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 311293 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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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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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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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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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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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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 * 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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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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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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------------------------------------------------------------------------------
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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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-----------
98
-----------
86
 * 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
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 * 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
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 * 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
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   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
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   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
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   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
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 * 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.
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 * Added support for G.719 media streams.
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 * Added support for 16khz signed linear media streams.
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 * 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,
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   RTP has been outfitted with the same abilities.
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   RTP has been outfitted with the same abilities.
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 * 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.
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 * Addition of the 'subscribe_network_change' option for turning on and off
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 * Addition of the 'subscribe_network_change' option for turning on and off
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   res_stun_monitor module support in chan_sip.
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   res_stun_monitor module support in chan_sip.
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 * Addition of the 'auth_options_requests' option for turning on and off
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 * Addition of the 'auth_options_requests' option for turning on and off
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   authentication for OPTIONS requests in chan_sip.
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   authentication for OPTIONS requests in chan_sip.
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IAX2 Changes
164
IAX2 Changes
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-----------
165
-----------
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 * 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.
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 * Added the ability for chan_iax2 to inform the dialplan whether or not
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 * 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
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   encryption is being used. This interoperates with the SIP SRTP implementation
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   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".
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 * Addition of the 'subscribe_network_change' option for turning on and off
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 * Addition of the 'subscribe_network_change' option for turning on and off
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   res_stun_monitor module support in chan_iax.
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   res_stun_monitor module support in chan_iax.
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MGCP Changes
176
MGCP Changes
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------------
177
------------
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 * Added ability to preset channel variables on indicated lines with the setvar
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 * Added ability to preset channel variables on indicated lines with the setvar
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   configuration option.  Also, clearvars=all resets the list of variables back
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   configuration option.  Also, clearvars=all resets the list of variables back
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   to none.
180
   to none.
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 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
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 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
169
   See configs/res_pktccops.conf for more information.
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   See configs/res_pktccops.conf for more information.
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XMPP Google Talk/Jingle changes
184
XMPP Google Talk/Jingle changes
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-------------------------------
185
-------------------------------
173
  * Added the externip option to gtalk.conf.
186
  * Added the externip option to gtalk.conf.
174
  * Added the stunaddr option to gtalk.conf which allows for the automatic
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  * Added the stunaddr option to gtalk.conf which allows for the automatic
175
    retrieval of the external ip from a stun server.
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    retrieval of the external ip from a stun server.
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Applications
190
Applications
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------------
191
------------
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 * Added 'p' option to PickupChan() to allow for picking up channel by the first
192
 * Added 'p' option to PickupChan() to allow for picking up channel by the first
180
   match to a partial channel name.
193
   match to a partial channel name.
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 * Added .m3u support for Mp3Player application.
194
 * Added .m3u support for Mp3Player application.
182
 * Added progress option to the app_dial D() option.  When progress DTMF is
195
 * Added progress option to the app_dial D() option.  When progress DTMF is
183
   present, those values are sent immediately upon receiving a PROGRESS message
196
   present, those values are sent immediately upon receiving a PROGRESS message
184
   regardless if the call has been answered or not.
197
   regardless if the call has been answered or not.
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 * Added functionality to the app_dial F() option to continue with execution
198
 * Added functionality to the app_dial F() option to continue with execution
186
   at the current location when no parameters are provided.
199
   at the current location when no parameters are provided.
187
 * Added the 'a' option to app_dial to answer the calling channel before any
200
 * Added the 'a' option to app_dial to answer the calling channel before any
188
   announcements or macros are executed.
201
   announcements or macros are executed.
189
 * Modified app_dial to set answertime when the called channel answers even if
202
 * Modified app_dial to set answertime when the called channel answers even if
190
   the called channel hangs up during playback of an announcement.
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   the called channel hangs up during playback of an announcement.
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 * Modified app_dial 'r' option to support an additional parameter to play an
204
 * Modified app_dial 'r' option to support an additional parameter to play an
192
   indication tone from indications.conf
205
   indication tone from indications.conf
193
 * Added c() option to app_chanspy. This option allows custom DTMF to be set
206
 * Added c() option to app_chanspy. This option allows custom DTMF to be set
194
   to cycle through the next available channel.  By default this is still '*'.
207
   to cycle through the next available channel.  By default this is still '*'.
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 * Added x() option to app_chanspy.  This option allows DTMF to be set to
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 * Added x() option to app_chanspy.  This option allows DTMF to be set to
196
   exit the application.
209
   exit the application.
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 * The Voicemail application has been improved to automatically ignore messages
210
 * The Voicemail application has been improved to automatically ignore messages
198
   that only contain silence.
211
   that only contain silence.
199
 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
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 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
200
   associated mailbox(es) to be greetings-only.
213
   associated mailbox(es) to be greetings-only.
201
 * The ChanSpy application now has the 'S' option, which makes the application
214
 * The ChanSpy application now has the 'S' option, which makes the application
202
   automatically exit once it hits a point where no more channels are available
215
   automatically exit once it hits a point where no more channels are available
203
   to spy on.
216
   to spy on.
204
 * The ChanSpy application also now has the 'E' option, which spies on a single
217
 * The ChanSpy application also now has the 'E' option, which spies on a single
205
   channel and exits when that channel hangs up.
218
   channel and exits when that channel hangs up.
206
 * The MeetMe application now turns on the DENOISE() function by default, for
219
 * The MeetMe application now turns on the DENOISE() function by default, for
207
   each participant.  In our tests, this has significantly decreased background
220
   each participant.  In our tests, this has significantly decreased background
208
   noise (especially noisy data centers).
221
   noise (especially noisy data centers).
209
 * Voicemail now permits storage of secrets in a separate file, located in the
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 * Voicemail now permits storage of secrets in a separate file, located in the
210
   spool directory of each individual user.  The control for this is located in
223
   spool directory of each individual user.  The control for this is located in
211
   the "passwordlocation" option in voicemail.conf.  Please see the sample
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   the "passwordlocation" option in voicemail.conf.  Please see the sample
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   configuration for more information.
225
   configuration for more information.
213
 * The ChanIsAvail application now exposes the returned cause code using a separate
226
 * The ChanIsAvail application now exposes the returned cause code using a separate
214
   variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
227
   variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
215
 * Added 'd' option to app_followme.  This option disables the "Please hold"
228
 * Added 'd' option to app_followme.  This option disables the "Please hold"
216
   announcement.
229
   announcement.
217
 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
230
 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
218
   received will terminate recording.
231
   received will terminate recording.
219
 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
232
 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
220
   Previously the folder could only be set per context, but has now been extended 
233
   Previously the folder could only be set per context, but has now been extended 
221
   using the imapfolder option.
234
   using the imapfolder option.
222
 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
235
 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
223
 * Voicemail now allows the pager date format to be specified separately from the
236
 * Voicemail now allows the pager date format to be specified separately from the
224
   email date format.
237
   email date format.
225
 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
238
 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
226
   to allow joining, leaving, and sending text to group chats.
239
   to allow joining, leaving, and sending text to group chats.
227
 * MeetMe has a new option 'G' to play an announcement before joining a conference.
240
 * MeetMe has a new option 'G' to play an announcement before joining a conference.
228
 * Page has a new option 'A(x)' which will playback an announcement simultaneously
241
 * Page has a new option 'A(x)' which will playback an announcement simultaneously
229
   to all paged phones (and optionally excluding the caller's one using the new
242
   to all paged phones (and optionally excluding the caller's one using the new
230
   option 'n') before the call is bridged.
243
   option 'n') before the call is bridged.
231
 * The 'f' option to Dial has been augmented to take an optional argument. If no
244
 * The 'f' option to Dial has been augmented to take an optional argument. If no
232
   argument is provided, the 'f' option works as it always has. If an argument is
245
   argument is provided, the 'f' option works as it always has. If an argument is
233
   provided, then the connected party information of all outgoing channels created
246
   provided, then the connected party information of all outgoing channels created
234
   during the Dial will be set to the argument passed to the 'f' option.
247
   during the Dial will be set to the argument passed to the 'f' option.
235
 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
248
 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
236
   Gosub on the peer.
249
   Gosub on the peer.
237
 * The OSP lookup application adds in/outbound network ID, optional security,
250
 * The OSP lookup application adds in/outbound network ID, optional security,
238
   number portability, QoS reporting, destination IP port, custom info and service
251
   number portability, QoS reporting, destination IP port, custom info and service
239
   type features.
252
   type features.
240
 * Added new application VMSayName that will play the recorded name of the voicemail
253
 * Added new application VMSayName that will play the recorded name of the voicemail
241
   user if it exists, otherwise will play the mailbox number.
254
   user if it exists, otherwise will play the mailbox number.
242
 * Added custom device states to ConfBridge bridges.  Use 'confbridge:<name>' to
255
 * Added custom device states to ConfBridge bridges.  Use 'confbridge:<name>' to
243
   retrieve state for a particular bridge, where <name> is the conference name
256
   retrieve state for a particular bridge, where <name> is the conference name
244
 * app_directory now allows exiting at any time using the operator or pound key.
257
 * app_directory now allows exiting at any time using the operator or pound key.
245
 * Voicemail now supports setting a locale per-mailbox.
258
 * Voicemail now supports setting a locale per-mailbox.
246
 * Two new applications are provided for declining counting phrases in multiple
259
 * Two new applications are provided for declining counting phrases in multiple
247
   languages.  See the application notes for SayCountedNoun and SayCountedAdj for
260
   languages.  See the application notes for SayCountedNoun and SayCountedAdj for
248
   more information.
261
   more information.
249
 * Voicemail now runs the externnotify script when pollmailboxes is activated and
262
 * Voicemail now runs the externnotify script when pollmailboxes is activated and
250
   notices a change.
263
   notices a change.
251
 * Voicemail now includes rdnis within msgXXXX.txt file.
264
 * Voicemail now includes rdnis within msgXXXX.txt file.
252
 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
265
 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
253
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
266
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
254
   a MeetMe conference
267
   a MeetMe conference
255

    
   
268

   
256
Dialplan Functions
269
Dialplan Functions
257
------------------
270
------------------
258
 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
271
 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
259
   over SRV records associated with a specific service. From the CLI, type
272
   over SRV records associated with a specific service. From the CLI, type
260
   'core show function SRVQUERY' and 'core show function SRVRESULT' for more
273
   'core show function SRVQUERY' and 'core show function SRVRESULT' for more
261
   details on how these may be used.
274
   details on how these may be used.
262
 * PITCH_SHIFT dialplan function added. This function can be used to modify the
275
 * PITCH_SHIFT dialplan function added. This function can be used to modify the
263
   pitch of a channel's tx and rx audio streams.
276
   pitch of a channel's tx and rx audio streams.
264
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
277
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
265
   setting various connected line and redirecting party information.
278
   setting various connected line and redirecting party information.
266
 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
279
 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
267
   support ISDN subaddressing.
280
   support ISDN subaddressing.
268
 * The CHANNEL() function now supports the "name" and "checkhangup" options.
281
 * The CHANNEL() function now supports the "name" and "checkhangup" options.
269
 * For DAHDI channels, the CHANNEL() dialplan function now allows
282
 * For DAHDI channels, the CHANNEL() dialplan function now allows
270
   the dialplan to request changes in the configuration of the active
283
   the dialplan to request changes in the configuration of the active
271
   echo canceller on the channel (if any), for the current call only.
284
   echo canceller on the channel (if any), for the current call only.
272
   The syntax is:
285
   The syntax is:
273

    
   
286

   
274
   exten => s,n,Set(CHANNEL(echocan_mode)=off)
287
   exten => s,n,Set(CHANNEL(echocan_mode)=off)
275

    
   
288

   
276
   The possible values are:
289
   The possible values are:
277

    
   
290

   
278
     on - normal mode (the echo canceller is actually reinitialized)
291
     on - normal mode (the echo canceller is actually reinitialized)
279
     off - disabled
292
     off - disabled
280
     fax - FAX/data mode (NLP disabled if possible, otherwise completely
293
     fax - FAX/data mode (NLP disabled if possible, otherwise completely
281
           disabled)
294
           disabled)
282
     voice - voice mode (returns from FAX mode, reverting the changes that
295
     voice - voice mode (returns from FAX mode, reverting the changes that
283
             were made when FAX mode was requested)
296
             were made when FAX mode was requested)
284
 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
297
 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
285
   and setting variables on the channel which created the current channel.
298
   and setting variables on the channel which created the current channel.
286
   Administrators should take care to avoid naming conflicts, when multiple
299
   Administrators should take care to avoid naming conflicts, when multiple
287
   channels are dialled at once, especially when used with the Local channel
300
   channels are dialled at once, especially when used with the Local channel
288
   construct (which all could set variables on the master channel).  Usage
301
   construct (which all could set variables on the master channel).  Usage
289
   of the HASH() dialplan function, with the key set to the name of the slave
302
   of the HASH() dialplan function, with the key set to the name of the slave
290
   channel, is one approach that will avoid conflicts.
303
   channel, is one approach that will avoid conflicts.
291
 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
304
 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
292
   audio in a channel.
305
   audio in a channel.
293
 * func_odbc now allows multiple row results to be retrieved without using
306
 * func_odbc now allows multiple row results to be retrieved without using
294
   mode=multirow.  If rowlimit is set, then additional rows may be retrieved
307
   mode=multirow.  If rowlimit is set, then additional rows may be retrieved
295
   from the same query by using the name of the function which retrieved the
308
   from the same query by using the name of the function which retrieved the
296
   first row as an argument to ODBC_FETCH().
309
   first row as an argument to ODBC_FETCH().
297
 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
310
 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
298
   dialplan. This function returns the content of the received message.
311
   dialplan. This function returns the content of the received message.
299
 * Added REPLACE, which searches a given variable name for a set of characters,
312
 * Added REPLACE, which searches a given variable name for a set of characters,
300
   then either replaces them with a single character or deletes them.
313
   then either replaces them with a single character or deletes them.
301
 * Added PASSTHRU, which literally passes the same argument back as its return
314
 * Added PASSTHRU, which literally passes the same argument back as its return
302
   value.  The intent is to be able to use a literal string argument to
315
   value.  The intent is to be able to use a literal string argument to
303
   functions that currently require a variable name as an argument.
316
   functions that currently require a variable name as an argument.
304
 * HASH-associated variables now can be inherited across channel creation, by
317
 * HASH-associated variables now can be inherited across channel creation, by
305
   prefixing the name of the hash at assignment with the appropriate number of
318
   prefixing the name of the hash at assignment with the appropriate number of
306
   underscores, just like variables.
319
   underscores, just like variables.
307
 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
320
 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
308
 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
321
 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
309
   whether or not channels that are bridged to the current channel will be
322
   whether or not channels that are bridged to the current channel will be
310
   required to have secure signaling and/or media.
323
   required to have secure signaling and/or media.
311
 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
324
 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
312
   the current channel has secure signaling and/or media.
325
   the current channel has secure signaling and/or media.
313
 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
326
 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
314
   "no_media_path" option.
327
   "no_media_path" option.
315
   Returns "0" if there is a B channel associated with the call.
328
   Returns "0" if there is a B channel associated with the call.
316
   Returns "1" if no B channel is associated with the call.  The call is either
329
   Returns "1" if no B channel is associated with the call.  The call is either
317
   on hold or is a call waiting call.
330
   on hold or is a call waiting call.
318
 * Added option to dialplan function CDR(), the 'f' option
331
 * Added option to dialplan function CDR(), the 'f' option
319
   allows for high resolution times for billsec and duration fields.
332
   allows for high resolution times for billsec and duration fields.
320
 * FILE() now supports line-mode and writing.
333
 * FILE() now supports line-mode and writing.
321
 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
334
 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
322
 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
335
 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
323

    
   
336

   
324
Dialplan Variables
337
Dialplan Variables
325
------------------
338
------------------
326
 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
339
 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
327
 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
340
 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
328
   and is set when a dynamic feature is triggered.
341
   and is set when a dynamic feature is triggered.
329
 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
342
 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
330
   to dynamically create a new parking lot matching the value this varible is
343
   to dynamically create a new parking lot matching the value this varible is
331
   set to.
344
   set to.
332
 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
345
 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
333
   features.conf that should be the base for dynamic parkinglots.
346
   features.conf that should be the base for dynamic parkinglots.
334
 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
347
 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
335
   parkinglot should have.
348
   parkinglot should have.
336
 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
349
 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
337
   should have.
350
   should have.
338

    
   
351

   
339
Queue changes
352
Queue changes
340
-------------
353
-------------
341
 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
354
 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
342
   timeout has expired.
355
   timeout has expired.
343
 * Added 'R' option to app_queue.  This option stops moh and indicates ringing
356
 * Added 'R' option to app_queue.  This option stops moh and indicates ringing
344
   to the caller when an Agent's phone is ringing.  This can be used to indicate
357
   to the caller when an Agent's phone is ringing.  This can be used to indicate
345
   to the caller that their call is about to be picked up, which is nice when
358
   to the caller that their call is about to be picked up, which is nice when
346
   one has been on hold for an extened period of time.
359
   one has been on hold for an extened period of time.
347
 * A new config option, penaltymemberslimit, has been added to queues.conf.
360
 * A new config option, penaltymemberslimit, has been added to queues.conf.
348
   When set this option will disregard penalty settings when a queue has too
361
   When set this option will disregard penalty settings when a queue has too
349
   few members.
362
   few members.
350
 * A new option, 'I' has been added to both app_queue and app_dial.
363
 * A new option, 'I' has been added to both app_queue and app_dial.
351
   By setting this option, Asterisk will not update the caller with
364
   By setting this option, Asterisk will not update the caller with
352
   connected line changes or redirecting party changes when they occur.
365
   connected line changes or redirecting party changes when they occur.
353
 * A 'relative-peroidic-announce' option has been added to queues.conf.  When
366
 * A 'relative-peroidic-announce' option has been added to queues.conf.  When
354
   enabled, this option will cause periodic announce times to be calculated
367
   enabled, this option will cause periodic announce times to be calculated
355
   from the end of announcements rather than from the beginning.
368
   from the end of announcements rather than from the beginning.
356
 * The autopause option in queues.conf can be passed a new value, "all." The
369
 * The autopause option in queues.conf can be passed a new value, "all." The
357
   result is that if a member becomes auto-paused, he will be paused in all
370
   result is that if a member becomes auto-paused, he will be paused in all
358
   queues for which he is a member, not just the queue that failed to reach
371
   queues for which he is a member, not just the queue that failed to reach
359
   the member.
372
   the member.
360
 * Added dialplan function QUEUE_EXISTS to check if a queue exists
373
 * Added dialplan function QUEUE_EXISTS to check if a queue exists
361
 * The queue logger now allows events to optionally propagate to a file,
374
 * The queue logger now allows events to optionally propagate to a file,
362
   even when realtime logging is turned on.  Additionally, realtime logging
375
   even when realtime logging is turned on.  Additionally, realtime logging
363
   supports sending the event arguments to 5 individual fields, although it
376
   supports sending the event arguments to 5 individual fields, although it
364
   will fallback to the previous data definition, if the new table layout is
377
   will fallback to the previous data definition, if the new table layout is
365
   not found.
378
   not found.
366

