CVS commit: pkgsrc/comms/asterisk16

[John Nemeth <jnemeth%netbsd.org@localhost>]

Module Name:    pkgsrc
Committed By:   jnemeth
Date:           Tue Jul  7 08:14:43 UTC 2009

Modified Files:
        pkgsrc/comms/asterisk16: Makefile PLIST distinfo
        pkgsrc/comms/asterisk16/patches: patch-aa patch-am patch-aq patch-ax
            patch-ay patch-az patch-ba patch-bb

Log Message:
Update to Asterisk 1.6.1.1

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
------------------------------------------------------------------------------

Device State Handling
---------------------
 * The event infrastructure in Asterisk got another big update to help support
    distributed events.  It currently supports distributed device state and
    distributed Voicemail MWI (Message Waiting Indication).  A new module has
    been merged, res_ais, which facilitates communicating events between 
servers.
    It uses the SAForum AIS (Service Availability Forum Application Interface
    Specification) CLM (Cluster Management) and EVT (Event) services to maintain
    a cluster of Asterisk servers, and to share events between them.  For more
    information on setting this up, see doc/distributed_devstate.txt.

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

DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
 * Channels can now be configured using named sections in chan_dahdi.conf, just
   like other channel drivers, including the use of templates.
 * The default for pridialplan has changed from 'national' to 'unknown'.

PBX Changes
-----------
 * It is now possible to specify a pattern match as a hint. Once a phone 
subscribes
   to something that matches the pattern a hint will be created using the 
contents
   and variables evaluated.
 * Dialplan matching has been extended to allow an extension to return to the
   PBX core to wait for more digits.  This is done by using the new dialplan
   application called "Incomplete".  This will permit a whole new level of
   extension control, by giving the administrator more control over early
   matches employing one of the short-circuit pattern match operators.  Note
   that custom applications can trigger this same behavior by returning the
   special value AST_PBX_INCOMPLETE.

The dial() application
----------------------
 * Dial has a new option: F(context^extension^pri), which permits a callee to
   continue in the dialplan, at the specified label, if the caller hangs up.
 * The Dial() application no longer copies the language used by the caller to 
the callee's
   channel. If you desire for the caller's channel's language to be used for 
file playback
   to the callee, then the file specified may be prepended with 
"${CHANNEL(language)}/" .

The chanspy() application
-------------------------
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This 
is
   like the pre-existing whisper mode, except that the spy can also talk to the
   participant on the bridged channel as well.
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
   to be spoken instead of the channel name or number. For more information on 
the
   use of this option, issue the command "core show application ChanSpy" from 
the
   Asterisk CLI.
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap 
between
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In 
other
   words, if using the 'd' option, it is not possible to enter a number to 
append to
   the first argument to Chanspy(). Pressing 4 will change to spy mode, 
pressing 5 will
   change to whisper mode, and pressing 6 will change to barge mode.

Other Application Changes
-------------------------
 * Directory now permits both first and last names to be matched at the same
   time.  In addition, the number of digits to enter of the name can be set in
   the arguments to Directory; previously, you could enter only 3, regardless
   of how many names are in your company.  For large companies, this should be
   quite helpful.
 * Voicemail now permits a mailbox setting to wrap around from first to last
   messages, if the "messagewrap" option is set to a true value.
 * Voicemail now permits an external script to be run, for password validation.
   The script should output "VALID" or "INVALID" on stdout, depending upon the
   wish to validate or invalidate the password given.  Arguments are:
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
   more details
 * The voicemail externnotify script now accepts an additional (last) parameter
   containing the number of urgent messages in the INBOX.
 * The Jack application now has a c() option to supply a custom client name.
 * ExternalIVR now takes several options that affect the way it performs, as
   well as having several new commands.  Please see doc/externalivr.txt for the
   complete documentation.
 * Added ability to communicate over a TCP socket instead of forking a child 
process for the
   ExternalIVR application.
 * ChanIsAvail has a new option, 'a', which will return all available channels 
instead
   of just the first one if you give the function more then one channel to 
check.
 * PrivacyManager now takes an option where you can specify a context where the
   given number will be matched. This way you have more control over who is 
allowed
   and it stops the people who blindly enter 10 digits.
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' 
locks
   answer times, disposition, on orig CDR against updates; 'D' Copies the 
disposition
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' 
adds/changes
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), 
ast_cdr_answer(),
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() 
func.
 * SendImage() no longer hangs up the channel on error; instead, it sets the
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
   applications.
 * Park has a new option, 's', which silences the announcement of the parking 
space number.
 * A non-numeric, zero, or negative timeout specified to Dial() will now be 
interpreted as
   invalid input and will be assumed to mean that no timeout is desired.

