[asterisk-commits] oej: branch oej/pinequeue-trunk r387411 - in /team/oej/pinequeue-trunk: ./ ad...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 2 06:46:15 CDT 2013
Author: oej
Date: Thu May 2 06:45:29 2013
New Revision: 387411
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=387411
Log:
Update, resolve conflicts, reset automerge.
Added:
team/oej/pinequeue-trunk/UPGRADE-11.txt
- copied unchanged from r387369, trunk/UPGRADE-11.txt
team/oej/pinequeue-trunk/apps/app_stasis.c
- copied unchanged from r387369, trunk/apps/app_stasis.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_empty.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_empty.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_inactive.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_inactive.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_multi.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_multi.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_multi_marked.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_multi_marked.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_single.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_single.c
team/oej/pinequeue-trunk/apps/confbridge/conf_state_single_marked.c
- copied unchanged from r387369, trunk/apps/confbridge/conf_state_single_marked.c
team/oej/pinequeue-trunk/apps/confbridge/include/conf_state.h
- copied unchanged from r387369, trunk/apps/confbridge/include/conf_state.h
team/oej/pinequeue-trunk/build_tools/find_missing_support_level
- copied unchanged from r387369, trunk/build_tools/find_missing_support_level
team/oej/pinequeue-trunk/build_tools/get_documentation.py
- copied unchanged from r387369, trunk/build_tools/get_documentation.py
team/oej/pinequeue-trunk/build_tools/post_process_documentation.py
- copied unchanged from r387369, trunk/build_tools/post_process_documentation.py
team/oej/pinequeue-trunk/channels/chan_gulp.c
- copied unchanged from r387369, trunk/channels/chan_gulp.c
team/oej/pinequeue-trunk/channels/chan_motif.c
- copied unchanged from r387369, trunk/channels/chan_motif.c
team/oej/pinequeue-trunk/channels/iax2/ (props changed)
- copied from r387369, trunk/channels/iax2/
team/oej/pinequeue-trunk/doc/CODING-GUIDELINES
- copied unchanged from r387369, trunk/doc/CODING-GUIDELINES
team/oej/pinequeue-trunk/funcs/func_hangupcause.c
- copied unchanged from r387369, trunk/funcs/func_hangupcause.c
team/oej/pinequeue-trunk/funcs/func_presencestate.c
- copied unchanged from r387369, trunk/funcs/func_presencestate.c
team/oej/pinequeue-trunk/tests/test_abstract_jb.c
- copied unchanged from r387369, trunk/tests/test_abstract_jb.c
team/oej/pinequeue-trunk/tests/test_astobj2_thrash.c
- copied unchanged from r387369, trunk/tests/test_astobj2_thrash.c
team/oej/pinequeue-trunk/tests/test_hashtab_thrash.c
- copied unchanged from r387369, trunk/tests/test_hashtab_thrash.c
team/oej/pinequeue-trunk/tests/test_json.c
- copied unchanged from r387369, trunk/tests/test_json.c
team/oej/pinequeue-trunk/tests/test_res_stasis.c
- copied unchanged from r387369, trunk/tests/test_res_stasis.c
team/oej/pinequeue-trunk/tests/test_scoped_lock.c
- copied unchanged from r387369, trunk/tests/test_scoped_lock.c
team/oej/pinequeue-trunk/tests/test_sorcery.c
- copied unchanged from r387369, trunk/tests/test_sorcery.c
team/oej/pinequeue-trunk/tests/test_sorcery_astdb.c
- copied unchanged from r387369, trunk/tests/test_sorcery_astdb.c
team/oej/pinequeue-trunk/tests/test_sorcery_realtime.c
- copied unchanged from r387369, trunk/tests/test_sorcery_realtime.c
team/oej/pinequeue-trunk/tests/test_stasis.c
- copied unchanged from r387369, trunk/tests/test_stasis.c