    
   
379

   
367
mISDN channel driver (chan_misdn) changes
380
mISDN channel driver (chan_misdn) changes
368
----------------------------------------
381
----------------------------------------
369
 * Added display_connected parameter to misdn.conf to put a display string
382
 * Added display_connected parameter to misdn.conf to put a display string
370
   in the CONNECT message containing the connected name and/or number if
383
   in the CONNECT message containing the connected name and/or number if
371
   the presentation setting permits it.
384
   the presentation setting permits it.
372
 * Added display_setup parameter to misdn.conf to put a display string
385
 * Added display_setup parameter to misdn.conf to put a display string
373
   in the SETUP message containing the caller name and/or number if the
386
   in the SETUP message containing the caller name and/or number if the
374
   presentation setting permits it.
387
   presentation setting permits it.
375
 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
388
 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
376
   indicate the dialplan settings are to be obtained from the asterisk
389
   indicate the dialplan settings are to be obtained from the asterisk
377
   channel.
390
   channel.
378
 * Made misdn.conf parameter callerid accept the "name" <number> format
391
 * Made misdn.conf parameter callerid accept the "name" <number> format
379
   used by the rest of the system.
392
   used by the rest of the system.
380
 * Made use the nationalprefix and internationalprefix misdn.conf
393
 * Made use the nationalprefix and internationalprefix misdn.conf
381
   parameters to prefix any received number from the ISDN link if that
394
   parameters to prefix any received number from the ISDN link if that
382
   number has the corresponding Type-Of-Number.  NOTE:  This includes
395
   number has the corresponding Type-Of-Number.  NOTE:  This includes
383
   comparing the incoming call's dialed number against the MSN list.
396
   comparing the incoming call's dialed number against the MSN list.
384
 * Added the following new parameters: unknownprefix, netspecificprefix,
397
 * Added the following new parameters: unknownprefix, netspecificprefix,
385
   subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
398
   subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
386
   received number from the ISDN link if that number has the corresponding
399
   received number from the ISDN link if that number has the corresponding
387
   Type-Of-Number.
400
   Type-Of-Number.
388
 * Added new dialplan application misdn_command which permits controlling
401
 * Added new dialplan application misdn_command which permits controlling
389
   the CCBS/CCNR functionality.
402
   the CCBS/CCNR functionality.
390
 * Added new dialplan function mISDN_CC which permits retrieval of various
403
 * Added new dialplan function mISDN_CC which permits retrieval of various
391
   values from an active call completion record.
404
   values from an active call completion record.
392
 * For PTP, you should manually send the COLR of the redirected-to party
405
 * For PTP, you should manually send the COLR of the redirected-to party
393
   for an incomming redirected call if the incoming call could experience
406
   for an incomming redirected call if the incoming call could experience
394
   further redirects.  Just set the REDIRECTING(to-num,i) = ${EXTEN} and
407
   further redirects.  Just set the REDIRECTING(to-num,i) = ${EXTEN} and
395
   set the REDIRECTING(to-pres) to the COLR.  A call has been redirected
408
   set the REDIRECTING(to-pres) to the COLR.  A call has been redirected
396
   if the REDIRECTING(from-num) is not empty.
409
   if the REDIRECTING(from-num) is not empty.
397
 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
410
 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
398
   option on all of the REDIRECTING statements before dialing the
411
   option on all of the REDIRECTING statements before dialing the
399
   redirected-to party.  You still have to set the REDIRECTING(to-xxx,i)
412
   redirected-to party.  You still have to set the REDIRECTING(to-xxx,i)
400
   and the REDIRECTING(from-xxx,i) values.  The PTP call will update the
413
   and the REDIRECTING(from-xxx,i) values.  The PTP call will update the
401
   redirecting-to presentation (COLR) when it becomes available.
414
   redirecting-to presentation (COLR) when it becomes available.
402
 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
415
 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
403
   information.
416
   information.
404

    
   
417

   
405
thirdparty mISDN enhancements
418
thirdparty mISDN enhancements
406
-----------------------------
419
-----------------------------
407
mISDN has been modified by Digium, Inc. to greatly expand facility message
420
mISDN has been modified by Digium, Inc. to greatly expand facility message
408
support to allow:
421
support to allow:
409
  * Enhanced COLP support for call diversion and transfer.
422
  * Enhanced COLP support for call diversion and transfer.
410
  * CCBS/CCNR support.
423
  * CCBS/CCNR support.
411

    
   
424

   
412
The latest modified mISDN v1.1.x based version is available at:
425
The latest modified mISDN v1.1.x based version is available at:
413
http://svn.digium.com/svn/thirdparty/mISDN/trunk
426
http://svn.digium.com/svn/thirdparty/mISDN/trunk
414
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
427
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
415

    
   
428

   
416
Tagged versions of the modified mISDN code are available under:
429
Tagged versions of the modified mISDN code are available under:
417
http://svn.digium.com/svn/thirdparty/mISDN/tags
430
http://svn.digium.com/svn/thirdparty/mISDN/tags
418
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
431
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
419

    
   
432

   
420
libpri channel driver (chan_dahdi) DAHDI changes
433
libpri channel driver (chan_dahdi) DAHDI changes
421
-------------------------------------------
434
-------------------------------------------
422
 * The channel variable PRIREDIRECTREASON is now just a status variable
435
 * The channel variable PRIREDIRECTREASON is now just a status variable
423
   and it is also deprecated.  Use the REDIRECTING(reason) dialplan function
436
   and it is also deprecated.  Use the REDIRECTING(reason) dialplan function
424
   to read and alter the reason.
437
   to read and alter the reason.
425
 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
438
 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
426
   redirected-to party for an incomming redirected call if the incoming call
439
   redirected-to party for an incomming redirected call if the incoming call
427
   could experience further redirects.  Just set the
440
   could experience further redirects.  Just set the
428
   REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
441
   REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
429
   to the COLR.  A call has been redirected if the REDIRECTING(count) is not
442
   to the COLR.  A call has been redirected if the REDIRECTING(count) is not
430
   zero.
443
   zero.
431
 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
444
 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
432
   use the inhibit(i) option on all of the REDIRECTING statements before
445
   use the inhibit(i) option on all of the REDIRECTING statements before
433
   dialing the redirected-to party.  You still have to set the
446
   dialing the redirected-to party.  You still have to set the
434
   REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values.  The call
447
   REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values.  The call
435
   will update the redirecting-to presentation (COLR) when it becomes available.
448
   will update the redirecting-to presentation (COLR) when it becomes available.
436
 * Added the ability to ignore calls that are not in a Multiple Subscriber
449
 * Added the ability to ignore calls that are not in a Multiple Subscriber
437
   Number (MSN) list for PTMP CPE interfaces.
450
   Number (MSN) list for PTMP CPE interfaces.
438
 * Added dynamic range compression support for dahdi channels.  It is
451
 * Added dynamic range compression support for dahdi channels.  It is
439
   configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
452
   configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
440
 * Added support for ISDN calling and called subaddress with partial support
453
 * Added support for ISDN calling and called subaddress with partial support
441
   for connected line subaddress.
454
   for connected line subaddress.
442
 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
455
 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
443
 * Added handling of received HOLD/RETRIEVE messages and the optional ability
456
 * Added handling of received HOLD/RETRIEVE messages and the optional ability
444
   to transfer a held call on disconnect similar to an analog phone.
457
   to transfer a held call on disconnect similar to an analog phone.
445
 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
458
 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
446
   Will reroute/deflect an outgoing call when receive the message.
459
   Will reroute/deflect an outgoing call when receive the message.
447
   Can use the DAHDISendCallreroutingFacility to send the message for the
460
   Can use the DAHDISendCallreroutingFacility to send the message for the
448
   supported switches.
461
   supported switches.
449
 * Added standard location to add options to chan_dahdi dialing:
462
 * Added standard location to add options to chan_dahdi dialing:
450
   Dial(DAHDI/g1[/extension[/options]])
463
   Dial(DAHDI/g1[/extension[/options]])
451
   Current options:
464
   Current options:
452
   K(<keypad_digits>)
465
   K(<keypad_digits>)
453
   R Reverse charging indication
466
   R Reverse charging indication
454
 * Added Reverse Charging Indication (Collect calls) send/receive option.
467
 * Added Reverse Charging Indication (Collect calls) send/receive option.
455
   Send reverse charging in SETUP message with the chan_dahdi R dialing option.
468
   Send reverse charging in SETUP message with the chan_dahdi R dialing option.
456
   Dial(DAHDI/g1/extension/R)
469
   Dial(DAHDI/g1/extension/R)
457
   Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
470
   Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
458
   (requires latest LibPRI)
471
   (requires latest LibPRI)
459
 * Added ability to send/receive keypad digits in the SETUP message.
472
 * Added ability to send/receive keypad digits in the SETUP message.
460
   Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
473
   Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
461
   dialing option.  Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
474
   dialing option.  Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
462
   Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
475
   Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
463
   (requires latest LibPRI)
476
   (requires latest LibPRI)
464
 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
477
 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
465
   to eliminate tromboned calls.  A tromboned call goes out an interface and comes
478
   to eliminate tromboned calls.  A tromboned call goes out an interface and comes
466
   back into the same interface.  Tromboned calls happen because of call routing,
479
   back into the same interface.  Tromboned calls happen because of call routing,
467
   call deflection, call forwarding, and call transfer.
480
   call deflection, call forwarding, and call transfer.
468
 * Added the ability to send and receive ETSI Advice-Of-Charge messages. 
481
 * Added the ability to send and receive ETSI Advice-Of-Charge messages. 
469
 * Added the ability to support call waiting calls.  (The SETUP has no B channel
482
 * Added the ability to support call waiting calls.  (The SETUP has no B channel
470
   assigned.)
483
   assigned.)
471
 * Added Malicious Call ID (MCID) event to the AMI call event class.
484
 * Added Malicious Call ID (MCID) event to the AMI call event class.
472
 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
485
 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
473

    
   
486

   
474
Asterisk Manager Interface
487
Asterisk Manager Interface
475
--------------------------
488
--------------------------
476
 * The Hangup action now accepts a Cause header which may be used to
489
 * The Hangup action now accepts a Cause header which may be used to
477
   set the channel's hangup cause.
490
   set the channel's hangup cause.
478
 * sslprivatekey option added to manager.conf and http.conf.  Adds the ability
491
 * sslprivatekey option added to manager.conf and http.conf.  Adds the ability
479
   to specify a separate .pem file to hold a private key.  By default sslcert
492
   to specify a separate .pem file to hold a private key.  By default sslcert
480
   is used to hold both the public and private key.
493
   is used to hold both the public and private key.
481
 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
494
 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
482
   for options containing the 'tls' prefix.  For example, 'sslenable' is now
495
   for options containing the 'tls' prefix.  For example, 'sslenable' is now
483
   'tlsenable'.  This has been done in effort to keep ssl and tls options consistent
496
   'tlsenable'.  This has been done in effort to keep ssl and tls options consistent
484
   across all .conf files. All affected sample.conf files have been modified to
497
   across all .conf files. All affected sample.conf files have been modified to
485
   reflect this change.  Previous options such as 'sslenable' still work,
498
   reflect this change.  Previous options such as 'sslenable' still work,
486
   but options with the 'tls' prefix are preferred.
499
   but options with the 'tls' prefix are preferred.
487
 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
500
 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
488
   in a channel. (res_mutestream.so)
501
   in a channel. (res_mutestream.so)
489
 * The configuration file manager.conf now supports a channelvars option, which
502
 * The configuration file manager.conf now supports a channelvars option, which
490
   specifies a list of channel variables to include in each channel-oriented
503
   specifies a list of channel variables to include in each channel-oriented
491
   event.
504
   event.
492
 * The redirect command now has new parameters ExtraContext, ExtraExtension, 
505
 * The redirect command now has new parameters ExtraContext, ExtraExtension, 
493
   and ExtraPriority to allow redirecting the second channel to a different
506
   and ExtraPriority to allow redirecting the second channel to a different
494
   location than the first.
507
   location than the first.
495
 * Added new event "JabberStatus" in the Jabber module to monitor buddies
508
 * Added new event "JabberStatus" in the Jabber module to monitor buddies
496
   status.
509
   status.
497
 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
510
 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
498
   in a MixMonitor recording.
511
   in a MixMonitor recording.
499
 * The 'iax2 show peers' output is now similar to the expected output of
512
 * The 'iax2 show peers' output is now similar to the expected output of
500
   'sip show peers'.
513
   'sip show peers'.
501
 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
514
 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
502
   aoc event class.
515
   aoc event class.
503
 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
516
 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
504
   AOC-E messages on a channel.
517
   AOC-E messages on a channel.
505
 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
518
 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
506
   conform more closely to similar events.
519
   conform more closely to similar events.
507
 * Added a new eventfilter option per user to allow whitelisting and blacklisting
520
 * Added a new eventfilter option per user to allow whitelisting and blacklisting
508
   of events.
521
   of events.
509
 * Added optional parkinglot variable for park command.
522
 * Added optional parkinglot variable for park command.
510

    
   
523

   
511
Channel Event Logging
524
Channel Event Logging
512
---------------------
525
---------------------
513
 * A new interface, CEL, is introduced here. CEL logs single events, much like
526
 * A new interface, CEL, is introduced here. CEL logs single events, much like
514
   the AMI, but it differs from the AMI in that it logs to db backends much
527
   the AMI, but it differs from the AMI in that it logs to db backends much
515
   like CDR does; is based on the event subsystem introduced by Russell, and
528
   like CDR does; is based on the event subsystem introduced by Russell, and
516
   can share in all its benefits; allows multiple backends to operate like CDR;
529
   can share in all its benefits; allows multiple backends to operate like CDR;
517
   is specialized to event data that would be of concern to billing sytems,
530
   is specialized to event data that would be of concern to billing sytems,
518
   like CDR. Backends for logging and accounting calls have been produced,
531
   like CDR. Backends for logging and accounting calls have been produced,
519
   but a new CDR backend is still in development.
532
   but a new CDR backend is still in development.
520

    
   
533

   
521
CDR
534
CDR
522
---
535
---
523
 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
536
 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
524
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
537
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
525
   etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
538
   etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
526
 * Multiple files and formats can now be specified in cdr_custom.conf.
539
 * Multiple files and formats can now be specified in cdr_custom.conf.
527
 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
540
 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
528
   See configs/cdr_syslog.conf.sample for more information.
541
   See configs/cdr_syslog.conf.sample for more information.
529
 * A 'sequence' field has been added to CDRs which can be combined with
542
 * A 'sequence' field has been added to CDRs which can be combined with
530
   linkedid or uniqueid to uniquely identify a CDR.
543
   linkedid or uniqueid to uniquely identify a CDR.
531
 * Handling of billsec and duration field has changed. If your table definition
544
 * Handling of billsec and duration field has changed. If your table definition
532
   specifies those fields as float,double or similar they will now be logged with
545
   specifies those fields as float,double or similar they will now be logged with
533
   microsecond accuracy instead of a whole integer.
546
   microsecond accuracy instead of a whole integer.
534

    
   
547

   
535
Calendaring for Asterisk
548
Calendaring for Asterisk
536
------------------------
549
------------------------
537
 * A new set of modules were added supporing calendar integration with Asterisk.
550
 * A new set of modules were added supporing calendar integration with Asterisk.
538
   Dialplan functions for reading from and writing to calendars are included,
551
   Dialplan functions for reading from and writing to calendars are included,
539
   as well as the ability to execute dialplan logic upon calendar event notifications.
552
   as well as the ability to execute dialplan logic upon calendar event notifications.
540
   iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
553
   iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
541
   Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
554
   Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
542
   Exchange Server 2007+ with full write and attendee support) are supported (Exchange
555
   Exchange Server 2007+ with full write and attendee support) are supported (Exchange
543
   2003 support does not support forms-based authentication).
556
   2003 support does not support forms-based authentication).
544

    
   
557

   
545
Call Completion Supplementary Services for Asterisk
558
Call Completion Supplementary Services for Asterisk
546
---------------------------------------------------
559
---------------------------------------------------
547
 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
560
 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
548
   DAHDI/ISDN supports call completion for the following switch types:
561
   DAHDI/ISDN supports call completion for the following switch types:
549
   EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
562
   EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
550
   See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
563
   See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
551

    
   
564

   
552
Multicast RTP Support
565
Multicast RTP Support
553
---------------------
566
---------------------
554
 * A new RTP engine and channel driver have been added which supports Multicast RTP.
567
 * A new RTP engine and channel driver have been added which supports Multicast RTP.
555
   The channel driver can be used with the Page application to perform multicast RTP
568
   The channel driver can be used with the Page application to perform multicast RTP
556
   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
569
   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
557
   Type can be either basic or linksys.
570
   Type can be either basic or linksys.
558
   Destination is the IP address and port for the RTP packets.
571
   Destination is the IP address and port for the RTP packets.
559
   Control address is specific to the linksys type and is used for sending the control
572
   Control address is specific to the linksys type and is used for sending the control
560
   packets unique to them.
573
   packets unique to them.
561

    
   
