versions listed below. This file does NOT include
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=== changes in behavior that would not be backwards compatible with
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=== changes in behavior that would not be backwards compatible with
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=== previous versions; for that information see the UPGRADE.txt file
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=== previous versions; for that information see the UPGRADE.txt file
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=== and the other UPGRADE files for older releases.
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=== and the other UPGRADE files for older releases.
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===
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===
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==============================================================================
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==============================================================================
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
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--- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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Text Messaging

    
   
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--------------

    
   
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 * Asterisk now has protocol independent support for processing text messages

    
   
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   outside of a call.  Messages are routed through the Asterisk dialplan.

    
   
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   SIP MESSAGE and XMPP are currently supported.  There are options in

    
   
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   jabber.conf and sip.conf to allow enabling these features.

    
   
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     -> jabber.conf: see the "sendtodialplan" and "context" options.

    
   
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     -> sip.conf: see the "accept_outofcall_message" and "auth_message_requests"

    
   
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        options.

    
   
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   The MESSAGE() dialplan function and MessageSend() application have been

    
   
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   added to go along with this functionality.  More detailed usage information

    
   
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   can be found on the Asterisk wiki (http://wiki.asterisk.org/).

    
   
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Parking
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Parking
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-------
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-------
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 * parkedmusicclass can now be set for non-default parking lots.
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 * parkedmusicclass can now be set for non-default parking lots.
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 * ParkedCall application can now specify a specific parkinglot.
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 * ParkedCall application can now specify a specific parkinglot.
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Asterisk Manager Interface
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Asterisk Manager Interface
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--------------------------
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--------------------------
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 * PeerStatus now includes Address and Port.
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 * PeerStatus now includes Address and Port.
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 * Added Hold events for when the remote party puts the call on and off hold
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 * Added Hold events for when the remote party puts the call on and off hold
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   for chan_dahdi ISDN channels.
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   for chan_dahdi ISDN channels.
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 * Added new action MeetmeListRooms to list active conferences (shows same
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 * Added new action MeetmeListRooms to list active conferences (shows same
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   data as "meetme list" at the CLI).
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   data as "meetme list" at the CLI).
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 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
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 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
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   Description field that is set by 'description' in the channel configuration
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   Description field that is set by 'description' in the channel configuration
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   file.
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   file.
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 * Added Uniqueid header to UserEvent.
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 * Added Uniqueid header to UserEvent.
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Asterisk HTTP Server
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Asterisk HTTP Server
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--------------------------
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--------------------------
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 * The HTTP Server can bind to IPv6 addresses.
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 * The HTTP Server can bind to IPv6 addresses.
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chan_dahdi
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chan_dahdi
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--------------------------
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--------------------------
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 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
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 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
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   with busydetect.  usage example: busypattern=200,200,200,600
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   with busydetect.  usage example: busypattern=200,200,200,600
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CLI Changes
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CLI Changes
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--------------------------
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--------------------------
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 * New 'gtalk show settings' command showing the current settings loaded from
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 * New 'gtalk show settings' command showing the current settings loaded from
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   gtalk.conf.
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   gtalk.conf.
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 * The 'logger reload' command now supports an optional argument, specifying an
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 * The 'logger reload' command now supports an optional argument, specifying an
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   alternate configuration file to use.
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   alternate configuration file to use.
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 * 'dialplan add extension' command will now automatically create a context if
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 * 'dialplan add extension' command will now automatically create a context if
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   the specified context does not exist with a message indicated it did so.
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   the specified context does not exist with a message indicated it did so.
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 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
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 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
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   Description field which can be populated with 'description' in the channel
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   Description field which can be populated with 'description' in the channel
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   configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
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   configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
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CDR
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CDR
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--------------------------
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--------------------------
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 * The filter option in cdr_adaptive_odbc now supports negating the argument,
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 * The filter option in cdr_adaptive_odbc now supports negating the argument,
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   thus allowing records which do NOT match the specified filter.
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   thus allowing records which do NOT match the specified filter.
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CODECS
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CODECS
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--------------------------
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--------------------------
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 * Ability to define custom SILK formats in codecs.conf.
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 * Ability to define custom SILK formats in codecs.conf.
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 * Addition of speex32 audio format with translation.
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 * Addition of speex32 audio format with translation.
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ConfBridge
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ConfBridge
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--------------------------
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--------------------------
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 * New highly optimized and customizable ConfBridge application capable of
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 * New highly optimized and customizable ConfBridge application capable of
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   mixing audio at sample rates ranging from 8khz-96khz.
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   mixing audio at sample rates ranging from 8khz-96khz.
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 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
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 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
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   and bridge profiles on a channel.
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   and bridge profiles on a channel.
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Dialplan Variables
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Dialplan Variables
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------------------
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------------------
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 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
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 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
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   ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
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   ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
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   variables from asterisk.conf.
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   variables from asterisk.conf.
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Dialplan Functions
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Dialplan Functions
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------------------
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------------------
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 * Addition of the JITTERBUFFER dialplan function. This function allows
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 * Addition of the JITTERBUFFER dialplan function. This function allows
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   for jitterbuffering to occur on the read side of a channel.  By using
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   for jitterbuffering to occur on the read side of a channel.  By using
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   this function conference applications such as ConfBridge and MeetMe can
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   this function conference applications such as ConfBridge and MeetMe can
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   have the rx streams jitterbuffered before conference mixing occurs.
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   have the rx streams jitterbuffered before conference mixing occurs.
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 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
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 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
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   hierarchy.
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   hierarchy.
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 * Added STRREPLACE function.  This function let's the user search a variable
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 * Added STRREPLACE function.  This function let's the user search a variable
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   for a given string to replace with another string as many times as the
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   for a given string to replace with another string as many times as the
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   user specifies or just throughout the whole string.
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   user specifies or just throughout the whole string.
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 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
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 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
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libpri channel driver (chan_dahdi) DAHDI changes
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libpri channel driver (chan_dahdi) DAHDI changes
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--------------------------
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--------------------------
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 * Added moh_signaling option to specify what to do when the channel's bridged
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 * Added moh_signaling option to specify what to do when the channel's bridged
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   peer puts the ISDN channel on hold.
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   peer puts the ISDN channel on hold.
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 * Added display_send and display_receive options to control how the display ie
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 * Added display_send and display_receive options to control how the display ie
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   is handled.  To send display text from the dialplan use the SendText()
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   is handled.  To send display text from the dialplan use the SendText()
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   application when the option is enabled.
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   application when the option is enabled.
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 * Added mcid_send option to allow sending a MCID request on a span.
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 * Added mcid_send option to allow sending a MCID request on a span.
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Calendaring
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Calendaring
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--------------------------
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--------------------------
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 * Added setvar option to calendar.conf to allow setting channel variables on
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 * Added setvar option to calendar.conf to allow setting channel variables on
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   notification channels.
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   notification channels.
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 * Added "calendar show types" CLI command to list registered calendar
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 * Added "calendar show types" CLI command to list registered calendar
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   connectors.
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   connectors.
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MixMonitor
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MixMonitor
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--------------------------
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--------------------------
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 * Added two new options, r and t with file name arguments to record 
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 * Added two new options, r and t with file name arguments to record 
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   single direction (unmixed) audio recording separate from the bidirectional
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   single direction (unmixed) audio recording separate from the bidirectional
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   (mixed) recording.  The mixed file name argument is optional now as long
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   (mixed) recording.  The mixed file name argument is optional now as long
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   as at least one recording option is used.
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   as at least one recording option is used.
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FollowMe
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FollowMe
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--------------------------
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--------------------------
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 * Added a new option, l, which will disable local call optimization for
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 * Added a new option, l, which will disable local call optimization for
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   channels involved with the FollowMe thread.  Use this option to improve
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   channels involved with the FollowMe thread.  Use this option to improve
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   compatability for a FollowMe call with certain dialplan apps, options, and
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   compatability for a FollowMe call with certain dialplan apps, options, and
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   functions.
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   functions.
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CEL
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CEL
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--------------------------
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--------------------------
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 * cel_pgsql now supports the 'extra' column for data added using the
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 * cel_pgsql now supports the 'extra' column for data added using the
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   CELGenUserEvent() application.
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   CELGenUserEvent() application.
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pbx_lua
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pbx_lua
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--------------------------
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--------------------------
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 * Support for defining hints has been added to pbx_lua.  See the 'hints' table
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 * Support for defining hints has been added to pbx_lua.  See the 'hints' table
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   in the sample extensions.lua file for syntax details.
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   in the sample extensions.lua file for syntax details.
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 * Applications that perform jumps in the dialplan such as Goto will now
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 * Applications that perform jumps in the dialplan such as Goto will now
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   execute properly.  When pbx_lua detects that the context, extension, or
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   execute properly.  When pbx_lua detects that the context, extension, or
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   priority we are executing on has changed it will immediatly return control
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   priority we are executing on has changed it will immediatly return control
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   to the asterisk PBX engine.  Currently the engine cannot detect a Goto to
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   to the asterisk PBX engine.  Currently the engine cannot detect a Goto to
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   the priority after the currently executing priority.
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   the priority after the currently executing priority.
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 * An autoservice is now started by default for pbx_lua channels.  It can be
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 * An autoservice is now started by default for pbx_lua channels.  It can be
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   stopped and restarted using the autoservice_stop() and autoservice_start()
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   stopped and restarted using the autoservice_stop() and autoservice_start()
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   functions.
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   functions.
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
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--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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SIP Changes
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SIP Changes
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-----------
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-----------
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 * Added preferred_codec_only option in sip.conf. This feature limits the joint
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 * Added preferred_codec_only option in sip.conf. This feature limits the joint
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   codecs sent in response to an INVITE to the single most preferred codec.
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   codecs sent in response to an INVITE to the single most preferred codec.
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 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
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 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
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   to be used for the outgoing call. It must be one of the codecs configured
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   to be used for the outgoing call. It must be one of the codecs configured
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   for the device.
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   for the device.
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 * Added tlsprivatekey option to sip.conf.  This allows a separate .pem file
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 * Added tlsprivatekey option to sip.conf.  This allows a separate .pem file
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   to be used for holding a private key.  If tlsprivatekey is not specified,
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   to be used for holding a private key.  If tlsprivatekey is not specified,
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   tlscertfile is searched for both public and private key.
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   tlscertfile is searched for both public and private key.
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 * Added tlsclientmethod option to sip.conf.  This allows the protocol for
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 * Added tlsclientmethod option to sip.conf.  This allows the protocol for
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   outbound client connections to be specified.
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   outbound client connections to be specified.
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 * The sendrpid parameter has been expanded to include the options
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 * The sendrpid parameter has been expanded to include the options
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   'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
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   'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
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   header to be sent (equivalent to setting sendrpid=yes) and setting
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   header to be sent (equivalent to setting sendrpid=yes) and setting
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   sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
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   sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
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 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
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 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
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   is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
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   is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
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   since the call will fail if Asterisk does not process the incoming SDP, Asterisk
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   since the call will fail if Asterisk does not process the incoming SDP, Asterisk
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   will accept the SDP even if the SDP version number is not properly incremented,
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   will accept the SDP even if the SDP version number is not properly incremented,
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   but will generate a warning in the log indicating that the SIP peer that sent
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   but will generate a warning in the log indicating that the SIP peer that sent
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   the SDP should have the 'ignoresdpversion' option set.
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   the SDP should have the 'ignoresdpversion' option set.
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 * The 'nat' option has now been been changed to have yes, no, force_rport, and
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 * The 'nat' option has now been been changed to have yes, no, force_rport, and
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   comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
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   comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
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   symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
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   symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
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   remote side requests it and disables symmetric RTP support. Setting it to
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   remote side requests it and disables symmetric RTP support. Setting it to
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   force_rport forces RFC 3581 behavior and disables symmetric RTP support.
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   force_rport forces RFC 3581 behavior and disables symmetric RTP support.
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   Setting it to comedia enables RFC 3581 behavior if the remote side requests it
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   Setting it to comedia enables RFC 3581 behavior if the remote side requests it
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   and enables symmetric RTP support.
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   and enables symmetric RTP support.
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 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
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 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
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   response.  This permits the master channel to know how each channel dialled
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   response.  This permits the master channel to know how each channel dialled
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   in a multi-channel setup resolved in an individual way.
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   in a multi-channel setup resolved in an individual way.
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 * Added 'externtcpport' and 'externtlsport' options to allow custom port
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 * Added 'externtcpport' and 'externtlsport' options to allow custom port
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   configuration for the externip and externhost options when tcp or tls is used.
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   configuration for the externip and externhost options when tcp or tls is used.
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 * Added support for message body (stored in content variable) to SIP NOTIFY message
188
 * Added support for message body (stored in content variable) to SIP NOTIFY message
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   accessible via AMI and CLI.
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   accessible via AMI and CLI.
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 * Added 'media_address' configuration option which can be used to explicitly specify
190
 * Added 'media_address' configuration option which can be used to explicitly specify
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   the IP address to use in the SDP for media (audio, video, and text) streams.
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   the IP address to use in the SDP for media (audio, video, and text) streams.
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 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
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 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
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   that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
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   that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
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   received.
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   received.
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 * Added 'use_q850_reason' configuration option for generating and parsing
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 * Added 'use_q850_reason' configuration option for generating and parsing
183
   if available  Reason: Q.850;cause=<cause code> header. It is implemented
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   if available  Reason: Q.850;cause=<cause code> header. It is implemented
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   in some gateways for better passing PRI/SS7 cause codes via SIP.
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   in some gateways for better passing PRI/SS7 cause codes via SIP.
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 * When dialing SIP peers, a new component may be added to the end of the dialstring
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 * When dialing SIP peers, a new component may be added to the end of the dialstring
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   to indicate that a specific remote IP address or host should be used when dialing
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   to indicate that a specific remote IP address or host should be used when dialing
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   the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
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   the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
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 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
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 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
189
   ability to selectively force bridged channels to also be encrypted is also
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   ability to selectively force bridged channels to also be encrypted is also
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   implemented. Branching in the dialplan can be done based on whether or not
203
   implemented. Branching in the dialplan can be done based on whether or not
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   a channel has secure media and/or signaling.
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   a channel has secure media and/or signaling.
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 * Added directmediapermit/directmediadeny to limit which peers can send direct media
205
 * Added directmediapermit/directmediadeny to limit which peers can send direct media
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   to each other
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   to each other
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 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
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 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
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   Charge messages to snom phones.
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   Charge messages to snom phones.
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 * Added support for G.719 media streams.
209
 * Added support for G.719 media streams.
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 * Added support for 16khz signed linear media streams.
210
 * Added support for 16khz signed linear media streams.
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 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
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 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
199
   RTP has been outfitted with the same abilities.
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   RTP has been outfitted with the same abilities.
200
 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
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 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
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   available in device configurations as well as in the dial plan.
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   available in device configurations as well as in the dial plan.
202
 * Addition of the 'subscribe_network_change' option for turning on and off
215
 * Addition of the 'subscribe_network_change' option for turning on and off
203
   res_stun_monitor module support in chan_sip.
216
   res_stun_monitor module support in chan_sip.
204
 * Addition of the 'auth_options_requests' option for turning on and off
217
 * Addition of the 'auth_options_requests' option for turning on and off
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   authentication for OPTIONS requests in chan_sip.
218
   authentication for OPTIONS requests in chan_sip.
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 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
219
 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
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IAX2 Changes
222
IAX2 Changes
210
-----------
223
-----------
211
 * Added rtsavesysname option into iax.conf to allow the systname to be saved
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 * Added rtsavesysname option into iax.conf to allow the systname to be saved
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   on realtime updates.
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   on realtime updates.
213
 * Added the ability for chan_iax2 to inform the dialplan whether or not
226
 * Added the ability for chan_iax2 to inform the dialplan whether or not
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   encryption is being used. This interoperates with the SIP SRTP implementation
227
   encryption is being used. This interoperates with the SIP SRTP implementation
215
   so that a secure SIP call can be bridged to a secure IAX call when the
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   so that a secure SIP call can be bridged to a secure IAX call when the
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   dialplan requires bridged channels to be "secure".
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   dialplan requires bridged channels to be "secure".
217
 * Addition of the 'subscribe_network_change' option for turning on and off
230
 * Addition of the 'subscribe_network_change' option for turning on and off
218
   res_stun_monitor module support in chan_iax.
231
   res_stun_monitor module support in chan_iax.
219

    
   
232

   
220

    
   
233

   
221
MGCP Changes
234
MGCP Changes
222
------------
235
------------
223
 * Added ability to preset channel variables on indicated lines with the setvar
236
 * Added ability to preset channel variables on indicated lines with the setvar
224
   configuration option.  Also, clearvars=all resets the list of variables back
237
   configuration option.  Also, clearvars=all resets the list of variables back
225
   to none.
238
   to none.
226
 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
239
 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
227
   See configs/res_pktccops.conf for more information.
240
   See configs/res_pktccops.conf for more information.
228

    
   
241

   
229
XMPP Google Talk/Jingle changes
242
XMPP Google Talk/Jingle changes
230
-------------------------------
243
-------------------------------
231
  * Added the externip option to gtalk.conf.
244
  * Added the externip option to gtalk.conf.
232
  * Added the stunaddr option to gtalk.conf which allows for the automatic
245
  * Added the stunaddr option to gtalk.conf which allows for the automatic
233
    retrieval of the external ip from a stun server.
246
    retrieval of the external ip from a stun server.
234

    
   
