UPGRADE.txt

===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
===
===========================================================

from 1.8.31.0 to 1.8.31.1:

* Due to the POODLE vulnerability (see 
  https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-3566), the
  default TLS method for TLS clients will no longer allow SSLv3. As
  SSLv2 was already deprecated, it is no longer allowed by default as
  well. TLS servers no longer allow SSLv2 or SSLv3 connections. This
  affects the chan_sip channel driver, AMI, and the Asterisk HTTP server.

* The res_jabber resource module no longer uses SSLv3 to connect to an
  XMPP server. It will now only use TLSv1 or later methods.

from 1.8.28.2 to 1.8.29.0:
* Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
  deal with switches that don't send an inband progress indication in the
  SETUP ACKNOWLEDGE message.

from 1.8.28.0 to 1.8.28.1:
* Added http.conf session_inactivity timer option to close HTTP connections
  that aren't doing anything.

* Removed the undocumented manager.conf block-sockets option.  It interferes with
  TCP/TLS inactivity timeouts.

from 1.8.27.0 to 1.8.28.0:
* The asterisk command line -I option and the asterisk.conf internal_timing
  option are removed and always enabled if any timing module is loaded.
* SIP (chan_sip) accounts dialed through a Local channel will now properly
  hide the "1 missed call" if one of the other dialed accounts picks up the
  call.

* Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
  objects will emit additional debug information to the refs log file located
  in the standard Asterisk log file directory. This log file is useful in
  tracking down object leaks and other reference counting issues. Prior to
  this version, this option was only available by modifying the source code
  directly. This change also includes a new script, refcounter.py, in the
  contrib folder that will process the refs log file.

from 1.8.26.0 to 1.8.27.0:
* res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
  Because of this the default settings would not load, so the minrate (minimum
  transmission rate) option was changed to default to 4800 since that is the
  minimum rate for v.27 which is included in the default modem options.

* When communicating with a peer on an Asterisk 1.4 or earlier system, the
  chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
  prevents an incompatible connected line frame from an Astersik 1.8 or later
  system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
  this particular incompatibility has always existed between 1.4 and 1.8 and
  later versions; this upgrade note is simply informing users of its existance.

* A compatibility setting, allow_empty_string_in_nontext, has been added to
  res_odbc.conf. When enabled (default behavior), empty column values are
  stored as empty strings during realtime updates. Disabling this option
  causes empty column values to be stored as NULLs for non-text columns.

  Disable it for PostgreSQL backends in order to avoid errors caused by
  updating integer columns with an empty string instead of NULL
  (sipppeers,sipregs)

from 1.8.23.0 to 1.8.24.0:
* res_agi will now properly indicate if there was an error in streaming an
  audio file.  The result code will be -1 and the result returned from the
  the function will be RESULT_FAILURE instead of the prior behavior of always
  returning RESULT_SUCCESS even if there was an error. 

* The option "register_retry_403" has been added to chan_sip to work around
  servers that are known to erroneously send 403 in response to valid
  REGISTER requests and allows Asterisk to continue attepmting to connect.
  Due to a failed merge, this option is present, but non-functional until 1.8.26.0.

* Certain dialplan functions have been marked as 'dangerous', and may only be
  executed from the dialplan. Execution from extenal sources (AMI's GetVar and
  SetVar actions; etc.) may be inhibited by setting live_dangerously in the
  [options] section of asterisk.conf to no. SHELL(), channel locking, and direct
  file read/write functions are marked as dangerous. DB_DELETE() and
  REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
  accept writes (which ignore the provided value).

from 1.8.22.0 to 1.8.23.0:
* The default settings for chan_sip are now overriden properly by the general
  settings in sip.conf.  Please look over your settings upon upgrading.

* It is now possible to play the Queue prompts to the first user waiting in a call queue.
  Note that this may impact the ability for agents to talk with users, as a prompt may
  still be playing when an agent connects to the user. This ability is disabled by
  default but can be enabled on an individual queue using the 'announce-to-first-user'
  option.

from 1.8.21.0 to 1.8.22.0:
* Added the 'n' option to MeetMe to prevent application of the DENOISE function
  to a channel joining a conference. Some channel drivers that vary the number
  of audio samples in a voice frame will experience significant quality problems
  if a denoiser is attached to the channel; this option gives them the ability
  to remove the denoiser without having to unload func_speex.