574

   
562
Security Events Framework
575
Security Events Framework
563
-------------------------
576
-------------------------
564
 * Asterisk has a new C API for reporting security events.  The module res_security_log
577
 * Asterisk has a new C API for reporting security events.  The module res_security_log
565
   sends these events to the "security" logger level.  Currently, AMI is the only
578
   sends these events to the "security" logger level.  Currently, AMI is the only
566
   Asterisk component that reports security events.  However, SIP support will be
579
   Asterisk component that reports security events.  However, SIP support will be
567
   coming soon.  For more information on the security events framework, see the
580
   coming soon.  For more information on the security events framework, see the
568
   "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
581
   "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
569

    
   
582

   
570
Fax
583
Fax
571
---
584
---
572
 * A technology independent fax frontend (res_fax) has been added to Asterisk.
585
 * A technology independent fax frontend (res_fax) has been added to Asterisk.
573
 * A spandsp based fax backend (res_fax_spandsp) has been added.
586
 * A spandsp based fax backend (res_fax_spandsp) has been added.
574
 * The app_fax module has been deprecated in favor of the res_fax module and
587
 * The app_fax module has been deprecated in favor of the res_fax module and
575
   the new res_fax_spandsp backend.
588
   the new res_fax_spandsp backend.
576
 * The SendFAX and ReceiveFAX applications now send their log messages to a
589
 * The SendFAX and ReceiveFAX applications now send their log messages to a
577
   'fax' logger level, instead of to the generic logger levels. To see these
590
   'fax' logger level, instead of to the generic logger levels. To see these
578
   messages, the system's logger.conf file will need to direct the 'fax' logger
591
   messages, the system's logger.conf file will need to direct the 'fax' logger
579
   level to one or more destinations; the logger.conf.sample file includes an
592
   level to one or more destinations; the logger.conf.sample file includes an
580
   example of how to do this. Note that if the 'fax' logger level is *not*
593
   example of how to do this. Note that if the 'fax' logger level is *not*
581
   directed to at least one destination, log messages generated by these
594
   directed to at least one destination, log messages generated by these
582
   applications will be lost, and that if the 'fax' logger level is directed to
595
   applications will be lost, and that if the 'fax' logger level is directed to
583
   the console, the 'core set verbose' and 'core set debug' CLI commands will
596
   the console, the 'core set verbose' and 'core set debug' CLI commands will
584
   have no effect on whether the messages appear on the console or not.
597
   have no effect on whether the messages appear on the console or not.
585

    
   
598

   
586
Miscellaneous
599
Miscellaneous
587
-------------
600
-------------
588
 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
601
 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
589
   Now, in order to enable transmitting silence during record the transmit_silence
602
   Now, in order to enable transmitting silence during record the transmit_silence
590
   option should be used.  transmit_silence_during_record remains a valid option, but
603
   option should be used.  transmit_silence_during_record remains a valid option, but
591
   defaults to the behavior of the transmit_silence option.
604
   defaults to the behavior of the transmit_silence option.
592
 * Addition of the Unit Test Framework API for managing registration and execution
605
 * Addition of the Unit Test Framework API for managing registration and execution
593
   of unit tests with the purpose of verifying the operation of C functions.
606
   of unit tests with the purpose of verifying the operation of C functions.
594
 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
607
 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
595
   XMPP text messages to the remote JID.
608
   XMPP text messages to the remote JID.
596
 * Modules.conf has a new option - "require" - that marks a module as critical for 
609
 * Modules.conf has a new option - "require" - that marks a module as critical for 
597
   the execution of Asterisk.
610
   the execution of Asterisk.
598
   If one of the required modules fail to load, Asterisk will exit with a return
611
   If one of the required modules fail to load, Asterisk will exit with a return
599
   code set to 2.
612
   code set to 2.
600
 * An 'X' option has been added to the asterisk application which enables #exec support.
613
 * An 'X' option has been added to the asterisk application which enables #exec support.
601
   This allows #exec to be used in asterisk.conf.
614
   This allows #exec to be used in asterisk.conf.
602
 * jabber.conf supports a new option auth_policy that toggles auto user registration.
615
 * jabber.conf supports a new option auth_policy that toggles auto user registration.
603
 * A new lockconfdir option has been added to asterisk.conf to protect the
616
 * A new lockconfdir option has been added to asterisk.conf to protect the
604
   configuration directory (/etc/asterisk by default) during reloads.
617
   configuration directory (/etc/asterisk by default) during reloads.
605
 * The parkeddynamic option has been added to features.conf to enable the creation
618
 * The parkeddynamic option has been added to features.conf to enable the creation
606
   of dynamic parkinglots.
619
   of dynamic parkinglots.
607
 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
620
 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
608
   the reportalarms config option.
621
   the reportalarms config option.
609
 * chan_dahdi supports dialing configuring and dialing by device file name.
622
 * chan_dahdi supports dialing configuring and dialing by device file name.
610
   DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
623
   DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
611
   it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
624
   it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
612
 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
625
 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
613
   False by default. If set, chan_dahdi will ignore failed 'channel' entries.
626
   False by default. If set, chan_dahdi will ignore failed 'channel' entries.
614
   Handy for the above name-based syntax as it does not depend on
627
   Handy for the above name-based syntax as it does not depend on
615
   initialization order.
628
   initialization order.
616
 * The Realtime dialplan switch now caches entries for 1 second.  This provides a
629
 * The Realtime dialplan switch now caches entries for 1 second.  This provides a
617
   significant increase in performance (about 3X) for installations using this switchtype.
630
   significant increase in performance (about 3X) for installations using this switchtype.
618
 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
631
 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
619
   AIS.  For more information, please see doc/distributed_devstate-XMPP.txt
632
   AIS.  For more information, please see doc/distributed_devstate-XMPP.txt
620
 * The addition of G.719 pass-through support.
633
 * The addition of G.719 pass-through support.
621
 * Added support for 16khz Speex audio.  This can be enabled by using 'allow=speex16'
634
 * Added support for 16khz Speex audio.  This can be enabled by using 'allow=speex16'
622
   during device configuration.
635
   during device configuration.
623
 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
636
 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
624
   have less than 3 lines on the LCD.
637
   have less than 3 lines on the LCD.
625
 * Realtime now supports database failover.  See the sample extconfig.conf for details.
638
 * Realtime now supports database failover.  See the sample extconfig.conf for details.
626
 * The addition of improved translation path building for wideband codecs.  Sample
639
 * The addition of improved translation path building for wideband codecs.  Sample
627
   rate changes during translation are now avoided unless absolutely necessary.
640
   rate changes during translation are now avoided unless absolutely necessary.
628
 * The addition of the res_stun_monitor module for monitoring and reacting to network
641
 * The addition of the res_stun_monitor module for monitoring and reacting to network
629
   changes while behind a NAT.
642
   changes while behind a NAT.
630

    
   
643

   
631
CLI Changes
644
CLI Changes
632
-----------
645
-----------
633
 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
646
 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
634
   optionally accept a filename, to apply the setting only to the code generated from
647
   optionally accept a filename, to apply the setting only to the code generated from
635
   that source file when Asterisk was built. However, there are some modules in Asterisk
648
   that source file when Asterisk was built. However, there are some modules in Asterisk
636
   that are composed of multiple source files, so this did not result in the behavior
649
   that are composed of multiple source files, so this did not result in the behavior
637
   that users expected. In this version, 'core set debug' and 'core set verbose'
650
   that users expected. In this version, 'core set debug' and 'core set verbose'
638
   can optionally accept *module* names instead (with or without the .so extension),
651
   can optionally accept *module* names instead (with or without the .so extension),
639
   which applies the setting to the entire module specified, regardless of which source
652
   which applies the setting to the entire module specified, regardless of which source
640
   files it was built from.
653
   files it was built from.
641
 * New 'manager show settings' command showing the current settings loaded from
654
 * New 'manager show settings' command showing the current settings loaded from
642
   manager.conf. 
655
   manager.conf. 
643
 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
656
 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
644
   the channel hangup request to all channels.
657
   the channel hangup request to all channels.
645
 * Added a "core reload" CLI command that executes a global reload of Asterisk.
658
 * Added a "core reload" CLI command that executes a global reload of Asterisk.
646

    
   
659

   
647
------------------------------------------------------------------------------
660
------------------------------------------------------------------------------
648
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
661
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
649
------------------------------------------------------------------------------
662
------------------------------------------------------------------------------
650

    
   
663

   
651
SIP Changes
664
SIP Changes
652
-----------
665
-----------
653
 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
666
 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
654
   Snom phones use this for call pickup of extensions that the phone is
667
   Snom phones use this for call pickup of extensions that the phone is
655
   subscribed to.
668
   subscribed to.
656
 * Added support for setting the domain in the URI for caller of an
669
 * Added support for setting the domain in the URI for caller of an
657
   outbound call by using the SIPFROMDOMAIN channel variable.
670
   outbound call by using the SIPFROMDOMAIN channel variable.
658
 * Added a new configuration option "remotesecret" for authentication to
671
 * Added a new configuration option "remotesecret" for authentication to
659
   remote services. For backwards compatibility, "secret" still has the
672
   remote services. For backwards compatibility, "secret" still has the
660
   same function as before, but now you can configure both a remote secret and a
673
   same function as before, but now you can configure both a remote secret and a
661
   local secret for mutual authentication.
674
   local secret for mutual authentication.
662
 * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set, 
675
 * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set, 
663
   the sound will be played to the target of an attended transfer
676
   the sound will be played to the target of an attended transfer
664
 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
677
 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
665
   finer control over how many peers Asterisk will qualify and the gap between them
678
   finer control over how many peers Asterisk will qualify and the gap between them
666
   when all peers need to be qualified at the same time.
679
   when all peers need to be qualified at the same time.
667
 * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
680
 * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
668
   (either globally or for a specific peer), chan_sip will treat any SDP data
681
   (either globally or for a specific peer), chan_sip will treat any SDP data
669
   it receives as new data and update the media stream accordingly.  By
682
   it receives as new data and update the media stream accordingly.  By
670
   default, Asterisk will only modify the media stream if the SDP session
683
   default, Asterisk will only modify the media stream if the SDP session
671
   version received is different from the current SDP session version.  This
684
   version received is different from the current SDP session version.  This
672
   option is required to interoperate with devices that have non-standard SDP
685
   option is required to interoperate with devices that have non-standard SDP
673
   session version implementations (observed with Microsoft OCS).  This option
686
   session version implementations (observed with Microsoft OCS).  This option
674
   is disabled by default.
687
   is disabled by default.
675
 * The parsing of register => lines in sip.conf has been modified to allow a port
688
 * The parsing of register => lines in sip.conf has been modified to allow a port
676
   to be present in the "user" portion. Please see the sip.conf.sample file for more
689
   to be present in the "user" portion. Please see the sip.conf.sample file for more
677
   information
690
   information
678
 * Added support for subscribing to MWI on a remote server and making the status available
691
 * Added support for subscribing to MWI on a remote server and making the status available
679
   as a mailbox. Please see the sip.conf.sample file for more information.
692
   as a mailbox. Please see the sip.conf.sample file for more information.
680
 * Added a function to remove SIP headers added in the dialplan before the
693
 * Added a function to remove SIP headers added in the dialplan before the
681
   first INVITE is generated - SIPRemoveHeader()
694
   first INVITE is generated - SIPRemoveHeader()
682
 * Channel variables set with setvar= in a device configuration is now 
695
 * Channel variables set with setvar= in a device configuration is now 
683
   set both for inbound and outbound calls.
696
   set both for inbound and outbound calls.
684
 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
697
 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
685

    
   
698

   
686
IAX2 changes
699
IAX2 changes
687
------------
700
------------
688
  * Added immediate option to iax.conf
701
  * Added immediate option to iax.conf
689
  * Added forceencryption option to iax.conf
702
  * Added forceencryption option to iax.conf
690
  * Added Encryption and Trunk status to manager command "iaxpeers"
703
  * Added Encryption and Trunk status to manager command "iaxpeers"
691

    
   
704

   
692
Skinny Changes
705
Skinny Changes
693
--------------
706
--------------
694
 * The configuration file now holds separate sections for devices and lines.
707
 * The configuration file now holds separate sections for devices and lines.
695
   Please have a look at configs/skinny.conf.sample and change your skinny.conf
708
   Please have a look at configs/skinny.conf.sample and change your skinny.conf
696
   accordingly.
709
   accordingly.
697

    
   
710

   
698
DAHDI Changes
711
DAHDI Changes
699
-------------
712
-------------
700
 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
713
 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
701
   support for LibOpenR2.  http://www.libopenr2.org/
714
   support for LibOpenR2.  http://www.libopenr2.org/
702
 * The UK option waitfordialtone has been added for use with BT analog
715
 * The UK option waitfordialtone has been added for use with BT analog
703
   lines.
716
   lines.
704
 * Added a 'faxbuffers' configuration option to chan_dahdi.conf.  This option
717
 * Added a 'faxbuffers' configuration option to chan_dahdi.conf.  This option
705
   is used in conjunction with the 'faxdetect' configuration option.  When
718
   is used in conjunction with the 'faxdetect' configuration option.  When
706
   'faxbuffers' is used and fax tones are detected, the channel will dynamically
719
   'faxbuffers' is used and fax tones are detected, the channel will dynamically
707
   switch to the configured faxbuffers policy.  For example, to use 6 buffers
720
   switch to the configured faxbuffers policy.  For example, to use 6 buffers
708
   and a 'full' buffer policy for a fax transmission, add:
721
   and a 'full' buffer policy for a fax transmission, add:
709
     faxbuffers=>6,full
722
     faxbuffers=>6,full
710
   The faxbuffers configuration will be in affect until the call is torn down.
723
   The faxbuffers configuration will be in affect until the call is torn down.
711
 * Added service message support for 4ESS/5ESS switches.
724
 * Added service message support for 4ESS/5ESS switches.
712

    
   
725

   
713
Dialplan Functions
726
Dialplan Functions
714
------------------
727
------------------
715
 * For DAHDI channels, the CHANNEL() dialplan function now
728
 * For DAHDI channels, the CHANNEL() dialplan function now
716
   supports changing the channel's buffer policy (for the current
729
   supports changing the channel's buffer policy (for the current
717
   call only), using this syntax:
730
   call only), using this syntax:
718

    
   
731

   
719
   exten => s,n,Set(CHANNEL(buffers)=6,full)
732
   exten => s,n,Set(CHANNEL(buffers)=6,full)
720

    
   
733

   
721
   This would change the channel to the 'full' buffer policy and
734
   This would change the channel to the 'full' buffer policy and
722
   6 (six) buffers. Possible options for this setting are the same
735
   6 (six) buffers. Possible options for this setting are the same
723
   as those in chan_dahdi.conf.
736
   as those in chan_dahdi.conf.
724
 * Added a new dialplan function, CURLOPT, which permits setting various
737
 * Added a new dialplan function, CURLOPT, which permits setting various
725
   options that may be useful with the CURL dialplan function, such as
738
   options that may be useful with the CURL dialplan function, such as
726
   cookies, proxies, connection timeouts, passwords, etc.
739
   cookies, proxies, connection timeouts, passwords, etc.
727
 * Permit the syntax and synopsis fields of the corresponding dialplan
740
 * Permit the syntax and synopsis fields of the corresponding dialplan
728
   functions to be individually set from func_odbc.conf.
741
   functions to be individually set from func_odbc.conf.
729
 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
742
 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
730
 * func_odbc now may specify an insert query to execute, when the write query
743
 * func_odbc now may specify an insert query to execute, when the write query
731
   affects 0 rows (usually indicating that no such row exists).
744
   affects 0 rows (usually indicating that no such row exists).
732
 * Added a new dialplan function, LISTFILTER, which permits removing elements
745
 * Added a new dialplan function, LISTFILTER, which permits removing elements
733
   from a set list, by name.  Uses the same general syntax as the existing CUT
746
   from a set list, by name.  Uses the same general syntax as the existing CUT
734
   and FIELDQTY dialplan functions, which also manage lists.
747
   and FIELDQTY dialplan functions, which also manage lists.
735
 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
748
 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
736
   obtaining realtime data from the dialplan.
749
   obtaining realtime data from the dialplan.
737
 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
750
 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
738
   a subroutine when using the GoSub() and Return() applications.
751
   a subroutine when using the GoSub() and Return() applications.
739
 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
752
 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
740
   of "core show function AUDIOHOOK_INHERIT" from the CLI
753
   of "core show function AUDIOHOOK_INHERIT" from the CLI
741
 * Added AES_ENCRYPT. For information on its use, please see the output
754
 * Added AES_ENCRYPT. For information on its use, please see the output
742
   of "core show function AES_ENCRYPT" from the CLI
755
   of "core show function AES_ENCRYPT" from the CLI
743
 * Added AES_DECRYPT. For information on its use, please see the output
756
 * Added AES_DECRYPT. For information on its use, please see the output
744
   of "core show function AES_DECRYPT" from the CLI
757
   of "core show function AES_DECRYPT" from the CLI
745
 * func_odbc now supports database transactions across multiple queries.
758
 * func_odbc now supports database transactions across multiple queries.
746

    
   
759

   
747
Applications
760
Applications
748
------------
761
------------
749
 * Scheduled meetme conferences may now have their end times extended by
762
 * Scheduled meetme conferences may now have their end times extended by
750
   using MeetMeAdmin.
763
   using MeetMeAdmin.
751
 * app_authenticate now gives the ability to select a prompt other than
764
 * app_authenticate now gives the ability to select a prompt other than
752
   the default.
765
   the default.
753
 * app_directory now pays attention to the searchcontexts setting in
766
 * app_directory now pays attention to the searchcontexts setting in
754
   voicemail.conf and will look through all contexts, if no context is
767
   voicemail.conf and will look through all contexts, if no context is
755
   specified in the initial argument.
768
   specified in the initial argument.
756
 * A new application, Originate, has been introduced, that allows asynchronous
769
 * A new application, Originate, has been introduced, that allows asynchronous
757
   call origination from the dialplan.
770
   call origination from the dialplan.
758
 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
771
 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
759
   in addition to the setting in the "general" context.
772
   in addition to the setting in the "general" context.
760
 * Added ConfBridge dialplan application which does conference bridges without
773
 * Added ConfBridge dialplan application which does conference bridges without
761
   DAHDI. For information on its use, please see the output of
774
   DAHDI. For information on its use, please see the output of
762
   "core show application ConfBridge" from the CLI.
775
   "core show application ConfBridge" from the CLI.
763