team/oej/pinequeue-trunk/tests/test_stasis_channels.c
- copied unchanged from r387369, trunk/tests/test_stasis_channels.c
team/oej/pinequeue-trunk/tests/test_stasis_http.c
- copied unchanged from r387369, trunk/tests/test_stasis_http.c
team/oej/pinequeue-trunk/tests/test_taskprocessor.c
- copied unchanged from r387369, trunk/tests/test_taskprocessor.c
team/oej/pinequeue-trunk/tests/test_threadpool.c
- copied unchanged from r387369, trunk/tests/test_threadpool.c
team/oej/pinequeue-trunk/tests/test_uuid.c
- copied unchanged from r387369, trunk/tests/test_uuid.c
team/oej/pinequeue-trunk/tests/test_voicemail_api.c
- copied unchanged from r387369, trunk/tests/test_voicemail_api.c
team/oej/pinequeue-trunk/tests/test_xml_escape.c
- copied unchanged from r387369, trunk/tests/test_xml_escape.c
Removed:
team/oej/pinequeue-trunk/channels/iax2-parser.c
team/oej/pinequeue-trunk/channels/iax2-parser.h
team/oej/pinequeue-trunk/channels/iax2-provision.c
team/oej/pinequeue-trunk/channels/iax2-provision.h
team/oej/pinequeue-trunk/channels/iax2.h
Modified:
team/oej/pinequeue-trunk/ (props changed)
team/oej/pinequeue-trunk/CHANGES
team/oej/pinequeue-trunk/CREDITS
team/oej/pinequeue-trunk/Makefile
team/oej/pinequeue-trunk/README-SERIOUSLY.bestpractices.txt
team/oej/pinequeue-trunk/UPGRADE-10.txt
team/oej/pinequeue-trunk/UPGRADE.txt
team/oej/pinequeue-trunk/addons/Makefile
team/oej/pinequeue-trunk/addons/app_mysql.c
team/oej/pinequeue-trunk/addons/cdr_mysql.c
team/oej/pinequeue-trunk/addons/chan_mobile.c
team/oej/pinequeue-trunk/addons/chan_ooh323.c
team/oej/pinequeue-trunk/addons/format_mp3.c
team/oej/pinequeue-trunk/addons/ooh323c/src/decode.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooCalls.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooCalls.h
team/oej/pinequeue-trunk/addons/ooh323c/src/ooCapability.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooCmdChannel.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooGkClient.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooLogChan.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooLogChan.h
team/oej/pinequeue-trunk/addons/ooh323c/src/ooSocket.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooStackCmds.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooStackCmds.h
team/oej/pinequeue-trunk/addons/ooh323c/src/ooTimer.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooh245.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooh245.h
team/oej/pinequeue-trunk/addons/ooh323c/src/ooh323.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooh323ep.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooq931.c
team/oej/pinequeue-trunk/addons/ooh323c/src/ooq931.h
team/oej/pinequeue-trunk/addons/ooh323c/src/perutil.c
team/oej/pinequeue-trunk/addons/ooh323cDriver.c
team/oej/pinequeue-trunk/addons/res_config_mysql.c
team/oej/pinequeue-trunk/agi/Makefile
team/oej/pinequeue-trunk/apps/Makefile
team/oej/pinequeue-trunk/apps/app_adsiprog.c
team/oej/pinequeue-trunk/apps/app_alarmreceiver.c
team/oej/pinequeue-trunk/apps/app_amd.c
team/oej/pinequeue-trunk/apps/app_celgenuserevent.c
team/oej/pinequeue-trunk/apps/app_chanspy.c
team/oej/pinequeue-trunk/apps/app_confbridge.c
team/oej/pinequeue-trunk/apps/app_controlplayback.c
team/oej/pinequeue-trunk/apps/app_db.c
team/oej/pinequeue-trunk/apps/app_dial.c
team/oej/pinequeue-trunk/apps/app_dictate.c
team/oej/pinequeue-trunk/apps/app_directed_pickup.c
team/oej/pinequeue-trunk/apps/app_directory.c
team/oej/pinequeue-trunk/apps/app_externalivr.c
team/oej/pinequeue-trunk/apps/app_fax.c
team/oej/pinequeue-trunk/apps/app_festival.c
team/oej/pinequeue-trunk/apps/app_followme.c
team/oej/pinequeue-trunk/apps/app_getcpeid.c
team/oej/pinequeue-trunk/apps/app_ices.c