* The Registry AMI event for SIP registrations will now always include the
  Username field. A previous bug fix missed an instance where it was not
  included; that has been corrected in this release.

from 1.8.20.0 to 1.8.20.1:
* Asterisk would previously not output certain error messages when a remote
  console attempted to connect to Asterisk and no instance of Asterisk was
  running. This error message is displayed on stderr; as a result, some
  initialization scripts that used remote consoles to test for the presence
  of a running Asterisk instance started to display erroneous error messages.
  The init.d scripts and the safe_asterisk have been updated in the contrib
  folder to account for this.

from 1.8.19 to 1.8.20:
* Asterisk has always had code to ignore dash '-' characters that are not
  part of a character set in the dialplan extensions.  The code now
  consistently ignores these characters when matching dialplan extensions.

from 1.8.18 to 1.8.19:
* Queue strategy rrmemory now has a predictable order similar to strategy
  rrordered. Members will be called in the order that they are added to the
  queue.

From 1.8.13 to 1.8.14:
* permitdirectmedia/denydirectmedia now controls whether peers can be
  bridged via directmedia by comparing the ACL to the bridging peer's
  address rather than its own address.

From 1.8.12 to 1.8.13:
* The complex processor detection and optimization has been removed from
  the makefile in favor of using native optimization suppport when available.
  BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".

From 1.8.11 to 1.8.12:
* In AEL dialplans, the "h" extension will now be inherited from prior
  calling contexts, just as it had in 1.4.  If you have created an AEL
  dialplan from scratch in earlier versions of 1.8, you may want to
  check that the execution of "h" extensions in earlier contexts is what
  you want.  If you want to interrupt this functionality, simply placing
  an "h" extension in the macro where you want no inheritance to take
  place should be sufficient.

From 1.8.10 to 1.8.11:

* The BLINDTRANSFER channel variable is deleted from a channel when it is
  bridged to prevent subtle bugs in the parking feature.  The channel
  variable is used by Asterisk internally for the Park application to work
  properly.  If you were using it for your own purposes, copy it to your
  own channel variable before the channel is bridged.

* If no transport is specified in sip.conf, transport will default to UDP.
  Also, if multiple transport= lines are used, only the last will be used.

From 1.6.2 to 1.8:

* When using TLS with Manager and the HTTP server, the desired port
  must be specified in the tlsbindaddr setting. If no port is specified,
  then the default port will be used. See the sample config file to know
  the default ports. Settings like "sslbindport" and "tlsbindport" have
  no effect.

* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
  This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
  This carries a performance penalty.

* Asterisk now requires libpri 1.4.11+ for PRI support.

* A couple of CLI commands in res_ais were changed back to their original form:
    "ais show clm members" --> "ais clm show members"
    "ais show evt event channels" --> "ais evt show event channels"

* The default value for 'autofill' and 'shared_lastcall' in queues.conf has
  been changed to 'yes'.

* The default value for the alwaysauthreject option in sip.conf has been changed
  from "no" to "yes".

* The behavior of the 'parkedcallstimeout' has changed slightly.  The formulation
  of the extension name that a timed out parked call is delivered to when this
  option is set to 'no' was modified such that instead of converting '/' to '0',
  the '/' is converted to an underscore '_'.  See the updated documentation in
  features.conf.sample for more information on the behavior of the
  'parkedcallstimeout' option.

* Asterisk-addons no longer exists as an independent package.  Those modules
  now live in the addons directory of the main Asterisk source tree.  They
  are not enabled by default.  For more information about why modules live in
  addons, see README-addons.txt.

* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
  users of this channel in the tree have been converted to LOG_NOTICE or removed
  (in cases where the same message was already generated to another channel).

* The usage of RTP inside of Asterisk has now become modularized. This means
  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
  If you are not using autoload=yes in modules.conf you will need to ensure
  it is set to load. If not, then any module which uses RTP (such as chan_sip)
  will not be able to send or receive calls.

* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still 
  remains. It now exists within app_chanspy.c and retains the exact same 
  functionality as before. 

* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
  1.6 behavior by default, if there is no [compat] section in asterisk.conf.  In
  prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
  Specifically, that means that pbx_realtime and res_agi expect you to use commas
  to separate arguments in applications, and Set only takes a single pair of
  a variable name/value.  The old 1.4 behavior may still be obtained by setting
  app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
  asterisk.conf.

* The PRI channels in chan_dahdi can no longer change the channel name if a
  different B channel is selected during call negotiation.  To prevent using
  the channel name to infer what B channel a call is using and to avoid name
  collisions, the channel name format is changed.
  The new channel naming for PRI channels is:
  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>

* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type)
  so the dialplan can determine the B channel currently in use by the channel.
  Use CHANNEL(no_media_path) to determine if the channel even has a B channel.

* Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk
  channel so AMI applications can passively determine the B channel currently
  in use.  Calls with "no-media" as the DAHDIChannel do not have an associated
  B channel.  No-media calls are either on hold or call-waiting.