    
   
776

   
764
Miscellaneous
777
Miscellaneous
765
-------------
778
-------------
766
 * The Asterisk CLI has a new command, "channel redirect", which is similar in
779
 * The Asterisk CLI has a new command, "channel redirect", which is similar in
767
   operation to the AMI Redirect action.
780
   operation to the AMI Redirect action.
768
 * extensions.conf now allows you to use keyword "same" to define an extension
781
 * extensions.conf now allows you to use keyword "same" to define an extension
769
   without actually specifying an extension.  It uses exactly the same pattern
782
   without actually specifying an extension.  It uses exactly the same pattern
770
   as previously used on the last "exten" line.  For example:
783
   as previously used on the last "exten" line.  For example:
771
     exten => 123,1,NoOp(something)
784
     exten => 123,1,NoOp(something)
772
     same  =>     n,SomethingElse()
785
     same  =>     n,SomethingElse()
773
 * musiconhold.conf classes of type 'files' can now use relative directory paths,
786
 * musiconhold.conf classes of type 'files' can now use relative directory paths,
774
   which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
787
   which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
775
 * All deprecated CLI commands are removed from the sourcecode. They are now handled
788
 * All deprecated CLI commands are removed from the sourcecode. They are now handled
776
   by the new clialiases module. See cli_aliases.conf.sample file.
789
   by the new clialiases module. See cli_aliases.conf.sample file.
777
 * Times within timespecs are now accurate down to the minute.  This is a change
790
 * Times within timespecs are now accurate down to the minute.  This is a change
778
   from historical Asterisk, which only provided timespecs rounded to the nearest
791
   from historical Asterisk, which only provided timespecs rounded to the nearest
779
   even (read: evenly divisible by 2) minute mark.
792
   even (read: evenly divisible by 2) minute mark.
780
 * The realtime switch now supports an option flag, 'p', which disables searches for
793
 * The realtime switch now supports an option flag, 'p', which disables searches for
781
   pattern matches.
794
   pattern matches.
782
 * In addition to a time range and date range, timespecs now accept a 5th optional
795
 * In addition to a time range and date range, timespecs now accept a 5th optional
783
   argument, timezone.  This allows you to perform time checks on alternate
796
   argument, timezone.  This allows you to perform time checks on alternate
784
   timezones, especially if those daylight savings time ranges vary from your
797
   timezones, especially if those daylight savings time ranges vary from your
785
   machine's native timezone.  See GotoIfTime, ExecIfTime, IFTIME(), and timed
798
   machine's native timezone.  See GotoIfTime, ExecIfTime, IFTIME(), and timed
786
   includes.
799
   includes.
787
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
800
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
788
   give you the correct output for an asterisk box behind nat. It will give you the
801
   give you the correct output for an asterisk box behind nat. It will give you the
789
   externhost and localnet settings.
802
   externhost and localnet settings.
790
 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
803
 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
791
   can connect calls in passthrough mode, as well as record and play back files.
804
   can connect calls in passthrough mode, as well as record and play back files.
792
 * Successful and unsuccessful call pickup can now be alerted through sounds, by
805
 * Successful and unsuccessful call pickup can now be alerted through sounds, by
793
   using pickupsound and pickupfailsound in features.conf.
806
   using pickupsound and pickupfailsound in features.conf.
794
 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
807
 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
795
   This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
808
   This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
796
   instead of the /var/run/asterisk.pid where it used to be. This will make
809
   instead of the /var/run/asterisk.pid where it used to be. This will make
797
   installs as non-root easier to manage.
810
   installs as non-root easier to manage.
798

    
   
811

   
799
CDR
812
CDR
800
---
813
---
801

    
   
814

   
802
* The cdr.conf file must exist and be correctly programmed in order for CDR records to
815
* The cdr.conf file must exist and be correctly programmed in order for CDR records to
803
  be written; they will no longer be explicitly written.
816
  be written; they will no longer be explicitly written.
804

    
   
817

   
805
Asterisk Manager Interface
818
Asterisk Manager Interface
806
--------------------------
819
--------------------------
807
 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
820
 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
808
   a non-empty value) in your request. If you do this, any pending AMI events will
821
   a non-empty value) in your request. If you do this, any pending AMI events will
809
   *not* be included in the response to your request as they would normally, but
822
   *not* be included in the response to your request as they would normally, but
810
   will be left in the event queue for the next request you make to retrieve. For
823
   will be left in the event queue for the next request you make to retrieve. For
811
   some applications, this will allow you to guarantee that you will only see
824
   some applications, this will allow you to guarantee that you will only see
812
   events in responses to 'WaitEvent' actions, and can better know when to expect them.
825
   events in responses to 'WaitEvent' actions, and can better know when to expect them.
813
   To know whether the Asterisk server supports this header or not, your client can
826
   To know whether the Asterisk server supports this header or not, your client can
814
   inspect the first response back from the server to see if it includes this header:
827
   inspect the first response back from the server to see if it includes this header:
815

    
   
828

   
816
   Pragma: SuppressEvents
829
   Pragma: SuppressEvents
817

    
   
830

   
818
   If this is included, the server supports event suppression.
831
   If this is included, the server supports event suppression.
819

    
   
832

   
820
 * Added 4 new Actions to list skinny device(s) and line(s)
833
 * Added 4 new Actions to list skinny device(s) and line(s)
821
   SKINNYdevices
834
   SKINNYdevices
822
   SKINNYshowdevice
835
   SKINNYshowdevice
823
   SKINNYlines
836
   SKINNYlines
824
   SKINNYshowline
837
   SKINNYshowline
825

    
   
838

   
826
LDAP Schema File Additions
839
LDAP Schema File Additions
827
--------------------------
840
--------------------------
828
 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox  objectClasses
841
 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox  objectClasses
829
   to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
842
   to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
830
 * Added new Fields:
843
 * Added new Fields:
831
   - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
844
   - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
832
   - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
845
   - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
833
   - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
846
   - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
834
 * Removed redundant IPaddr (there's already IPAddress)
847
 * Removed redundant IPaddr (there's already IPAddress)
835
   - Gives more configuration Flags for SIP-Users available (tested)
848
   - Gives more configuration Flags for SIP-Users available (tested)
836
   - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
849
   - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
837
     without extensibleObject (which really should be the last resort); gives
850
     without extensibleObject (which really should be the last resort); gives
838
     also additional possibilities for LDAP-filter 
851
     also additional possibilities for LDAP-filter 
839

    
   
852

   
840
------------------------------------------------------------------------------
853
------------------------------------------------------------------------------
841
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
854
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
842
------------------------------------------------------------------------------
855
------------------------------------------------------------------------------
843

    
   
856

   
844
Device State Handling
857
Device State Handling
845
---------------------
858
---------------------
846
 * The event infrastructure in Asterisk got another big update to help support
859
 * The event infrastructure in Asterisk got another big update to help support
847
    distributed events.  It currently supports distributed device state and
860
    distributed events.  It currently supports distributed device state and
848
    distributed Voicemail MWI (Message Waiting Indication).  A new module has
861
    distributed Voicemail MWI (Message Waiting Indication).  A new module has
849
    been merged, res_ais, which facilitates communicating events between servers.
862
    been merged, res_ais, which facilitates communicating events between servers.
850
    It uses the SAForum AIS (Service Availability Forum Application Interface
863
    It uses the SAForum AIS (Service Availability Forum Application Interface
851
    Specification) CLM (Cluster Management) and EVT (Event) services to maintain
864
    Specification) CLM (Cluster Management) and EVT (Event) services to maintain
852
    a cluster of Asterisk servers, and to share events between them.  For more
865
    a cluster of Asterisk servers, and to share events between them.  For more
853
    information on setting this up, see doc/distributed_devstate.txt.
866
    information on setting this up, see doc/distributed_devstate.txt.
854

    
   
867

   
855
Dialplan Functions
868
Dialplan Functions
856
------------------
869
------------------
857
 * Added a new dialplan function, AST_CONFIG(), which allows you to access
870
 * Added a new dialplan function, AST_CONFIG(), which allows you to access
858
   variables from an Asterisk configuration file.
871
   variables from an Asterisk configuration file.
859
 * The JACK_HOOK function now has a c() option to supply a custom client name.
872
 * The JACK_HOOK function now has a c() option to supply a custom client name.
860
 * Added two new dialplan functions from libspeex for audio gain control and 
873
 * Added two new dialplan functions from libspeex for audio gain control and 
861
   denoise, AGC() and DENOISE(). Both functions can be applied to the tx and 
874
   denoise, AGC() and DENOISE(). Both functions can be applied to the tx and 
862
   rx directions of a channel from the dialplan.
875
   rx directions of a channel from the dialplan.
863
 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
876
 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
864
   based on other parameters.  The default is still to search based on the
877
   based on other parameters.  The default is still to search based on the
865
   forwarding station ID.  However, there are new options that allow you to search
878
   forwarding station ID.  However, there are new options that allow you to search
866
   based on the message desk terminal ID, or the message desk number.
879
   based on the message desk terminal ID, or the message desk number.
867
 * TIMEOUT() has been modified to be accurate down to the millisecond.
880
 * TIMEOUT() has been modified to be accurate down to the millisecond.
868
 * ENUM*() functions now include the following new options:
881
 * ENUM*() functions now include the following new options:
869
     - 'u' returns the full URI and does not strip off the URI-scheme.
882
     - 'u' returns the full URI and does not strip off the URI-scheme.
870
     - 's' triggers ISN specific rewriting
883
     - 's' triggers ISN specific rewriting
871
     - 'i' looks for branches into an Infrastructure ENUM tree
884
     - 'i' looks for branches into an Infrastructure ENUM tree
872
     - 'd' for a direct DNS lookup without any flipping of digits.
885
     - 'd' for a direct DNS lookup without any flipping of digits.
873
 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
886
 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
874
 * CHANNEL() now has options for the maximum, minimum, and standard or normal
887
 * CHANNEL() now has options for the maximum, minimum, and standard or normal
875
   deviation of jitter, rtt, and loss for a call using chan_sip.
888
   deviation of jitter, rtt, and loss for a call using chan_sip.
876

    
   
889

   
877
DAHDI channel driver (chan_dahdi) Changes
890
DAHDI channel driver (chan_dahdi) Changes
878
----------------------------------------
891
----------------------------------------
879
 * Channels can now be configured using named sections in chan_dahdi.conf, just
892
 * Channels can now be configured using named sections in chan_dahdi.conf, just
880
   like other channel drivers, including the use of templates.
893
   like other channel drivers, including the use of templates.
881
 * The default for pridialplan has changed from 'national' to 'unknown'.
894
 * The default for pridialplan has changed from 'national' to 'unknown'.
882

    
   
895

   
883
PBX Changes
896
PBX Changes
884
-----------
897
-----------
885
 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
898
 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
886
   to something that matches the pattern a hint will be created using the contents
899
   to something that matches the pattern a hint will be created using the contents
887
   and variables evaluated.
900
   and variables evaluated.
888
 * Dialplan matching has been extended to allow an extension to return to the
901
 * Dialplan matching has been extended to allow an extension to return to the
889
   PBX core to wait for more digits.  This is done by using the new dialplan
902
   PBX core to wait for more digits.  This is done by using the new dialplan
890
   application called "Incomplete".  This will permit a whole new level of
903
   application called "Incomplete".  This will permit a whole new level of
891
   extension control, by giving the administrator more control over early
904
   extension control, by giving the administrator more control over early
892
   matches employing one of the short-circuit pattern match operators.  Note
905
   matches employing one of the short-circuit pattern match operators.  Note
893
   that custom applications can trigger this same behavior by returning the
906
   that custom applications can trigger this same behavior by returning the
894
   special value AST_PBX_INCOMPLETE.
907
   special value AST_PBX_INCOMPLETE.
895

    
   
908

   
896
Application Changes
909
Application Changes
897
-------------------
910
-------------------
898
 * Directory now permits both first and last names to be matched at the same
911
 * Directory now permits both first and last names to be matched at the same
899
   time.  In addition, the number of digits to enter of the name can be set in
912
   time.  In addition, the number of digits to enter of the name can be set in
900
   the arguments to Directory; previously, you could enter only 3, regardless
913
   the arguments to Directory; previously, you could enter only 3, regardless
901
   of how many names are in your company.  For large companies, this should be
914
   of how many names are in your company.  For large companies, this should be
902
   quite helpful.
915
   quite helpful.
903
 * Voicemail now permits a mailbox setting to wrap around from first to last
916
 * Voicemail now permits a mailbox setting to wrap around from first to last
904
   messages, if the "messagewrap" option is set to a true value.
917
   messages, if the "messagewrap" option is set to a true value.
905
 * Voicemail now permits an external script to be run, for password validation.
918
 * Voicemail now permits an external script to be run, for password validation.
906
   The script should output "VALID" or "INVALID" on stdout, depending upon the
919
   The script should output "VALID" or "INVALID" on stdout, depending upon the
907
   wish to validate or invalidate the password given.  Arguments are:
920
   wish to validate or invalidate the password given.  Arguments are:
908
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
921
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
909
   more details
922
   more details
910
 * Dial has a new option: F(context^extension^pri), which permits a callee to
923
 * Dial has a new option: F(context^extension^pri), which permits a callee to
911
   continue in the dialplan, at the specified label, if the caller hangs up.
924
   continue in the dialplan, at the specified label, if the caller hangs up.
912
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
925
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
913
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
926
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
914
 * The Jack application now has a c() option to supply a custom client name.
927
 * The Jack application now has a c() option to supply a custom client name.
915
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
928
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
916
   like the pre-existing whisper mode, except that the spy can also talk to the
929
   like the pre-existing whisper mode, except that the spy can also talk to the
917
   participant on the bridged channel as well.
930
   participant on the bridged channel as well.
918
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
931
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
919
   to be spoken instead of the channel name or number. For more information on the
932
   to be spoken instead of the channel name or number. For more information on the
920
   use of this option, issue the command "core show application ChanSpy" from the 
933
   use of this option, issue the command "core show application ChanSpy" from the 
921
   Asterisk CLI.
934
   Asterisk CLI.
922
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
935
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
923
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
936
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
924
   words, if using the 'd' option, it is not possible to enter a number to append to
937
   words, if using the 'd' option, it is not possible to enter a number to append to
925
   the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
938
   the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
926
   change to whisper mode, and pressing 6 will change to barge mode.
939
   change to whisper mode, and pressing 6 will change to barge mode.
927
 * ExternalIVR now takes several options that affect the way it performs, as
940
 * ExternalIVR now takes several options that affect the way it performs, as
928
   well as having several new commands.  Please see doc/externalivr.txt for the
941
   well as having several new commands.  Please see doc/externalivr.txt for the
929
   complete documentation.
942
   complete documentation.
930
 * Added ability to communicate over a TCP socket instead of forking a child process for the 
943
 * Added ability to communicate over a TCP socket instead of forking a child process for the 
931
   ExternalIVR application.
944
   ExternalIVR application.
932
 * ChanIsAvail has a new option, 'a', which will return all available channels instead
945
 * ChanIsAvail has a new option, 'a', which will return all available channels instead
933
   of just the first one if you give the function more then one channel to check.
946
   of just the first one if you give the function more then one channel to check.
934
 * PrivacyManager now takes an option where you can specify a context where the 
947
 * PrivacyManager now takes an option where you can specify a context where the 
935
   given number will be matched. This way you have more control over who is allowed
948
   given number will be matched. This way you have more control over who is allowed
936
   and it stops the people who blindly enter 10 digits.
949
   and it stops the people who blindly enter 10 digits.
937
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
950
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
938
   answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
951
   answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
939
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
952
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
940
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
953
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
941
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
954
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
942
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
955
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
943
 * The Dial() application no longer copies the language used by the caller to the callee's
956
 * The Dial() application no longer copies the language used by the caller to the callee's
944
   channel. If you desire for the caller's channel's language to be used for file playback
957
   channel. If you desire for the caller's channel's language to be used for file playback
945
   to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
958
   to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
946
 * SendImage() no longer hangs up the channel on error; instead, it sets the
959
 * SendImage() no longer hangs up the channel on error; instead, it sets the
947
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
960
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
948
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
961
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
949
   applications.
962
   applications.
950
 * Park has a new option, 's', which silences the announcement of the parking space number.
963
 * Park has a new option, 's', which silences the announcement of the parking space number.
951
 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
964
 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
952
   invalid input and will be assumed to mean that no timeout is desired.
965
   invalid input and will be assumed to mean that no timeout is desired.
953

    
   
966

   
954
SIP Changes
967
SIP Changes
955
-----------
968
-----------
956
 * Added DNS manager support to registrations for peers referencing peer entries.
969
 * Added DNS manager support to registrations for peers referencing peer entries.
957
   DNS manager runs in the background which allows DNS lookups to be run asynchronously 
970
   DNS manager runs in the background which allows DNS lookups to be run asynchronously 
958
   as well as periodically updating the IP address. These properties allow for
971
   as well as periodically updating the IP address. These properties allow for
959
   better performance as well as recovery in the event of an IP change.
972
   better performance as well as recovery in the event of an IP change.
960
 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve 
973
 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve 
961
   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
974
   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
962
   These changes also provide performance improvements for call setup and tear down.
975
   These changes also provide performance improvements for call setup and tear down.
963
 * Added ability to specify registration expiry time on a per registration basis in
976
 * Added ability to specify registration expiry time on a per registration basis in
964
   the register line.
977
   the register line.
965
 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
978
 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
966
   lost packets.
979
   lost packets.
967
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
980
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
968
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
981
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
969
 * 'sip show peers' and 'sip show users' display their entries sorted in
982
 * 'sip show peers' and 'sip show users' display their entries sorted in
970
    alphabetical order, as opposed to the order they were in, in the config 
983
    alphabetical order, as opposed to the order they were in, in the config 
971
    file or database. 
984
    file or database. 
972
 * Videosupport now supports an additional option, "always", which always sets
985
 * Videosupport now supports an additional option, "always", which always sets
973
    up video RTP ports, even on clients that don't support it.  This helps with
986
    up video RTP ports, even on clients that don't support it.  This helps with
974
    callfiles and certain transfers to ensure that if two video phones are
987
    callfiles and certain transfers to ensure that if two video phones are
975
    connected, they will always share video feeds.
988
    connected, they will always share video feeds.
976