team/oej/pinequeue-trunk/apps/app_jack.c
team/oej/pinequeue-trunk/apps/app_macro.c
team/oej/pinequeue-trunk/apps/app_meetme.c
team/oej/pinequeue-trunk/apps/app_minivm.c
team/oej/pinequeue-trunk/apps/app_mixmonitor.c
team/oej/pinequeue-trunk/apps/app_originate.c
team/oej/pinequeue-trunk/apps/app_osplookup.c
team/oej/pinequeue-trunk/apps/app_page.c
team/oej/pinequeue-trunk/apps/app_parkandannounce.c
team/oej/pinequeue-trunk/apps/app_playback.c
team/oej/pinequeue-trunk/apps/app_queue.c
team/oej/pinequeue-trunk/apps/app_record.c
team/oej/pinequeue-trunk/apps/app_saycounted.c (props changed)
team/oej/pinequeue-trunk/apps/app_senddtmf.c
team/oej/pinequeue-trunk/apps/app_skel.c
team/oej/pinequeue-trunk/apps/app_sms.c
team/oej/pinequeue-trunk/apps/app_speech_utils.c
team/oej/pinequeue-trunk/apps/app_stack.c
team/oej/pinequeue-trunk/apps/app_system.c
team/oej/pinequeue-trunk/apps/app_userevent.c
team/oej/pinequeue-trunk/apps/app_voicemail.c
team/oej/pinequeue-trunk/apps/app_waitforring.c
team/oej/pinequeue-trunk/apps/app_while.c
team/oej/pinequeue-trunk/apps/confbridge/conf_config_parser.c
team/oej/pinequeue-trunk/apps/confbridge/include/confbridge.h
team/oej/pinequeue-trunk/bridges/Makefile
team/oej/pinequeue-trunk/bridges/bridge_builtin_features.c
team/oej/pinequeue-trunk/bridges/bridge_multiplexed.c
team/oej/pinequeue-trunk/bridges/bridge_simple.c
team/oej/pinequeue-trunk/bridges/bridge_softmix.c
team/oej/pinequeue-trunk/build_tools/cflags-devmode.xml
team/oej/pinequeue-trunk/build_tools/cflags.xml
team/oej/pinequeue-trunk/build_tools/make_buildopts_h
team/oej/pinequeue-trunk/build_tools/make_linker_version_script
team/oej/pinequeue-trunk/build_tools/make_version
team/oej/pinequeue-trunk/build_tools/menuselect-deps.in
team/oej/pinequeue-trunk/build_tools/mkpkgconfig
team/oej/pinequeue-trunk/build_tools/sha1sum-sh (props changed)
team/oej/pinequeue-trunk/cel/Makefile
team/oej/pinequeue-trunk/cel/cel_odbc.c
team/oej/pinequeue-trunk/cel/cel_pgsql.c
team/oej/pinequeue-trunk/cel/cel_radius.c
team/oej/pinequeue-trunk/cel/cel_sqlite3_custom.c
team/oej/pinequeue-trunk/channels/Makefile
team/oej/pinequeue-trunk/channels/chan_agent.c
team/oej/pinequeue-trunk/channels/chan_alsa.c
team/oej/pinequeue-trunk/channels/chan_bridge.c
team/oej/pinequeue-trunk/channels/chan_console.c
team/oej/pinequeue-trunk/channels/chan_dahdi.c
team/oej/pinequeue-trunk/channels/chan_gtalk.c
team/oej/pinequeue-trunk/channels/chan_h323.c
team/oej/pinequeue-trunk/channels/chan_iax2.c
team/oej/pinequeue-trunk/channels/chan_jingle.c
team/oej/pinequeue-trunk/channels/chan_local.c
team/oej/pinequeue-trunk/channels/chan_mgcp.c
team/oej/pinequeue-trunk/channels/chan_misdn.c
team/oej/pinequeue-trunk/channels/chan_multicast_rtp.c (props changed)
team/oej/pinequeue-trunk/channels/chan_oss.c
team/oej/pinequeue-trunk/channels/chan_phone.c
team/oej/pinequeue-trunk/channels/chan_sip.c
team/oej/pinequeue-trunk/channels/chan_skinny.c
team/oej/pinequeue-trunk/channels/chan_unistim.c
team/oej/pinequeue-trunk/channels/chan_vpb.cc
team/oej/pinequeue-trunk/channels/console_board.c
team/oej/pinequeue-trunk/channels/console_gui.c
team/oej/pinequeue-trunk/channels/console_video.c
team/oej/pinequeue-trunk/channels/misdn/chan_misdn_config.h
team/oej/pinequeue-trunk/channels/misdn/ie.c
team/oej/pinequeue-trunk/channels/misdn/isdn_lib.c
team/oej/pinequeue-trunk/channels/misdn/isdn_lib.h
team/oej/pinequeue-trunk/channels/misdn/isdn_msg_parser.c
team/oej/pinequeue-trunk/channels/misdn/portinfo.c
team/oej/pinequeue-trunk/channels/misdn_config.c
team/oej/pinequeue-trunk/channels/sig_analog.c
team/oej/pinequeue-trunk/channels/sig_analog.h
team/oej/pinequeue-trunk/channels/sig_pri.c