* The ChanIsAvail application has been changed so the AVAILSTATUS variable
  no longer contains both the device state and cause code. The cause code
  is now available in the AVAILCAUSECODE variable. If existing dialplan logic
  is written to expect AVAILSTATUS to contain the cause code it needs to be
  changed to use AVAILCAUSECODE.

* ExternalIVR will now send Z events for invalid or missing files, T events
  now include the interrupted file and bugs in argument parsing have been
  fixed so there may be arguments specified in incorrect ways that were
  working that will no longer work. Please see 
  https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.

* OSP lookup application changes following variable names:
  OSPPEERIP to OSPINPEERIP
  OSPTECH to OSPOUTTECH
  OSPDEST to OSPDESTINATION
  OSPCALLING to OSPOUTCALLING
  OSPCALLED to OSPOUTCALLED
  OSPRESULTS to OSPDESTREMAILS

* The Manager event 'iax2 show peers' output has been updated.  It now has a
  similar output of 'sip show peers'.

* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
  of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
  the current dialplan context.

* The CALLERPRES() dialplan function is deprecated in favor of
  CALLERID(num-pres) and CALLERID(name-pres).

* Environment variables that start with "AST_" are reserved to the system and
  may no longer be set from the dialplan.

* When a call is redirected inside of a Dial, the app and appdata fields of the
  CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.

* The CDR handling of billsec and duration field has changed. If your table
  definition specifies those fields as float,double or similar they will now
  be logged with microsecond accuracy instead of a whole integer.

* chan_sip will no longer set up a local call forward when receiving a
  482 Loop Detected response. The dialplan will just continue from where it
  left off.

* The 'stunaddr' option has been removed from chan_sip.  This feature did not
  behave as expected, had no correct use case, and was not RFC compliant. The
  removal of this feature will hopefully be followed by a correct RFC compliant
  STUN implementation in chan_sip in the future.

* The default value for the pedantic option in sip.conf has been changed
  from "no" to "yes".

* The ConnectedLineNum and ConnectedLineName headers were added to many AMI
  events/responses if the CallerIDNum/CallerIDName headers were also present.
  The addition of connected line support changes the behavior of the channel
  caller ID somewhat.  The channel caller ID value no longer time shares with
  the connected line ID on outgoing call legs.  The timing of some AMI
  events/responses output the connected line ID as caller ID.  These party ID's
  are now separate.

* The Dial application d and H options do not automatically answer the call
  anymore.  It broke DTMF attended transfers.  Since many SIP and ISDN phones
  cannot send DTMF before a call is connected, you need to answer the call
  leg to those phones before using Dial with these options for them to have
  any effect before the dialed party answers.

* The outgoing directory (where .call files are read) now uses inotify to
  detect file changes instead of polling the directory on a regular basis.
  If your outgoing folder is on a NFS mount or another network file system,
  changes to the files will not be detected.  You can revert to polling the
  directory by specifying --without-inotify to configure before compiling.

* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
  table with type 'user' for user type objects.

* The sip.conf allowoverlap option now accepts 'dtmf' as a value.  If you
  are using the early media DTMF overlap dialing method you now need to set
  allowoverlap=dtmf.

From 1.6.1 to 1.6.2:

* SIP no longer sends the 183 progress message for early media by
  default.  Applications requiring early media should use the
  progress() dialplan app to generate the progress message. 

* The firmware for the IAXy has been removed from Asterisk.  It can be
  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
  install the firmware into its proper location, place the firmware in the
  contrib/firmware/iax/ directory in the Asterisk source tree before running
  "make install".

* T.38 FAX error correction mode can no longer be configured in udptl.conf;
  instead, it is configured on a per-peer (or global) basis in sip.conf, with
  the same default as was present in udptl.conf.sample.

* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
  instead, it is either supplied by the application servicing the T.38 channel
  (for a FAX send or receive) or calculated from the bridged endpoint's
  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
  allows for overriding the value supplied by a remote endpoint, which is useful
  when T.38 connections are made to gateways that supply incorrectly-calculated
  maximum datagram sizes.

* There have been some changes to the IAX2 protocol to address the security
  concerns documented in the security advisory AST-2009-006.  Please see the
  IAX2 security document, doc/IAX2-security.pdf, for information regarding
  backwards compatibility with versions of Asterisk that do not contain these
  changes to IAX2.

* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
  has been renamed to 'directmedia', to better reflect what it actually does.
  In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
  starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
  option never had any effect on these cases, it only affected the re-INVITEs
  used for direct media path setup. For MGCP and Skinny, the option was poorly
  named because those protocols don't even use INVITE messages at all. For
  backwards compatibility, the old option is still supported in both normal
  and Realtime configuration files, but all of the sample configuration files,
  Realtime/LDAP schemas, and other documentation refer to it using the new name.