    
   
989

   
977
IAX Changes
990
IAX Changes
978
-----------
991
-----------
979
 * Existing DNS manager lookups extended to check for SRV records.
992
 * Existing DNS manager lookups extended to check for SRV records.
980
 * IAX2 encryption support has been improved to support periodic key rotation
993
 * IAX2 encryption support has been improved to support periodic key rotation
981
   within a call for enhanced security.  The option "keyrotate" has been
994
   within a call for enhanced security.  The option "keyrotate" has been
982
   provided to disable this functionality to preserve backwards compatibility
995
   provided to disable this functionality to preserve backwards compatibility
983
   with older versions of IAX2 that do not support key rotation.
996
   with older versions of IAX2 that do not support key rotation.
984

    
   
997

   
985
CLI Changes
998
CLI Changes
986
-----------
999
-----------
987
  * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1000
  * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
988
     data tree based on the given <path>.
1001
     data tree based on the given <path>.
989
  * New CLI command "data show providers" that will display all the registered
1002
  * New CLI command "data show providers" that will display all the registered
990
     callbacks.
1003
     callbacks.
991
  * New CLI command, "config reload <file.conf>" which reloads any module that
1004
  * New CLI command, "config reload <file.conf>" which reloads any module that
992
     references that particular configuration file.  Also added "config list"
1005
     references that particular configuration file.  Also added "config list"
993
     which shows which configuration files are in use.
1006
     which shows which configuration files are in use.
994
  * New CLI commands, "pri show version" and "ss7 show version" that will
1007
  * New CLI commands, "pri show version" and "ss7 show version" that will
995
     display which version of libpri and libss7 are being used, respectively.
1008
     display which version of libpri and libss7 are being used, respectively.
996
     A new API call was added so trunk will now have to be compiled against
1009
     A new API call was added so trunk will now have to be compiled against
997
     a versions of libpri and libss7 that have them or it will not know that
1010
     a versions of libpri and libss7 that have them or it will not know that
998
     these libraries exist.
1011
     these libraries exist.
999
  * The commands "core show globals", "core set global" and "core set chanvar" has
1012
  * The commands "core show globals", "core set global" and "core set chanvar" has
1000
     been deprecated in favor of the more semanticly correct "dialplan show globals",
1013
     been deprecated in favor of the more semanticly correct "dialplan show globals",
1001
     "dialplan set chanvar" and "dialplan set global".
1014
     "dialplan set chanvar" and "dialplan set global".
1002
  * New CLI command "dialplan show chanvar" to list all variables associated
1015
  * New CLI command "dialplan show chanvar" to list all variables associated
1003
    with a given channel.
1016
    with a given channel.
1004

    
   
1017

   
1005
DNS manager changes
1018
DNS manager changes
1006
-------------------
1019
-------------------
1007
  * Addresses managed by DNS manager now can check to see if there is a DNS
1020
  * Addresses managed by DNS manager now can check to see if there is a DNS
1008
    SRV record for a given domain and will use that hostname/port if present.
1021
    SRV record for a given domain and will use that hostname/port if present.
1009

    
   
1022

   
1010
AMI - The manager (TCP/TLS/HTTP)
1023
AMI - The manager (TCP/TLS/HTTP)
1011
--------------------------------
1024
--------------------------------
1012
  * The Status command now takes an optional list of variables to display
1025
  * The Status command now takes an optional list of variables to display
1013
    along with channel status.
1026
    along with channel status.
1014
  * The QueueEntry event now also includes the channel's uniqueid
1027
  * The QueueEntry event now also includes the channel's uniqueid
1015

    
   
1028

   
1016
ODBC Changes
1029
ODBC Changes
1017
------------
1030
------------
1018
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
1031
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
1019
    as some people were running into this limit.  This limit has been increased
1032
    as some people were running into this limit.  This limit has been increased
1020
    to 4.2 billion.
1033
    to 4.2 billion.
1021

    
   
1034

   
1022
Queue changes
1035
Queue changes
1023
-------------
1036
-------------
1024
  * The TRANSFER queue log entry now includes the the caller's original
1037
  * The TRANSFER queue log entry now includes the the caller's original
1025
    position in the transferred-from queue.
1038
    position in the transferred-from queue.
1026
  * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1039
  * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1027
    "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1040
    "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1028
    as well as an explanation about timeout options in general
1041
    as well as an explanation about timeout options in general
1029
  * Added a new option - C - for forcing the "answered elsewhere" flag on
1042
  * Added a new option - C - for forcing the "answered elsewhere" flag on
1030
    cancellation of calls in to members of the queue. This is to avoid the
1043
    cancellation of calls in to members of the queue. This is to avoid the
1031
    call to a member of a queue having the call listed as a "missed call".
1044
    call to a member of a queue having the call listed as a "missed call".
1032

    
   
1045

   
1033
Realtime changes
1046
Realtime changes
1034
----------------
1047
----------------
1035
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1048
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1036
    adaptive capabilities.  What this means in practical terms is that if your
1049
    adaptive capabilities.  What this means in practical terms is that if your
1037
    realtime table lacks critical fields, Asterisk will now emit warnings to
1050
    realtime table lacks critical fields, Asterisk will now emit warnings to
1038
    that effect.  Also, some of the realtime drivers have the ability (if
1051
    that effect.  Also, some of the realtime drivers have the ability (if
1039
    configured) to automatically add those columns to the table with the
1052
    configured) to automatically add those columns to the table with the
1040
    correct type and length.
1053
    correct type and length.
1041

    
   
1054

   
1042
Miscellaneous
1055
Miscellaneous
1043
-------------
1056
-------------
1044
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1057
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1045
    the 'setvar' option to cause a given audio file to be played upon completion
1058
    the 'setvar' option to cause a given audio file to be played upon completion
1046
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
1059
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
1047
    Skinny channels only.
1060
    Skinny channels only.
1048
  * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
1061
  * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
1049
    for more information.
1062
    for more information.
1050
  * Config file variables may now be appended to, by using the '+=' append
1063
  * Config file variables may now be appended to, by using the '+=' append
1051
    operator.  This is most helpful when working with long SQL queries in
1064
    operator.  This is most helpful when working with long SQL queries in
1052
    func_odbc.conf, as the queries no longer need to be specified on a single
1065
    func_odbc.conf, as the queries no longer need to be specified on a single
1053
    line.
1066
    line.
1054
  * CDR config file, cdr.conf, has an added option, "initiatedseconds", 
1067
  * CDR config file, cdr.conf, has an added option, "initiatedseconds", 
1055
    which will add a second to the billsec when the ending
1068
    which will add a second to the billsec when the ending
1056
    time is set, if the number in the microseconds field of the end time is 
1069
    time is set, if the number in the microseconds field of the end time is 
1057
    greater than the number of microseconds in the answer time. This allows
1070
    greater than the number of microseconds in the answer time. This allows
1058
    users to count the 'initiated' seconds in their billing records. 
1071
    users to count the 'initiated' seconds in their billing records. 
1059

    
   
1072

   
1060
------------------------------------------------------------------------------
1073
------------------------------------------------------------------------------
1061
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
1074
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
1062
------------------------------------------------------------------------------
1075
------------------------------------------------------------------------------
1063

    
   
1076

   
1064
AMI - The manager (TCP/TLS/HTTP)
1077
AMI - The manager (TCP/TLS/HTTP)
1065
--------------------------------
1078
--------------------------------
1066
  * Manager has undergone a lot of changes, all of them documented
1079
  * Manager has undergone a lot of changes, all of them documented
1067
    in doc/manager_1_1.txt
1080
    in doc/manager_1_1.txt
1068
  * Manager version has changed to 1.1
1081
  * Manager version has changed to 1.1
1069
  * Added a new action 'CoreShowChannels' to list currently defined channels
1082
  * Added a new action 'CoreShowChannels' to list currently defined channels
1070
     and some information about them. 
1083
     and some information about them. 
1071
  * Added a new action 'SIPshowregistry' to list SIP registrations.
1084
  * Added a new action 'SIPshowregistry' to list SIP registrations.
1072
  * Added TLS support for the manager interface and HTTP server
1085
  * Added TLS support for the manager interface and HTTP server
1073
  * Added the URI redirect option for the built-in HTTP server
1086
  * Added the URI redirect option for the built-in HTTP server
1074
  * The output of CallerID in Manager events is now more consistent.
1087
  * The output of CallerID in Manager events is now more consistent.
1075
     CallerIDNum is used for number and CallerIDName for name.
1088
     CallerIDNum is used for number and CallerIDName for name.
1076
  * Enable https support for builtin web server.
1089
  * Enable https support for builtin web server.
1077
     See configs/http.conf.sample for details.
1090
     See configs/http.conf.sample for details.
1078
  * Added a new action, GetConfigJSON, which can return the contents of an
1091
  * Added a new action, GetConfigJSON, which can return the contents of an
1079
     Asterisk configuration file in JSON format.  This is intended to help
1092
     Asterisk configuration file in JSON format.  This is intended to help
1080
     improve the performance of AJAX applications using the manager interface
1093
     improve the performance of AJAX applications using the manager interface
1081
     over HTTP.
1094
     over HTTP.
1082
  * SIP and IAX manager events now use "ChannelType" in all cases where we 
1095
  * SIP and IAX manager events now use "ChannelType" in all cases where we 
1083
     indicate channel driver. Previously, we used a mixture of "Channel"
1096
     indicate channel driver. Previously, we used a mixture of "Channel"
1084
     and "ChannelDriver" headers.
1097
     and "ChannelDriver" headers.
1085
  * Added a "Bridge" action which allows you to bridge any two channels that
1098
  * Added a "Bridge" action which allows you to bridge any two channels that
1086
     are currently active on the system.
1099
     are currently active on the system.
1087
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1100
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1088
     the voicemail users setup.
1101
     the voicemail users setup.
1089
  * Added 'DBDel' and 'DBDelTree' manager commands.
1102
  * Added 'DBDel' and 'DBDelTree' manager commands.
1090
  * cdr_manager now reports events via the "cdr" level, separating it from
1103
  * cdr_manager now reports events via the "cdr" level, separating it from
1091
     the very verbose "call" level.
1104
     the very verbose "call" level.
1092
  * Manager users are now stored in memory. If you change the manager account
1105
  * Manager users are now stored in memory. If you change the manager account
1093
    list (delete or add accounts) you need to reload manager.
1106
    list (delete or add accounts) you need to reload manager.
1094
  * Added Masquerade manager event for when a masquerade happens between
1107
  * Added Masquerade manager event for when a masquerade happens between
1095
     two channels.
1108
     two channels.
1096
  * Added "manager reload" command for the CLI
1109
  * Added "manager reload" command for the CLI
1097
  * Lots of commands that only provided information are now allowed under the
1110
  * Lots of commands that only provided information are now allowed under the
1098
     Reporting privilege, instead of only under Call or System.
1111
     Reporting privilege, instead of only under Call or System.
1099
  * The IAX* commands now require either System or Reporting privilege, to
1112
  * The IAX* commands now require either System or Reporting privilege, to
1100
     mirror the privileges of the SIP* commands.
1113
     mirror the privileges of the SIP* commands.
1101
  * Added ability to retrieve list of categories in a config file.
1114
  * Added ability to retrieve list of categories in a config file.
1102
  * Added ability to retrieve the content of a particular category.
1115
  * Added ability to retrieve the content of a particular category.
1103
  * Added ability to empty a context.
1116
  * Added ability to empty a context.
1104
  * Created new action to create a new file.
1117
  * Created new action to create a new file.
1105
  * Updated delete action to allow deletion by line number with respect to category.
1118
  * Updated delete action to allow deletion by line number with respect to category.
1106
  * Added new action insert to add new variable to category at specified line.
1119
  * Added new action insert to add new variable to category at specified line.
1107
  * Updated action newcat to allow new category to be inserted in file above another
1120
  * Updated action newcat to allow new category to be inserted in file above another
1108
    existing category.
1121
    existing category.
1109
  * Added new event "JitterBufStats" in the IAX2 channel
1122
  * Added new event "JitterBufStats" in the IAX2 channel
1110
  * Originate now requires the Originate privilege and, if you want to call out
1123
  * Originate now requires the Originate privilege and, if you want to call out
1111
    to a subshell, it requires the System privilege, as well.  This was done to
1124
    to a subshell, it requires the System privilege, as well.  This was done to
1112
    enhance manager security.
1125
    enhance manager security.
1113
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" 
1126
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" 
1114
  * New command: Atxfer. See doc/manager_1_1.txt for more details or 
1127
  * New command: Atxfer. See doc/manager_1_1.txt for more details or 
1115
    manager show command Atxfer from the CLI
1128
    manager show command Atxfer from the CLI
1116
  * New command: IAXregistry. See doc/manager_1_1.txt for more details or
1129
  * New command: IAXregistry. See doc/manager_1_1.txt for more details or
1117
    manager show command IAXregistry from the CLI
1130
    manager show command IAXregistry from the CLI
1118

    
   
1131

   
1119
Dialplan functions
1132
Dialplan functions
1120
------------------
1133
------------------
1121
  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1134
  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1122
     state in the dialplan, as well as creating custom device states that are
1135
     state in the dialplan, as well as creating custom device states that are
1123
     controllable from the dialplan.
1136
     controllable from the dialplan.
1124
  * Extend CALLERID() function with "pres" and "ton" parameters to
1137
  * Extend CALLERID() function with "pres" and "ton" parameters to
1125
     fetch string representation of calling number presentation indicator
1138
     fetch string representation of calling number presentation indicator
1126
     and numeric representation of type of calling number value.
1139
     and numeric representation of type of calling number value.
1127
  * MailboxExists converted to dialplan function
1140
  * MailboxExists converted to dialplan function
1128
  * A new option to Dial() for telling IP phones not to count the call
1141
  * A new option to Dial() for telling IP phones not to count the call
1129
     as "missed" when dial times out and cancels.
1142
     as "missed" when dial times out and cancels.
1130
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1143
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1131
     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
1144
     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
1132
     held for any given channel.  Also, locks are automatically freed when a
1145
     held for any given channel.  Also, locks are automatically freed when a
1133
     channel is hung up.
1146
     channel is hung up.
1134
  * Added HINT() dialplan function that allows retrieving hint information.
1147
  * Added HINT() dialplan function that allows retrieving hint information.
1135
     Hints are mappings between extensions and devices for the sake of 
1148
     Hints are mappings between extensions and devices for the sake of 
1136
     determining the state of an extension.  This function can retrieve the list
1149
     determining the state of an extension.  This function can retrieve the list
1137
     of devices or the name associated with a hint.
1150
     of devices or the name associated with a hint.
1138
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1151
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1139
    of any extension.
1152
    of any extension.
1140
  * Added SYSINFO() dialplan function which allows retrieval of system information
1153
  * Added SYSINFO() dialplan function which allows retrieval of system information
1141
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1154
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1142
     the existence of a dialplan target.
1155
     the existence of a dialplan target.
1143
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1156
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1144
     upper and lower case, respectively.
1157
     upper and lower case, respectively.
1145
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1158
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1146
     ID for the call (not the Asterisk call ID or unique ID), provided that the
1159
     ID for the call (not the Asterisk call ID or unique ID), provided that the
1147
     channel driver supports this. For SIP, you get the SIP call-ID for the
1160
     channel driver supports this. For SIP, you get the SIP call-ID for the
1148
     bridged channel which you can store in the CDR with a custom field.
1161
     bridged channel which you can store in the CDR with a custom field.
1149

    
   
1162

   
1150
CLI Changes
1163
CLI Changes
1151
-----------
1164
-----------
1152
  * Added CLI permissions, config file: cli_permissions.conf
1165
  * Added CLI permissions, config file: cli_permissions.conf
1153
     default is to allow all commands for every local user/group.
1166
     default is to allow all commands for every local user/group.
1154
     Also this new feature added three new CLI commands:
1167
     Also this new feature added three new CLI commands:
1155
      - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1168
      - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1156
      - cli reload permissions
1169
      - cli reload permissions
1157
      - cli show permissions
1170
      - cli show permissions
1158
  * New CLI command "core show hint" (usage: core show hint <exten>)
1171
  * New CLI command "core show hint" (usage: core show hint <exten>)
1159
  * New CLI command "core show settings"
1172
  * New CLI command "core show settings"
1160
  * Added 'core show channels count' CLI command.
1173
  * Added 'core show channels count' CLI command.
1161
  * Added the ability to set the core debug and verbose values on a per-file basis.
1174
  * Added the ability to set the core debug and verbose values on a per-file basis.
1162
  * Added 'queue pause member' and 'queue unpause member' CLI commands
1175
  * Added 'queue pause member' and 'queue unpause member' CLI commands
1163
  * Ability to set process limits ("ulimit") without restarting Asterisk
1176
  * Ability to set process limits ("ulimit") without restarting Asterisk
1164
  * Enhanced "agi debug" to print the channel name as a prefix to the debug
1177
  * Enhanced "agi debug" to print the channel name as a prefix to the debug
1165
     output to make debugging on busy systems much easier.
1178
     output to make debugging on busy systems much easier.
1166
  * New CLI commands "dialplan set extenpatternmatching true/false"
1179
  * New CLI commands "dialplan set extenpatternmatching true/false"
1167
  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1180
  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1168
  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
1181
  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
1169
    listed in the startup_commands section of cli.conf will get executed.
1182
    listed in the startup_commands section of cli.conf will get executed.
1170
  * Added a CLI command, "devstate change", which allows you to set custom device
1183
  * Added a CLI command, "devstate change", which allows you to set custom device
1171
     states from the func_devstate module that provides the DEVICE_STATE() function
1184
     states from the func_devstate module that provides the DEVICE_STATE() function
1172
     and handling of the "Custom:" devices.
1185
     and handling of the "Custom:" devices.
1173
  * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1186
  * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1174
    sorted into the different possible callbacks, with the number of entries
1187
    sorted into the different possible callbacks, with the number of entries
1175
    currently scheduled for each. Gives you a feel for how busy the sip channel
1188
    currently scheduled for each. Gives you a feel for how busy the sip channel
1176
    driver is.
1189
    driver is.
1177
  * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1190
  * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1178
  * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1191
  * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1179
    (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1192
    (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1180