247

   
235
Applications
248
Applications
236
------------
249
------------
237
 * Added 'p' option to PickupChan() to allow for picking up channel by the first
250
 * Added 'p' option to PickupChan() to allow for picking up channel by the first
238
   match to a partial channel name.
251
   match to a partial channel name.
239
 * Added .m3u support for Mp3Player application.
252
 * Added .m3u support for Mp3Player application.
240
 * Added progress option to the app_dial D() option.  When progress DTMF is
253
 * Added progress option to the app_dial D() option.  When progress DTMF is
241
   present, those values are sent immediately upon receiving a PROGRESS message
254
   present, those values are sent immediately upon receiving a PROGRESS message
242
   regardless if the call has been answered or not.
255
   regardless if the call has been answered or not.
243
 * Added functionality to the app_dial F() option to continue with execution
256
 * Added functionality to the app_dial F() option to continue with execution
244
   at the current location when no parameters are provided.
257
   at the current location when no parameters are provided.
245
 * Added the 'a' option to app_dial to answer the calling channel before any
258
 * Added the 'a' option to app_dial to answer the calling channel before any
246
   announcements or macros are executed.
259
   announcements or macros are executed.
247
 * Modified app_dial to set answertime when the called channel answers even if
260
 * Modified app_dial to set answertime when the called channel answers even if
248
   the called channel hangs up during playback of an announcement.
261
   the called channel hangs up during playback of an announcement.
249
 * Modified app_dial 'r' option to support an additional parameter to play an
262
 * Modified app_dial 'r' option to support an additional parameter to play an
250
   indication tone from indications.conf
263
   indication tone from indications.conf
251
 * Added c() option to app_chanspy. This option allows custom DTMF to be set
264
 * Added c() option to app_chanspy. This option allows custom DTMF to be set
252
   to cycle through the next available channel.  By default this is still '*'.
265
   to cycle through the next available channel.  By default this is still '*'.
253
 * Added x() option to app_chanspy.  This option allows DTMF to be set to
266
 * Added x() option to app_chanspy.  This option allows DTMF to be set to
254
   exit the application.
267
   exit the application.
255
 * The Voicemail application has been improved to automatically ignore messages
268
 * The Voicemail application has been improved to automatically ignore messages
256
   that only contain silence.
269
   that only contain silence.
257
 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
270
 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
258
   associated mailbox(es) to be greetings-only.
271
   associated mailbox(es) to be greetings-only.
259
 * The ChanSpy application now has the 'S' option, which makes the application
272
 * The ChanSpy application now has the 'S' option, which makes the application
260
   automatically exit once it hits a point where no more channels are available
273
   automatically exit once it hits a point where no more channels are available
261
   to spy on.
274
   to spy on.
262
 * The ChanSpy application also now has the 'E' option, which spies on a single
275
 * The ChanSpy application also now has the 'E' option, which spies on a single
263
   channel and exits when that channel hangs up.
276
   channel and exits when that channel hangs up.
264
 * The MeetMe application now turns on the DENOISE() function by default, for
277
 * The MeetMe application now turns on the DENOISE() function by default, for
265
   each participant.  In our tests, this has significantly decreased background
278
   each participant.  In our tests, this has significantly decreased background
266
   noise (especially noisy data centers).
279
   noise (especially noisy data centers).
267
 * Voicemail now permits storage of secrets in a separate file, located in the
280
 * Voicemail now permits storage of secrets in a separate file, located in the
268
   spool directory of each individual user.  The control for this is located in
281
   spool directory of each individual user.  The control for this is located in
269
   the "passwordlocation" option in voicemail.conf.  Please see the sample
282
   the "passwordlocation" option in voicemail.conf.  Please see the sample
270
   configuration for more information.
283
   configuration for more information.
271
 * The ChanIsAvail application now exposes the returned cause code using a separate
284
 * The ChanIsAvail application now exposes the returned cause code using a separate
272
   variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
285
   variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
273
 * Added 'd' option to app_followme.  This option disables the "Please hold"
286
 * Added 'd' option to app_followme.  This option disables the "Please hold"
274
   announcement.
287
   announcement.
275
 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
288
 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
276
   received will terminate recording.
289
   received will terminate recording.
277
 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
290
 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
278
   Previously the folder could only be set per context, but has now been extended 
291
   Previously the folder could only be set per context, but has now been extended 
279
   using the imapfolder option.
292
   using the imapfolder option.
280
 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
293
 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
281
 * Voicemail now allows the pager date format to be specified separately from the
294
 * Voicemail now allows the pager date format to be specified separately from the
282
   email date format.
295
   email date format.
283
 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
296
 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
284
   to allow joining, leaving, and sending text to group chats.
297
   to allow joining, leaving, and sending text to group chats.
285
 * MeetMe has a new option 'G' to play an announcement before joining a conference.
298
 * MeetMe has a new option 'G' to play an announcement before joining a conference.
286
 * Page has a new option 'A(x)' which will playback an announcement simultaneously
299
 * Page has a new option 'A(x)' which will playback an announcement simultaneously
287
   to all paged phones (and optionally excluding the caller's one using the new
300
   to all paged phones (and optionally excluding the caller's one using the new
288
   option 'n') before the call is bridged.
301
   option 'n') before the call is bridged.
289
 * The 'f' option to Dial has been augmented to take an optional argument. If no
302
 * The 'f' option to Dial has been augmented to take an optional argument. If no
290
   argument is provided, the 'f' option works as it always has. If an argument is
303
   argument is provided, the 'f' option works as it always has. If an argument is
291
   provided, then the connected party information of all outgoing channels created
304
   provided, then the connected party information of all outgoing channels created
292
   during the Dial will be set to the argument passed to the 'f' option.
305
   during the Dial will be set to the argument passed to the 'f' option.
293
 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
306
 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
294
   Gosub on the peer.
307
   Gosub on the peer.
295
 * The OSP lookup application adds in/outbound network ID, optional security,
308
 * The OSP lookup application adds in/outbound network ID, optional security,
296
   number portability, QoS reporting, destination IP port, custom info and service
309
   number portability, QoS reporting, destination IP port, custom info and service
297
   type features.
310
   type features.
298
 * Added new application VMSayName that will play the recorded name of the voicemail
311
 * Added new application VMSayName that will play the recorded name of the voicemail
299
   user if it exists, otherwise will play the mailbox number.
312
   user if it exists, otherwise will play the mailbox number.
300
 * Added custom device states to ConfBridge bridges.  Use 'confbridge:<name>' to
313
 * Added custom device states to ConfBridge bridges.  Use 'confbridge:<name>' to
301
   retrieve state for a particular bridge, where <name> is the conference name
314
   retrieve state for a particular bridge, where <name> is the conference name
302
 * app_directory now allows exiting at any time using the operator or pound key.
315
 * app_directory now allows exiting at any time using the operator or pound key.
303
 * Voicemail now supports setting a locale per-mailbox.
316
 * Voicemail now supports setting a locale per-mailbox.
304
 * Two new applications are provided for declining counting phrases in multiple
317
 * Two new applications are provided for declining counting phrases in multiple
305
   languages.  See the application notes for SayCountedNoun and SayCountedAdj for
318
   languages.  See the application notes for SayCountedNoun and SayCountedAdj for
306
   more information.
319
   more information.
307
 * Voicemail now runs the externnotify script when pollmailboxes is activated and
320
 * Voicemail now runs the externnotify script when pollmailboxes is activated and
308
   notices a change.
321
   notices a change.
309
 * Voicemail now includes rdnis within msgXXXX.txt file.
322
 * Voicemail now includes rdnis within msgXXXX.txt file.
310
 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
323
 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
311
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
324
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
312
   a MeetMe conference
325
   a MeetMe conference
313
 * Added ability to include '@parkinglot' to ParkedCall extension in order to specify
326
 * Added ability to include '@parkinglot' to ParkedCall extension in order to specify
314
   a specific parkinglot on which to search the extension.
327
   a specific parkinglot on which to search the extension.
315

    
   
328

   
316
Dialplan Functions
329
Dialplan Functions
317
------------------
330
------------------
318
 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
331
 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
319
   over SRV records associated with a specific service. From the CLI, type
332
   over SRV records associated with a specific service. From the CLI, type
320
   'core show function SRVQUERY' and 'core show function SRVRESULT' for more
333
   'core show function SRVQUERY' and 'core show function SRVRESULT' for more
321
   details on how these may be used.
334
   details on how these may be used.
322
 * PITCH_SHIFT dialplan function added. This function can be used to modify the
335
 * PITCH_SHIFT dialplan function added. This function can be used to modify the
323
   pitch of a channel's tx and rx audio streams.
336
   pitch of a channel's tx and rx audio streams.
324
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
337
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
325
   setting various connected line and redirecting party information.
338
   setting various connected line and redirecting party information.
326
 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
339
 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
327
   support ISDN subaddressing.
340
   support ISDN subaddressing.
328
 * The CHANNEL() function now supports the "name" and "checkhangup" options.
341
 * The CHANNEL() function now supports the "name" and "checkhangup" options.
329
 * For DAHDI channels, the CHANNEL() dialplan function now allows
342
 * For DAHDI channels, the CHANNEL() dialplan function now allows
330
   the dialplan to request changes in the configuration of the active
343
   the dialplan to request changes in the configuration of the active
331
   echo canceller on the channel (if any), for the current call only.
344
   echo canceller on the channel (if any), for the current call only.
332
   The syntax is:
345
   The syntax is:
333

    
   
346

   
334
   exten => s,n,Set(CHANNEL(echocan_mode)=off)
347
   exten => s,n,Set(CHANNEL(echocan_mode)=off)
335

    
   
348

   
336
   The possible values are:
349
   The possible values are:
337

    
   
350

   
338
     on - normal mode (the echo canceller is actually reinitialized)
351
     on - normal mode (the echo canceller is actually reinitialized)
339
     off - disabled
352
     off - disabled
340
     fax - FAX/data mode (NLP disabled if possible, otherwise completely
353
     fax - FAX/data mode (NLP disabled if possible, otherwise completely
341
           disabled)
354
           disabled)
342
     voice - voice mode (returns from FAX mode, reverting the changes that
355
     voice - voice mode (returns from FAX mode, reverting the changes that
343
             were made when FAX mode was requested)
356
             were made when FAX mode was requested)
344
 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
357
 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
345
   and setting variables on the channel which created the current channel.
358
   and setting variables on the channel which created the current channel.
346
   Administrators should take care to avoid naming conflicts, when multiple
359
   Administrators should take care to avoid naming conflicts, when multiple
347
   channels are dialled at once, especially when used with the Local channel
360
   channels are dialled at once, especially when used with the Local channel
348
   construct (which all could set variables on the master channel).  Usage
361
   construct (which all could set variables on the master channel).  Usage
349
   of the HASH() dialplan function, with the key set to the name of the slave
362
   of the HASH() dialplan function, with the key set to the name of the slave
350
   channel, is one approach that will avoid conflicts.
363
   channel, is one approach that will avoid conflicts.
351
 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
364
 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
352
   audio in a channel.
365
   audio in a channel.
353
 * func_odbc now allows multiple row results to be retrieved without using
366
 * func_odbc now allows multiple row results to be retrieved without using
354
   mode=multirow.  If rowlimit is set, then additional rows may be retrieved
367
   mode=multirow.  If rowlimit is set, then additional rows may be retrieved
355
   from the same query by using the name of the function which retrieved the
368
   from the same query by using the name of the function which retrieved the
356
   first row as an argument to ODBC_FETCH().
369
   first row as an argument to ODBC_FETCH().
357
 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
370
 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
358
   dialplan. This function returns the content of the received message.
371
   dialplan. This function returns the content of the received message.
359
 * Added REPLACE, which searches a given variable name for a set of characters,
372
 * Added REPLACE, which searches a given variable name for a set of characters,
360
   then either replaces them with a single character or deletes them.
373
   then either replaces them with a single character or deletes them.
361
 * Added PASSTHRU, which literally passes the same argument back as its return
374
 * Added PASSTHRU, which literally passes the same argument back as its return
362
   value.  The intent is to be able to use a literal string argument to
375
   value.  The intent is to be able to use a literal string argument to
363
   functions that currently require a variable name as an argument.
376
   functions that currently require a variable name as an argument.
364
 * HASH-associated variables now can be inherited across channel creation, by
377
 * HASH-associated variables now can be inherited across channel creation, by
365
   prefixing the name of the hash at assignment with the appropriate number of
378
   prefixing the name of the hash at assignment with the appropriate number of
366
   underscores, just like variables.
379
   underscores, just like variables.
367
 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
380
 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
368
 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
381
 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
369
   whether or not channels that are bridged to the current channel will be
382
   whether or not channels that are bridged to the current channel will be
370
   required to have secure signaling and/or media.
383
   required to have secure signaling and/or media.
371
 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
384
 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
372
   the current channel has secure signaling and/or media.
385
   the current channel has secure signaling and/or media.
373
 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
386
 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
374
   "no_media_path" option.
387
   "no_media_path" option.
375
   Returns "0" if there is a B channel associated with the call.
388
   Returns "0" if there is a B channel associated with the call.
376
   Returns "1" if no B channel is associated with the call.  The call is either
389
   Returns "1" if no B channel is associated with the call.  The call is either
377
   on hold or is a call waiting call.
390
   on hold or is a call waiting call.
378
 * Added option to dialplan function CDR(), the 'f' option
391
 * Added option to dialplan function CDR(), the 'f' option
379
   allows for high resolution times for billsec and duration fields.
392
   allows for high resolution times for billsec and duration fields.
380
 * FILE() now supports line-mode and writing.
393
 * FILE() now supports line-mode and writing.
381
 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
394
 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
382
 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
395
 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
383

    
   
396

   
384
Dialplan Variables
397
Dialplan Variables
385
------------------
398
------------------
386
 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
399
 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
387
 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
400
 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
388
   and is set when a dynamic feature is triggered.
401
   and is set when a dynamic feature is triggered.
389
 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
402
 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
390
   to dynamically create a new parking lot matching the value this varible is
403
   to dynamically create a new parking lot matching the value this varible is
391
   set to.
404
   set to.
392
 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
405
 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
393
   features.conf that should be the base for dynamic parkinglots.
406
   features.conf that should be the base for dynamic parkinglots.
394
 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
407
 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
395
   parkinglot should have.
408
   parkinglot should have.
396
 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
409
 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
397
   should have.
410
   should have.
398

    
   
411

   
399
Queue changes
412
Queue changes
400
-------------
413
-------------
401
 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
414
 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
402
   timeout has expired.
415
   timeout has expired.
403
 * Added 'R' option to app_queue.  This option stops moh and indicates ringing
416
 * Added 'R' option to app_queue.  This option stops moh and indicates ringing
404
   to the caller when an Agent's phone is ringing.  This can be used to indicate
417
   to the caller when an Agent's phone is ringing.  This can be used to indicate
405
   to the caller that their call is about to be picked up, which is nice when
418
   to the caller that their call is about to be picked up, which is nice when
406
   one has been on hold for an extened period of time.
419
   one has been on hold for an extened period of time.
407
 * A new config option, penaltymemberslimit, has been added to queues.conf.
420
 * A new config option, penaltymemberslimit, has been added to queues.conf.
408
   When set this option will disregard penalty settings when a queue has too
421
   When set this option will disregard penalty settings when a queue has too
409
   few members.
422
   few members.
410
 * A new option, 'I' has been added to both app_queue and app_dial.
423
 * A new option, 'I' has been added to both app_queue and app_dial.
411
   By setting this option, Asterisk will not update the caller with
424
   By setting this option, Asterisk will not update the caller with
412
   connected line changes or redirecting party changes when they occur.
425
   connected line changes or redirecting party changes when they occur.
413
 * A 'relative-peroidic-announce' option has been added to queues.conf.  When
426
 * A 'relative-peroidic-announce' option has been added to queues.conf.  When
414
   enabled, this option will cause periodic announce times to be calculated
427
   enabled, this option will cause periodic announce times to be calculated
415
   from the end of announcements rather than from the beginning.
428
   from the end of announcements rather than from the beginning.
416
 * The autopause option in queues.conf can be passed a new value, "all." The
429
 * The autopause option in queues.conf can be passed a new value, "all." The
417
   result is that if a member becomes auto-paused, he will be paused in all
430
   result is that if a member becomes auto-paused, he will be paused in all
418
   queues for which he is a member, not just the queue that failed to reach
431
   queues for which he is a member, not just the queue that failed to reach
419
   the member.
432
   the member.
420
 * Added dialplan function QUEUE_EXISTS to check if a queue exists
433
 * Added dialplan function QUEUE_EXISTS to check if a queue exists
421
 * The queue logger now allows events to optionally propagate to a file,
434
 * The queue logger now allows events to optionally propagate to a file,
422
   even when realtime logging is turned on.  Additionally, realtime logging
435
   even when realtime logging is turned on.  Additionally, realtime logging
423
   supports sending the event arguments to 5 individual fields, although it
436
   supports sending the event arguments to 5 individual fields, although it
424
   will fallback to the previous data definition, if the new table layout is
437
   will fallback to the previous data definition, if the new table layout is
425
   not found.
438
   not found.
426

    
   
439

   
427
mISDN channel driver (chan_misdn) changes
440
mISDN channel driver (chan_misdn) changes
428
----------------------------------------
441
----------------------------------------
429
 * Added display_connected parameter to misdn.conf to put a display string
442
 * Added display_connected parameter to misdn.conf to put a display string
430
   in the CONNECT message containing the connected name and/or number if
443
   in the CONNECT message containing the connected name and/or number if
431
   the presentation setting permits it.
444
   the presentation setting permits it.
432
 * Added display_setup parameter to misdn.conf to put a display string
445
 * Added display_setup parameter to misdn.conf to put a display string
433
   in the SETUP message containing the caller name and/or number if the
446
   in the SETUP message containing the caller name and/or number if the
434
   presentation setting permits it.
447
   presentation setting permits it.
435
 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
448
 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
436
   indicate the dialplan settings are to be obtained from the asterisk
449
   indicate the dialplan settings are to be obtained from the asterisk
437
   channel.
450
   channel.
438
 * Made misdn.conf parameter callerid accept the "name" <number> format
451
 * Made misdn.conf parameter callerid accept the "name" <number> format
439
   used by the rest of the system.
452
   used by the rest of the system.
440
 * Made use the nationalprefix and internationalprefix misdn.conf
453
 * Made use the nationalprefix and internationalprefix misdn.conf
441
   parameters to prefix any received number from the ISDN link if that
454
   parameters to prefix any received number from the ISDN link if that
442
   number has the corresponding Type-Of-Number.  NOTE:  This includes
455
   number has the corresponding Type-Of-Number.  NOTE:  This includes
443
   comparing the incoming call's dialed number against the MSN list.
456
   comparing the incoming call's dialed number against the MSN list.
444
 * Added the following new parameters: unknownprefix, netspecificprefix,
457
 * Added the following new parameters: unknownprefix, netspecificprefix,
445
   subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
458
   subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
446
   received number from the ISDN link if that number has the corresponding
459
   received number from the ISDN link if that number has the corresponding
447
   Type-Of-Number.
460
   Type-Of-Number.
448
 * Added new dialplan application misdn_command which permits controlling
461
 * Added new dialplan application misdn_command which permits controlling
449
   the CCBS/CCNR functionality.
462
   the CCBS/CCNR functionality.
450
 * Added new dialplan function mISDN_CC which permits retrieval of various
463
 * Added new dialplan function mISDN_CC which permits retrieval of various
451
   values from an active call completion record.
464
   values from an active call completion record.
452
 * For PTP, you should manually send the COLR of the redirected-to party
465
 * For PTP, you should manually send the COLR of the redirected-to party
453
   for an incomming redirected call if the incoming call could experience
466
   for an incomming redirected call if the incoming call could experience
454
   further redirects.  Just set the REDIRECTING(to-num,i) = ${EXTEN} and
467
   further redirects.  Just set the REDIRECTING(to-num,i) = ${EXTEN} and
455
   set the REDIRECTING(to-pres) to the COLR.  A call has been redirected
468
   set the REDIRECTING(to-pres) to the COLR.  A call has been redirected
456
   if the REDIRECTING(from-num) is not empty.
469
   if the REDIRECTING(from-num) is not empty.
457
 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
470
 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
458
   option on all of the REDIRECTING statements before dialing the
471
   option on all of the REDIRECTING statements before dialing the
459
   redirected-to party.  You still have to set the REDIRECTING(to-xxx,i)
472
   redirected-to party.  You still have to set the REDIRECTING(to-xxx,i)
460
   and the REDIRECTING(from-xxx,i) values.  The PTP call will update the
473
   and the REDIRECTING(from-xxx,i) values.  The PTP call will update the
461
   redirecting-to presentation (COLR) when it becomes available.
474
   redirecting-to presentation (COLR) when it becomes available.
462
 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
475
 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
463
   information.
476
   information.
464

    
   
477

   
465
thirdparty mISDN enhancements
478
thirdparty mISDN enhancements
466
-----------------------------
479
-----------------------------
467
mISDN has been modified by Digium, Inc. to greatly expand facility message
480
mISDN has been modified by Digium, Inc. to greatly expand facility message
468
support to allow:
481
support to allow:
469
  * Enhanced COLP support for call diversion and transfer.
482
  * Enhanced COLP support for call diversion and transfer.
470
  * CCBS/CCNR support.
483
  * CCBS/CCNR support.
471

    
   
484

   
472
The latest modified mISDN v1.1.x based version is available at:
485
The latest modified mISDN v1.1.x based version is available at:
473
http://svn.digium.com/svn/thirdparty/mISDN/trunk
486
http://svn.digium.com/svn/thirdparty/mISDN/trunk
474
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
487
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
475

    
   
488

   
476
Tagged versions of the modified mISDN code are available under:
489
Tagged versions of the modified mISDN code are available under:
477
http://svn.digium.com/svn/thirdparty/mISDN/tags
490
http://svn.digium.com/svn/thirdparty/mISDN/tags
478
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
491
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
479

    
   