team/oej/pinequeue-trunk/channels/sig_pri.h
team/oej/pinequeue-trunk/channels/sig_ss7.c (contents, props changed)
team/oej/pinequeue-trunk/channels/sig_ss7.h (contents, props changed)
team/oej/pinequeue-trunk/channels/sip/config_parser.c
team/oej/pinequeue-trunk/channels/sip/dialplan_functions.c
team/oej/pinequeue-trunk/channels/sip/include/dialog.h
team/oej/pinequeue-trunk/channels/sip/include/reqresp_parser.h
team/oej/pinequeue-trunk/channels/sip/include/sdp_crypto.h
team/oej/pinequeue-trunk/channels/sip/include/security_events.h (props changed)
team/oej/pinequeue-trunk/channels/sip/include/sip.h
team/oej/pinequeue-trunk/channels/sip/include/srtp.h
team/oej/pinequeue-trunk/channels/sip/reqresp_parser.c
team/oej/pinequeue-trunk/channels/sip/sdp_crypto.c
team/oej/pinequeue-trunk/channels/sip/security_events.c (contents, props changed)
team/oej/pinequeue-trunk/channels/sip/srtp.c
team/oej/pinequeue-trunk/channels/sip/utils.c
team/oej/pinequeue-trunk/channels/vcodecs.c
team/oej/pinequeue-trunk/channels/vgrabbers.c
team/oej/pinequeue-trunk/config.guess
team/oej/pinequeue-trunk/config.sub
team/oej/pinequeue-trunk/configure
team/oej/pinequeue-trunk/configure.ac
team/oej/pinequeue-trunk/doc/Makefile (props changed)
team/oej/pinequeue-trunk/doc/README.txt
team/oej/pinequeue-trunk/doc/appdocsxml.dtd
team/oej/pinequeue-trunk/funcs/Makefile
team/oej/pinequeue-trunk/funcs/func_audiohookinherit.c
team/oej/pinequeue-trunk/funcs/func_callerid.c
team/oej/pinequeue-trunk/funcs/func_channel.c
team/oej/pinequeue-trunk/funcs/func_curl.c
team/oej/pinequeue-trunk/funcs/func_cut.c
team/oej/pinequeue-trunk/funcs/func_devstate.c
team/oej/pinequeue-trunk/funcs/func_dialgroup.c
team/oej/pinequeue-trunk/funcs/func_frame_trace.c
team/oej/pinequeue-trunk/funcs/func_global.c
team/oej/pinequeue-trunk/funcs/func_jitterbuffer.c
team/oej/pinequeue-trunk/funcs/func_lock.c
team/oej/pinequeue-trunk/funcs/func_logic.c
team/oej/pinequeue-trunk/funcs/func_math.c
team/oej/pinequeue-trunk/funcs/func_odbc.c
team/oej/pinequeue-trunk/funcs/func_realtime.c
team/oej/pinequeue-trunk/funcs/func_shell.c
team/oej/pinequeue-trunk/funcs/func_speex.c
team/oej/pinequeue-trunk/funcs/func_strings.c
team/oej/pinequeue-trunk/funcs/func_volume.c
team/oej/pinequeue-trunk/pbx/Makefile
team/oej/pinequeue-trunk/pbx/dundi-parser.c
team/oej/pinequeue-trunk/pbx/pbx_ael.c
team/oej/pinequeue-trunk/pbx/pbx_config.c
team/oej/pinequeue-trunk/pbx/pbx_dundi.c
team/oej/pinequeue-trunk/pbx/pbx_lua.c
team/oej/pinequeue-trunk/pbx/pbx_realtime.c
team/oej/pinequeue-trunk/pbx/pbx_spool.c
team/oej/pinequeue-trunk/sounds/Makefile
team/oej/pinequeue-trunk/sounds/sounds.xml
team/oej/pinequeue-trunk/static-http/ajamdemo.html
team/oej/pinequeue-trunk/static-http/astman.css
team/oej/pinequeue-trunk/static-http/mantest.html
team/oej/pinequeue-trunk/tests/Makefile
team/oej/pinequeue-trunk/tests/test_acl.c
team/oej/pinequeue-trunk/tests/test_astobj2.c
team/oej/pinequeue-trunk/tests/test_config.c
team/oej/pinequeue-trunk/tests/test_db.c
team/oej/pinequeue-trunk/tests/test_devicestate.c
team/oej/pinequeue-trunk/tests/test_event.c
team/oej/pinequeue-trunk/tests/test_expr.c (props changed)
team/oej/pinequeue-trunk/tests/test_func_file.c (props changed)
team/oej/pinequeue-trunk/tests/test_gosub.c
team/oej/pinequeue-trunk/tests/test_linkedlists.c
team/oej/pinequeue-trunk/tests/test_locale.c (props changed)
team/oej/pinequeue-trunk/tests/test_poll.c (props changed)
team/oej/pinequeue-trunk/tests/test_stringfields.c
team/oej/pinequeue-trunk/tests/test_strings.c
Propchange: team/oej/pinequeue-trunk/
------------------------------------------------------------------------------
automerge = Is-there-life-off-net?