    
   
1193

   
1181
SIP changes
1194
SIP changes
1182
-----------
1195
-----------
1183
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
1196
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
1184
    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1197
    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1185
    for a received call.  If it is detected, the channel will jump to the 
1198
    for a received call.  If it is detected, the channel will jump to the 
1186
    'fax' extension in the dialplan.
1199
    'fax' extension in the dialplan.
1187
  * The default SIP useragent= identifier now includes the Asterisk version
1200
  * The default SIP useragent= identifier now includes the Asterisk version
1188
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1201
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1189
     If set, and the incoming request carries authentication info,
1202
     If set, and the incoming request carries authentication info,
1190
     the username to match in the users list is taken from the Digest header
1203
     the username to match in the users list is taken from the Digest header
1191
     rather than from the From: field. This feature is considered experimental.
1204
     rather than from the From: field. This feature is considered experimental.
1192
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1205
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1193
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1206
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1194
  * The "localmask" setting was removed in version 1.2 and the reminder about it
1207
  * The "localmask" setting was removed in version 1.2 and the reminder about it
1195
     being removed is now also removed.
1208
     being removed is now also removed.
1196
  * A new option "busylevel" for setting a level of calls where asterisk reports
1209
  * A new option "busylevel" for setting a level of calls where asterisk reports
1197
     a device as busy, to separate it from call-limit. This value is also added
1210
     a device as busy, to separate it from call-limit. This value is also added
1198
     to the SIP_PEER dialplan function.
1211
     to the SIP_PEER dialplan function.
1199
  * A new realtime family called "sipregs" is now supported to store SIP registration
1212
  * A new realtime family called "sipregs" is now supported to store SIP registration
1200
     data. If this family is defined, "sippeers" will be used for configuration and
1213
     data. If this family is defined, "sippeers" will be used for configuration and
1201
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1214
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1202
     registration data, as before.
1215
     registration data, as before.
1203
  * The SIPPEER function have new options for port address, call and pickup groups
1216
  * The SIPPEER function have new options for port address, call and pickup groups
1204
  * Added support for T.140 realtime text in SIP/RTP
1217
  * Added support for T.140 realtime text in SIP/RTP
1205
  * The "checkmwi" option has been removed from sip.conf, as it is no longer
1218
  * The "checkmwi" option has been removed from sip.conf, as it is no longer
1206
     required due to the restructuring of how MWI is handled.  See the descriptions 
1219
     required due to the restructuring of how MWI is handled.  See the descriptions 
1207
     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
1220
     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
1208
     for more information.
1221
     for more information.
1209
  * Added rtpdest option to CHANNEL() dialplan function.
1222
  * Added rtpdest option to CHANNEL() dialplan function.
1210
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1223
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1211
  * SIP now adds a header to the CANCEL if the call was answered by another phone
1224
  * SIP now adds a header to the CANCEL if the call was answered by another phone
1212
     in the same dial command, or if the new c option in dial() is used.
1225
     in the same dial command, or if the new c option in dial() is used.
1213
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1226
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1214
     states it is not needed. For phones, however, that do require it the "registertrying" option
1227
     states it is not needed. For phones, however, that do require it the "registertrying" option
1215
     has been added so it can be enabled. 
1228
     has been added so it can be enabled. 
1216
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
1229
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
1217
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1230
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1218
     used to enable this functionality).
1231
     used to enable this functionality).
1219
  * New settings for timer T1 and timer B on a global level or per device. This makes it 
1232
  * New settings for timer T1 and timer B on a global level or per device. This makes it 
1220
     possible to force timeout faster on non-responsive SIP servers. These settings are
1233
     possible to force timeout faster on non-responsive SIP servers. These settings are
1221
     considered advanced, so don't use them unless you have a problem.
1234
     considered advanced, so don't use them unless you have a problem.
1222
  * Added a dial string option to be able to set the To: header in an INVITE to any
1235
  * Added a dial string option to be able to set the To: header in an INVITE to any
1223
     SIP uri.
1236
     SIP uri.
1224
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1237
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1225
     the qualify frequency.
1238
     the qualify frequency.
1226
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
1239
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
1227
     were not properly torn down due to network or endpoint failures during an established
1240
     were not properly torn down due to network or endpoint failures during an established
1228
     SIP session.
1241
     SIP session.
1229
  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
1242
  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
1230
     configs/sip.conf.sample for more information on how it is used.
1243
     configs/sip.conf.sample for more information on how it is used.
1231
  * Added a new configuration option "authfailureevents" that enables manager events when
1244
  * Added a new configuration option "authfailureevents" that enables manager events when
1232
    a peer can't authenticate properly. 
1245
    a peer can't authenticate properly. 
1233
  * Added DNS manager support to registrations for peers not referencing a peer entry.
1246
  * Added DNS manager support to registrations for peers not referencing a peer entry.
1234

    
   
1247

   
1235
IAX2 changes
1248
IAX2 changes
1236
------------
1249
------------
1237
  * Added the trunkmaxsize configuration option to chan_iax2.
1250
  * Added the trunkmaxsize configuration option to chan_iax2.
1238
  * Added the srvlookup option to iax.conf
1251
  * Added the srvlookup option to iax.conf
1239
  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
1252
  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
1240
     dialplan function.
1253
     dialplan function.
1241

    
   
1254

   
1242
XMPP Google Talk/Jingle changes
1255
XMPP Google Talk/Jingle changes
1243
-------------------------------
1256
-------------------------------
1244
  * Added the bindaddr option to gtalk.conf.
1257
  * Added the bindaddr option to gtalk.conf.
1245

    
   
1258

   
1246
Skinny changes
1259
Skinny changes
1247
-------------
1260
-------------
1248
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1261
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1249
  * Proper codec support in chan_skinny.
1262
  * Proper codec support in chan_skinny.
1250
  * Added settings for IP and Ethernet QoS requests
1263
  * Added settings for IP and Ethernet QoS requests
1251

    
   
1264

   
1252
MGCP changes
1265
MGCP changes
1253
------------
1266
------------
1254
  * Added separate settings for media QoS in mgcp.conf
1267
  * Added separate settings for media QoS in mgcp.conf
1255

    
   
1268

   
1256
Console Channel Driver changes
1269
Console Channel Driver changes
1257
------------------------------
1270
------------------------------
1258
  * Added experimental support for video send & receive to chan_oss.
1271
  * Added experimental support for video send & receive to chan_oss.
1259
    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1272
    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1260
    a video source.
1273
    a video source.
1261

    
   
1274

   
1262
Phone channel changes (chan_phone)
1275
Phone channel changes (chan_phone)
1263
----------------------------------
1276
----------------------------------
1264
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1277
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1265

    
   
1278

   
1266
H.323 channel Changes
1279
H.323 channel Changes
1267
---------------------
1280
---------------------
1268
  * H323 remote hold notification support added (by NOTIFY message
1281
  * H323 remote hold notification support added (by NOTIFY message
1269
     and/or H.450 supplementary service)
1282
     and/or H.450 supplementary service)
1270

    
   
1283

   
1271
Local channel changes
1284
Local channel changes
1272
---------------------
1285
---------------------
1273
  * The device state functionality in the Local channel driver has been updated
1286
  * The device state functionality in the Local channel driver has been updated
1274
     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1287
     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1275
     to just UNKNOWN if the extension exists.
1288
     to just UNKNOWN if the extension exists.
1276
  * Added jitterbuffer support for chan_local.  This allows you to use the
1289
  * Added jitterbuffer support for chan_local.  This allows you to use the
1277
     generic jitterbuffer on incoming calls going to Asterisk applications.
1290
     generic jitterbuffer on incoming calls going to Asterisk applications.
1278
     For example, this would allow you to use a jitterbuffer for an incoming
1291
     For example, this would allow you to use a jitterbuffer for an incoming
1279
     SIP call to Voicemail by putting a Local channel in the middle.  This
1292
     SIP call to Voicemail by putting a Local channel in the middle.  This
1280
     feature is enabled by using the 'j' option in the Dial string to the Local
1293
     feature is enabled by using the 'j' option in the Dial string to the Local
1281
     channel in conjunction with the existing 'n' option for local channels.
1294
     channel in conjunction with the existing 'n' option for local channels.
1282
  * A 'b' option has been added which causes chan_local to return the actual channel
1295
  * A 'b' option has been added which causes chan_local to return the actual channel
1283
     that is behind it when queried. This is useful for transfer scenarios as the
1296
     that is behind it when queried. This is useful for transfer scenarios as the
1284
     actual channel will be transferred, not the Local channel.
1297
     actual channel will be transferred, not the Local channel.
1285

    
   
1298

   
1286
Agent channel changes
1299
Agent channel changes
1287
----------------------
1300
----------------------
1288
  * The ackcall and endcall options are now supplemented with options acceptdtmf
1301
  * The ackcall and endcall options are now supplemented with options acceptdtmf
1289
    and enddtmf. These allow for the DTMF keypress to be configurable. The options
1302
    and enddtmf. These allow for the DTMF keypress to be configurable. The options
1290
    default to their old hard-coded values ('#' and '*' respectively) so this should
1303
    default to their old hard-coded values ('#' and '*' respectively) so this should
1291
    not break any existing agent installations.
1304
    not break any existing agent installations.
1292

    
   
1305

   
1293
DAHDI channel driver (chan_dahdi) Changes
1306
DAHDI channel driver (chan_dahdi) Changes
1294
----------------------------------------
1307
----------------------------------------
1295
  * SS7 support (via libss7 library)
1308
  * SS7 support (via libss7 library)
1296
  * In India, some carriers transmit CID via dtmf. Some code has been added
1309
  * In India, some carriers transmit CID via dtmf. Some code has been added
1297
     that will handle some situations. The cidstart=polarity_IN choice has been added for
1310
     that will handle some situations. The cidstart=polarity_IN choice has been added for
1298
     those carriers that transmit CID via dtmf after a polarity change.
1311
     those carriers that transmit CID via dtmf after a polarity change.
1299
  * CID matching information is now shown when doing 'dialplan show'.
1312
  * CID matching information is now shown when doing 'dialplan show'.
1300
  * Added dahdi show version CLI command.
1313
  * Added dahdi show version CLI command.
1301
  * Added setvar support to chan_dahdi.conf channel entries.
1314
  * Added setvar support to chan_dahdi.conf channel entries.
1302
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
1315
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
1303
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
1316
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
1304
     the script specified in the mwimonitornotify option is executed.  An internal
1317
     the script specified in the mwimonitornotify option is executed.  An internal
1305
     event indicating the new state of the mailbox is also generated, so that
1318
     event indicating the new state of the mailbox is also generated, so that
1306
     the normal MWI facilities in Asterisk work as usual.
1319
     the normal MWI facilities in Asterisk work as usual.
1307
  * Added signalling type 'auto', which attempts to use the same signalling type
1320
  * Added signalling type 'auto', which attempts to use the same signalling type
1308
     for a channel as configured in DAHDI. This is primarily designed for analog
1321
     for a channel as configured in DAHDI. This is primarily designed for analog
1309
     ports, but will also work for digital ports that are configured for FXS or FXO
1322
     ports, but will also work for digital ports that are configured for FXS or FXO
1310
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
1323
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
1311
     does not specify signalling for a channel (which is unlikely as the sample
1324
     does not specify signalling for a channel (which is unlikely as the sample
1312
     configuration file has always recommended specifying it for every channel) then
1325
     configuration file has always recommended specifying it for every channel) then
1313
     the 'auto' mode will be used for that channel if possible.
1326
     the 'auto' mode will be used for that channel if possible.
1314
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1327
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1315
     state for a channel; also ensured that the DNDState Manager event is
1328
     state for a channel; also ensured that the DNDState Manager event is
1316
     emitted no matter how the DND state is set or cleared.
1329
     emitted no matter how the DND state is set or cleared.
1317

    
   
1330

   
1318
New Channel Drivers
1331
New Channel Drivers
1319
-------------------
1332
-------------------
1320
  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
1333
  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
1321
     configs/unistim.conf.sample for details.  This new channel driver allows
1334
     configs/unistim.conf.sample for details.  This new channel driver allows
1322
     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1335
     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1323
  * Added a new channel driver, chan_console, which uses portaudio as a cross
1336
  * Added a new channel driver, chan_console, which uses portaudio as a cross
1324
     platform audio interface.  It was written as a channel driver that would
1337
     platform audio interface.  It was written as a channel driver that would
1325
     work with Mac CoreAudio, but portaudio supports a number of other audio
1338
     work with Mac CoreAudio, but portaudio supports a number of other audio
1326
     interfaces, as well. Note that this channel driver requires v19 or higher
1339
     interfaces, as well. Note that this channel driver requires v19 or higher
1327
     of portaudio; older versions have a different API.
1340
     of portaudio; older versions have a different API.
1328
 
1341
 
1329
DUNDi changes
1342
DUNDi changes
1330
-------------
1343
-------------
1331
  * Added the ability to specify arguments to the Dial application when using
1344
  * Added the ability to specify arguments to the Dial application when using
1332
     the DUNDi switch in the dialplan.
1345
     the DUNDi switch in the dialplan.
1333
  * Added the ability to set weights for responses dynamically.  This can be
1346
  * Added the ability to set weights for responses dynamically.  This can be
1334
     done using a global variable or a dialplan function.  Using the SHELL()
1347
     done using a global variable or a dialplan function.  Using the SHELL()
1335
     function would allow you to have an external script set the weight for
1348
     function would allow you to have an external script set the weight for
1336
     each response.
1349
     each response.
1337
  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
1350
  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
1338
     functions will allow you to initiate a DUNDi query from the dialplan,
1351
     functions will allow you to initiate a DUNDi query from the dialplan,
1339
     find out how many results there are, and access each one.
1352
     find out how many results there are, and access each one.
1340
  * Added the ability to specifiy a port for a dundi peer.
1353
  * Added the ability to specifiy a port for a dundi peer.
1341

    
   
1354

   
1342
ENUM changes
1355
ENUM changes
1343
------------
1356
------------
1344
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
1357
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
1345
     functions will allow you to initiate an ENUM lookup from the dialplan,
1358
     functions will allow you to initiate an ENUM lookup from the dialplan,
1346
     and Asterisk will cache the results.  ENUMRESULT can be used to access
1359
     and Asterisk will cache the results.  ENUMRESULT can be used to access
1347
     the results without doing multiple DNS queries.
1360
     the results without doing multiple DNS queries.
1348

    
   
1361

   
1349
Voicemail Changes
1362
Voicemail Changes
1350
-----------------
1363
-----------------
1351
  * Added the ability to customize which sound files are used for some of the
1364
  * Added the ability to customize which sound files are used for some of the
1352
     prompts within the Voicemail application by changing them in voicemail.conf
1365
     prompts within the Voicemail application by changing them in voicemail.conf
1353
  * Added the ability for the "voicemail show users" CLI command to show users
1366
  * Added the ability for the "voicemail show users" CLI command to show users
1354
     configured by the dynamic realtime configuration method.
1367
     configured by the dynamic realtime configuration method.
1355
  * MWI (Message Waiting Indication) handling has been significantly
1368
  * MWI (Message Waiting Indication) handling has been significantly
1356
     restructured internally to Asterisk.  It is now totally event based
1369
     restructured internally to Asterisk.  It is now totally event based
1357
     instead of polling based.  The voicemail application will notify other
1370
     instead of polling based.  The voicemail application will notify other
1358
     modules that have subscribed to MWI events when something in the mailbox
1371
     modules that have subscribed to MWI events when something in the mailbox
1359
     changes.
1372
     changes.
1360
    This also means that if any other entity outside of Asterisk is changing
1373
    This also means that if any other entity outside of Asterisk is changing
1361
     the contents of mailboxes, then the voicemail application still needs to
1374
     the contents of mailboxes, then the voicemail application still needs to
1362
     poll for changes.  Examples of situations that would require this option
1375
     poll for changes.  Examples of situations that would require this option
1363
     are web interfaces to voicemail or an email client in the case of using
1376
     are web interfaces to voicemail or an email client in the case of using
1364
     IMAP storage.  So, two new options have been added to voicemail.conf
1377
     IMAP storage.  So, two new options have been added to voicemail.conf
1365
     to account for this: "pollmailboxes" and "pollfreq".  See the sample
1378
     to account for this: "pollmailboxes" and "pollfreq".  See the sample
1366
     configuration file for details.
1379
     configuration file for details.
1367
  * Added "tw" language support
1380
  * Added "tw" language support
1368
  * Added support for storage of greetings using an IMAP server
1381
  * Added support for storage of greetings using an IMAP server
1369
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
1382
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
1370
  * SMDI is now enabled in voicemail using the smdienable option.
1383
  * SMDI is now enabled in voicemail using the smdienable option.
1371
  * A "lockmode" option has been added to asterisk.conf to configure the file
1384
  * A "lockmode" option has been added to asterisk.conf to configure the file
1372
     locking method used for voicemail, and potentially other things in the
1385
     locking method used for voicemail, and potentially other things in the
1373
     future.  The default is the old behavior, lockfile.  However, there is a
1386
     future.  The default is the old behavior, lockfile.  However, there is a
1374
     new method, "flock", that uses a different method for situations where the
1387
     new method, "flock", that uses a different method for situations where the
1375
     lockfile will not work, such as on SMB/CIFS mounts.
1388
     lockfile will not work, such as on SMB/CIFS mounts.
1376
  * Added the ability to backup deleted messages, to ease recovery in the case
1389
  * Added the ability to backup deleted messages, to ease recovery in the case
1377
     that a user accidentally deletes a message, and discovers that they need it.
1390
     that a user accidentally deletes a message, and discovers that they need it.
1378
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
1391
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
1379
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
1392
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
1380
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1393
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1381
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
1394
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
1382
     outside entity is modifying the state of the mailbox (such as IMAP storage or
1395
     outside entity is modifying the state of the mailbox (such as IMAP storage or
1383
     a web interface of some kind).
1396
     a web interface of some kind).
1384
  * Added the support for marking messages as "urgent." There are two methods to accomplish
1397
  * Added the support for marking messages as "urgent." There are two methods to accomplish
1385
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1398
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1386
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1399
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1387
     the message as urgent after he has recorded a voicemail by following the voice instructions.
1400
     the message as urgent after he has recorded a voicemail by following the voice instructions.
1388
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1401
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1389
     messages
1402
     messages
1390