492

   
480
libpri channel driver (chan_dahdi) DAHDI changes
493
libpri channel driver (chan_dahdi) DAHDI changes
481
-------------------------------------------
494
-------------------------------------------
482
 * The channel variable PRIREDIRECTREASON is now just a status variable
495
 * The channel variable PRIREDIRECTREASON is now just a status variable
483
   and it is also deprecated.  Use the REDIRECTING(reason) dialplan function
496
   and it is also deprecated.  Use the REDIRECTING(reason) dialplan function
484
   to read and alter the reason.
497
   to read and alter the reason.
485
 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
498
 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
486
   redirected-to party for an incomming redirected call if the incoming call
499
   redirected-to party for an incomming redirected call if the incoming call
487
   could experience further redirects.  Just set the
500
   could experience further redirects.  Just set the
488
   REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
501
   REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
489
   to the COLR.  A call has been redirected if the REDIRECTING(count) is not
502
   to the COLR.  A call has been redirected if the REDIRECTING(count) is not
490
   zero.
503
   zero.
491
 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
504
 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
492
   use the inhibit(i) option on all of the REDIRECTING statements before
505
   use the inhibit(i) option on all of the REDIRECTING statements before
493
   dialing the redirected-to party.  You still have to set the
506
   dialing the redirected-to party.  You still have to set the
494
   REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values.  The call
507
   REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values.  The call
495
   will update the redirecting-to presentation (COLR) when it becomes available.
508
   will update the redirecting-to presentation (COLR) when it becomes available.
496
 * Added the ability to ignore calls that are not in a Multiple Subscriber
509
 * Added the ability to ignore calls that are not in a Multiple Subscriber
497
   Number (MSN) list for PTMP CPE interfaces.
510
   Number (MSN) list for PTMP CPE interfaces.
498
 * Added dynamic range compression support for dahdi channels.  It is
511
 * Added dynamic range compression support for dahdi channels.  It is
499
   configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
512
   configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
500
 * Added support for ISDN calling and called subaddress with partial support
513
 * Added support for ISDN calling and called subaddress with partial support
501
   for connected line subaddress.
514
   for connected line subaddress.
502
 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
515
 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
503
 * Added handling of received HOLD/RETRIEVE messages and the optional ability
516
 * Added handling of received HOLD/RETRIEVE messages and the optional ability
504
   to transfer a held call on disconnect similar to an analog phone.
517
   to transfer a held call on disconnect similar to an analog phone.
505
 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
518
 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
506
   Will reroute/deflect an outgoing call when receive the message.
519
   Will reroute/deflect an outgoing call when receive the message.
507
   Can use the DAHDISendCallreroutingFacility to send the message for the
520
   Can use the DAHDISendCallreroutingFacility to send the message for the
508
   supported switches.
521
   supported switches.
509
 * Added standard location to add options to chan_dahdi dialing:
522
 * Added standard location to add options to chan_dahdi dialing:
510
   Dial(DAHDI/g1[/extension[/options]])
523
   Dial(DAHDI/g1[/extension[/options]])
511
   Current options:
524
   Current options:
512
   K(<keypad_digits>)
525
   K(<keypad_digits>)
513
   R Reverse charging indication
526
   R Reverse charging indication
514
 * Added Reverse Charging Indication (Collect calls) send/receive option.
527
 * Added Reverse Charging Indication (Collect calls) send/receive option.
515
   Send reverse charging in SETUP message with the chan_dahdi R dialing option.
528
   Send reverse charging in SETUP message with the chan_dahdi R dialing option.
516
   Dial(DAHDI/g1/extension/R)
529
   Dial(DAHDI/g1/extension/R)
517
   Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
530
   Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
518
   (requires latest LibPRI)
531
   (requires latest LibPRI)
519
 * Added ability to send/receive keypad digits in the SETUP message.
532
 * Added ability to send/receive keypad digits in the SETUP message.
520
   Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
533
   Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
521
   dialing option.  Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
534
   dialing option.  Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
522
   Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
535
   Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
523
   (requires latest LibPRI)
536
   (requires latest LibPRI)
524
 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
537
 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
525
   to eliminate tromboned calls.  A tromboned call goes out an interface and comes
538
   to eliminate tromboned calls.  A tromboned call goes out an interface and comes
526
   back into the same interface.  Tromboned calls happen because of call routing,
539
   back into the same interface.  Tromboned calls happen because of call routing,
527
   call deflection, call forwarding, and call transfer.
540
   call deflection, call forwarding, and call transfer.
528
 * Added the ability to send and receive ETSI Advice-Of-Charge messages. 
541
 * Added the ability to send and receive ETSI Advice-Of-Charge messages. 
529
 * Added the ability to support call waiting calls.  (The SETUP has no B channel
542
 * Added the ability to support call waiting calls.  (The SETUP has no B channel
530
   assigned.)
543
   assigned.)
531
 * Added Malicious Call ID (MCID) event to the AMI call event class.
544
 * Added Malicious Call ID (MCID) event to the AMI call event class.
532
 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
545
 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
533

    
   
546

   
534
Asterisk Manager Interface
547
Asterisk Manager Interface
535
--------------------------
548
--------------------------
536
 * The Hangup action now accepts a Cause header which may be used to
549
 * The Hangup action now accepts a Cause header which may be used to
537
   set the channel's hangup cause.
550
   set the channel's hangup cause.
538
 * sslprivatekey option added to manager.conf and http.conf.  Adds the ability
551
 * sslprivatekey option added to manager.conf and http.conf.  Adds the ability
539
   to specify a separate .pem file to hold a private key.  By default sslcert
552
   to specify a separate .pem file to hold a private key.  By default sslcert
540
   is used to hold both the public and private key.
553
   is used to hold both the public and private key.
541
 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
554
 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
542
   for options containing the 'tls' prefix.  For example, 'sslenable' is now
555
   for options containing the 'tls' prefix.  For example, 'sslenable' is now
543
   'tlsenable'.  This has been done in effort to keep ssl and tls options consistent
556
   'tlsenable'.  This has been done in effort to keep ssl and tls options consistent
544
   across all .conf files. All affected sample.conf files have been modified to
557
   across all .conf files. All affected sample.conf files have been modified to
545
   reflect this change.  Previous options such as 'sslenable' still work,
558
   reflect this change.  Previous options such as 'sslenable' still work,
546
   but options with the 'tls' prefix are preferred.
559
   but options with the 'tls' prefix are preferred.
547
 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
560
 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
548
   in a channel. (res_mutestream.so)
561
   in a channel. (res_mutestream.so)
549
 * The configuration file manager.conf now supports a channelvars option, which
562
 * The configuration file manager.conf now supports a channelvars option, which
550
   specifies a list of channel variables to include in each channel-oriented
563
   specifies a list of channel variables to include in each channel-oriented
551
   event.
564
   event.
552
 * The redirect command now has new parameters ExtraContext, ExtraExtension, 
565
 * The redirect command now has new parameters ExtraContext, ExtraExtension, 
553
   and ExtraPriority to allow redirecting the second channel to a different
566
   and ExtraPriority to allow redirecting the second channel to a different
554
   location than the first.
567
   location than the first.
555
 * Added new event "JabberStatus" in the Jabber module to monitor buddies
568
 * Added new event "JabberStatus" in the Jabber module to monitor buddies
556
   status.
569
   status.
557
 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
570
 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
558
   in a MixMonitor recording.
571
   in a MixMonitor recording.
559
 * The 'iax2 show peers' output is now similar to the expected output of
572
 * The 'iax2 show peers' output is now similar to the expected output of
560
   'sip show peers'.
573
   'sip show peers'.
561
 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
574
 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
562
   aoc event class.
575
   aoc event class.
563
 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
576
 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
564
   AOC-E messages on a channel.
577
   AOC-E messages on a channel.
565
 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
578
 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
566
   conform more closely to similar events.
579
   conform more closely to similar events.
567
 * Added a new eventfilter option per user to allow whitelisting and blacklisting
580
 * Added a new eventfilter option per user to allow whitelisting and blacklisting
568
   of events.
581
   of events.
569
 * Added optional parkinglot variable for park command.
582
 * Added optional parkinglot variable for park command.
570
 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
583
 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
571
   if CallerIDNum and CallerIDName headers are also present.
584
   if CallerIDNum and CallerIDName headers are also present.
572

    
   
585

   
573
Channel Event Logging
586
Channel Event Logging
574
---------------------
587
---------------------
575
 * A new interface, CEL, is introduced here. CEL logs single events, much like
588
 * A new interface, CEL, is introduced here. CEL logs single events, much like
576
   the AMI, but it differs from the AMI in that it logs to db backends much
589
   the AMI, but it differs from the AMI in that it logs to db backends much
577
   like CDR does; is based on the event subsystem introduced by Russell, and
590
   like CDR does; is based on the event subsystem introduced by Russell, and
578
   can share in all its benefits; allows multiple backends to operate like CDR;
591
   can share in all its benefits; allows multiple backends to operate like CDR;
579
   is specialized to event data that would be of concern to billing sytems,
592
   is specialized to event data that would be of concern to billing sytems,
580
   like CDR. Backends for logging and accounting calls have been produced,
593
   like CDR. Backends for logging and accounting calls have been produced,
581
   but a new CDR backend is still in development.
594
   but a new CDR backend is still in development.
582

    
   
595

   
583
CDR
596
CDR
584
---
597
---
585
 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
598
 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
586
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
599
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
587
   etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
600
   etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
588
 * Multiple files and formats can now be specified in cdr_custom.conf.
601
 * Multiple files and formats can now be specified in cdr_custom.conf.
589
 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
602
 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
590
   See configs/cdr_syslog.conf.sample for more information.
603
   See configs/cdr_syslog.conf.sample for more information.
591
 * A 'sequence' field has been added to CDRs which can be combined with
604
 * A 'sequence' field has been added to CDRs which can be combined with
592
   linkedid or uniqueid to uniquely identify a CDR.
605
   linkedid or uniqueid to uniquely identify a CDR.
593
 * Handling of billsec and duration field has changed. If your table definition
606
 * Handling of billsec and duration field has changed. If your table definition
594
   specifies those fields as float,double or similar they will now be logged with
607
   specifies those fields as float,double or similar they will now be logged with
595
   microsecond accuracy instead of a whole integer.
608
   microsecond accuracy instead of a whole integer.
596

    
   
609

   
597
Calendaring for Asterisk
610
Calendaring for Asterisk
598
------------------------
611
------------------------
599
 * A new set of modules were added supporing calendar integration with Asterisk.
612
 * A new set of modules were added supporing calendar integration with Asterisk.
600
   Dialplan functions for reading from and writing to calendars are included,
613
   Dialplan functions for reading from and writing to calendars are included,
601
   as well as the ability to execute dialplan logic upon calendar event notifications.
614
   as well as the ability to execute dialplan logic upon calendar event notifications.
602
   iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
615
   iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
603
   Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
616
   Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
604
   Exchange Server 2007+ with full write and attendee support) are supported (Exchange
617
   Exchange Server 2007+ with full write and attendee support) are supported (Exchange
605
   2003 support does not support forms-based authentication).
618
   2003 support does not support forms-based authentication).
606

    
   
619

   
607
Call Completion Supplementary Services for Asterisk
620
Call Completion Supplementary Services for Asterisk
608
---------------------------------------------------
621
---------------------------------------------------
609
 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
622
 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
610
   DAHDI/ISDN supports call completion for the following switch types:
623
   DAHDI/ISDN supports call completion for the following switch types:
611
   EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
624
   EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
612
   See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
625
   See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
613

    
   
626

   
614
Multicast RTP Support
627
Multicast RTP Support
615
---------------------
628
---------------------
616
 * A new RTP engine and channel driver have been added which supports Multicast RTP.
629
 * A new RTP engine and channel driver have been added which supports Multicast RTP.
617
   The channel driver can be used with the Page application to perform multicast RTP
630
   The channel driver can be used with the Page application to perform multicast RTP
618
   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
631
   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
619
   Type can be either basic or linksys.
632
   Type can be either basic or linksys.
620
   Destination is the IP address and port for the RTP packets.
633
   Destination is the IP address and port for the RTP packets.
621
   Control address is specific to the linksys type and is used for sending the control
634
   Control address is specific to the linksys type and is used for sending the control
622
   packets unique to them.
635
   packets unique to them.
623

    
   
636

   
624
Security Events Framework
637
Security Events Framework
625
-------------------------
638
-------------------------
626
 * Asterisk has a new C API for reporting security events.  The module res_security_log
639
 * Asterisk has a new C API for reporting security events.  The module res_security_log
627
   sends these events to the "security" logger level.  Currently, AMI is the only
640
   sends these events to the "security" logger level.  Currently, AMI is the only
628
   Asterisk component that reports security events.  However, SIP support will be
641
   Asterisk component that reports security events.  However, SIP support will be
629
   coming soon.  For more information on the security events framework, see the
642
   coming soon.  For more information on the security events framework, see the
630
   "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
643
   "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
631

    
   
644

   
632
Fax
645
Fax
633
---
646
---
634
 * A technology independent fax frontend (res_fax) has been added to Asterisk.
647
 * A technology independent fax frontend (res_fax) has been added to Asterisk.
635
 * A spandsp based fax backend (res_fax_spandsp) has been added.
648
 * A spandsp based fax backend (res_fax_spandsp) has been added.
636
 * The app_fax module has been deprecated in favor of the res_fax module and
649
 * The app_fax module has been deprecated in favor of the res_fax module and
637
   the new res_fax_spandsp backend.
650
   the new res_fax_spandsp backend.
638
 * The SendFAX and ReceiveFAX applications now send their log messages to a
651
 * The SendFAX and ReceiveFAX applications now send their log messages to a
639
   'fax' logger level, instead of to the generic logger levels. To see these
652
   'fax' logger level, instead of to the generic logger levels. To see these
640
   messages, the system's logger.conf file will need to direct the 'fax' logger
653
   messages, the system's logger.conf file will need to direct the 'fax' logger
641
   level to one or more destinations; the logger.conf.sample file includes an
654
   level to one or more destinations; the logger.conf.sample file includes an
642
   example of how to do this. Note that if the 'fax' logger level is *not*
655
   example of how to do this. Note that if the 'fax' logger level is *not*
643
   directed to at least one destination, log messages generated by these
656
   directed to at least one destination, log messages generated by these
644
   applications will be lost, and that if the 'fax' logger level is directed to
657
   applications will be lost, and that if the 'fax' logger level is directed to
645
   the console, the 'core set verbose' and 'core set debug' CLI commands will
658
   the console, the 'core set verbose' and 'core set debug' CLI commands will
646
   have no effect on whether the messages appear on the console or not.
659
   have no effect on whether the messages appear on the console or not.
647

    
   
660

   
648
Miscellaneous
661
Miscellaneous
649
-------------
662
-------------
650
 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
663
 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
651
   Now, in order to enable transmitting silence during record the transmit_silence
664
   Now, in order to enable transmitting silence during record the transmit_silence
652
   option should be used.  transmit_silence_during_record remains a valid option, but
665
   option should be used.  transmit_silence_during_record remains a valid option, but
653
   defaults to the behavior of the transmit_silence option.
666
   defaults to the behavior of the transmit_silence option.
654
 * Addition of the Unit Test Framework API for managing registration and execution
667
 * Addition of the Unit Test Framework API for managing registration and execution
655
   of unit tests with the purpose of verifying the operation of C functions.
668
   of unit tests with the purpose of verifying the operation of C functions.
656
 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
669
 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
657
   XMPP text messages to the remote JID.
670
   XMPP text messages to the remote JID.
658
 * Modules.conf has a new option - "require" - that marks a module as critical for 
671
 * Modules.conf has a new option - "require" - that marks a module as critical for 
659
   the execution of Asterisk.
672
   the execution of Asterisk.
660
   If one of the required modules fail to load, Asterisk will exit with a return
673
   If one of the required modules fail to load, Asterisk will exit with a return
661
   code set to 2.
674
   code set to 2.
662
 * An 'X' option has been added to the asterisk application which enables #exec support.
675
 * An 'X' option has been added to the asterisk application which enables #exec support.
663
   This allows #exec to be used in asterisk.conf.
676
   This allows #exec to be used in asterisk.conf.
664
 * jabber.conf supports a new option auth_policy that toggles auto user registration.
677
 * jabber.conf supports a new option auth_policy that toggles auto user registration.
665
 * A new lockconfdir option has been added to asterisk.conf to protect the
678
 * A new lockconfdir option has been added to asterisk.conf to protect the
666
   configuration directory (/etc/asterisk by default) during reloads.
679
   configuration directory (/etc/asterisk by default) during reloads.
667
 * The parkeddynamic option has been added to features.conf to enable the creation
680
 * The parkeddynamic option has been added to features.conf to enable the creation
668
   of dynamic parkinglots.
681
   of dynamic parkinglots.
669
 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
682
 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
670
   the reportalarms config option.
683
   the reportalarms config option.
671
 * chan_dahdi supports dialing configuring and dialing by device file name.
684
 * chan_dahdi supports dialing configuring and dialing by device file name.
672
   DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
685
   DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
673
   it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
686
   it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
674
 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
687
 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
675
   False by default. If set, chan_dahdi will ignore failed 'channel' entries.
688
   False by default. If set, chan_dahdi will ignore failed 'channel' entries.
676
   Handy for the above name-based syntax as it does not depend on
689
   Handy for the above name-based syntax as it does not depend on
677
   initialization order.
690
   initialization order.
678
 * The Realtime dialplan switch now caches entries for 1 second.  This provides a
691
 * The Realtime dialplan switch now caches entries for 1 second.  This provides a
679
   significant increase in performance (about 3X) for installations using this switchtype.
692
   significant increase in performance (about 3X) for installations using this switchtype.
680
 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
693
 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
681
   AIS.  For more information, please see doc/distributed_devstate-XMPP.txt
694
   AIS.  For more information, please see doc/distributed_devstate-XMPP.txt
682
 * The addition of G.719 pass-through support.
695
 * The addition of G.719 pass-through support.
683
 * Added support for 16khz Speex audio.  This can be enabled by using 'allow=speex16'
696
 * Added support for 16khz Speex audio.  This can be enabled by using 'allow=speex16'
684
   during device configuration.
697
   during device configuration.
685
 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
698
 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
686
   have less than 3 lines on the LCD.
699
   have less than 3 lines on the LCD.
687
 * Realtime now supports database failover.  See the sample extconfig.conf for details.
700
 * Realtime now supports database failover.  See the sample extconfig.conf for details.
688
 * The addition of improved translation path building for wideband codecs.  Sample
701
 * The addition of improved translation path building for wideband codecs.  Sample
689
   rate changes during translation are now avoided unless absolutely necessary.
702
   rate changes during translation are now avoided unless absolutely necessary.
690
 * The addition of the res_stun_monitor module for monitoring and reacting to network
703
 * The addition of the res_stun_monitor module for monitoring and reacting to network
691
   changes while behind a NAT.
704
   changes while behind a NAT.
692

    
   
705

   
693
CLI Changes
706
CLI Changes
694
-----------
707
-----------
695
 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
708
 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
696
   optionally accept a filename, to apply the setting only to the code generated from
709
   optionally accept a filename, to apply the setting only to the code generated from
697
   that source file when Asterisk was built. However, there are some modules in Asterisk
710
   that source file when Asterisk was built. However, there are some modules in Asterisk
698
   that are composed of multiple source files, so this did not result in the behavior
711
   that are composed of multiple source files, so this did not result in the behavior
699
   that users expected. In this version, 'core set debug' and 'core set verbose'
712
   that users expected. In this version, 'core set debug' and 'core set verbose'
700
   can optionally accept *module* names instead (with or without the .so extension),
713
   can optionally accept *module* names instead (with or without the .so extension),
701
   which applies the setting to the entire module specified, regardless of which source
714
   which applies the setting to the entire module specified, regardless of which source
702
   files it was built from.
715
   files it was built from.
703
 * New 'manager show settings' command showing the current settings loaded from
716
 * New 'manager show settings' command showing the current settings loaded from
704
   manager.conf. 
717
   manager.conf. 
705
 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
718
 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
706
   the channel hangup request to all channels.
719
   the channel hangup request to all channels.
707
 * Added a "core reload" CLI command that executes a global reload of Asterisk.
720
 * Added a "core reload" CLI command that executes a global reload of Asterisk.
708

    
   
721

   
709
------------------------------------------------------------------------------
722
------------------------------------------------------------------------------
710
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
723
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
711
------------------------------------------------------------------------------
724
------------------------------------------------------------------------------
712

    
   