Propchange: team/oej/pinequeue-trunk/
('svnmerge-integrated' removed)
Modified: team/oej/pinequeue-trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-trunk/CHANGES?view=diff&rev=387411&r1=387410&r2=387411
==============================================================================
--- team/oej/pinequeue-trunk/CHANGES (original)
+++ team/oej/pinequeue-trunk/CHANGES Thu May 2 06:45:29 2013
@@ -7,15 +7,160 @@
=== and the other UPGRADE files for older releases.
===
==============================================================================
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
+------------------------------------------------------------------------------
+
+
+AMI (Asterisk Manager Interface)
+------------------
+ * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
+ in its response if the peer has a subscribe context set.
+
+ * The SIPqualifypeer action now acknowledges the request once it has established
+ that the request is against a known peer. It also issues a new event,
+ 'SIPqualifypeerdone', once the qualify action has been completed.
+
+ * The PlayDTMF action now supports an optional 'Duration' parameter. This
+ specifies the duration of the digit to be played, in milliseconds.
+
+ * Added VoicemailRefresh action to allow an external entity to trigger mailbox
+ updates when changes occur instead of requiring the use of pollmailboxes.
+
+ * CLI Command 'Manager Show Commands' no longer truncates command names longer
+ than 15 characters and no longer shows authorization requirement for commands.
+ 'Manager Show Command' now displays the privileges needed for using a given
+ manager command instead.
+
+ * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
+ client to manipulate audio currently being played back on a channel. The
+ supported operations depend on the application being used to send audio to
+ the channel. When the audio playback was initiated using the ControlPlayback
+ application or CONTROL STREAM FILE AGI command, the audio can be paused,
+ stopped, restarted, reversed, or skipped forward. When initiated by other
+ mechanisms (such as the Playback application), the audio can be stopped,
+ reversed, or skipped forward.
+
+ * Channel related events now contain a snapshot of channel state, adding new
+ fields to many of these events.
+
+ * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
+ in a future release. Please use the common 'Exten' field instead.
+
+ * The AMI event 'UserEvent' from app_userevent now contains the channel state
+ fields. The channel state fields will come before the body fields.
+
+ * The deprecated use of | (pipe) as a separator in the channelvars setting in
+ manager.conf has been removed.
+
+ * Channel Variables conveyed with a channel no longer contain the name of the
+ channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
+ ChanVariable: bar=baz. When multiple channels are present in a single AMI
+ event, the various ChanVariable fields will contain a suffix that specifies
+ which channel they correspond to.
+
+Channel Drivers
+------------------
+
+chan_mobile
+------------------
+ * Added general support for busy detection.
+
+ * Added ECAM command support for Sony Ericsson phones.
+
+chan_sip
+------------------
+ * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
+ using the 'supportpath' setting, either on a global basis or on a peer basis.
+ This setting enables Asterisk to route outgoing out-of-dialog requests via a
+ set of proxies by using a pre-loaded route-set defined by the Path headers in
+ the REGISTER request. See Realtime updates for more configuration information.
+
+Features
+-------------------
+ * The BRIDGE_FEATURES channel variable would previously only set features for
+ the calling party and would set this feature regardless of whether the
+ feature was in caps or in lowercase. Use of a caps feature for a letter
+ will now apply the feature to the calling party while use of a lowercase
+ letter will apply that feature to the called party.
+
+ * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
+
+ * PARKINGSLOT and PARKEDLOT channel variables will now be set for a parked
+ channel even when comebactoorigin=yes
+
+ * You can now have the settings for a channel updated using the FEATURE()
+ and FEATUREMAP() functions inherited to child channels by setting
+ FEATURE(inherit)=yes.
+
+Logging
+-------------------
+ * When performing queue pause/unpause on an interface without specifying an
+ individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
+ least one member of any queue exists for that interface.
+
+ * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
+ for realtime queue log entries.
+
+MeetMe
+-------------------
+* 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.
+
+Queue
+-------------------
+ * Add queue available hint. exten => 8501,hint,Queue:markq_avail
+ Note: the suffix '_avail' after the queuename.
+ Reports 'InUse' for no logged in agents or no free agents.
+ Reports 'Idle' when an agent is free.