    
   
1403

   
1391
Queue changes
1404
Queue changes
1392
-------------
1405
-------------
1393
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1406
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1394
     used across multiple queues.
1407
     used across multiple queues.
1395
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
1408
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
1396
     setqueueentryvar options for each queue, see queues.conf.sample for details.
1409
     setqueueentryvar options for each queue, see queues.conf.sample for details.
1397
  * Added keepstats option to queues.conf which will keep queue
1410
  * Added keepstats option to queues.conf which will keep queue
1398
     statistics during a reload.
1411
     statistics during a reload.
1399
  * setinterfacevar option in queues.conf also now sets a variable
1412
  * setinterfacevar option in queues.conf also now sets a variable
1400
     called MEMBERNAME which contains the member's name.
1413
     called MEMBERNAME which contains the member's name.
1401
  * Added 'Strategy' field to manager event QueueParams which represents
1414
  * Added 'Strategy' field to manager event QueueParams which represents
1402
     the queue strategy in use. 
1415
     the queue strategy in use. 
1403
  * Added option to run macro when a queue member is connected to a caller, 
1416
  * Added option to run macro when a queue member is connected to a caller, 
1404
     see queues.conf.sample for details.
1417
     see queues.conf.sample for details.
1405
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1418
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1406
     does not count paused queue members as unavailable.
1419
     does not count paused queue members as unavailable.
1407
  * Added min-announce-frequency option to queues.conf which allows you to control the
1420
  * Added min-announce-frequency option to queues.conf which allows you to control the
1408
     minimum amount of time between queue announcements for use when the caller's queue
1421
     minimum amount of time between queue announcements for use when the caller's queue
1409
     position changes frequently.
1422
     position changes frequently.
1410
  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1423
  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1411
     queue log.
1424
     queue log.
1412
  * Added ability for non-realtime queues to have realtime members
1425
  * Added ability for non-realtime queues to have realtime members
1413
  * Added the "linear" strategy to queues.
1426
  * Added the "linear" strategy to queues.
1414
  * Added the "wrandom" strategy to queues.
1427
  * Added the "wrandom" strategy to queues.
1415
  * Added new channel variable QUEUE_MIN_PENALTY
1428
  * Added new channel variable QUEUE_MIN_PENALTY
1416
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1429
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1417
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
1430
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
1418
  * Added a new parameter for member definition, called state_interface. This may be
1431
  * Added a new parameter for member definition, called state_interface. This may be
1419
    used so that a member may be called via one interface but have a different interface's
1432
    used so that a member may be called via one interface but have a different interface's
1420
    device state reported.
1433
    device state reported.
1421
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1434
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1422
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1435
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1423
    "manager show command QueueReset."
1436
    "manager show command QueueReset."
1424
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1437
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1425
    specified by the periodic-announce option, then one will be chosen randomly when it is time
1438
    specified by the periodic-announce option, then one will be chosen randomly when it is time
1426
    to play a periodic announcment
1439
    to play a periodic announcment
1427
  * New configuration options: announce-position now takes two more values in addition to "yes" and
1440
  * New configuration options: announce-position now takes two more values in addition to "yes" and
1428
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1441
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1429
    announce-position-limit. By setting announce-position to "limit" callers will only have their
1442
    announce-position-limit. By setting announce-position to "limit" callers will only have their
1430
    position announced if their position is less than what is specified by announce-position-limit.
1443
    position announced if their position is less than what is specified by announce-position-limit.
1431
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1444
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1432
    will be told that their are more than announce-position-limit callers waiting.
1445
    will be told that their are more than announce-position-limit callers waiting.
1433
  * Two new queue log events have been added. An ADDMEMBER event will be logged
1446
  * Two new queue log events have been added. An ADDMEMBER event will be logged
1434
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
1447
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
1435
    when a realtime queue member is removed. Since there is no calling channel associated
1448
    when a realtime queue member is removed. Since there is no calling channel associated
1436
    with these events, the string "REALTIME" is placed where the channel's unique id
1449
    with these events, the string "REALTIME" is placed where the channel's unique id
1437
    is typically placed.
1450
    is typically placed.
1438
  * The configuration method for the "joinempty" and "leavewhenempty" options has
1451
  * The configuration method for the "joinempty" and "leavewhenempty" options has
1439
    changed to a comma-separated list of methods of determining member availability
1452
    changed to a comma-separated list of methods of determining member availability
1440
    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1453
    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1441
    values are still accepted for backwards-compatibility, though.
1454
    values are still accepted for backwards-compatibility, though.
1442
  * The average talktime is now calculated on queues. This information is reported via the
1455
  * The average talktime is now calculated on queues. This information is reported via the
1443
    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1456
    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1444
    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1457
    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1445
    the queue.
1458
    the queue.
1446

    
   
1459

   
1447
MeetMe Changes
1460
MeetMe Changes
1448
--------------
1461
--------------
1449
  * The 'o' option to provide an optimization has been removed and its functionality 
1462
  * The 'o' option to provide an optimization has been removed and its functionality 
1450
     has been enabled by default.
1463
     has been enabled by default.
1451
  * When a conference is created, the UNIQUEID of the channel that caused it to be
1464
  * When a conference is created, the UNIQUEID of the channel that caused it to be
1452
     created is stored.  Then, every channel that joins the conference will have the
1465
     created is stored.  Then, every channel that joins the conference will have the
1453
     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
1466
     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
1454
     callers that come and go from long standing conferences.
1467
     callers that come and go from long standing conferences.
1455
  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1468
  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1456
     except it does operations on a channel by name, instead of number in a conference.
1469
     except it does operations on a channel by name, instead of number in a conference.
1457
     This is a very useful feature in combination with the 'X' option to ChanSpy.
1470
     This is a very useful feature in combination with the 'X' option to ChanSpy.
1458
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1471
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1459
     when kicked out.
1472
     when kicked out.
1460
  * Added new RealTime functionality to provide support for scheduled conferencing.
1473
  * Added new RealTime functionality to provide support for scheduled conferencing.
1461
     This includes optional messages to the caller if they attempt to join before
1474
     This includes optional messages to the caller if they attempt to join before
1462
     the schedule start time, or to allow the caller to join the conference early.
1475
     the schedule start time, or to allow the caller to join the conference early.
1463
     Also included is optional support for limiting the number of callers per
1476
     Also included is optional support for limiting the number of callers per
1464
     RealTime conference.
1477
     RealTime conference.
1465
  * Added the S() and L() options to the MeetMe application.  These are pretty
1478
  * Added the S() and L() options to the MeetMe application.  These are pretty
1466
     much identical to the S() and L() options to Dial().  They let you set
1479
     much identical to the S() and L() options to Dial().  They let you set
1467
     timeouts for the conference, as well as have warning sounds played to
1480
     timeouts for the conference, as well as have warning sounds played to
1468
     let the caller know how much time is left, and when it is running out.
1481
     let the caller know how much time is left, and when it is running out.
1469
  * Added the ability to do "meetme concise" with the "meetme" CLI command.
1482
  * Added the ability to do "meetme concise" with the "meetme" CLI command.
1470
     This extends the concise capabilities of this CLI command to include
1483
     This extends the concise capabilities of this CLI command to include
1471
     listing all conferences, instead of an addition to the other sub commands
1484
     listing all conferences, instead of an addition to the other sub commands
1472
     for the "meetme" command.
1485
     for the "meetme" command.
1473
  * Added the ability to specify the music on hold class used to play into the
1486
  * Added the ability to specify the music on hold class used to play into the
1474
     conference when there is only one member and the M option is used.
1487
     conference when there is only one member and the M option is used.
1475
  * Added MEETME_INFO dialplan function which provides a way to query
1488
  * Added MEETME_INFO dialplan function which provides a way to query
1476
     various properties of a Meetme conference.
1489
     various properties of a Meetme conference.
1477
  * Added new admin features: *81: Roll call, *82: eject all, *83: mute all, 
1490
  * Added new admin features: *81: Roll call, *82: eject all, *83: mute all, 
1478
     and *84: record in-conf
1491
     and *84: record in-conf
1479

    
   
1492

   
1480
Other Dialplan Application Changes
1493
Other Dialplan Application Changes
1481
----------------------------------
1494
----------------------------------
1482
  * Argument support for Gosub application
1495
  * Argument support for Gosub application
1483
  * From the to-do lists: straighten out the app timeout args:
1496
  * From the to-do lists: straighten out the app timeout args:
1484
     Wait() app now really does 0.3 seconds- was truncating arg to an int.
1497
     Wait() app now really does 0.3 seconds- was truncating arg to an int.
1485
     WaitExten() same as Wait().
1498
     WaitExten() same as Wait().
1486
     Congestion() - Now takes floating pt. argument.
1499
     Congestion() - Now takes floating pt. argument.
1487
     Busy() - now takes floating pt. argument.
1500
     Busy() - now takes floating pt. argument.
1488
     Read() - timeout now can be floating pt.
1501
     Read() - timeout now can be floating pt.
1489
     WaitForRing() now takes floating pt timeout arg.
1502
     WaitForRing() now takes floating pt timeout arg.
1490
     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1503
     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1491
  * Added 's' option to Page application.
1504
  * Added 's' option to Page application.
1492
  * Added an optional timeout argument to the Page application.
1505
  * Added an optional timeout argument to the Page application.
1493
  * Added 'E', 'V', and 'P' commands to ExternalIVR.
1506
  * Added 'E', 'V', and 'P' commands to ExternalIVR.
1494
  * Added 'o' and 'X' options to Chanspy.
1507
  * Added 'o' and 'X' options to Chanspy.
1495
  * Added a new dialplan application, Bridge, which allows you to bridge the
1508
  * Added a new dialplan application, Bridge, which allows you to bridge the
1496
     calling channel to any other active channel on the system.
1509
     calling channel to any other active channel on the system.
1497
  * Added the ability to specify a music on hold class to play instead of ringing
1510
  * Added the ability to specify a music on hold class to play instead of ringing
1498
     for the SLATrunk application.
1511
     for the SLATrunk application.
1499
  * The Read application no longer exits the dialplan on error.  Instead, it sets
1512
  * The Read application no longer exits the dialplan on error.  Instead, it sets
1500
     READSTATUS to ERROR, which you can catch and handle separately.
1513
     READSTATUS to ERROR, which you can catch and handle separately.
1501
  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1514
  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1502
     of asking for verification of each name, one at a time.
1515
     of asking for verification of each name, one at a time.
1503
  * Privacy() no longer uses privacy.conf, as all options are specifyable as
1516
  * Privacy() no longer uses privacy.conf, as all options are specifyable as
1504
     direct options to the app.
1517
     direct options to the app.
1505
  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1518
  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1506
     for more details
1519
     for more details
1507
  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1520
  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1508
  * The ChannelRedirect application no longer exits the dialplan if the given channel
1521
  * The ChannelRedirect application no longer exits the dialplan if the given channel
1509
     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1522
     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1510
     or NOCHANNEL if the given channel was not found.
1523
     or NOCHANNEL if the given channel was not found.
1511
  * The silencethreshold setting that was previously configurable in multiple
1524
  * The silencethreshold setting that was previously configurable in multiple
1512
     applications is now settable globally via dsp.conf.
1525
     applications is now settable globally via dsp.conf.
1513

    
   
1526

   
1514
Music On Hold Changes
1527
Music On Hold Changes
1515
---------------------
1528
---------------------
1516
  * A new option, "digit", has been added for music on hold classes in 
1529
  * A new option, "digit", has been added for music on hold classes in 
1517
     musiconhold.conf.  If this is set for a music on hold class, a caller
1530
     musiconhold.conf.  If this is set for a music on hold class, a caller
1518
     listening to music on hold can press this digit to switch to listening
1531
     listening to music on hold can press this digit to switch to listening
1519
     to this music on hold class.
1532
     to this music on hold class.
1520
  * Support for realtime music on hold has been added.
1533
  * Support for realtime music on hold has been added.
1521
  * In conjunction with the realtime music on hold, a general section has
1534
  * In conjunction with the realtime music on hold, a general section has
1522
     been added to musiconhold.conf, its sole variable is cachertclasses. If this
1535
     been added to musiconhold.conf, its sole variable is cachertclasses. If this
1523
     is set, then music on hold classes found in realtime will be cached in memory.
1536
     is set, then music on hold classes found in realtime will be cached in memory.
1524

    
   
1537

   
1525
AEL Changes
1538
AEL Changes
1526
-----------
1539
-----------
1527
  * AEL upgraded to use the Gosub with Arguments instead
1540
  * AEL upgraded to use the Gosub with Arguments instead
1528
     of Macro application, to hopefully reduce the problems
1541
     of Macro application, to hopefully reduce the problems
1529
     seen with the artificially low stack ceiling that 
1542
     seen with the artificially low stack ceiling that 
1530
     Macro bumps into. Macros can only call other Macros
1543
     Macro bumps into. Macros can only call other Macros
1531
     to a depth of 7. Tests run using gosub, show depths
1544
     to a depth of 7. Tests run using gosub, show depths
1532
     limited only by virtual memory. A small test demonstrated
1545
     limited only by virtual memory. A small test demonstrated
1533
     recursive call depths of 100,000 without problems.
1546
     recursive call depths of 100,000 without problems.
1534
     -- in addition to this, all apps that allowed a macro
1547
     -- in addition to this, all apps that allowed a macro
1535
     to be called, as in Dial, queues, etc, are now allowing
1548
     to be called, as in Dial, queues, etc, are now allowing
1536
     a gosub call in similar fashion.
1549
     a gosub call in similar fashion.
1537
  * AEL now generates LOCAL(argname) declarations when it
1550
  * AEL now generates LOCAL(argname) declarations when it
1538
     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1551
     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1539
     etc. That makes the arguments local in scope. The user
1552
     etc. That makes the arguments local in scope. The user
1540
     can define their own local variables in macros, now,
1553
     can define their own local variables in macros, now,
1541
     by saying "local myvar=someval;"  or using Set() in this
1554
     by saying "local myvar=someval;"  or using Set() in this
1542
     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
1555
     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
1543
     an AEL keyword).
1556
     an AEL keyword).
1544
  * utils/conf2ael introduced. Will convert an extensions.conf
1557
  * utils/conf2ael introduced. Will convert an extensions.conf
1545
     file into extensions.ael. Very crude and unfinished, but 
1558
     file into extensions.ael. Very crude and unfinished, but 
1546
     will be improved as time goes by. Should be useful for a
1559
     will be improved as time goes by. Should be useful for a
1547
     first pass at conversion.
1560
     first pass at conversion.
1548
  * aelparse will now read extensions.conf to see if a referenced
1561
  * aelparse will now read extensions.conf to see if a referenced
1549
     macro or context is there before issueing a warning.
1562
     macro or context is there before issueing a warning.
1550
  * AEL parser sets a local channel variable ~~EXTEN~~, to 
1563
  * AEL parser sets a local channel variable ~~EXTEN~~, to 
1551
    preserve the value of ${EXTEN} thru switch statements.
1564
    preserve the value of ${EXTEN} thru switch statements.
1552
  * New operator in $[...] expressions: the ~~ operator serves
1565
  * New operator in $[...] expressions: the ~~ operator serves
1553
    as a concatenation operator. AT THE MOMENT, it is really only
1566
    as a concatenation operator. AT THE MOMENT, it is really only
1554
    necessary and useful in AEL, especially in if() expressions.
1567
    necessary and useful in AEL, especially in if() expressions.
1555
    Operation: ${a} ~~ ${b|  with force both a and b to strings, strip 
1568
    Operation: ${a} ~~ ${b|  with force both a and b to strings, strip 
1556
    any enclosing double-quotes, and evaluate to the value of a
1569
    any enclosing double-quotes, and evaluate to the value of a
1557
    concatenated with the value of b.  For example if a is set to
1570
    concatenated with the value of b.  For example if a is set to
1558
    "xyz"  and b has the value "abc", then ${a} ~~ ${b| would
1571
    "xyz"  and b has the value "abc", then ${a} ~~ ${b| would
1559
    evaluate to xyzabc .
1572
    evaluate to xyzabc .
1560

    
   
1573

   
1561

    
   