725

   
713
SIP Changes
726
SIP Changes
714
-----------
727
-----------
715
 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
728
 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
716
   Snom phones use this for call pickup of extensions that the phone is
729
   Snom phones use this for call pickup of extensions that the phone is
717
   subscribed to.
730
   subscribed to.
718
 * Added support for setting the domain in the URI for caller of an
731
 * Added support for setting the domain in the URI for caller of an
719
   outbound call by using the SIPFROMDOMAIN channel variable.
732
   outbound call by using the SIPFROMDOMAIN channel variable.
720
 * Added a new configuration option "remotesecret" for authentication to
733
 * Added a new configuration option "remotesecret" for authentication to
721
   remote services. For backwards compatibility, "secret" still has the
734
   remote services. For backwards compatibility, "secret" still has the
722
   same function as before, but now you can configure both a remote secret and a
735
   same function as before, but now you can configure both a remote secret and a
723
   local secret for mutual authentication.
736
   local secret for mutual authentication.
724
 * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set, 
737
 * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set, 
725
   the sound will be played to the target of an attended transfer
738
   the sound will be played to the target of an attended transfer
726
 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
739
 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
727
   finer control over how many peers Asterisk will qualify and the gap between them
740
   finer control over how many peers Asterisk will qualify and the gap between them
728
   when all peers need to be qualified at the same time.
741
   when all peers need to be qualified at the same time.
729
 * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
742
 * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
730
   (either globally or for a specific peer), chan_sip will treat any SDP data
743
   (either globally or for a specific peer), chan_sip will treat any SDP data
731
   it receives as new data and update the media stream accordingly.  By
744
   it receives as new data and update the media stream accordingly.  By
732
   default, Asterisk will only modify the media stream if the SDP session
745
   default, Asterisk will only modify the media stream if the SDP session
733
   version received is different from the current SDP session version.  This
746
   version received is different from the current SDP session version.  This
734
   option is required to interoperate with devices that have non-standard SDP
747
   option is required to interoperate with devices that have non-standard SDP
735
   session version implementations (observed with Microsoft OCS).  This option
748
   session version implementations (observed with Microsoft OCS).  This option
736
   is disabled by default.
749
   is disabled by default.
737
 * The parsing of register => lines in sip.conf has been modified to allow a port
750
 * The parsing of register => lines in sip.conf has been modified to allow a port
738
   to be present in the "user" portion. Please see the sip.conf.sample file for more
751
   to be present in the "user" portion. Please see the sip.conf.sample file for more
739
   information
752
   information
740
 * Added support for subscribing to MWI on a remote server and making the status available
753
 * Added support for subscribing to MWI on a remote server and making the status available
741
   as a mailbox. Please see the sip.conf.sample file for more information.
754
   as a mailbox. Please see the sip.conf.sample file for more information.
742
 * Added a function to remove SIP headers added in the dialplan before the
755
 * Added a function to remove SIP headers added in the dialplan before the
743
   first INVITE is generated - SIPRemoveHeader()
756
   first INVITE is generated - SIPRemoveHeader()
744
 * Channel variables set with setvar= in a device configuration is now 
757
 * Channel variables set with setvar= in a device configuration is now 
745
   set both for inbound and outbound calls.
758
   set both for inbound and outbound calls.
746
 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
759
 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
747

    
   
760

   
748
IAX2 changes
761
IAX2 changes
749
------------
762
------------
750
  * Added immediate option to iax.conf
763
  * Added immediate option to iax.conf
751
  * Added forceencryption option to iax.conf
764
  * Added forceencryption option to iax.conf
752
  * Added Encryption and Trunk status to manager command "iaxpeers"
765
  * Added Encryption and Trunk status to manager command "iaxpeers"
753

    
   
766

   
754
Skinny Changes
767
Skinny Changes
755
--------------
768
--------------
756
 * The configuration file now holds separate sections for devices and lines.
769
 * The configuration file now holds separate sections for devices and lines.
757
   Please have a look at configs/skinny.conf.sample and change your skinny.conf
770
   Please have a look at configs/skinny.conf.sample and change your skinny.conf
758
   accordingly.
771
   accordingly.
759

    
   
772

   
760
DAHDI Changes
773
DAHDI Changes
761
-------------
774
-------------
762
 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
775
 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
763
   support for LibOpenR2.  http://www.libopenr2.org/
776
   support for LibOpenR2.  http://www.libopenr2.org/
764
 * The UK option waitfordialtone has been added for use with BT analog
777
 * The UK option waitfordialtone has been added for use with BT analog
765
   lines.
778
   lines.
766
 * Added a 'faxbuffers' configuration option to chan_dahdi.conf.  This option
779
 * Added a 'faxbuffers' configuration option to chan_dahdi.conf.  This option
767
   is used in conjunction with the 'faxdetect' configuration option.  When
780
   is used in conjunction with the 'faxdetect' configuration option.  When
768
   'faxbuffers' is used and fax tones are detected, the channel will dynamically
781
   'faxbuffers' is used and fax tones are detected, the channel will dynamically
769
   switch to the configured faxbuffers policy.  For example, to use 6 buffers
782
   switch to the configured faxbuffers policy.  For example, to use 6 buffers
770
   and a 'full' buffer policy for a fax transmission, add:
783
   and a 'full' buffer policy for a fax transmission, add:
771
     faxbuffers=>6,full
784
     faxbuffers=>6,full
772
   The faxbuffers configuration will be in affect until the call is torn down.
785
   The faxbuffers configuration will be in affect until the call is torn down.
773
 * Added service message support for 4ESS/5ESS switches.
786
 * Added service message support for 4ESS/5ESS switches.
774

    
   
787

   
775
Dialplan Functions
788
Dialplan Functions
776
------------------
789
------------------
777
 * For DAHDI channels, the CHANNEL() dialplan function now
790
 * For DAHDI channels, the CHANNEL() dialplan function now
778
   supports changing the channel's buffer policy (for the current
791
   supports changing the channel's buffer policy (for the current
779
   call only), using this syntax:
792
   call only), using this syntax:
780

    
   
793

   
781
   exten => s,n,Set(CHANNEL(buffers)=6,full)
794
   exten => s,n,Set(CHANNEL(buffers)=6,full)
782

    
   
795

   
783
   This would change the channel to the 'full' buffer policy and
796
   This would change the channel to the 'full' buffer policy and
784
   6 (six) buffers. Possible options for this setting are the same
797
   6 (six) buffers. Possible options for this setting are the same
785
   as those in chan_dahdi.conf.
798
   as those in chan_dahdi.conf.
786
 * Added a new dialplan function, CURLOPT, which permits setting various
799
 * Added a new dialplan function, CURLOPT, which permits setting various
787
   options that may be useful with the CURL dialplan function, such as
800
   options that may be useful with the CURL dialplan function, such as
788
   cookies, proxies, connection timeouts, passwords, etc.
801
   cookies, proxies, connection timeouts, passwords, etc.
789
 * Permit the syntax and synopsis fields of the corresponding dialplan
802
 * Permit the syntax and synopsis fields of the corresponding dialplan
790
   functions to be individually set from func_odbc.conf.
803
   functions to be individually set from func_odbc.conf.
791
 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
804
 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
792
 * func_odbc now may specify an insert query to execute, when the write query
805
 * func_odbc now may specify an insert query to execute, when the write query
793
   affects 0 rows (usually indicating that no such row exists).
806
   affects 0 rows (usually indicating that no such row exists).
794
 * Added a new dialplan function, LISTFILTER, which permits removing elements
807
 * Added a new dialplan function, LISTFILTER, which permits removing elements
795
   from a set list, by name.  Uses the same general syntax as the existing CUT
808
   from a set list, by name.  Uses the same general syntax as the existing CUT
796
   and FIELDQTY dialplan functions, which also manage lists.
809
   and FIELDQTY dialplan functions, which also manage lists.
797
 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
810
 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
798
   obtaining realtime data from the dialplan.
811
   obtaining realtime data from the dialplan.
799
 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
812
 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
800
   a subroutine when using the GoSub() and Return() applications.
813
   a subroutine when using the GoSub() and Return() applications.
801
 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
814
 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
802
   of "core show function AUDIOHOOK_INHERIT" from the CLI
815
   of "core show function AUDIOHOOK_INHERIT" from the CLI
803
 * Added AES_ENCRYPT. For information on its use, please see the output
816
 * Added AES_ENCRYPT. For information on its use, please see the output
804
   of "core show function AES_ENCRYPT" from the CLI
817
   of "core show function AES_ENCRYPT" from the CLI
805
 * Added AES_DECRYPT. For information on its use, please see the output
818
 * Added AES_DECRYPT. For information on its use, please see the output
806
   of "core show function AES_DECRYPT" from the CLI
819
   of "core show function AES_DECRYPT" from the CLI
807
 * func_odbc now supports database transactions across multiple queries.
820
 * func_odbc now supports database transactions across multiple queries.
808

    
   
821

   
809
Applications
822
Applications
810
------------
823
------------
811
 * Scheduled meetme conferences may now have their end times extended by
824
 * Scheduled meetme conferences may now have their end times extended by
812
   using MeetMeAdmin.
825
   using MeetMeAdmin.
813
 * app_authenticate now gives the ability to select a prompt other than
826
 * app_authenticate now gives the ability to select a prompt other than
814
   the default.
827
   the default.
815
 * app_directory now pays attention to the searchcontexts setting in
828
 * app_directory now pays attention to the searchcontexts setting in
816
   voicemail.conf and will look through all contexts, if no context is
829
   voicemail.conf and will look through all contexts, if no context is
817
   specified in the initial argument.
830
   specified in the initial argument.
818
 * A new application, Originate, has been introduced, that allows asynchronous
831
 * A new application, Originate, has been introduced, that allows asynchronous
819
   call origination from the dialplan.
832
   call origination from the dialplan.
820
 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
833
 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
821
   in addition to the setting in the "general" context.
834
   in addition to the setting in the "general" context.
822
 * Added ConfBridge dialplan application which does conference bridges without
835
 * Added ConfBridge dialplan application which does conference bridges without
823
   DAHDI. For information on its use, please see the output of
836
   DAHDI. For information on its use, please see the output of
824
   "core show application ConfBridge" from the CLI.
837
   "core show application ConfBridge" from the CLI.
825

    
   
838

   
826
Miscellaneous
839
Miscellaneous
827
-------------
840
-------------
828
 * The Asterisk CLI has a new command, "channel redirect", which is similar in
841
 * The Asterisk CLI has a new command, "channel redirect", which is similar in
829
   operation to the AMI Redirect action.
842
   operation to the AMI Redirect action.
830
 * extensions.conf now allows you to use keyword "same" to define an extension
843
 * extensions.conf now allows you to use keyword "same" to define an extension
831
   without actually specifying an extension.  It uses exactly the same pattern
844
   without actually specifying an extension.  It uses exactly the same pattern
832
   as previously used on the last "exten" line.  For example:
845
   as previously used on the last "exten" line.  For example:
833
     exten => 123,1,NoOp(something)
846
     exten => 123,1,NoOp(something)
834
     same  =>     n,SomethingElse()
847
     same  =>     n,SomethingElse()
835
 * musiconhold.conf classes of type 'files' can now use relative directory paths,
848
 * musiconhold.conf classes of type 'files' can now use relative directory paths,
836
   which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
849
   which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
837
 * All deprecated CLI commands are removed from the sourcecode. They are now handled
850
 * All deprecated CLI commands are removed from the sourcecode. They are now handled
838
   by the new clialiases module. See cli_aliases.conf.sample file.
851
   by the new clialiases module. See cli_aliases.conf.sample file.
839
 * Times within timespecs are now accurate down to the minute.  This is a change
852
 * Times within timespecs are now accurate down to the minute.  This is a change
840
   from historical Asterisk, which only provided timespecs rounded to the nearest
853
   from historical Asterisk, which only provided timespecs rounded to the nearest
841
   even (read: evenly divisible by 2) minute mark.
854
   even (read: evenly divisible by 2) minute mark.
842
 * The realtime switch now supports an option flag, 'p', which disables searches for
855
 * The realtime switch now supports an option flag, 'p', which disables searches for
843
   pattern matches.
856
   pattern matches.
844
 * In addition to a time range and date range, timespecs now accept a 5th optional
857
 * In addition to a time range and date range, timespecs now accept a 5th optional
845
   argument, timezone.  This allows you to perform time checks on alternate
858
   argument, timezone.  This allows you to perform time checks on alternate
846
   timezones, especially if those daylight savings time ranges vary from your
859
   timezones, especially if those daylight savings time ranges vary from your
847
   machine's native timezone.  See GotoIfTime, ExecIfTime, IFTIME(), and timed
860
   machine's native timezone.  See GotoIfTime, ExecIfTime, IFTIME(), and timed
848
   includes.
861
   includes.
849
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
862
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
850
   give you the correct output for an asterisk box behind nat. It will give you the
863
   give you the correct output for an asterisk box behind nat. It will give you the
851
   externhost and localnet settings.
864
   externhost and localnet settings.
852
 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
865
 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
853
   can connect calls in passthrough mode, as well as record and play back files.
866
   can connect calls in passthrough mode, as well as record and play back files.
854
 * Successful and unsuccessful call pickup can now be alerted through sounds, by
867
 * Successful and unsuccessful call pickup can now be alerted through sounds, by
855
   using pickupsound and pickupfailsound in features.conf.
868
   using pickupsound and pickupfailsound in features.conf.
856
 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
869
 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
857
   This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
870
   This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
858
   instead of the /var/run/asterisk.pid where it used to be. This will make
871
   instead of the /var/run/asterisk.pid where it used to be. This will make
859
   installs as non-root easier to manage.
872
   installs as non-root easier to manage.
860

    
   
873

   
861
CDR
874
CDR
862
---
875
---
863

    
   
876

   
864
* The cdr.conf file must exist and be correctly programmed in order for CDR records to
877
* The cdr.conf file must exist and be correctly programmed in order for CDR records to
865
  be written; they will no longer be explicitly written.
878
  be written; they will no longer be explicitly written.
866

    
   
879

   
867
Asterisk Manager Interface
880
Asterisk Manager Interface
868
--------------------------
881
--------------------------
869
 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
882
 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
870
   a non-empty value) in your request. If you do this, any pending AMI events will
883
   a non-empty value) in your request. If you do this, any pending AMI events will
871
   *not* be included in the response to your request as they would normally, but
884
   *not* be included in the response to your request as they would normally, but
872
   will be left in the event queue for the next request you make to retrieve. For
885
   will be left in the event queue for the next request you make to retrieve. For
873
   some applications, this will allow you to guarantee that you will only see
886
   some applications, this will allow you to guarantee that you will only see
874
   events in responses to 'WaitEvent' actions, and can better know when to expect them.
887
   events in responses to 'WaitEvent' actions, and can better know when to expect them.
875
   To know whether the Asterisk server supports this header or not, your client can
888
   To know whether the Asterisk server supports this header or not, your client can
876
   inspect the first response back from the server to see if it includes this header:
889
   inspect the first response back from the server to see if it includes this header:
877

    
   
890

   
878
   Pragma: SuppressEvents
891
   Pragma: SuppressEvents
879

    
   
892

   
880
   If this is included, the server supports event suppression.
893
   If this is included, the server supports event suppression.
881

    
   
894

   
882
 * Added 4 new Actions to list skinny device(s) and line(s)
895
 * Added 4 new Actions to list skinny device(s) and line(s)
883
   SKINNYdevices
896
   SKINNYdevices
884
   SKINNYshowdevice
897
   SKINNYshowdevice
885
   SKINNYlines
898
   SKINNYlines
886
   SKINNYshowline
899
   SKINNYshowline
887

    
   
900

   
888
LDAP Schema File Additions
901
LDAP Schema File Additions
889
--------------------------
902
--------------------------
890
 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox  objectClasses
903
 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox  objectClasses
891
   to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
904
   to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
892
 * Added new Fields:
905
 * Added new Fields:
893
   - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
906
   - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
894
   - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
907
   - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
895
   - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
908
   - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
896
 * Removed redundant IPaddr (there's already IPAddress)
909
 * Removed redundant IPaddr (there's already IPAddress)
897
   - Gives more configuration Flags for SIP-Users available (tested)
910
   - Gives more configuration Flags for SIP-Users available (tested)
898
   - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
911
   - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
899
     without extensibleObject (which really should be the last resort); gives
912
     without extensibleObject (which really should be the last resort); gives
900
     also additional possibilities for LDAP-filter 
913
     also additional possibilities for LDAP-filter 
901

    
   
914

   
902
------------------------------------------------------------------------------
915
------------------------------------------------------------------------------
903
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
916
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
904
------------------------------------------------------------------------------
917
------------------------------------------------------------------------------
905

    
   
918

   
906
Device State Handling
919
Device State Handling
907
---------------------
920
---------------------
908
 * The event infrastructure in Asterisk got another big update to help support
921
 * The event infrastructure in Asterisk got another big update to help support
909
    distributed events.  It currently supports distributed device state and
922
    distributed events.  It currently supports distributed device state and
910
    distributed Voicemail MWI (Message Waiting Indication).  A new module has
923
    distributed Voicemail MWI (Message Waiting Indication).  A new module has
911
    been merged, res_ais, which facilitates communicating events between servers.
924
    been merged, res_ais, which facilitates communicating events between servers.
912
    It uses the SAForum AIS (Service Availability Forum Application Interface
925
    It uses the SAForum AIS (Service Availability Forum Application Interface
913
    Specification) CLM (Cluster Management) and EVT (Event) services to maintain
926
    Specification) CLM (Cluster Management) and EVT (Event) services to maintain
914
    a cluster of Asterisk servers, and to share events between them.  For more
927
    a cluster of Asterisk servers, and to share events between them.  For more
915
    information on setting this up, see doc/distributed_devstate.txt.
928
    information on setting this up, see doc/distributed_devstate.txt.
916

    
   
929

   
917
Dialplan Functions
930
Dialplan Functions
918
------------------
931
------------------
919
 * Added a new dialplan function, AST_CONFIG(), which allows you to access
932
 * Added a new dialplan function, AST_CONFIG(), which allows you to access
920
   variables from an Asterisk configuration file.
933
   variables from an Asterisk configuration file.
921
 * The JACK_HOOK function now has a c() option to supply a custom client name.
934
 * The JACK_HOOK function now has a c() option to supply a custom client name.
922
 * Added two new dialplan functions from libspeex for audio gain control and 
935
 * Added two new dialplan functions from libspeex for audio gain control and 
923
   denoise, AGC() and DENOISE(). Both functions can be applied to the tx and 
936
   denoise, AGC() and DENOISE(). Both functions can be applied to the tx and 
924
   rx directions of a channel from the dialplan.
937
   rx directions of a channel from the dialplan.
925
 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
938
 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
926
   based on other parameters.  The default is still to search based on the
939
   based on other parameters.  The default is still to search based on the
927
   forwarding station ID.  However, there are new options that allow you to search
940
   forwarding station ID.  However, there are new options that allow you to search
928
   based on the message desk terminal ID, or the message desk number.
941
   based on the message desk terminal ID, or the message desk number.
929
 * TIMEOUT() has been modified to be accurate down to the millisecond.
942
 * TIMEOUT() has been modified to be accurate down to the millisecond.
930
 * ENUM*() functions now include the following new options:
943
 * ENUM*() functions now include the following new options:
931
     - 'u' returns the full URI and does not strip off the URI-scheme.
944
     - 'u' returns the full URI and does not strip off the URI-scheme.
932
     - 's' triggers ISN specific rewriting
945
     - 's' triggers ISN specific rewriting
933
     - 'i' looks for branches into an Infrastructure ENUM tree
946
     - 'i' looks for branches into an Infrastructure ENUM tree
934
     - 'd' for a direct DNS lookup without any flipping of digits.
947
     - 'd' for a direct DNS lookup without any flipping of digits.
935
 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
948
 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
936
 * CHANNEL() now has options for the maximum, minimum, and standard or normal
949
 * CHANNEL() now has options for the maximum, minimum, and standard or normal
937
   deviation of jitter, rtt, and loss for a call using chan_sip.
950
   deviation of jitter, rtt, and loss for a call using chan_sip.
938

    
   
951

   
939
DAHDI channel driver (chan_dahdi) Changes
952
DAHDI channel driver (chan_dahdi) Changes
940
----------------------------------------
953
----------------------------------------
941
 * Channels can now be configured using named sections in chan_dahdi.conf, just
954
 * Channels can now be configured using named sections in chan_dahdi.conf, just
942
   like other channel drivers, including the use of templates.
955
   like other channel drivers, including the use of templates.
943
 * The default for pridialplan has changed from 'national' to 'unknown'.
956
 * The default for pridialplan has changed from 'national' to 'unknown'.
944