+
+Core
+------------------
+ * Redirecting reasons can now be set to arbitrary strings. This means
+ that the REDIRECTING dialplan function can be used to set the redirecting
+ reason to any string. It also allows for custom strings to be read as the
+ redirecting reason from SIP Diversion headers.
+
+Realtime
+------------------
+ * Dynamic realtime tables for SIP Users can now include a 'path' field. This
+ will store the path information for that peer when it registers. Realtime
+ tables can also use the 'supportpath' field to enable Path header support.
+
+ * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
+ objectIdentifier. This maps to the supportpath option in sip.conf.
+
+RTP
+------------------
+ * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
+ them, an Asterisk-specific version of pjproject needs to be installed.
+ Tarballs are available from https://github.com/asterisk/pjproject/tags/.
+
+XMPP
+------------------
+ * Device state for XMPP buddies is now available using the following format:
+ XMPP/<client name>/<buddy address>
+ If any resource is available the device state is considered to be not in use.
+ If no resources exist or all are unavailable the device state is considered
+ to be unavailable.
+
+Sorcery
+------------------
+ * All future modules which utilize Sorcery for object persistence must have a
+ column named "id" within their schema when using the Sorcery realtime module.
+ This column must be able to contain a string of up to 128 characters in length.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
-Core
-----
- * The expression parser now recognizes the ABS() absolute value function,
- which will convert negative floating point values to positive values.
+
+
+Build System
+-------------------
* The Asterisk build system will now build and install a shared library
(libasteriskssl.so) used to wrap various initialization and shutdown functions
from the libssl and libcrypto libraries provided by OpenSSL. This is done so
@@ -23,179 +168,593 @@
modules that are loaded into Asterisk, since they should only be called once
in any single process. If desired, this feature can be disabled by supplying
the "--disable-asteriskssl" option to the configure script.
- * Threads belonging to a particular call are now linked with callids which get
- added to any log messages produced by those threads. Log messages can now be
- easily identified as involved with a certain call by looking at their call id.
- This feature can be disabled in logger.conf with the display_callids option.
- * The minimum DTMF duration can now be configured in asterisk.conf
- as "mindtmfduration". The default value is (as before) set to 80 ms.
- (previously it was only available in source code)
-
-CLI Changes
--------------------
- * mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
- of all running mixmonitors on a channel.
- * The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
- numeric instead of 0, 1, or 2.
+
+ * A new make target, 'full', has been added to the Makefile. This performs
+ the same compilation actions as make all, but will also scan the entirety of
+ each source file for documentation. This option is needed to generate AMI
+ event documentation. Note that your system must have Python in order for
+ this make target to succeed.
+
+ * The optimization portion of the build system has been reworked to avoid
+ broken builds on certain architectures. All architecture-specific
+ optimization has been removed in favor of using -march=native to allow gcc
+ to detect the environment in which it is running when possible. This can
+ be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
+
+ * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
+ make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
+
+ * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
+ previously parsed the header file to obtain the version of Asterisk, you
+ will now have to go through Asterisk to get the version information.
+
+
+Applications
+-------------------
+
+Bridge
+-------------------
+ * Added 'F()' option. Similar to the dial option, this can be supplied with
+ arguments indicating where the callee should go after the caller is hung up,
+ or without options specified, the priority after the Queue will be used.
+
ConfBridge
-------------------
* Added menu action admin_toggle_mute_participants. This will mute / unmute
- all non-admin participants on a conference. The confbridge configuration file
- also allows for the default sounds played to all conference users when this
- occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
- * Added menu action participant_count. This will playback the number of current
- participants in a conference.
- * Added announcement configuration option to user profile. If set the sound file will
- be played to the user, and only the user, upon joining the conference bridge.
+ all non-admin participants on a conference. The confbridge configuration
+ file also allows for the default sounds played to all conference users when
+ this occurs to be overriden using sound_participants_unmuted and
+ sound_participants_muted.
+
+ * Added menu action participant_count. This will playback the number of
+ current participants in a conference.
+
+ * Added announcement configuration option to user profile. If set the sound
+ file will be played to the user, and only the user, upon joining the
+ conference bridge.
+
+ * Added record_file_append option that defaults to "yes", but if set to no
+ will create a new file between each start/stop recording.
+
+
+Dial
+-------------------
+ * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
+ channels respectively before the callee channels are called.
+
+
+ExternalIVR
+-------------------
+ * Added support for IPv6.
+
+ * Add interrupt ('I') command to ExternalIVR. Sending this command from an
+ external process will cause the current playlist to be cleared, including
+ stopping any audio file that is currently playing. This is useful when you
+ want to interrupt audio playback only when specific DTMF is entered by the
+ caller.