1574

   
1562
Call Features (res_features) Changes
1575
Call Features (res_features) Changes
1563
------------------------------------
1576
------------------------------------
1564
  * Added the parkedcalltransfers option to features.conf
1577
  * Added the parkedcalltransfers option to features.conf
1565
  * Added parkedcallparking option to control one touch parking w/ parking
1578
  * Added parkedcallparking option to control one touch parking w/ parking
1566
    pickup
1579
    pickup
1567
  * Added parkedcallhangup option to control disconnect feature w/ parking
1580
  * Added parkedcallhangup option to control disconnect feature w/ parking
1568
    pickup
1581
    pickup
1569
  * Added parkedcallrecording option to control one-touch record w/ parking
1582
  * Added parkedcallrecording option to control one-touch record w/ parking
1570
    pickup
1583
    pickup
1571
  * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
1584
  * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
1572
    parkedcalltransfers option support for multiple parking lots.
1585
    parkedcalltransfers option support for multiple parking lots.
1573
  * Added BRIDGE_FEATURES variable to set available features for a channel
1586
  * Added BRIDGE_FEATURES variable to set available features for a channel
1574
  * The built-in method for doing attended transfers has been updated to
1587
  * The built-in method for doing attended transfers has been updated to
1575
     include some new options that allow you to have the transferee sent
1588
     include some new options that allow you to have the transferee sent
1576
     back to the person that did the transfer if the transfer is not successful.
1589
     back to the person that did the transfer if the transfer is not successful.
1577
     See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1590
     See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1578
     in features.conf.sample.
1591
     in features.conf.sample.
1579
  * Added support for configuring named groups of custom call features in
1592
  * Added support for configuring named groups of custom call features in
1580
     features.conf.  This means that features can be written a single time, and
1593
     features.conf.  This means that features can be written a single time, and
1581
     then mapped into groups of features for different key mappings or easier
1594
     then mapped into groups of features for different key mappings or easier
1582
     access control.
1595
     access control.
1583
  * Updated the ParkedCall application to allow you to not specify a parking
1596
  * Updated the ParkedCall application to allow you to not specify a parking
1584
     extension.  If you don't specify a parking space to pick up, it will grab
1597
     extension.  If you don't specify a parking space to pick up, it will grab
1585
     the first one available.
1598
     the first one available.
1586
  * Added cli command 'features reload' to reload call features from features.conf
1599
  * Added cli command 'features reload' to reload call features from features.conf
1587
  * Moved into core asterisk binary.
1600
  * Moved into core asterisk binary.
1588
  * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1601
  * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1589
  * Added the ability for custom parking lots to be configured with their own
1602
  * Added the ability for custom parking lots to be configured with their own
1590
    parking extension with the parkext option.
1603
    parking extension with the parkext option.
1591

    
   
1604

   
1592
Language Support Changes
1605
Language Support Changes
1593
------------------------
1606
------------------------
1594
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1607
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1595
  * Added support for the Hungarian language for saying numbers, dates, and times.
1608
  * Added support for the Hungarian language for saying numbers, dates, and times.
1596

    
   
1609

   
1597
AGI Changes
1610
AGI Changes
1598
-----------
1611
-----------
1599
  * Added SPEECH commands for speech recognition. A complete listing can be found
1612
  * Added SPEECH commands for speech recognition. A complete listing can be found
1600
    using agi show.
1613
    using agi show.
1601
  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1614
  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1602
    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
1615
    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
1603
    does not behave as expected; the native command needs to be used, instead.
1616
    does not behave as expected; the native command needs to be used, instead.
1604
  * Added the ability to perform SRV lookups on fast AGI calls. To use this
1617
  * Added the ability to perform SRV lookups on fast AGI calls. To use this
1605
    feature, simply use hagi: instead of agi: as the protocol portion
1618
    feature, simply use hagi: instead of agi: as the protocol portion
1606
    of the URI parameter to the AGI function call in your dial plan. Also note
1619
    of the URI parameter to the AGI function call in your dial plan. Also note
1607
    that specifying a port number in the AGI URI will disable SRV lookups,
1620
    that specifying a port number in the AGI URI will disable SRV lookups,
1608
    even if you use the hagi: protocol.
1621
    even if you use the hagi: protocol.
1609
  * No longer support MSG_OOB flag on HANGUP.
1622
  * No longer support MSG_OOB flag on HANGUP.
1610

    
   
1623

   
1611
Logger changes
1624
Logger changes
1612
--------------
1625
--------------
1613
  * Added rotatestrategy option to logger.conf, along with two new options:
1626
  * Added rotatestrategy option to logger.conf, along with two new options:
1614
     "timestamp" which will use the time to name the logger files instead of
1627
     "timestamp" which will use the time to name the logger files instead of
1615
     sequence number; and "rotate", which rotates the names of the log files,
1628
     sequence number; and "rotate", which rotates the names of the log files,
1616
     similar to the way syslog rotates files.
1629
     similar to the way syslog rotates files.
1617
  * Added exec_after_rotate option to logger.conf, which allows a system
1630
  * Added exec_after_rotate option to logger.conf, which allows a system
1618
     command to be run after rotation.  This is primarily useful with
1631
     command to be run after rotation.  This is primarily useful with
1619
     rotatestrategy=rotate, to allow a limit on the number of log files kept
1632
     rotatestrategy=rotate, to allow a limit on the number of log files kept
1620
     and to ensure that the oldest log file gets deleted.
1633
     and to ensure that the oldest log file gets deleted.
1621
  * Added realtime support for the queue log
1634
  * Added realtime support for the queue log
1622

    
   
1635

   
1623
Call Detail Records 
1636
Call Detail Records 
1624
-------------------
1637
-------------------
1625
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
1638
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
1626
    to add fields to the manager event from the CDR variables.
1639
    to add fields to the manager event from the CDR variables.
1627
  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1640
  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1628
     backend database CDR table.  Specifically, additional, non-standard
1641
     backend database CDR table.  Specifically, additional, non-standard
1629
     columns are supported, merely by setting the corresponding CDR variable in
1642
     columns are supported, merely by setting the corresponding CDR variable in
1630
     your dialplan.  In addition, you may alias any column to another name (for
1643
     your dialplan.  In addition, you may alias any column to another name (for
1631
     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1644
     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1632
     simply "alias src => ANI" in the configuration file).  Records may be
1645
     simply "alias src => ANI" in the configuration file).  Records may be
1633
     posted to more than one backend, simply by specifying multiple categories
1646
     posted to more than one backend, simply by specifying multiple categories
1634
     in the configuration file.  And finally, you may filter which CDRs get
1647
     in the configuration file.  And finally, you may filter which CDRs get
1635
     posted to each backend, by specifying a filter (which the record must
1648
     posted to each backend, by specifying a filter (which the record must
1636
     match) for the particular category.  Filters are additive (meaning all
1649
     match) for the particular category.  Filters are additive (meaning all
1637
     rules must match to post that CDR).
1650
     rules must match to post that CDR).
1638
  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1651
  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1639
     module.  Specifically, you may add additional columns into the table and
1652
     module.  Specifically, you may add additional columns into the table and
1640
     they will be set, if you set the corresponding CDR variable name.  Also,
1653
     they will be set, if you set the corresponding CDR variable name.  Also,
1641
     if you omit columns in your database table, they will be silently skipped
1654
     if you omit columns in your database table, they will be silently skipped
1642
     (but a record will still be inserted, based on what columns remain).  Note
1655
     (but a record will still be inserted, based on what columns remain).  Note
1643
     that the other two features from cdr_adaptive_odbc (alias and filter) are
1656
     that the other two features from cdr_adaptive_odbc (alias and filter) are
1644
     not currently supported.
1657
     not currently supported.
1645
  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1658
  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1646
     has been disabled using the NoCDR application.
1659
     has been disabled using the NoCDR application.
1647

    
   
1660

   
1648
Miscellaneous New Modules
1661
Miscellaneous New Modules
1649
-------------------------
1662
-------------------------
1650
  * Added a new CDR module, cdr_sqlite3_custom.
1663
  * Added a new CDR module, cdr_sqlite3_custom.
1651
  * Added a new realtime configuration module, res_config_sqlite
1664
  * Added a new realtime configuration module, res_config_sqlite
1652
  * Added a new codec translation module, codec_resample, which re-samples
1665
  * Added a new codec translation module, codec_resample, which re-samples
1653
     signed linear audio between 8 kHz and 16 kHz to help support wideband
1666
     signed linear audio between 8 kHz and 16 kHz to help support wideband
1654
     codecs.
1667
     codecs.
1655
  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1668
  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1656
     based on configuration templates that use Asterisk dialplan function and
1669
     based on configuration templates that use Asterisk dialplan function and
1657
     variable substitution.  It should be possible to create phone profiles and
1670
     variable substitution.  It should be possible to create phone profiles and
1658
     templates that work for the majority of phones provisioned over http. It
1671
     templates that work for the majority of phones provisioned over http. It
1659
     is currently only intended to provision a single user account per phone.
1672
     is currently only intended to provision a single user account per phone.
1660
     An example profile and set of templates for Polycom phones is provided.
1673
     An example profile and set of templates for Polycom phones is provided.
1661
     NOTE: Polycom firmware is not included, but should be placed in
1674
     NOTE: Polycom firmware is not included, but should be placed in
1662
     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1675
     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1663
  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1676
  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1664
     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
1677
     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
1665
     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
1678
     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
1666
     interfaces create an input and output JACK port.  The application makes
1679
     interfaces create an input and output JACK port.  The application makes
1667
     these ports the endpoint of the call.  The audio coming from the channel
1680
     these ports the endpoint of the call.  The audio coming from the channel
1668
     goes out the output port and whatever comes back in on the input port is
1681
     goes out the output port and whatever comes back in on the input port is
1669
     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
1682
     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
1670
     audiohook on the channel.  This lets you run the audio coming from a
1683
     audiohook on the channel.  This lets you run the audio coming from a
1671
     channel through JACK, and whatever comes back in is what gets forwarded
1684
     channel through JACK, and whatever comes back in is what gets forwarded
1672
     on as the channel's audio.  This is very useful for building custom
1685
     on as the channel's audio.  This is very useful for building custom
1673
     vocoders or doing recording or analysis of the channel's audio in another
1686
     vocoders or doing recording or analysis of the channel's audio in another
1674
     application.
1687
     application.
1675
  * Added a new module, res_config_curl, which permits using a HTTP POST url
1688
  * Added a new module, res_config_curl, which permits using a HTTP POST url
1676
     to retrieve, create, update, and delete realtime information from a remote
1689
     to retrieve, create, update, and delete realtime information from a remote
1677
     web server.  Note that this module requires func_curl.so to be loaded for
1690
     web server.  Note that this module requires func_curl.so to be loaded for
1678
     backend functionality.
1691
     backend functionality.
1679
  * Added a new module, res_config_ldap, which permits the use of an LDAP
1692
  * Added a new module, res_config_ldap, which permits the use of an LDAP
1680
     server for realtime data access.
1693
     server for realtime data access.
1681
  * Added support for writing and running your dialplan in lua using the pbx_lua
1694
  * Added support for writing and running your dialplan in lua using the pbx_lua
1682
     module.  See configs/extensions.lua.sample for examples of how to do this.
1695
     module.  See configs/extensions.lua.sample for examples of how to do this.
1683

    
   
1696

   
1684
Miscellaneous 
1697
Miscellaneous 
1685
-------------
1698
-------------
1686
  * Ability to use libcap to set high ToS bits when non-root
1699
  * Ability to use libcap to set high ToS bits when non-root
1687
     on Linux. If configure is unable to find libcap then you
1700
     on Linux. If configure is unable to find libcap then you
1688
     can use --with-cap to specify the path.
1701
     can use --with-cap to specify the path.
1689
  * Added maxfiles option to options section of asterisk.conf which allows you to specify
1702
  * Added maxfiles option to options section of asterisk.conf which allows you to specify
1690
     what Asterisk should set as the maximum number of open files when it loads.
1703
     what Asterisk should set as the maximum number of open files when it loads.
1691
  * Added the jittertargetextra configuration option.
1704
  * Added the jittertargetextra configuration option.
1692
  * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
1705
  * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
1693
     configuration files for the IP channel drivers.  The new option is "cos".
1706
     configuration files for the IP channel drivers.  The new option is "cos".
1694
     This information is also documented in doc/qos.tex, or the IP Quality of Service
1707
     This information is also documented in doc/qos.tex, or the IP Quality of Service
1695
     section of asterisk.pdf.
1708
     section of asterisk.pdf.
1696
  * When originating a call using AMI or pbx_spool that fails the reason for failure
1709
  * When originating a call using AMI or pbx_spool that fails the reason for failure
1697
     will now be available in the failed extension using the REASON dialplan variable.
1710
     will now be available in the failed extension using the REASON dialplan variable.
1698
  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1711
  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1699
     It allows you to configure a prefix for auto-monitor recordings.
1712
     It allows you to configure a prefix for auto-monitor recordings.
1700
  * A new extension pattern matching algorithm, based on a trie, is introduced
1713
  * A new extension pattern matching algorithm, based on a trie, is introduced
1701
     here, that could noticeably speed up mid-sized to large dialplans.
1714
     here, that could noticeably speed up mid-sized to large dialplans.
1702
     It is NOT used by default, as duplicating the behaviour of the old pattern
1715
     It is NOT used by default, as duplicating the behaviour of the old pattern
1703
     matcher is still under development. A config file option, in extensions.conf,
1716
     matcher is still under development. A config file option, in extensions.conf,
1704
     in the [general] section, called "extenpatternmatchingnew", is by default
1717
     in the [general] section, called "extenpatternmatchingnew", is by default
1705
     set to false; setting that to true will force the use of the new algorithm.
1718
     set to false; setting that to true will force the use of the new algorithm.
1706
     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1719
     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1707
     be used to switch the algorithms at run time.
1720
     be used to switch the algorithms at run time.
1708
  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1721
  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1709
     specifying which socket to use to connect to the running Asterisk daemon
1722
     specifying which socket to use to connect to the running Asterisk daemon
1710
     (-s)
1723
     (-s)
1711
  * Performance enhancements to the sched facility, which is used in
1724
  * Performance enhancements to the sched facility, which is used in
1712
    the channel drivers, etc. Added hashtabs and doubly-linked lists
1725
    the channel drivers, etc. Added hashtabs and doubly-linked lists
1713
    to speed up deletion; start at the beginning or end of list to
1726
    to speed up deletion; start at the beginning or end of list to
1714
    speed up insertion.
1727
    speed up insertion.
1715
  * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1728
  * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1716
    dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1729
    dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1717
    Added regression tests to the tests/ dir, also.
1730
    Added regression tests to the tests/ dir, also.
1718
  * Added a refcount trace feature to astobj2 for those trying to balance
1731
  * Added a refcount trace feature to astobj2 for those trying to balance
1719
    object creation, deletion; work, play; space and time. See the
1732
    object creation, deletion; work, play; space and time. See the
1720
    notes in astobj2.h. Also, see utils/refcounter as well, as a
1733
    notes in astobj2.h. Also, see utils/refcounter as well, as a
1721
    quick way to find unbalanced refcounts in what could be a sea
1734
    quick way to find unbalanced refcounts in what could be a sea
1722
    of objects that were balanced.
1735
    of objects that were balanced.
1723
  * Added logging to 'make update' command.  See update.log
1736
  * Added logging to 'make update' command.  See update.log
1724
  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1737
  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1725
     do not come from the remote party.
1738
     do not come from the remote party.
1726
  * Added the 'n' option to the SpeechBackground application to tell it to not
1739
  * Added the 'n' option to the SpeechBackground application to tell it to not
1727
     answer the channel if it has not already been answered.
1740
     answer the channel if it has not already been answered.
1728
  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1741
  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1729
     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
1742
     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
1730
     dialplan debugging.
1743
     dialplan debugging.
1731
  * iLBC source code no longer included (see UPGRADE.txt for details)
1744
  * iLBC source code no longer included (see UPGRADE.txt for details)
1732
  * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if 
1745
  * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if 
1733
     deadlock is detected, a backtrace of the stack which led to the lock calls
1746
     deadlock is detected, a backtrace of the stack which led to the lock calls
1734
     will be output to the CLI.
1747
     will be output to the CLI.
1735
  * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1748
  * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1736
     the "core show locks" CLI command will give lock information output as well
1749
     the "core show locks" CLI command will give lock information output as well
1737
     as a backtrace of the stack which led to the lock calls.
1750
     as a backtrace of the stack which led to the lock calls.
1738
  * users.conf now sports an optional alternateexts property, which permits
1751
  * users.conf now sports an optional alternateexts property, which permits
1739
    allocation of additional extensions which will reach the specified user.
1752
    allocation of additional extensions which will reach the specified user.
1740
  * A new option for the configure script, --enable-internal-poll, has been added
1753
  * A new option for the configure script, --enable-internal-poll, has been added
1741
    for use with systems which may have a buggy implementation of the poll system
1754
    for use with systems which may have a buggy implementation of the poll system
1742
    call. If you notice odd behavior such as the CLI being unresponsive on remote
1755
    call. If you notice odd behavior such as the CLI being unresponsive on remote
1743
    consoles, you may want to try using this option. This option is enabled by default
1756
    consoles, you may want to try using this option. This option is enabled by default
1744
    on Darwin systems since it is known that the Darwin poll() implementation has
1757
    on Darwin systems since it is known that the Darwin poll() implementation has
1745
    odd issues.
1758
    odd issues.
1746

    
   
1759

   
1747
Timer Changes
1760
Timer Changes
1748
--------------------
1761
--------------------
1749
* In addition to timing from DAHDI, there is a new timing module called
1762
* In addition to timing from DAHDI, there is a new timing module called
1750
  res_timing_timerfd. In order to use this, you must be running Linux with
1763
  res_timing_timerfd. In order to use this, you must be running Linux with
1751
  a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1764
  a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1752
  script will be able to tell if you have the requirements. From menuselect, select
1765
  script will be able to tell if you have the requirements. From menuselect, select
1753
  res_timing_timerfd from the Resource Modules menu.
1766
  res_timing_timerfd from the Resource Modules menu.
/trunk/channels/chan_sip.c
Diff Revision 3 Diff Revision 6
 
/trunk/channels/sip/include/sip.h
Diff Revision 3 Diff Revision 6
 
/trunk/configs/jabber.conf.sample
Diff Revision 3 Diff Revision 6
 
/trunk/configs/sip.conf.sample
Diff Revision 3 Diff Revision 6
 
/trunk/doc/asterisk-messaging.txt
Diff Revision 3 Diff Revision 6
 
/trunk/include/asterisk/_private.h
Diff Revision 3 Diff Revision 6
 
/trunk/include/asterisk/channel.h
Diff Revision 3 Diff Revision 6
 
/trunk/include/asterisk/jabber.h
Diff Revision 3 Diff Revision 6
 
/trunk/include/asterisk/message.h
Diff Revision 3 Diff Revision 6
 
/trunk/main/asterisk.c
Diff Revision 3 Diff Revision 6
 
/trunk/main/channel.c
Diff Revision 3 Diff Revision 6
 
/trunk/main/message.c
Diff Revision 3 Diff Revision 6
 
/trunk/res/res_jabber.c
Diff Revision 3 Diff Revision 6
 
  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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