    
   
957

   
945
PBX Changes
958
PBX Changes
946
-----------
959
-----------
947
 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
960
 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
948
   to something that matches the pattern a hint will be created using the contents
961
   to something that matches the pattern a hint will be created using the contents
949
   and variables evaluated.
962
   and variables evaluated.
950
 * Dialplan matching has been extended to allow an extension to return to the
963
 * Dialplan matching has been extended to allow an extension to return to the
951
   PBX core to wait for more digits.  This is done by using the new dialplan
964
   PBX core to wait for more digits.  This is done by using the new dialplan
952
   application called "Incomplete".  This will permit a whole new level of
965
   application called "Incomplete".  This will permit a whole new level of
953
   extension control, by giving the administrator more control over early
966
   extension control, by giving the administrator more control over early
954
   matches employing one of the short-circuit pattern match operators.  Note
967
   matches employing one of the short-circuit pattern match operators.  Note
955
   that custom applications can trigger this same behavior by returning the
968
   that custom applications can trigger this same behavior by returning the
956
   special value AST_PBX_INCOMPLETE.
969
   special value AST_PBX_INCOMPLETE.
957

    
   
970

   
958
Application Changes
971
Application Changes
959
-------------------
972
-------------------
960
 * Directory now permits both first and last names to be matched at the same
973
 * Directory now permits both first and last names to be matched at the same
961
   time.  In addition, the number of digits to enter of the name can be set in
974
   time.  In addition, the number of digits to enter of the name can be set in
962
   the arguments to Directory; previously, you could enter only 3, regardless
975
   the arguments to Directory; previously, you could enter only 3, regardless
963
   of how many names are in your company.  For large companies, this should be
976
   of how many names are in your company.  For large companies, this should be
964
   quite helpful.
977
   quite helpful.
965
 * Voicemail now permits a mailbox setting to wrap around from first to last
978
 * Voicemail now permits a mailbox setting to wrap around from first to last
966
   messages, if the "messagewrap" option is set to a true value.
979
   messages, if the "messagewrap" option is set to a true value.
967
 * Voicemail now permits an external script to be run, for password validation.
980
 * Voicemail now permits an external script to be run, for password validation.
968
   The script should output "VALID" or "INVALID" on stdout, depending upon the
981
   The script should output "VALID" or "INVALID" on stdout, depending upon the
969
   wish to validate or invalidate the password given.  Arguments are:
982
   wish to validate or invalidate the password given.  Arguments are:
970
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
983
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
971
   more details
984
   more details
972
 * Dial has a new option: F(context^extension^pri), which permits a callee to
985
 * Dial has a new option: F(context^extension^pri), which permits a callee to
973
   continue in the dialplan, at the specified label, if the caller hangs up.
986
   continue in the dialplan, at the specified label, if the caller hangs up.
974
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
987
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
975
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
988
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
976
 * The Jack application now has a c() option to supply a custom client name.
989
 * The Jack application now has a c() option to supply a custom client name.
977
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
990
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
978
   like the pre-existing whisper mode, except that the spy can also talk to the
991
   like the pre-existing whisper mode, except that the spy can also talk to the
979
   participant on the bridged channel as well.
992
   participant on the bridged channel as well.
980
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
993
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
981
   to be spoken instead of the channel name or number. For more information on the
994
   to be spoken instead of the channel name or number. For more information on the
982
   use of this option, issue the command "core show application ChanSpy" from the 
995
   use of this option, issue the command "core show application ChanSpy" from the 
983
   Asterisk CLI.
996
   Asterisk CLI.
984
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
997
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
985
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
998
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
986
   words, if using the 'd' option, it is not possible to enter a number to append to
999
   words, if using the 'd' option, it is not possible to enter a number to append to
987
   the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1000
   the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
988
   change to whisper mode, and pressing 6 will change to barge mode.
1001
   change to whisper mode, and pressing 6 will change to barge mode.
989
 * ExternalIVR now takes several options that affect the way it performs, as
1002
 * ExternalIVR now takes several options that affect the way it performs, as
990
   well as having several new commands.  Please see doc/externalivr.txt for the
1003
   well as having several new commands.  Please see doc/externalivr.txt for the
991
   complete documentation.
1004
   complete documentation.
992
 * Added ability to communicate over a TCP socket instead of forking a child process for the 
1005
 * Added ability to communicate over a TCP socket instead of forking a child process for the 
993
   ExternalIVR application.
1006
   ExternalIVR application.
994
 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1007
 * ChanIsAvail has a new option, 'a', which will return all available channels instead
995
   of just the first one if you give the function more then one channel to check.
1008
   of just the first one if you give the function more then one channel to check.
996
 * PrivacyManager now takes an option where you can specify a context where the 
1009
 * PrivacyManager now takes an option where you can specify a context where the 
997
   given number will be matched. This way you have more control over who is allowed
1010
   given number will be matched. This way you have more control over who is allowed
998
   and it stops the people who blindly enter 10 digits.
1011
   and it stops the people who blindly enter 10 digits.
999
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1012
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1000
   answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1013
   answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1001
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1014
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1002
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1015
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1003
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1016
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1004
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1017
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1005
 * The Dial() application no longer copies the language used by the caller to the callee's
1018
 * The Dial() application no longer copies the language used by the caller to the callee's
1006
   channel. If you desire for the caller's channel's language to be used for file playback
1019
   channel. If you desire for the caller's channel's language to be used for file playback
1007
   to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1020
   to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1008
 * SendImage() no longer hangs up the channel on error; instead, it sets the
1021
 * SendImage() no longer hangs up the channel on error; instead, it sets the
1009
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1022
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1010
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
1023
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
1011
   applications.
1024
   applications.
1012
 * Park has a new option, 's', which silences the announcement of the parking space number.
1025
 * Park has a new option, 's', which silences the announcement of the parking space number.
1013
 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1026
 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1014
   invalid input and will be assumed to mean that no timeout is desired.
1027
   invalid input and will be assumed to mean that no timeout is desired.
1015

    
   
1028

   
1016
SIP Changes
1029
SIP Changes
1017
-----------
1030
-----------
1018
 * Added DNS manager support to registrations for peers referencing peer entries.
1031
 * Added DNS manager support to registrations for peers referencing peer entries.
1019
   DNS manager runs in the background which allows DNS lookups to be run asynchronously 
1032
   DNS manager runs in the background which allows DNS lookups to be run asynchronously 
1020
   as well as periodically updating the IP address. These properties allow for
1033
   as well as periodically updating the IP address. These properties allow for
1021
   better performance as well as recovery in the event of an IP change.
1034
   better performance as well as recovery in the event of an IP change.
1022
 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve 
1035
 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve 
1023
   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1036
   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1024
   These changes also provide performance improvements for call setup and tear down.
1037
   These changes also provide performance improvements for call setup and tear down.
1025
 * Added ability to specify registration expiry time on a per registration basis in
1038
 * Added ability to specify registration expiry time on a per registration basis in
1026
   the register line.
1039
   the register line.
1027
 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1040
 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1028
   lost packets.
1041
   lost packets.
1029
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1042
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1030
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1043
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1031
 * 'sip show peers' and 'sip show users' display their entries sorted in
1044
 * 'sip show peers' and 'sip show users' display their entries sorted in
1032
    alphabetical order, as opposed to the order they were in, in the config 
1045
    alphabetical order, as opposed to the order they were in, in the config 
1033
    file or database. 
1046
    file or database. 
1034
 * Videosupport now supports an additional option, "always", which always sets
1047
 * Videosupport now supports an additional option, "always", which always sets
1035
    up video RTP ports, even on clients that don't support it.  This helps with
1048
    up video RTP ports, even on clients that don't support it.  This helps with
1036
    callfiles and certain transfers to ensure that if two video phones are
1049
    callfiles and certain transfers to ensure that if two video phones are
1037
    connected, they will always share video feeds.
1050
    connected, they will always share video feeds.
1038

    
   
1051

   
1039
IAX Changes
1052
IAX Changes
1040
-----------
1053
-----------
1041
 * Existing DNS manager lookups extended to check for SRV records.
1054
 * Existing DNS manager lookups extended to check for SRV records.
1042
 * IAX2 encryption support has been improved to support periodic key rotation
1055
 * IAX2 encryption support has been improved to support periodic key rotation
1043
   within a call for enhanced security.  The option "keyrotate" has been
1056
   within a call for enhanced security.  The option "keyrotate" has been
1044
   provided to disable this functionality to preserve backwards compatibility
1057
   provided to disable this functionality to preserve backwards compatibility
1045
   with older versions of IAX2 that do not support key rotation.
1058
   with older versions of IAX2 that do not support key rotation.
1046

    
   
1059

   
1047
CLI Changes
1060
CLI Changes
1048
-----------
1061
-----------
1049
  * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1062
  * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1050
     data tree based on the given <path>.
1063
     data tree based on the given <path>.
1051
  * New CLI command "data show providers" that will display all the registered
1064
  * New CLI command "data show providers" that will display all the registered
1052
     callbacks.
1065
     callbacks.
1053
  * New CLI command, "config reload <file.conf>" which reloads any module that
1066
  * New CLI command, "config reload <file.conf>" which reloads any module that
1054
     references that particular configuration file.  Also added "config list"
1067
     references that particular configuration file.  Also added "config list"
1055
     which shows which configuration files are in use.
1068
     which shows which configuration files are in use.
1056
  * New CLI commands, "pri show version" and "ss7 show version" that will
1069
  * New CLI commands, "pri show version" and "ss7 show version" that will
1057
     display which version of libpri and libss7 are being used, respectively.
1070
     display which version of libpri and libss7 are being used, respectively.
1058
     A new API call was added so trunk will now have to be compiled against
1071
     A new API call was added so trunk will now have to be compiled against
1059
     a versions of libpri and libss7 that have them or it will not know that
1072
     a versions of libpri and libss7 that have them or it will not know that
1060
     these libraries exist.
1073
     these libraries exist.
1061
  * The commands "core show globals", "core set global" and "core set chanvar" has
1074
  * The commands "core show globals", "core set global" and "core set chanvar" has
1062
     been deprecated in favor of the more semanticly correct "dialplan show globals",
1075
     been deprecated in favor of the more semanticly correct "dialplan show globals",
1063
     "dialplan set chanvar" and "dialplan set global".
1076
     "dialplan set chanvar" and "dialplan set global".
1064
  * New CLI command "dialplan show chanvar" to list all variables associated
1077
  * New CLI command "dialplan show chanvar" to list all variables associated
1065
    with a given channel.
1078
    with a given channel.
1066

    
   
1079

   
1067
DNS manager changes
1080
DNS manager changes
1068
-------------------
1081
-------------------
1069
  * Addresses managed by DNS manager now can check to see if there is a DNS
1082
  * Addresses managed by DNS manager now can check to see if there is a DNS
1070
    SRV record for a given domain and will use that hostname/port if present.
1083
    SRV record for a given domain and will use that hostname/port if present.
1071

    
   
1084

   
1072
AMI - The manager (TCP/TLS/HTTP)
1085
AMI - The manager (TCP/TLS/HTTP)
1073
--------------------------------
1086
--------------------------------
1074
  * The Status command now takes an optional list of variables to display
1087
  * The Status command now takes an optional list of variables to display
1075
    along with channel status.
1088
    along with channel status.
1076
  * The QueueEntry event now also includes the channel's uniqueid
1089
  * The QueueEntry event now also includes the channel's uniqueid
1077

    
   
1090

   
1078
ODBC Changes
1091
ODBC Changes
1079
------------
1092
------------
1080
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
1093
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
1081
    as some people were running into this limit.  This limit has been increased
1094
    as some people were running into this limit.  This limit has been increased
1082
    to 4.2 billion.
1095
    to 4.2 billion.
1083

    
   
1096

   
1084
Queue changes
1097
Queue changes
1085
-------------
1098
-------------
1086
  * The TRANSFER queue log entry now includes the the caller's original
1099
  * The TRANSFER queue log entry now includes the the caller's original
1087
    position in the transferred-from queue.
1100
    position in the transferred-from queue.
1088
  * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1101
  * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1089
    "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1102
    "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1090
    as well as an explanation about timeout options in general
1103
    as well as an explanation about timeout options in general
1091
  * Added a new option - C - for forcing the "answered elsewhere" flag on
1104
  * Added a new option - C - for forcing the "answered elsewhere" flag on
1092
    cancellation of calls in to members of the queue. This is to avoid the
1105
    cancellation of calls in to members of the queue. This is to avoid the
1093
    call to a member of a queue having the call listed as a "missed call".
1106
    call to a member of a queue having the call listed as a "missed call".
1094

    
   
1107

   
1095
Realtime changes
1108
Realtime changes
1096
----------------
1109
----------------
1097
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1110
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1098
    adaptive capabilities.  What this means in practical terms is that if your
1111
    adaptive capabilities.  What this means in practical terms is that if your
1099
    realtime table lacks critical fields, Asterisk will now emit warnings to
1112
    realtime table lacks critical fields, Asterisk will now emit warnings to
1100
    that effect.  Also, some of the realtime drivers have the ability (if
1113
    that effect.  Also, some of the realtime drivers have the ability (if
1101
    configured) to automatically add those columns to the table with the
1114
    configured) to automatically add those columns to the table with the
1102
    correct type and length.
1115
    correct type and length.
1103

    
   
1116

   
1104
Miscellaneous
1117
Miscellaneous
1105
-------------
1118
-------------
1106
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1119
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1107
    the 'setvar' option to cause a given audio file to be played upon completion
1120
    the 'setvar' option to cause a given audio file to be played upon completion
1108
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
1121
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
1109
    Skinny channels only.
1122
    Skinny channels only.
1110
  * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
1123
  * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
1111
    for more information.
1124
    for more information.
1112
  * Config file variables may now be appended to, by using the '+=' append
1125
  * Config file variables may now be appended to, by using the '+=' append
1113
    operator.  This is most helpful when working with long SQL queries in
1126
    operator.  This is most helpful when working with long SQL queries in
1114
    func_odbc.conf, as the queries no longer need to be specified on a single
1127
    func_odbc.conf, as the queries no longer need to be specified on a single
1115
    line.
1128
    line.
1116
  * CDR config file, cdr.conf, has an added option, "initiatedseconds", 
1129
  * CDR config file, cdr.conf, has an added option, "initiatedseconds", 
1117
    which will add a second to the billsec when the ending
1130
    which will add a second to the billsec when the ending
1118
    time is set, if the number in the microseconds field of the end time is 
1131
    time is set, if the number in the microseconds field of the end time is 
1119
    greater than the number of microseconds in the answer time. This allows
1132
    greater than the number of microseconds in the answer time. This allows
1120
    users to count the 'initiated' seconds in their billing records. 
1133
    users to count the 'initiated' seconds in their billing records. 
1121

    
   
1134

   
1122
------------------------------------------------------------------------------
1135
------------------------------------------------------------------------------
1123
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
1136
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
1124
------------------------------------------------------------------------------
1137
------------------------------------------------------------------------------
1125

    
   
1138

   
1126
AMI - The manager (TCP/TLS/HTTP)
1139
AMI - The manager (TCP/TLS/HTTP)
1127
--------------------------------
1140
--------------------------------
1128
  * Manager has undergone a lot of changes, all of them documented
1141
  * Manager has undergone a lot of changes, all of them documented
1129
    in doc/manager_1_1.txt
1142
    in doc/manager_1_1.txt
1130
  * Manager version has changed to 1.1
1143
  * Manager version has changed to 1.1
1131
  * Added a new action 'CoreShowChannels' to list currently defined channels
1144
  * Added a new action 'CoreShowChannels' to list currently defined channels
1132
     and some information about them. 
1145
     and some information about them. 
1133
  * Added a new action 'SIPshowregistry' to list SIP registrations.
1146
  * Added a new action 'SIPshowregistry' to list SIP registrations.
1134
  * Added TLS support for the manager interface and HTTP server
1147
  * Added TLS support for the manager interface and HTTP server
1135
  * Added the URI redirect option for the built-in HTTP server
1148
  * Added the URI redirect option for the built-in HTTP server
1136
  * The output of CallerID in Manager events is now more consistent.
1149
  * The output of CallerID in Manager events is now more consistent.
1137
     CallerIDNum is used for number and CallerIDName for name.
1150
     CallerIDNum is used for number and CallerIDName for name.
1138
  * Enable https support for builtin web server.
1151
  * Enable https support for builtin web server.
1139
     See configs/http.conf.sample for details.
1152
     See configs/http.conf.sample for details.
1140
  * Added a new action, GetConfigJSON, which can return the contents of an
1153
  * Added a new action, GetConfigJSON, which can return the contents of an
1141
     Asterisk configuration file in JSON format.  This is intended to help
1154
     Asterisk configuration file in JSON format.  This is intended to help
1142
     improve the performance of AJAX applications using the manager interface
1155
     improve the performance of AJAX applications using the manager interface
1143
     over HTTP.
1156
     over HTTP.
1144
  * SIP and IAX manager events now use "ChannelType" in all cases where we 
1157
  * SIP and IAX manager events now use "ChannelType" in all cases where we 
1145
     indicate channel driver. Previously, we used a mixture of "Channel"
1158
     indicate channel driver. Previously, we used a mixture of "Channel"
1146
     and "ChannelDriver" headers.
1159
     and "ChannelDriver" headers.
1147
  * Added a "Bridge" action which allows you to bridge any two channels that
1160
  * Added a "Bridge" action which allows you to bridge any two channels that
1148
     are currently active on the system.
1161
     are currently active on the system.
1149
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1162
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1150
     the voicemail users setup.
1163
     the voicemail users setup.
1151
  * Added 'DBDel' and 'DBDelTree' manager commands.
1164
  * Added 'DBDel' and 'DBDelTree' manager commands.
1152
  * cdr_manager now reports events via the "cdr" level, separating it from
1165
  * cdr_manager now reports events via the "cdr" level, separating it from
1153
     the very verbose "call" level.
1166
     the very verbose "call" level.
1154
  * Manager users are now stored in memory. If you change the manager account
1167
  * Manager users are now stored in memory. If you change the manager account
1155
    list (delete or add accounts) you need to reload manager.
1168
    list (delete or add accounts) you need to reload manager.
1156
  * Added Masquerade manager event for when a masquerade happens between
1169
  * Added Masquerade manager event for when a masquerade happens between
1157
     two channels.
1170
     two channels.
1158
  * Added "manager reload" command for the CLI
1171
  * Added "manager reload" command for the CLI
1159
  * Lots of commands that only provided information are now allowed under the
1172
  * Lots of commands that only provided information are now allowed under the
1160
     Reporting privilege, instead of only under Call or System.
1173
     Reporting privilege, instead of only under Call or System.
1161
  * The IAX* commands now require either System or Reporting privilege, to
1174
  * The IAX* commands now require either System or Reporting privilege, to
1162
     mirror the privileges of the SIP* commands.
1175
     mirror the privileges of the SIP* commands.
1163
  * Added ability to retrieve list of categories in a config file.
1176
  * Added ability to retrieve list of categories in a config file.
1164
  * Added ability to retrieve the content of a particular category.
1177
  * Added ability to retrieve the content of a particular category.
1165
  * Added ability to empty a context.
1178
  * Added ability to empty a context.
1166
  * Created new action to create a new file.
1179
  * Created new action to create a new file.
1167
  * Updated delete action to allow deletion by line number with respect to category.
1180
  * Updated delete action to allow deletion by line number with respect to category.
1168
  * Added new action insert to add new variable to category at specified line.
1181
  * Added new action insert to add new variable to category at specified line.
1169
  * Updated action newcat to allow new category to be inserted in file above another
1182
  * Updated action newcat to allow new category to be inserted in file above another
1170
    existing category.
1183
    existing category.
1171
  * Added new event "JitterBufStats" in the IAX2 channel
1184
  * Added new event "JitterBufStats" in the IAX2 channel
1172
  * Originate now requires the Originate privilege and, if you want to call out
1185
  * Originate now requires the Originate privilege and, if you want to call out
1173
    to a subshell, it requires the System privilege, as well.  This was done to
1186
    to a subshell, it requires the System privilege, as well.  This was done to
1174
    enhance manager security.
1187
    enhance manager security.
1175
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" 
1188
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" 
1176
  * New command: Atxfer. See doc/manager_1_1.txt for more details or 
1189
  * New command: Atxfer. See doc/manager_1_1.txt for more details or 
1177
    manager show command Atxfer from the CLI
1190
    manager show command Atxfer from the CLI
1178
  * New command: IAXregistry. See doc/manager_1_1.txt for more details or
1191
  * New command: IAXregistry. See doc/manager_1_1.txt for more details or
1179
    manager show command IAXregistry from the CLI
1192
    manager show command IAXregistry from the CLI
1180