* The default console now will use colors according to the default background
  color, instead of forcing the background color to black.  If you are using a
  light colored background for your console, you may wish to use the option
  flag '-W' to present better color choices for the various messages.  However,
  if you'd prefer the old method of forcing colors to white text on a black
  background, the compatibility option -B is provided for this purpose.

* SendImage() no longer hangs up the channel on transmission error or on
  any other error; in those cases, a FAILURE status is stored in
  SENDIMAGESTATUS and dialplan execution continues.  The possible
  return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
  UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
  has been replaced with 'UNSUPPORTED').  This change makes the
  SendImage application more consistent with other applications.

* skinny.conf now has separate sections for lines and devices.
  Please have a look at configs/skinny.conf.sample and update
  your skinny.conf.

* Queue names previously were treated in a case-sensitive manner,
  meaning that queues with names like "sales" and "sALeS" would be
  seen as unique queues. The parsing logic has changed to use
  case-insensitive comparisons now when originally hashing based on
  queue names, meaning that now the two queues mentioned as examples
  earlier will be seen as having the same name.

* The SPRINTF() dialplan function has been moved into its own module,
  func_sprintf, and is no longer included in func_strings. If you use this
  function and do not use 'autoload=yes' in modules.conf, you will need
  to explicitly load func_sprintf for it to be available.

* The res_indications module has been removed.  Its functionality was important
  enough that most of it has been moved into the Asterisk core.
  Two applications previously provided by res_indications, PlayTones and
  StopPlayTones, have been moved into a new module, app_playtones.

* Support for Taiwanese was incorrectly supported with the "tw" language code.
  In reality, the "tw" language code is reserved for the Twi language, native
  to Ghana.  If you were previously using the "tw" language code, you should
  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
  specific localizations.  Additionally, "mx" should be changed to "es_MX",
  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
  "cs", not "cz".

* DAHDISendCallreroutingFacility() parameters are now comma-separated,
  instead of the old pipe.

* res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
  that would end up being interpreted as a bug once Asterisk started removing 
  the contacts from a user list.

* The cdr.conf file must exist and be configured correctly in order for CDR
  records to be written.

* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,
  which should cover most uses of the extended ASCII set.  If your strings
  use a different encoding in Asterisk, the "encoding" parameter may be set
  to specify the correct character set.

From 1.6.0.1 to 1.6.1:

* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
  API calls were added in 1.6.0, so that modules that provide multiple
  AGI commands could register/unregister them all with a single
  step. However, these API calls were not implemented properly, and did
  not allow the caller to know whether registration or unregistration
  succeeded or failed. They have been redefined to now return success
  or failure, but this means any code using these functions will need
  be recompiled after upgrading to a version of Asterisk containing
  these changes. In addition, the source code using these functions
  should be reviewed to ensure it can properly react to failure
  of registration or unregistration of its API commands.

* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
  to better match what it really does, and the argument order has been
  changed to be consistent with other API calls that perform similar
  operations.

From 1.6.0.x to 1.6.1:

* In previous versions of Asterisk, due to the way objects were arranged in
  memory by chan_sip, the order of entries in sip.conf could be adjusted to
  control the behavior of matching against peers and users.  The way objects
  are managed has been significantly changed for reasons involving performance
  and stability.  A side effect of these changes is that the order of entries
  in sip.conf can no longer be relied upon to control behavior.

* The following core commands dealing with dialplan have been deprecated: 'core
  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
  instead.

* In the dialplan expression parser, the logical value of spaces
  immediately preceding a standalone 0 previously evaluated to
  true. It now evaluates to false.  This has confused a good many
  people in the past (typically because they failed to realize the
  space had any significance).  Since this violates the Principle of
  Least Surprise, it has been changed.

* While app_directory has always relied on having a voicemail.conf or users.conf file
  correctly set up, it now is dependent on app_voicemail being compiled as well.

* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
  and you should start using that function instead for retrieving information about
  the channel in a technology-agnostic way.

* If you have any third party modules which use a config file variable whose
  name ends in a '+', please note that the append capability added to this
  version may now conflict with that variable naming scheme.  An easy
  workaround is to ensure that a space occurs between the '+' and the '=',
  to differentiate your variable from the append operator.  This potential
  conflict is unlikely, but is documented here to be thorough.

* The "Join" event from app_queue now uses the CallerIDNum header instead of
  the CallerID header to indicate the CallerID number.

* If you use ODBC storage for voicemail, there is a new field called "flag"
  which should be a char(8) or larger.  This field specifies whether or not a
  message has been designated to be "Urgent", "PRIORITY", or not.