SIP Changes
-----------
 * Added DNS manager support to registrations for peers referencing peer 
entries.
   DNS manager runs in the background which allows DNS lookups to be run 
asynchronously
   as well as periodically updating the IP address. These properties allow for
   better performance as well as recovery in the event of an IP change.
 * Performance improvements via using hash tables (astobj2) and doubly-linked 
lists to improve
   load/reload of large numbers of peers/users by ~40x (for large lists of 
peers.
   Initially, we saw 4x improvement in call setup/destruction, but at the time
   of merging, this gain has disappeared; further research will be done to try
   and restore this performance improvement. Astobj2 refcounting is now used
   for users, peers, and dialogs.  Users are encouraged to assist in regression
   testing and problem reporting!
 * Added ability to specify registration expiry time on a per registration 
basis in
   the register line.
 * Added support for Realtime Text redundancy - T140 RED - in T.140 to
   prevent text loss due to lost packets.
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
 * 'sip show peers' and 'sip show users' display their entries sorted in
   alphabetical order, as opposed to the order they were in, in the config
   file or database.
 * Videosupport now supports an additional option, "always", which always sets
   up video RTP ports, even on clients that don't support it.  This helps with
   callfiles and certain transfers to ensure that if two video phones are
   connected, they will always share video feeds.

IAX Changes
-----------
 * Existing DNS manager lookups extended to check for SRV records.
 * IAX2 encryption support has been improved to support periodic key rotation
   within a call for enhanced security.  The option "keyrotate" has been
   provided to disable this functionality to preserve backwards compatibility
   with older versions of IAX2 that do not support key rotation.

CLI Changes
-----------
  * New CLI command, "config reload <file.conf>" which reloads any module that
    references that particular configuration file.  Also added "config list"
    which shows which configuration files are in use.
  * New CLI commands, "pri show version" and "ss7 show version" that will
    display which version of libpri and libss7 are being used, respectively.
    A new API call was added so trunk will now have to be compiled against
    a versions of libpri and libss7 that have them or it will not know that
    these libraries exist.
  * The commands "core show globals", "core set global" and "core set chanvar" 
has
    been deprecated in favor of the more semanticly correct "dialplan show 
globals",
    "dialplan set chanvar" and "dialplan set global".
  * New CLI command "dialplan show chanvar" to list all variables associated
    with a given channel.

DNS manager changes
-------------------
  * Addresses managed by DNS manager now can check to see if there is a DNS
    SRV record for a given domain and will use that hostname/port if present.

AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * The Status action now takes an optional list of variables to display
    along with channel status.

ODBC Changes
------------
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
    as some people were running into this limit.  This limit has been increased
    to 4.2 billion.

Queue changes
-------------
  * The TRANSFER queue log entry now includes the caller's original position in
    the transferred-from queue.
  * A new configuration option, "timeoutpriority" has been added. Please see 
the section
    labeled "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed 
explanation
    of the option as well as an explanation about timeout options in general

Realtime changes
----------------
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
    adaptive capabilities.  What this means in practical terms is that if your
    realtime table lacks critical fields, Asterisk will now emit warnings to
    that effect.  Also, some of the realtime drivers have the ability (if
    configured) to automatically add those columns to the table with the
    correct type and length.

Miscellaneous
-------------
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
    the 'setvar' option to cause a given audio file to be played upon completion
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
    Skinny channels only.
  * You can now compile Asterisk against the Hoard Memory Allocator, see 
doc/hoard.txt
    for more information.
  * Config file variables may now be appended to, by using the '+=' append
    operator.  This is most helpful when working with long SQL queries in
    func_odbc.conf, as the queries no longer need to be specified on a single
    line.


To generate a diff of this commit:
cvs rdiff -u -r1.1.1.1 -r1.2 pkgsrc/comms/asterisk16/Makefile \
    pkgsrc/comms/asterisk16/distinfo
cvs rdiff -u -r1.3 -r1.4 pkgsrc/comms/asterisk16/PLIST
cvs rdiff -u -r1.1.1.1 -r1.2 pkgsrc/comms/asterisk16/patches/patch-aa \
    pkgsrc/comms/asterisk16/patches/patch-am \
    pkgsrc/comms/asterisk16/patches/patch-aq \
    pkgsrc/comms/asterisk16/patches/patch-ax \
    pkgsrc/comms/asterisk16/patches/patch-ay \
    pkgsrc/comms/asterisk16/patches/patch-az \
    pkgsrc/comms/asterisk16/patches/patch-ba \
    pkgsrc/comms/asterisk16/patches/patch-bb

Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.