    
   
1193

   
1181
Dialplan functions
1194
Dialplan functions
1182
------------------
1195
------------------
1183
  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1196
  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1184
     state in the dialplan, as well as creating custom device states that are
1197
     state in the dialplan, as well as creating custom device states that are
1185
     controllable from the dialplan.
1198
     controllable from the dialplan.
1186
  * Extend CALLERID() function with "pres" and "ton" parameters to
1199
  * Extend CALLERID() function with "pres" and "ton" parameters to
1187
     fetch string representation of calling number presentation indicator
1200
     fetch string representation of calling number presentation indicator
1188
     and numeric representation of type of calling number value.
1201
     and numeric representation of type of calling number value.
1189
  * MailboxExists converted to dialplan function
1202
  * MailboxExists converted to dialplan function
1190
  * A new option to Dial() for telling IP phones not to count the call
1203
  * A new option to Dial() for telling IP phones not to count the call
1191
     as "missed" when dial times out and cancels.
1204
     as "missed" when dial times out and cancels.
1192
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1205
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1193
     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
1206
     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
1194
     held for any given channel.  Also, locks are automatically freed when a
1207
     held for any given channel.  Also, locks are automatically freed when a
1195
     channel is hung up.
1208
     channel is hung up.
1196
  * Added HINT() dialplan function that allows retrieving hint information.
1209
  * Added HINT() dialplan function that allows retrieving hint information.
1197
     Hints are mappings between extensions and devices for the sake of 
1210
     Hints are mappings between extensions and devices for the sake of 
1198
     determining the state of an extension.  This function can retrieve the list
1211
     determining the state of an extension.  This function can retrieve the list
1199
     of devices or the name associated with a hint.
1212
     of devices or the name associated with a hint.
1200
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1213
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1201
    of any extension.
1214
    of any extension.
1202
  * Added SYSINFO() dialplan function which allows retrieval of system information
1215
  * Added SYSINFO() dialplan function which allows retrieval of system information
1203
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1216
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1204
     the existence of a dialplan target.
1217
     the existence of a dialplan target.
1205
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1218
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1206
     upper and lower case, respectively.
1219
     upper and lower case, respectively.
1207
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1220
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1208
     ID for the call (not the Asterisk call ID or unique ID), provided that the
1221
     ID for the call (not the Asterisk call ID or unique ID), provided that the
1209
     channel driver supports this. For SIP, you get the SIP call-ID for the
1222
     channel driver supports this. For SIP, you get the SIP call-ID for the
1210
     bridged channel which you can store in the CDR with a custom field.
1223
     bridged channel which you can store in the CDR with a custom field.
1211

    
   
1224

   
1212
CLI Changes
1225
CLI Changes
1213
-----------
1226
-----------
1214
  * Added CLI permissions, config file: cli_permissions.conf
1227
  * Added CLI permissions, config file: cli_permissions.conf
1215
     default is to allow all commands for every local user/group.
1228
     default is to allow all commands for every local user/group.
1216
     Also this new feature added three new CLI commands:
1229
     Also this new feature added three new CLI commands:
1217
      - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1230
      - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1218
      - cli reload permissions
1231
      - cli reload permissions
1219
      - cli show permissions
1232
      - cli show permissions
1220
  * New CLI command "core show hint" (usage: core show hint <exten>)
1233
  * New CLI command "core show hint" (usage: core show hint <exten>)
1221
  * New CLI command "core show settings"
1234
  * New CLI command "core show settings"
1222
  * Added 'core show channels count' CLI command.
1235
  * Added 'core show channels count' CLI command.
1223
  * Added the ability to set the core debug and verbose values on a per-file basis.
1236
  * Added the ability to set the core debug and verbose values on a per-file basis.
1224
  * Added 'queue pause member' and 'queue unpause member' CLI commands
1237
  * Added 'queue pause member' and 'queue unpause member' CLI commands
1225
  * Ability to set process limits ("ulimit") without restarting Asterisk
1238
  * Ability to set process limits ("ulimit") without restarting Asterisk
1226
  * Enhanced "agi debug" to print the channel name as a prefix to the debug
1239
  * Enhanced "agi debug" to print the channel name as a prefix to the debug
1227
     output to make debugging on busy systems much easier.
1240
     output to make debugging on busy systems much easier.
1228
  * New CLI commands "dialplan set extenpatternmatching true/false"
1241
  * New CLI commands "dialplan set extenpatternmatching true/false"
1229
  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1242
  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1230
  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
1243
  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
1231
    listed in the startup_commands section of cli.conf will get executed.
1244
    listed in the startup_commands section of cli.conf will get executed.
1232
  * Added a CLI command, "devstate change", which allows you to set custom device
1245
  * Added a CLI command, "devstate change", which allows you to set custom device
1233
     states from the func_devstate module that provides the DEVICE_STATE() function
1246
     states from the func_devstate module that provides the DEVICE_STATE() function
1234
     and handling of the "Custom:" devices.
1247
     and handling of the "Custom:" devices.
1235
  * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1248
  * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1236
    sorted into the different possible callbacks, with the number of entries
1249
    sorted into the different possible callbacks, with the number of entries
1237
    currently scheduled for each. Gives you a feel for how busy the sip channel
1250
    currently scheduled for each. Gives you a feel for how busy the sip channel
1238
    driver is.
1251
    driver is.
1239
  * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1252
  * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1240
  * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1253
  * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1241
    (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1254
    (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1242

    
   
1255

   
1243
SIP changes
1256
SIP changes
1244
-----------
1257
-----------
1245
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
1258
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
1246
    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1259
    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1247
    for a received call.  If it is detected, the channel will jump to the 
1260
    for a received call.  If it is detected, the channel will jump to the 
1248
    'fax' extension in the dialplan.
1261
    'fax' extension in the dialplan.
1249
  * The default SIP useragent= identifier now includes the Asterisk version
1262
  * The default SIP useragent= identifier now includes the Asterisk version
1250
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1263
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1251
     If set, and the incoming request carries authentication info,
1264
     If set, and the incoming request carries authentication info,
1252
     the username to match in the users list is taken from the Digest header
1265
     the username to match in the users list is taken from the Digest header
1253
     rather than from the From: field. This feature is considered experimental.
1266
     rather than from the From: field. This feature is considered experimental.
1254
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1267
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1255
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1268
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1256
  * The "localmask" setting was removed in version 1.2 and the reminder about it
1269
  * The "localmask" setting was removed in version 1.2 and the reminder about it
1257
     being removed is now also removed.
1270
     being removed is now also removed.
1258
  * A new option "busylevel" for setting a level of calls where asterisk reports
1271
  * A new option "busylevel" for setting a level of calls where asterisk reports
1259
     a device as busy, to separate it from call-limit. This value is also added
1272
     a device as busy, to separate it from call-limit. This value is also added
1260
     to the SIP_PEER dialplan function.
1273
     to the SIP_PEER dialplan function.
1261
  * A new realtime family called "sipregs" is now supported to store SIP registration
1274
  * A new realtime family called "sipregs" is now supported to store SIP registration
1262
     data. If this family is defined, "sippeers" will be used for configuration and
1275
     data. If this family is defined, "sippeers" will be used for configuration and
1263
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1276
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1264
     registration data, as before.
1277
     registration data, as before.
1265
  * The SIPPEER function have new options for port address, call and pickup groups
1278
  * The SIPPEER function have new options for port address, call and pickup groups
1266
  * Added support for T.140 realtime text in SIP/RTP
1279
  * Added support for T.140 realtime text in SIP/RTP
1267
  * The "checkmwi" option has been removed from sip.conf, as it is no longer
1280
  * The "checkmwi" option has been removed from sip.conf, as it is no longer
1268
     required due to the restructuring of how MWI is handled.  See the descriptions 
1281
     required due to the restructuring of how MWI is handled.  See the descriptions 
1269
     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
1282
     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
1270
     for more information.
1283
     for more information.
1271
  * Added rtpdest option to CHANNEL() dialplan function.
1284
  * Added rtpdest option to CHANNEL() dialplan function.
1272
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1285
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1273
  * SIP now adds a header to the CANCEL if the call was answered by another phone
1286
  * SIP now adds a header to the CANCEL if the call was answered by another phone
1274
     in the same dial command, or if the new c option in dial() is used.
1287
     in the same dial command, or if the new c option in dial() is used.
1275
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1288
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1276
     states it is not needed. For phones, however, that do require it the "registertrying" option
1289
     states it is not needed. For phones, however, that do require it the "registertrying" option
1277
     has been added so it can be enabled. 
1290
     has been added so it can be enabled. 
1278
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
1291
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
1279
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1292
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1280
     used to enable this functionality).
1293
     used to enable this functionality).
1281
  * New settings for timer T1 and timer B on a global level or per device. This makes it 
1294
  * New settings for timer T1 and timer B on a global level or per device. This makes it 
1282
     possible to force timeout faster on non-responsive SIP servers. These settings are
1295
     possible to force timeout faster on non-responsive SIP servers. These settings are
1283
     considered advanced, so don't use them unless you have a problem.
1296
     considered advanced, so don't use them unless you have a problem.
1284
  * Added a dial string option to be able to set the To: header in an INVITE to any
1297
  * Added a dial string option to be able to set the To: header in an INVITE to any
1285
     SIP uri.
1298
     SIP uri.
1286
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1299
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1287
     the qualify frequency.
1300
     the qualify frequency.
1288
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
1301
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
1289
     were not properly torn down due to network or endpoint failures during an established
1302
     were not properly torn down due to network or endpoint failures during an established
1290
     SIP session.
1303
     SIP session.
1291
  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
1304
  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
1292
     configs/sip.conf.sample for more information on how it is used.
1305
     configs/sip.conf.sample for more information on how it is used.
1293
  * Added a new configuration option "authfailureevents" that enables manager events when
1306
  * Added a new configuration option "authfailureevents" that enables manager events when
1294
    a peer can't authenticate properly. 
1307
    a peer can't authenticate properly. 
1295
  * Added DNS manager support to registrations for peers not referencing a peer entry.
1308
  * Added DNS manager support to registrations for peers not referencing a peer entry.
1296

    
   
1309

   
1297
IAX2 changes
1310
IAX2 changes
1298
------------
1311
------------
1299
  * Added the trunkmaxsize configuration option to chan_iax2.
1312
  * Added the trunkmaxsize configuration option to chan_iax2.
1300
  * Added the srvlookup option to iax.conf
1313
  * Added the srvlookup option to iax.conf
1301
  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
1314
  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
1302
     dialplan function.
1315
     dialplan function.
1303

    
   
1316

   
1304
XMPP Google Talk/Jingle changes
1317
XMPP Google Talk/Jingle changes
1305
-------------------------------
1318
-------------------------------
1306
  * Added the bindaddr option to gtalk.conf.
1319
  * Added the bindaddr option to gtalk.conf.
1307

    
   
1320

   
1308
Skinny changes
1321
Skinny changes
1309
-------------
1322
-------------
1310
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1323
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1311
  * Proper codec support in chan_skinny.
1324
  * Proper codec support in chan_skinny.
1312
  * Added settings for IP and Ethernet QoS requests
1325
  * Added settings for IP and Ethernet QoS requests
1313

    
   
1326

   
1314
MGCP changes
1327
MGCP changes
1315
------------
1328
------------
1316
  * Added separate settings for media QoS in mgcp.conf
1329
  * Added separate settings for media QoS in mgcp.conf
1317

    
   
1330

   
1318
Console Channel Driver changes
1331
Console Channel Driver changes
1319
------------------------------
1332
------------------------------
1320
  * Added experimental support for video send & receive to chan_oss.
1333
  * Added experimental support for video send & receive to chan_oss.
1321
    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1334
    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1322
    a video source.
1335
    a video source.
1323

    
   
1336

   
1324
Phone channel changes (chan_phone)
1337
Phone channel changes (chan_phone)
1325
----------------------------------
1338
----------------------------------
1326
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1339
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1327

    
   
1340

   
1328
H.323 channel Changes
1341
H.323 channel Changes
1329
---------------------
1342
---------------------
1330
  * H323 remote hold notification support added (by NOTIFY message
1343
  * H323 remote hold notification support added (by NOTIFY message
1331
     and/or H.450 supplementary service)
1344
     and/or H.450 supplementary service)
1332

    
   
1345

   
1333
Local channel changes
1346
Local channel changes
1334
---------------------
1347
---------------------
1335
  * The device state functionality in the Local channel driver has been updated
1348
  * The device state functionality in the Local channel driver has been updated
1336
     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1349
     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1337
     to just UNKNOWN if the extension exists.
1350
     to just UNKNOWN if the extension exists.
1338
  * Added jitterbuffer support for chan_local.  This allows you to use the
1351
  * Added jitterbuffer support for chan_local.  This allows you to use the
1339
     generic jitterbuffer on incoming calls going to Asterisk applications.
1352
     generic jitterbuffer on incoming calls going to Asterisk applications.
1340
     For example, this would allow you to use a jitterbuffer for an incoming
1353
     For example, this would allow you to use a jitterbuffer for an incoming
1341
     SIP call to Voicemail by putting a Local channel in the middle.  This
1354
     SIP call to Voicemail by putting a Local channel in the middle.  This
1342
     feature is enabled by using the 'j' option in the Dial string to the Local
1355
     feature is enabled by using the 'j' option in the Dial string to the Local
1343
     channel in conjunction with the existing 'n' option for local channels.
1356
     channel in conjunction with the existing 'n' option for local channels.
1344
  * A 'b' option has been added which causes chan_local to return the actual channel
1357
  * A 'b' option has been added which causes chan_local to return the actual channel
1345
     that is behind it when queried. This is useful for transfer scenarios as the
1358
     that is behind it when queried. This is useful for transfer scenarios as the
1346
     actual channel will be transferred, not the Local channel.
1359
     actual channel will be transferred, not the Local channel.
1347

    
   
1360

   
1348
Agent channel changes
1361
Agent channel changes
1349
----------------------
1362
----------------------
1350
  * The ackcall and endcall options are now supplemented with options acceptdtmf
1363
  * The ackcall and endcall options are now supplemented with options acceptdtmf
1351
    and enddtmf. These allow for the DTMF keypress to be configurable. The options
1364
    and enddtmf. These allow for the DTMF keypress to be configurable. The options
1352
    default to their old hard-coded values ('#' and '*' respectively) so this should
1365
    default to their old hard-coded values ('#' and '*' respectively) so this should
1353
    not break any existing agent installations.
1366
    not break any existing agent installations.
1354

    
   
1367

   
1355
DAHDI channel driver (chan_dahdi) Changes
1368
DAHDI channel driver (chan_dahdi) Changes
1356
----------------------------------------
1369
----------------------------------------
1357
  * SS7 support (via libss7 library)
1370
  * SS7 support (via libss7 library)
1358
  * In India, some carriers transmit CID via dtmf. Some code has been added
1371
  * In India, some carriers transmit CID via dtmf. Some code has been added
1359
     that will handle some situations. The cidstart=polarity_IN choice has been added for
1372
     that will handle some situations. The cidstart=polarity_IN choice has been added for
1360
     those carriers that transmit CID via dtmf after a polarity change.
1373
     those carriers that transmit CID via dtmf after a polarity change.
1361
  * CID matching information is now shown when doing 'dialplan show'.
1374
  * CID matching information is now shown when doing 'dialplan show'.
1362
  * Added dahdi show version CLI command.
1375
  * Added dahdi show version CLI command.
1363
  * Added setvar support to chan_dahdi.conf channel entries.
1376
  * Added setvar support to chan_dahdi.conf channel entries.
1364
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
1377
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
1365
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
1378
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
1366
     the script specified in the mwimonitornotify option is executed.  An internal
1379
     the script specified in the mwimonitornotify option is executed.  An internal
1367
     event indicating the new state of the mailbox is also generated, so that
1380
     event indicating the new state of the mailbox is also generated, so that
1368
     the normal MWI facilities in Asterisk work as usual.
1381
     the normal MWI facilities in Asterisk work as usual.
1369
  * Added signalling type 'auto', which attempts to use the same signalling type
1382
  * Added signalling type 'auto', which attempts to use the same signalling type
1370
     for a channel as configured in DAHDI. This is primarily designed for analog
1383
     for a channel as configured in DAHDI. This is primarily designed for analog
1371
     ports, but will also work for digital ports that are configured for FXS or FXO
1384
     ports, but will also work for digital ports that are configured for FXS or FXO
1372
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
1385
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
1373
     does not specify signalling for a channel (which is unlikely as the sample
1386
     does not specify signalling for a channel (which is unlikely as the sample
1374
     configuration file has always recommended specifying it for every channel) then
1387
     configuration file has always recommended specifying it for every channel) then
1375
     the 'auto' mode will be used for that channel if possible.
1388
     the 'auto' mode will be used for that channel if possible.
1376
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1389
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1377
     state for a channel; also ensured that the DNDState Manager event is
1390
     state for a channel; also ensured that the DNDState Manager event is
1378
     emitted no matter how the DND state is set or cleared.
1391
     emitted no matter how the DND state is set or cleared.
1379

    
   
1392

   
1380
New Channel Drivers
1393
New Channel Drivers
1381
-------------------
1394
-------------------
1382
  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
1395
  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
1383
     configs/unistim.conf.sample for details.  This new channel driver allows
1396
     configs/unistim.conf.sample for details.  This new channel driver allows
1384
     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1397
     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1385
  * Added a new channel driver, chan_console, which uses portaudio as a cross
1398
  * Added a new channel driver, chan_console, which uses portaudio as a cross
1386
     platform audio interface.  It was written as a channel driver that would
1399
     platform audio interface.  It was written as a channel driver that would
1387
     work with Mac CoreAudio, but portaudio supports a number of other audio
1400
     work with Mac CoreAudio, but portaudio supports a number of other audio
1388
     interfaces, as well. Note that this channel driver requires v19 or higher
1401
     interfaces, as well. Note that this channel driver requires v19 or higher
1389
     of portaudio; older versions have a different API.
1402
     of portaudio; older versions have a different API.
1390
 
1403
 
1391
DUNDi changes
1404
DUNDi changes
1392
-------------
1405
-------------
1393
  * Added the ability to specify arguments to the Dial application when using
1406
  * Added the ability to specify arguments to the Dial application when using
1394
     the DUNDi switch in the dialplan.
1407
     the DUNDi switch in the dialplan.
1395
  * Added the ability to set weights for responses dynamically.  This can be
1408
  * Added the ability to set weights for responses dynamically.  This can be
1396
     done using a global variable or a dialplan function.  Using the SHELL()
1409
     done using a global variable or a dialplan function.  Using the SHELL()
1397
     function would allow you to have an external script set the weight for
1410
     function would allow you to have an external script set the weight for
1398
     each response.
1411
     each response.
1399
  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
1412
  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
1400
     functions will allow you to initiate a DUNDi query from the dialplan,
1413
     functions will allow you to initiate a DUNDi query from the dialplan,
1401
     find out how many results there are, and access each one.
1414
     find out how many results there are, and access each one.
1402
  * Added the ability to specifiy a port for a dundi peer.
1415
  * Added the ability to specifiy a port for a dundi peer.
1403

    
   
1416

   
1404
ENUM changes
1417
ENUM changes
1405
------------
1418
------------
1406
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
1419
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
1407
     functions will allow you to initiate an ENUM lookup from the dialplan,
1420
     functions will allow you to initiate an ENUM lookup from the dialplan,
1408
     and Asterisk will cache the results.  ENUMRESULT can be used to access
1421
     and Asterisk will cache the results.  ENUMRESULT can be used to access
1409
     the results without doing multiple DNS queries.
1422
     the results without doing multiple DNS queries.
1410