+
+
+FollowMe
+-------------------
+ * A new option, 'I' has been added to app_followme. By setting this option,
+ Asterisk will not update the caller with connected line changes when they
+ occur. This is similar to app_dial and app_queue.
+
+ * The 'N' option is now ignored if the call is already answered.
+
+ * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
+ and caller channels respectively before the callee channels are called.
+
+ * The winning FollowMe outgoing call is now put on hold if the caller put it on
+ hold.
+
+
+MixMonitor
+------------------
+ * MixMonitor hooks now have IDs associated with them which can be used to
+ assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
+ will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
+ now accepts that ID as an argument.
+
+ * Added 'm' option, which stores a copy of the recording as a voicemail in the
+ indicated mailboxes.
+
+
+MySQL
+-------------------
+ * The connect action in app_mysql now allows you to specify a port number to
+ connect to. This is useful if you run a MySQL server on a non-standard
+ port number.
+
+
+OSP Applications
+-------------------
+ * Increased the default number of allowed destinations from 5 to 12.
+
+
+Page
+-------------------
+ * The app_page application now no longer depends on DAHDI or app_meetme. It
+ has been re-architected to use app_confbridge internally.
+
+
+Queue
+-------------------
+ * Added queue options autopausebusy and autopauseunavail for automatically
+ pausing a queue member when their device reports busy or congestion.
+
+ * The 'ignorebusy' option for queue members has been deprecated in favor of
+ the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
+ added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
+ per interface basis. Individual ringinuse values can now be set in
+ queues.conf via an argument to member definitions. Lastly, the queue
+ 'ringinuse' setting now only determines defaults for the per member
+ 'ringinuse' setting and does not override per member settings like it does
+ in earlier versions.
+
+ * Added 'F()' option. Similar to the dial option, this can be supplied with
+ arguments indicating where the callee should go after the caller is hung up,
+ or without options specified, the priority after the Queue will be used.
+
+ * Added new option log_member_name_as_agent, which will cause the membername to
+ be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
+ state_interface has been set.
+
+ * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
+
+ * App_queue will now play periodic announcements for the caller that
+ holds the first position in the queue while waiting for answer.
+
+SayUnixTime
+------------------
+ * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
+ when receiving DTMF. Use the 'j' option to enable extension jumping. Also
+ changed arguments to SayUnixTime so that every option is truly optional even
+ when using multiple options (so that j option could be used without having to
+ manually specify timezone and format) There are other benefits, e.g., format
+ can now be used without specifying time zone as well.
+
Voicemail
------------------
- * Addition of the VM_INFO function - see Dialplan function changes
+ * Addition of the VM_INFO function - see Function changes.
+
* The imapserver, imapport, and imapflags configuration options can now be
overriden on a user by user basis.
-SIP Changes
------------
- * Asterisk will no longer substitute CID number for CID name into display
+ * When voicemail plays a message's envelope with saycid set to yes, when
+ reaching the caller id field it will play a recording of a file with the same
+ base name as the sender's callerid if there is a similarly named file in
+ <astspooldir>/recordings/callerids/
+
+ * Voicemails now contains a unique message identifier "msg_id", which is stored
+ in the message envelope with the sound files. IMAP backends will now store
+ the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
+ backends will store the message identifier in a "msg_id" column. See
+ UPGRADE.txt for more information.
+
+ * Added VoiceMailPlayMsg application. This application will play a single
+ voicemail message from a mailbox. The result of the application, SUCCESS or
+ FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
+
+
+Functions
+------------------
+ * Hangup handlers can be attached to channels using the CHANNEL() function.
+ Hangup handlers will run when the channel is hung up similar to the h
+ extension. The hangup_handler_push option will push a GoSub compatible
+ location in the dialplan onto the channel's hangup handler stack. The
+ hangup_handler_pop option will remove the last added location, and optionally
+ replace it with a new GoSub compatible location. The hangup_handler_wipe
+ option will remove all locations on the stack, and optionally add a new
+ location.
+
+ * The expression parser now recognizes the ABS() absolute value function,
+ which will convert negative floating point values to positive values.
+
+ * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
+ control of faxdetect.
+
+ * Addition of the VM_INFO function that can be used to retrieve voicemail
+ user information, such as the email address and full name.
+ The MAILBOX_EXISTS dialplan function has been deprecated in favour of
+ VM_INFO.
+
+ * The REDIRECTING function now supports the redirecting original party id
+ and reason.