    
   
1423

   
1411
Voicemail Changes
1424
Voicemail Changes
1412
-----------------
1425
-----------------
1413
  * Added the ability to customize which sound files are used for some of the
1426
  * Added the ability to customize which sound files are used for some of the
1414
     prompts within the Voicemail application by changing them in voicemail.conf
1427
     prompts within the Voicemail application by changing them in voicemail.conf
1415
  * Added the ability for the "voicemail show users" CLI command to show users
1428
  * Added the ability for the "voicemail show users" CLI command to show users
1416
     configured by the dynamic realtime configuration method.
1429
     configured by the dynamic realtime configuration method.
1417
  * MWI (Message Waiting Indication) handling has been significantly
1430
  * MWI (Message Waiting Indication) handling has been significantly
1418
     restructured internally to Asterisk.  It is now totally event based
1431
     restructured internally to Asterisk.  It is now totally event based
1419
     instead of polling based.  The voicemail application will notify other
1432
     instead of polling based.  The voicemail application will notify other
1420
     modules that have subscribed to MWI events when something in the mailbox
1433
     modules that have subscribed to MWI events when something in the mailbox
1421
     changes.
1434
     changes.
1422
    This also means that if any other entity outside of Asterisk is changing
1435
    This also means that if any other entity outside of Asterisk is changing
1423
     the contents of mailboxes, then the voicemail application still needs to
1436
     the contents of mailboxes, then the voicemail application still needs to
1424
     poll for changes.  Examples of situations that would require this option
1437
     poll for changes.  Examples of situations that would require this option
1425
     are web interfaces to voicemail or an email client in the case of using
1438
     are web interfaces to voicemail or an email client in the case of using
1426
     IMAP storage.  So, two new options have been added to voicemail.conf
1439
     IMAP storage.  So, two new options have been added to voicemail.conf
1427
     to account for this: "pollmailboxes" and "pollfreq".  See the sample
1440
     to account for this: "pollmailboxes" and "pollfreq".  See the sample
1428
     configuration file for details.
1441
     configuration file for details.
1429
  * Added "tw" language support
1442
  * Added "tw" language support
1430
  * Added support for storage of greetings using an IMAP server
1443
  * Added support for storage of greetings using an IMAP server
1431
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
1444
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
1432
  * SMDI is now enabled in voicemail using the smdienable option.
1445
  * SMDI is now enabled in voicemail using the smdienable option.
1433
  * A "lockmode" option has been added to asterisk.conf to configure the file
1446
  * A "lockmode" option has been added to asterisk.conf to configure the file
1434
     locking method used for voicemail, and potentially other things in the
1447
     locking method used for voicemail, and potentially other things in the
1435
     future.  The default is the old behavior, lockfile.  However, there is a
1448
     future.  The default is the old behavior, lockfile.  However, there is a
1436
     new method, "flock", that uses a different method for situations where the
1449
     new method, "flock", that uses a different method for situations where the
1437
     lockfile will not work, such as on SMB/CIFS mounts.
1450
     lockfile will not work, such as on SMB/CIFS mounts.
1438
  * Added the ability to backup deleted messages, to ease recovery in the case
1451
  * Added the ability to backup deleted messages, to ease recovery in the case
1439
     that a user accidentally deletes a message, and discovers that they need it.
1452
     that a user accidentally deletes a message, and discovers that they need it.
1440
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
1453
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
1441
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
1454
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
1442
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1455
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1443
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
1456
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
1444
     outside entity is modifying the state of the mailbox (such as IMAP storage or
1457
     outside entity is modifying the state of the mailbox (such as IMAP storage or
1445
     a web interface of some kind).
1458
     a web interface of some kind).
1446
  * Added the support for marking messages as "urgent." There are two methods to accomplish
1459
  * Added the support for marking messages as "urgent." There are two methods to accomplish
1447
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1460
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1448
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1461
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1449
     the message as urgent after he has recorded a voicemail by following the voice instructions.
1462
     the message as urgent after he has recorded a voicemail by following the voice instructions.
1450
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1463
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1451
     messages
1464
     messages
1452

    
   
1465

   
1453
Queue changes
1466
Queue changes
1454
-------------
1467
-------------
1455
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1468
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1456
     used across multiple queues.
1469
     used across multiple queues.
1457
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
1470
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
1458
     setqueueentryvar options for each queue, see queues.conf.sample for details.
1471
     setqueueentryvar options for each queue, see queues.conf.sample for details.
1459
  * Added keepstats option to queues.conf which will keep queue
1472
  * Added keepstats option to queues.conf which will keep queue
1460
     statistics during a reload.
1473
     statistics during a reload.
1461
  * setinterfacevar option in queues.conf also now sets a variable
1474
  * setinterfacevar option in queues.conf also now sets a variable
1462
     called MEMBERNAME which contains the member's name.
1475
     called MEMBERNAME which contains the member's name.
1463
  * Added 'Strategy' field to manager event QueueParams which represents
1476
  * Added 'Strategy' field to manager event QueueParams which represents
1464
     the queue strategy in use. 
1477
     the queue strategy in use. 
1465
  * Added option to run macro when a queue member is connected to a caller, 
1478
  * Added option to run macro when a queue member is connected to a caller, 
1466
     see queues.conf.sample for details.
1479
     see queues.conf.sample for details.
1467
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1480
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1468
     does not count paused queue members as unavailable.
1481
     does not count paused queue members as unavailable.
1469
  * Added min-announce-frequency option to queues.conf which allows you to control the
1482
  * Added min-announce-frequency option to queues.conf which allows you to control the
1470
     minimum amount of time between queue announcements for use when the caller's queue
1483
     minimum amount of time between queue announcements for use when the caller's queue
1471
     position changes frequently.
1484
     position changes frequently.
1472
  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1485
  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1473
     queue log.
1486
     queue log.
1474
  * Added ability for non-realtime queues to have realtime members
1487
  * Added ability for non-realtime queues to have realtime members
1475
  * Added the "linear" strategy to queues.
1488
  * Added the "linear" strategy to queues.
1476
  * Added the "wrandom" strategy to queues.
1489
  * Added the "wrandom" strategy to queues.
1477
  * Added new channel variable QUEUE_MIN_PENALTY
1490
  * Added new channel variable QUEUE_MIN_PENALTY
1478
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1491
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1479
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
1492
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
1480
  * Added a new parameter for member definition, called state_interface. This may be
1493
  * Added a new parameter for member definition, called state_interface. This may be
1481
    used so that a member may be called via one interface but have a different interface's
1494
    used so that a member may be called via one interface but have a different interface's
1482
    device state reported.
1495
    device state reported.
1483
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1496
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1484
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1497
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1485
    "manager show command QueueReset."
1498
    "manager show command QueueReset."
1486
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1499
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1487
    specified by the periodic-announce option, then one will be chosen randomly when it is time
1500
    specified by the periodic-announce option, then one will be chosen randomly when it is time
1488
    to play a periodic announcment
1501
    to play a periodic announcment
1489
  * New configuration options: announce-position now takes two more values in addition to "yes" and
1502
  * New configuration options: announce-position now takes two more values in addition to "yes" and
1490
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1503
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1491
    announce-position-limit. By setting announce-position to "limit" callers will only have their
1504
    announce-position-limit. By setting announce-position to "limit" callers will only have their
1492
    position announced if their position is less than what is specified by announce-position-limit.
1505
    position announced if their position is less than what is specified by announce-position-limit.
1493
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1506
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1494
    will be told that their are more than announce-position-limit callers waiting.
1507
    will be told that their are more than announce-position-limit callers waiting.
1495
  * Two new queue log events have been added. An ADDMEMBER event will be logged
1508
  * Two new queue log events have been added. An ADDMEMBER event will be logged
1496
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
1509
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
1497
    when a realtime queue member is removed. Since there is no calling channel associated
1510
    when a realtime queue member is removed. Since there is no calling channel associated
1498
    with these events, the string "REALTIME" is placed where the channel's unique id
1511
    with these events, the string "REALTIME" is placed where the channel's unique id
1499
    is typically placed.
1512
    is typically placed.
1500
  * The configuration method for the "joinempty" and "leavewhenempty" options has
1513
  * The configuration method for the "joinempty" and "leavewhenempty" options has
1501
    changed to a comma-separated list of methods of determining member availability
1514
    changed to a comma-separated list of methods of determining member availability
1502
    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1515
    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1503
    values are still accepted for backwards-compatibility, though.
1516
    values are still accepted for backwards-compatibility, though.
1504
  * The average talktime is now calculated on queues. This information is reported via the
1517
  * The average talktime is now calculated on queues. This information is reported via the
1505
    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1518
    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1506
    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1519
    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1507
    the queue.
1520
    the queue.
1508

    
   
1521

   
1509
MeetMe Changes
1522
MeetMe Changes
1510
--------------
1523
--------------
1511
  * The 'o' option to provide an optimization has been removed and its functionality 
1524
  * The 'o' option to provide an optimization has been removed and its functionality 
1512
     has been enabled by default.
1525
     has been enabled by default.
1513
  * When a conference is created, the UNIQUEID of the channel that caused it to be
1526
  * When a conference is created, the UNIQUEID of the channel that caused it to be
1514
     created is stored.  Then, every channel that joins the conference will have the
1527
     created is stored.  Then, every channel that joins the conference will have the
1515
     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
1528
     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
1516
     callers that come and go from long standing conferences.
1529
     callers that come and go from long standing conferences.
1517
  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1530
  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1518
     except it does operations on a channel by name, instead of number in a conference.
1531
     except it does operations on a channel by name, instead of number in a conference.
1519
     This is a very useful feature in combination with the 'X' option to ChanSpy.
1532
     This is a very useful feature in combination with the 'X' option to ChanSpy.
1520
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1533
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1521
     when kicked out.
1534
     when kicked out.
1522
  * Added new RealTime functionality to provide support for scheduled conferencing.
1535
  * Added new RealTime functionality to provide support for scheduled conferencing.
1523
     This includes optional messages to the caller if they attempt to join before
1536
     This includes optional messages to the caller if they attempt to join before
1524
     the schedule start time, or to allow the caller to join the conference early.
1537
     the schedule start time, or to allow the caller to join the conference early.
1525
     Also included is optional support for limiting the number of callers per
1538
     Also included is optional support for limiting the number of callers per
1526
     RealTime conference.
1539
     RealTime conference.
1527
  * Added the S() and L() options to the MeetMe application.  These are pretty
1540
  * Added the S() and L() options to the MeetMe application.  These are pretty
1528
     much identical to the S() and L() options to Dial().  They let you set
1541
     much identical to the S() and L() options to Dial().  They let you set
1529
     timeouts for the conference, as well as have warning sounds played to
1542
     timeouts for the conference, as well as have warning sounds played to
1530
     let the caller know how much time is left, and when it is running out.
1543
     let the caller know how much time is left, and when it is running out.
1531
  * Added the ability to do "meetme concise" with the "meetme" CLI command.
1544
  * Added the ability to do "meetme concise" with the "meetme" CLI command.
1532
     This extends the concise capabilities of this CLI command to include
1545
     This extends the concise capabilities of this CLI command to include
1533
     listing all conferences, instead of an addition to the other sub commands
1546
     listing all conferences, instead of an addition to the other sub commands
1534
     for the "meetme" command.
1547
     for the "meetme" command.
1535
  * Added the ability to specify the music on hold class used to play into the
1548
  * Added the ability to specify the music on hold class used to play into the
1536
     conference when there is only one member and the M option is used.
1549
     conference when there is only one member and the M option is used.
1537
  * Added MEETME_INFO dialplan function which provides a way to query
1550
  * Added MEETME_INFO dialplan function which provides a way to query
1538
     various properties of a Meetme conference.
1551
     various properties of a Meetme conference.
1539
  * Added new admin features: *81: Roll call, *82: eject all, *83: mute all, 
1552
  * Added new admin features: *81: Roll call, *82: eject all, *83: mute all, 
1540
     and *84: record in-conf
1553
     and *84: record in-conf
1541

    
   
1554

   
1542
Other Dialplan Application Changes
1555
Other Dialplan Application Changes
1543
----------------------------------
1556
----------------------------------
1544
  * Argument support for Gosub application
1557
  * Argument support for Gosub application
1545
  * From the to-do lists: straighten out the app timeout args:
1558
  * From the to-do lists: straighten out the app timeout args:
1546
     Wait() app now really does 0.3 seconds- was truncating arg to an int.
1559
     Wait() app now really does 0.3 seconds- was truncating arg to an int.
1547
     WaitExten() same as Wait().
1560
     WaitExten() same as Wait().
1548
     Congestion() - Now takes floating pt. argument.
1561
     Congestion() - Now takes floating pt. argument.
1549
     Busy() - now takes floating pt. argument.
1562
     Busy() - now takes floating pt. argument.
1550
     Read() - timeout now can be floating pt.
1563
     Read() - timeout now can be floating pt.
1551
     WaitForRing() now takes floating pt timeout arg.
1564
     WaitForRing() now takes floating pt timeout arg.
1552
     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1565
     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1553
  * Added 's' option to Page application.
1566
  * Added 's' option to Page application.
1554
  * Added an optional timeout argument to the Page application.
1567
  * Added an optional timeout argument to the Page application.
1555
  * Added 'E', 'V', and 'P' commands to ExternalIVR.
1568
  * Added 'E', 'V', and 'P' commands to ExternalIVR.
1556
  * Added 'o' and 'X' options to Chanspy.
1569
  * Added 'o' and 'X' options to Chanspy.
1557
  * Added a new dialplan application, Bridge, which allows you to bridge the
1570
  * Added a new dialplan application, Bridge, which allows you to bridge the
1558
     calling channel to any other active channel on the system.
1571
     calling channel to any other active channel on the system.
1559
  * Added the ability to specify a music on hold class to play instead of ringing
1572
  * Added the ability to specify a music on hold class to play instead of ringing
1560
     for the SLATrunk application.
1573
     for the SLATrunk application.
1561
  * The Read application no longer exits the dialplan on error.  Instead, it sets
1574
  * The Read application no longer exits the dialplan on error.  Instead, it sets
1562
     READSTATUS to ERROR, which you can catch and handle separately.
1575
     READSTATUS to ERROR, which you can catch and handle separately.
1563
  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1576
  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1564
     of asking for verification of each name, one at a time.
1577
     of asking for verification of each name, one at a time.
1565
  * Privacy() no longer uses privacy.conf, as all options are specifyable as
1578
  * Privacy() no longer uses privacy.conf, as all options are specifyable as
1566
     direct options to the app.
1579
     direct options to the app.
1567
  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1580
  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1568
     for more details
1581
     for more details
1569
  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1582
  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1570
  * The ChannelRedirect application no longer exits the dialplan if the given channel
1583
  * The ChannelRedirect application no longer exits the dialplan if the given channel
1571
     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1584
     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1572
     or NOCHANNEL if the given channel was not found.
1585
     or NOCHANNEL if the given channel was not found.
1573
  * The silencethreshold setting that was previously configurable in multiple
1586
  * The silencethreshold setting that was previously configurable in multiple
1574
     applications is now settable globally via dsp.conf.
1587
     applications is now settable globally via dsp.conf.
1575

    
   
1588

   
1576
Music On Hold Changes
1589
Music On Hold Changes
1577
---------------------
1590
---------------------
1578
  * A new option, "digit", has been added for music on hold classes in 
1591
  * A new option, "digit", has been added for music on hold classes in 
1579
     musiconhold.conf.  If this is set for a music on hold class, a caller
1592
     musiconhold.conf.  If this is set for a music on hold class, a caller
1580
     listening to music on hold can press this digit to switch to listening
1593
     listening to music on hold can press this digit to switch to listening
1581
     to this music on hold class.
1594
     to this music on hold class.
1582
  * Support for realtime music on hold has been added.
1595
  * Support for realtime music on hold has been added.
1583
  * In conjunction with the realtime music on hold, a general section has
1596
  * In conjunction with the realtime music on hold, a general section has
1584
     been added to musiconhold.conf, its sole variable is cachertclasses. If this
1597
     been added to musiconhold.conf, its sole variable is cachertclasses. If this
1585
     is set, then music on hold classes found in realtime will be cached in memory.
1598
     is set, then music on hold classes found in realtime will be cached in memory.
1586

    
   
1599

   
1587
AEL Changes
1600
AEL Changes
1588
-----------
1601
-----------
1589
  * AEL upgraded to use the Gosub with Arguments instead
1602
  * AEL upgraded to use the Gosub with Arguments instead
1590
     of Macro application, to hopefully reduce the problems
1603
     of Macro application, to hopefully reduce the problems
1591
     seen with the artificially low stack ceiling that 
1604
     seen with the artificially low stack ceiling that 
1592
     Macro bumps into. Macros can only call other Macros
1605
     Macro bumps into. Macros can only call other Macros
1593
     to a depth of 7. Tests run using gosub, show depths
1606
     to a depth of 7. Tests run using gosub, show depths
1594
     limited only by virtual memory. A small test demonstrated
1607
     limited only by virtual memory. A small test demonstrated
1595
     recursive call depths of 100,000 without problems.
1608
     recursive call depths of 100,000 without problems.
1596
     -- in addition to this, all apps that allowed a macro
1609
     -- in addition to this, all apps that allowed a macro
1597
     to be called, as in Dial, queues, etc, are now allowing
1610
     to be called, as in Dial, queues, etc, are now allowing
1598
     a gosub call in similar fashion.
1611
     a gosub call in similar fashion.
1599
  * AEL now generates LOCAL(argname) declarations when it
1612
  * AEL now generates LOCAL(argname) declarations when it
1600
     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1613
     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1601
     etc. That makes the arguments local in scope. The user
1614
     etc. That makes the arguments local in scope. The user
1602
     can define their own local variables in macros, now,
1615
     can define their own local variables in macros, now,
1603
     by saying "local myvar=someval;"  or using Set() in this
1616
     by saying "local myvar=someval;"  or using Set() in this
1604
     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
1617
     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
1605
     an AEL keyword).
1618
     an AEL keyword).
1606
  * utils/conf2ael introduced. Will convert an extensions.conf
1619
  * utils/conf2ael introduced. Will convert an extensions.conf
1607
     file into extensions.ael. Very crude and unfinished, but 
1620
     file into extensions.ael. Very crude and unfinished, but 
1608
     will be improved as time goes by. Should be useful for a
1621
     will be improved as time goes by. Should be useful for a
1609
     first pass at conversion.
1622
     first pass at conversion.
1610
  * aelparse will now read extensions.conf to see if a referenced
1623
  * aelparse will now read extensions.conf to see if a referenced
1611
     macro or context is there before issueing a warning.
1624
     macro or context is there before issueing a warning.
1612
  * AEL parser sets a local channel variable ~~EXTEN~~, to 
1625
  * AEL parser sets a local channel variable ~~EXTEN~~, to 
1613
    preserve the value of ${EXTEN} thru switch statements.
1626
    preserve the value of ${EXTEN} thru switch statements.
1614
  * New operator in $[...] expressions: the ~~ operator serves
1627
  * New operator in $[...] expressions: the ~~ operator serves
1615
    as a concatenation operator. AT THE MOMENT, it is really only
1628
    as a concatenation operator. AT THE MOMENT, it is really only
1616
    necessary and useful in AEL, especially in if() expressions.
1629
    necessary and useful in AEL, especially in if() expressions.
1617
    Operation: ${a} ~~ ${b|  with force both a and b to strings, strip 
1630
    Operation: ${a} ~~ ${b|  with force both a and b to strings, strip 
1618
    any enclosing double-quotes, and evaluate to the value of a
1631
    any enclosing double-quotes, and evaluate to the value of a
1619
    concatenated with the value of b.  For example if a is set to
1632
    concatenated with the value of b.  For example if a is set to
1620
    "xyz"  and b has the value "abc", then ${a} ~~ ${b| would
1633
    "xyz"  and b has the value "abc", then ${a} ~~ ${b| would
1621
    evaluate to xyzabc .
1634
    evaluate to xyzabc .
1622

    
   
1635

   
1623

    
   
1636

   
1624
Call Features (res_features) Changes
1637
Call Features (res_features) Changes
1625
------------------------------------
1638
------------------------------------
1626
  * Added the parkedcalltransfers option to features.conf
1639
  * Added the parkedcalltransfers option to features.conf
1627
  * Added parkedcallparking option to control one touch parking w/ parking
1640
  * Added parkedcallparking option to control one touch parking w/ parking
1628
    pickup
1641
    pickup
1629
  * Added parkedcallhangup option to control disconnect feature w/ parking
1642
  * Added parkedcallhangup option to control disconnect feature w/ parking
1630
    pickup
1643
    pickup
1631
  * Added parkedcallrecording option to control one-touch record w/ parking
1644
  * Added parkedcallrecording option to control one-touch record w/ parking
1632
    pickup
1645
    pickup
1633
  * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
1646
  * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
1634
    parkedcalltransfers option support for multiple parking lots.
1647
    parkedcalltransfers option support for multiple parking lots.
1635
  * Added BRIDGE_FEATURES variable to set available features for a channel
1648
  * Added BRIDGE_FEATURES variable to set available features for a channel
1636
  * The built-in method for doing attended transfers has been updated to
1649
  * The built-in method for doing attended transfers has been updated to
1637
     include some new options that allow you to have the transferee sent
1650
     include some new options that allow you to have the transferee sent
1638
     back to the person that did the transfer if the transfer is not successful.
1651
     back to the person that did the transfer if the transfer is not successful.
1639
     See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1652
     See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1640
     in features.conf.sample.
1653
     in features.conf.sample.
1641
  * Added support for configuring named groups of custom call features in
1654
  * Added support for configuring named groups of custom call features in
1642
     features.conf.  This means that features can be written a single time, and
1655
     features.conf.  This means that features can be written a single time, and
1643
     then mapped into groups of features for different key mappings or easier
1656
     then mapped into groups of features for different key mappings or easier
1644
     access control.
1657
     access control.
1645
  * Updated the ParkedCall application to allow you to not specify a parking
1658
  * Updated the ParkedCall application to allow you to not specify a parking
1646
     extension.  If you don't specify a parking space to pick up, it will grab
1659
     extension.  If you don't specify a parking space to pick up, it will grab
1647
     the first one available.
1660
     the first one available.
1648
  * Added cli command 'features reload' to reload call features from features.conf
1661
  * Added cli command 'features reload' to reload call features from features.conf
1649
  * Moved into core asterisk binary.
1662
  * Moved into core asterisk binary.
1650
  * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1663
  * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1651
  * Added the ability for custom parking lots to be configured with their own
1664
  * Added the ability for custom parking lots to be configured with their own
1652
    parking extension with the parkext option.
1665
    parking extension with the parkext option.
1653

    
   
1666

   
1654
Language Support Changes
1667
Language Support Changes
1655
------------------------
1668
------------------------
1656
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1669
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1657
  * Added support for the Hungarian language for saying numbers, dates, and times.
1670
  * Added support for the Hungarian language for saying numbers, dates, and times.
1658

    
   
1671

   
1659
AGI Changes
1672
AGI Changes
1660
-----------
1673
-----------
1661
  * Added SPEECH commands for speech recognition. A complete listing can be found
1674
  * Added SPEECH commands for speech recognition. A complete listing can be found
1662
    using agi show.
1675
    using agi show.
1663
  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1676
  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1664
    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
1677
    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
1665
    does not behave as expected; the native command needs to be used, instead.
1678
    does not behave as expected; the native command needs to be used, instead.
1666
  * Added the ability to perform SRV lookups on fast AGI calls. To use this
1679
  * Added the ability to perform SRV lookups on fast AGI calls. To use this
1667
    feature, simply use hagi: instead of agi: as the protocol portion
1680
    feature, simply use hagi: instead of agi: as the protocol portion
1668
    of the URI parameter to the AGI function call in your dial plan. Also note
1681
    of the URI parameter to the AGI function call in your dial plan. Also note
1669
    that specifying a port number in the AGI URI will disable SRV lookups,
1682
    that specifying a port number in the AGI URI will disable SRV lookups,
1670
    even if you use the hagi: protocol.
1683
    even if you use the hagi: protocol.
1671
  * No longer support MSG_OOB flag on HANGUP.
1684
  * No longer support MSG_OOB flag on HANGUP.
1672

    
   
1685

   
1673
Logger changes
1686
Logger changes
1674
--------------
1687
--------------
1675
  * Added rotatestrategy option to logger.conf, along with two new options:
1688
  * Added rotatestrategy option to logger.conf, along with two new options:
1676
     "timestamp" which will use the time to name the logger files instead of
1689
     "timestamp" which will use the time to name the logger files instead of
1677
     sequence number; and "rotate", which rotates the names of the log files,
1690
     sequence number; and "rotate", which rotates the names of the log files,
1678
     similar to the way syslog rotates files.
1691
     similar to the way syslog rotates files.
1679
  * Added exec_after_rotate option to logger.conf, which allows a system
1692
  * Added exec_after_rotate option to logger.conf, which allows a system
1680
     command to be run after rotation.  This is primarily useful with
1693
     command to be run after rotation.  This is primarily useful with
1681
     rotatestrategy=rotate, to allow a limit on the number of log files kept
1694
     rotatestrategy=rotate, to allow a limit on the number of log files kept
1682
     and to ensure that the oldest log file gets deleted.
1695
     and to ensure that the oldest log file gets deleted.
1683
  * Added realtime support for the queue log
1696
  * Added realtime support for the queue log
1684

    
   
1697

   
1685
Call Detail Records 
1698
Call Detail Records 
1686
-------------------
1699
-------------------
1687
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
1700
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
1688
    to add fields to the manager event from the CDR variables.
1701
    to add fields to the manager event from the CDR variables.
1689
  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1702
  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1690
     backend database CDR table.  Specifically, additional, non-standard
1703
     backend database CDR table.  Specifically, additional, non-standard
1691
     columns are supported, merely by setting the corresponding CDR variable in
1704
     columns are supported, merely by setting the corresponding CDR variable in
1692
     your dialplan.  In addition, you may alias any column to another name (for
1705
     your dialplan.  In addition, you may alias any column to another name (for
1693
     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1706
     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1694
     simply "alias src => ANI" in the configuration file).  Records may be
1707
     simply "alias src => ANI" in the configuration file).  Records may be
1695
     posted to more than one backend, simply by specifying multiple categories
1708
     posted to more than one backend, simply by specifying multiple categories
1696
     in the configuration file.  And finally, you may filter which CDRs get
1709
     in the configuration file.  And finally, you may filter which CDRs get
1697
     posted to each backend, by specifying a filter (which the record must
1710
     posted to each backend, by specifying a filter (which the record must
1698
     match) for the particular category.  Filters are additive (meaning all
1711
     match) for the particular category.  Filters are additive (meaning all
1699
     rules must match to post that CDR).
1712
     rules must match to post that CDR).
1700
  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1713
  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1701
     module.  Specifically, you may add additional columns into the table and
1714
     module.  Specifically, you may add additional columns into the table and
1702
     they will be set, if you set the corresponding CDR variable name.  Also,
1715
     they will be set, if you set the corresponding CDR variable name.  Also,
1703
     if you omit columns in your database table, they will be silently skipped
1716
     if you omit columns in your database table, they will be silently skipped
1704
     (but a record will still be inserted, based on what columns remain).  Note
1717
     (but a record will still be inserted, based on what columns remain).  Note
1705
     that the other two features from cdr_adaptive_odbc (alias and filter) are
1718
     that the other two features from cdr_adaptive_odbc (alias and filter) are
1706
     not currently supported.
1719
     not currently supported.
1707
  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1720
  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1708
     has been disabled using the NoCDR application.
1721
     has been disabled using the NoCDR application.
1709

    
   
1722

   
1710
Miscellaneous New Modules
1723
Miscellaneous New Modules
1711
-------------------------
1724
-------------------------
1712
  * Added a new CDR module, cdr_sqlite3_custom.
1725
  * Added a new CDR module, cdr_sqlite3_custom.
1713
  * Added a new realtime configuration module, res_config_sqlite
1726
  * Added a new realtime configuration module, res_config_sqlite
1714
  * Added a new codec translation module, codec_resample, which re-samples
1727
  * Added a new codec translation module, codec_resample, which re-samples
1715
     signed linear audio between 8 kHz and 16 kHz to help support wideband
1728
     signed linear audio between 8 kHz and 16 kHz to help support wideband
1716
     codecs.
1729
     codecs.
1717
  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1730
  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1718
     based on configuration templates that use Asterisk dialplan function and
1731
     based on configuration templates that use Asterisk dialplan function and
1719
     variable substitution.  It should be possible to create phone profiles and
1732
     variable substitution.  It should be possible to create phone profiles and
1720
     templates that work for the majority of phones provisioned over http. It
1733
     templates that work for the majority of phones provisioned over http. It
1721
     is currently only intended to provision a single user account per phone.
1734
     is currently only intended to provision a single user account per phone.
1722
     An example profile and set of templates for Polycom phones is provided.
1735
     An example profile and set of templates for Polycom phones is provided.
1723
     NOTE: Polycom firmware is not included, but should be placed in
1736
     NOTE: Polycom firmware is not included, but should be placed in
1724
     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1737
     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1725
  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1738
  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1726
     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
1739
     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
1727
     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
1740
     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
1728
     interfaces create an input and output JACK port.  The application makes
1741
     interfaces create an input and output JACK port.  The application makes
1729
     these ports the endpoint of the call.  The audio coming from the channel
1742
     these ports the endpoint of the call.  The audio coming from the channel
1730
     goes out the output port and whatever comes back in on the input port is
1743
     goes out the output port and whatever comes back in on the input port is
1731
     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
1744
     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
1732
     audiohook on the channel.  This lets you run the audio coming from a
1745
     audiohook on the channel.  This lets you run the audio coming from a
1733
     channel through JACK, and whatever comes back in is what gets forwarded
1746
     channel through JACK, and whatever comes back in is what gets forwarded
1734
     on as the channel's audio.  This is very useful for building custom
1747
     on as the channel's audio.  This is very useful for building custom
1735
     vocoders or doing recording or analysis of the channel's audio in another
1748
     vocoders or doing recording or analysis of the channel's audio in another
1736
     application.
1749
     application.
1737
  * Added a new module, res_config_curl, which permits using a HTTP POST url
1750
  * Added a new module, res_config_curl, which permits using a HTTP POST url
1738
     to retrieve, create, update, and delete realtime information from a remote
1751
     to retrieve, create, update, and delete realtime information from a remote
1739
     web server.  Note that this module requires func_curl.so to be loaded for
1752
     web server.  Note that this module requires func_curl.so to be loaded for
1740
     backend functionality.
1753
     backend functionality.
1741
  * Added a new module, res_config_ldap, which permits the use of an LDAP
1754
  * Added a new module, res_config_ldap, which permits the use of an LDAP
1742
     server for realtime data access.
1755
     server for realtime data access.
1743
  * Added support for writing and running your dialplan in lua using the pbx_lua
1756
  * Added support for writing and running your dialplan in lua using the pbx_lua
1744
     module.  See configs/extensions.lua.sample for examples of how to do this.
1757
     module.  See configs/extensions.lua.sample for examples of how to do this.
1745

    
   
1758

   
1746
Miscellaneous 
1759
Miscellaneous 
1747
-------------
1760
-------------
1748
  * Ability to use libcap to set high ToS bits when non-root
1761
  * Ability to use libcap to set high ToS bits when non-root
1749
     on Linux. If configure is unable to find libcap then you
1762
     on Linux. If configure is unable to find libcap then you
1750
     can use --with-cap to specify the path.
1763
     can use --with-cap to specify the path.
1751
  * Added maxfiles option to options section of asterisk.conf which allows you to specify
1764
  * Added maxfiles option to options section of asterisk.conf which allows you to specify
1752
     what Asterisk should set as the maximum number of open files when it loads.
1765
     what Asterisk should set as the maximum number of open files when it loads.
1753
  * Added the jittertargetextra configuration option.
1766
  * Added the jittertargetextra configuration option.
1754
  * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
1767
  * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
1755
     configuration files for the IP channel drivers.  The new option is "cos".
1768
     configuration files for the IP channel drivers.  The new option is "cos".
1756
     This information is also documented in doc/qos.tex, or the IP Quality of Service
1769
     This information is also documented in doc/qos.tex, or the IP Quality of Service
1757
     section of asterisk.pdf.
1770
     section of asterisk.pdf.
1758
  * When originating a call using AMI or pbx_spool that fails the reason for failure
1771
  * When originating a call using AMI or pbx_spool that fails the reason for failure
1759
     will now be available in the failed extension using the REASON dialplan variable.
1772
     will now be available in the failed extension using the REASON dialplan variable.
1760
  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1773
  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1761
     It allows you to configure a prefix for auto-monitor recordings.
1774
     It allows you to configure a prefix for auto-monitor recordings.
1762
  * A new extension pattern matching algorithm, based on a trie, is introduced
1775
  * A new extension pattern matching algorithm, based on a trie, is introduced
1763
     here, that could noticeably speed up mid-sized to large dialplans.
1776
     here, that could noticeably speed up mid-sized to large dialplans.
1764
     It is NOT used by default, as duplicating the behaviour of the old pattern
1777
     It is NOT used by default, as duplicating the behaviour of the old pattern
1765
     matcher is still under development. A config file option, in extensions.conf,
1778
     matcher is still under development. A config file option, in extensions.conf,
1766
     in the [general] section, called "extenpatternmatchingnew", is by default
1779
     in the [general] section, called "extenpatternmatchingnew", is by default
1767
     set to false; setting that to true will force the use of the new algorithm.
1780
     set to false; setting that to true will force the use of the new algorithm.
1768
     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1781
     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1769
     be used to switch the algorithms at run time.
1782
     be used to switch the algorithms at run time.
1770
  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1783
  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1771
     specifying which socket to use to connect to the running Asterisk daemon
1784
     specifying which socket to use to connect to the running Asterisk daemon
1772
     (-s)
1785
     (-s)
1773
  * Performance enhancements to the sched facility, which is used in
1786
  * Performance enhancements to the sched facility, which is used in
1774
    the channel drivers, etc. Added hashtabs and doubly-linked lists
1787
    the channel drivers, etc. Added hashtabs and doubly-linked lists
1775
    to speed up deletion; start at the beginning or end of list to
1788
    to speed up deletion; start at the beginning or end of list to
1776
    speed up insertion.
1789
    speed up insertion.
1777
  * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1790
  * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1778
    dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1791
    dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1779
    Added regression tests to the tests/ dir, also.
1792
    Added regression tests to the tests/ dir, also.
1780
  * Added a refcount trace feature to astobj2 for those trying to balance
1793
  * Added a refcount trace feature to astobj2 for those trying to balance
1781
    object creation, deletion; work, play; space and time. See the
1794
    object creation, deletion; work, play; space and time. See the
1782
    notes in astobj2.h. Also, see utils/refcounter as well, as a
1795
    notes in astobj2.h. Also, see utils/refcounter as well, as a
1783
    quick way to find unbalanced refcounts in what could be a sea
1796
    quick way to find unbalanced refcounts in what could be a sea
1784
    of objects that were balanced.
1797
    of objects that were balanced.
1785
  * Added logging to 'make update' command.  See update.log
1798
  * Added logging to 'make update' command.  See update.log
1786
  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1799
  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1787
     do not come from the remote party.
1800
     do not come from the remote party.
1788
  * Added the 'n' option to the SpeechBackground application to tell it to not
1801
  * Added the 'n' option to the SpeechBackground application to tell it to not
1789
     answer the channel if it has not already been answered.
1802
     answer the channel if it has not already been answered.
1790
  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1803
  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1791
     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
1804
     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
1792
     dialplan debugging.
1805
     dialplan debugging.
1793
  * iLBC source code no longer included (see UPGRADE.txt for details)
1806
  * iLBC source code no longer included (see UPGRADE.txt for details)
1794
  * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if 
1807
  * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if 
1795
     deadlock is detected, a backtrace of the stack which led to the lock calls
1808
     deadlock is detected, a backtrace of the stack which led to the lock calls
1796
     will be output to the CLI.
1809
     will be output to the CLI.
1797
  * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1810
  * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1798
     the "core show locks" CLI command will give lock information output as well
1811
     the "core show locks" CLI command will give lock information output as well
1799
     as a backtrace of the stack which led to the lock calls.
1812
     as a backtrace of the stack which led to the lock calls.
1800
  * users.conf now sports an optional alternateexts property, which permits
1813
  * users.conf now sports an optional alternateexts property, which permits
1801
    allocation of additional extensions which will reach the specified user.
1814
    allocation of additional extensions which will reach the specified user.
1802
  * A new option for the configure script, --enable-internal-poll, has been added
1815
  * A new option for the configure script, --enable-internal-poll, has been added
1803
    for use with systems which may have a buggy implementation of the poll system
1816
    for use with systems which may have a buggy implementation of the poll system
1804
    call. If you notice odd behavior such as the CLI being unresponsive on remote
1817
    call. If you notice odd behavior such as the CLI being unresponsive on remote
1805
    consoles, you may want to try using this option. This option is enabled by default
1818
    consoles, you may want to try using this option. This option is enabled by default
1806
    on Darwin systems since it is known that the Darwin poll() implementation has
1819
    on Darwin systems since it is known that the Darwin poll() implementation has
1807
    odd issues.
1820
    odd issues.
1808

    
   
1821

   
1809
Timer Changes
1822
Timer Changes
1810
--------------------
1823
--------------------
1811
* In addition to timing from DAHDI, there is a new timing module called
1824
* In addition to timing from DAHDI, there is a new timing module called
1812
  res_timing_timerfd. In order to use this, you must be running Linux with
1825
  res_timing_timerfd. In order to use this, you must be running Linux with
1813
  a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1826
  a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1814
  script will be able to tell if you have the requirements. From menuselect, select
1827
  script will be able to tell if you have the requirements. From menuselect, select
1815
  res_timing_timerfd from the Resource Modules menu.
1828
  res_timing_timerfd from the Resource Modules menu.
/trunk/channels/chan_sip.c
Diff Revision 3 Diff Revision 7
 
/trunk/channels/sip/include/sip.h
Diff Revision 3 Diff Revision 7
 
/trunk/configs/jabber.conf.sample
Diff Revision 3 Diff Revision 7
 
/trunk/configs/sip.conf.sample
Diff Revision 3 Diff Revision 7
 
/trunk/doc/asterisk-messaging.txt
Diff Revision 3 Diff Revision 7
 
/trunk/include/asterisk/_private.h
Diff Revision 3 Diff Revision 7
 
/trunk/include/asterisk/channel.h
Diff Revision 3 Diff Revision 7
 
/trunk/include/asterisk/jabber.h
Diff Revision 3 Diff Revision 7
 
/trunk/include/asterisk/message.h
Diff Revision 3 Diff Revision 7
 
/trunk/main/asterisk.c
Diff Revision 3 Diff Revision 7
 
/trunk/main/channel.c
Diff Revision 3 Diff Revision 7
 
/trunk/main/message.c
Diff Revision 3 Diff Revision 7
 
/trunk/res/res_jabber.c
Diff Revision 3 Diff Revision 7
 
  1. /trunk/CHANGES: Loading...
  2. /trunk/channels/chan_sip.c: Loading...
  3. /trunk/channels/sip/include/sip.h: Loading...
  4. /trunk/configs/jabber.conf.sample: Loading...
  5. /trunk/configs/sip.conf.sample: Loading...
  6. /trunk/doc/asterisk-messaging.txt: Loading...
  7. /trunk/include/asterisk/_private.h: Loading...
  8. /trunk/include/asterisk/channel.h: Loading...
  9. /trunk/include/asterisk/jabber.h: Loading...
  10. /trunk/include/asterisk/message.h: Loading...
  11. /trunk/main/asterisk.c: Loading...
  12. /trunk/main/channel.c: Loading...
  13. /trunk/main/message.c: Loading...
  14. /trunk/res/res_jabber.c: Loading...

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