+
+ * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
+ lets you set some of the configuration options from the [general] section
+ of features.conf on a per-channel basis. FEATUREMAP() lets you customize
+ the key sequence used to activate built-in features, such as blindxfer,
+ and automon. See the built-in documentation for details.
+
+ * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
+ instead of simply the uri. This is the format that MessageSend() can use
+ in the from parameter for outgoing SIP messages.
+
+ * Added the PRESENCE_STATE function. This allows retrieving presence state
+ information from any presence state provider. It also allows setting
+ presence state information from a CustomPresence presence state provider.
+ See AMI/CLI changes for related commands.
+
+ * Added the AMI_CLIENT function to make manager account attributes available
+ to the dialplan. It currently supports returning the current number of
+ active sessions for a given account.
+
+ * Added support for private party ID information to CALLERID, CONNECTEDLINE,
+ and the REDIRECTING functions.
+
+
+Channel Drivers
+------------------
+
+chan_local
+------------------
+ * Added a manager event "LocalBridge" for local channel call bridges between
+ the two pseudo-channels created.
+
+
+chan_dahdi
+------------------
+ * Added dialtone_detect option for analog ports to disconnect incoming
+ calls when dialtone is detected.
+
+ * Added option colp_send to send ISDN connected line information. Allowed
+ settings are block, to not send any connected line information; connect, to
+ send connected line information on initial connect; and update, to send
+ information on any update during a call. Default is update.
+
+ * Add options namedcallgroup and namedpickupgroup to support installations
+ where a higher number of groups (>64) is required.
+
+ * Added support to use private party ID information with PRI calls.
+
+
+chan_motif
+------------------
+ * A new channel driver named chan_motif has been added which provides support for
+ Google Talk and Jingle in a single channel driver. This new channel driver includes
+ support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
+ hold, unhold, and ringing notification. It is also compliant with the current Jingle
+ specification, current Google Jingle specification, and the original Google Talk
+ protocol.
+
+
+chan_ooh323
+------------------
+ * Added NAT support for RTP. Setting in config is 'nat', which can be set
+ globally and overriden on a peer by peer basis.
+
+ * Direct media functionality has been added. Options in config are:
+ directmedia (directrtp) and directrtpsetup (earlydirect)
+
+ * ChannelUpdate events now contain a CallRef header.
+
+
+chan_sip
+------------------
+ * Asterisk will no longer substitute CID number for CID name in the display
name field if CID number exists without a CID name. This change improves
compatibility with certain device features such as Avaya IP500's directory
lookup service.
+
* A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
created using that setting to not be removed during SIP reload.
- * Add support to realtime for the 'callbackextension' option
+
+ * Added settings recordonfeature and recordofffeature. When receiving an INFO
+ request with a "Record:" header, this will turn the requested feature on/off.
+ Allowed values are 'automon', 'automixmon', and blank to disable. Note that
+ dynamic features must be enabled and configured properly on the requesting
+ channel for this to function properly.
+
+ * Add support to realtime for the 'callbackextension' option.
+
* When multiple peers exist with the same address, but differing
callbackextension options, incoming requests that are matched by address
will be matched to the peer with the matching callbackextension if it is
available.
- * NAT settings are now a combinable list of options. The equivalent of the
- deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
+
* Two new NAT options, auto_force_rport and auto_comedia, have been added
which set the force_rport and comedia options automatically if Asterisk
detects that an incoming SIP request crossed a NAT after being sent by
the remote endpoint.
+
+ * The default global nat setting in sip.conf has been changed from force_rport
+ to auto_force_rport.
+
+ * NAT settings are now a combinable list of options. The equivalent of the
+ deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
+
* Adds an option send_diversion which can be disabled to prevent
- diversion headers from automatically being added to invites.
+ diversion headers from automatically being added to INVITE requests.
+
* Add support for lightweight NAT keepalive. If enabled a blank packet will
be sent to the remote host at a given interval to keep the NAT mapping open.
This can be enabled using the keepalive configuration option.
-Chan_local changes
-------------------
- * Added a manager event "LocalBridge" for local channel call bridges between
- the two pseudo-channels created.
-
-Chan_dahdi changes
-------------------
- * Added dialtone_detect option for analog ports to disconnect incoming
- calls when dialtone is detected.
-
-Chan_unistim changes
+ * Add option 'tonezone' to specify country code for indications. This option
+ can be set both globally and overridden for specific peers.
+
+ * The SIP Security Events Framework now supports IPv6.
+
+ * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
+ between multiple user agents. When set, for directmedia reinvites,
+ Asterisk will not send an immediate reinvite on an incoming call leg. This
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