[svn-commits] oej: branch group/edv-appleraisin-trunk r364041 - in /team/group/edv-applerai...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Apr 26 10:07:28 CDT 2012
Author: oej
Date: Thu Apr 26 10:06:55 2012
New Revision: 364041
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=364041
Log:
Hrrmmpf.
Added:
team/group/edv-appleraisin-trunk/UPGRADE-10.txt
- copied unchanged from r363877, trunk/UPGRADE-10.txt
team/group/edv-appleraisin-trunk/channels/sip/include/security_events.h
- copied unchanged from r363877, trunk/channels/sip/include/security_events.h
team/group/edv-appleraisin-trunk/channels/sip/security_events.c
- copied unchanged from r363877, trunk/channels/sip/security_events.c
team/group/edv-appleraisin-trunk/channels/sip/utils.c
- copied unchanged from r363877, trunk/channels/sip/utils.c
team/group/edv-appleraisin-trunk/tests/test_jitterbuf.c
- copied unchanged from r363877, trunk/tests/test_jitterbuf.c
team/group/edv-appleraisin-trunk/tests/test_linkedlists.c
- copied unchanged from r363877, trunk/tests/test_linkedlists.c
team/group/edv-appleraisin-trunk/tests/test_netsock2.c
- copied unchanged from r363877, trunk/tests/test_netsock2.c
Removed:
team/group/edv-appleraisin-trunk/apps/app_rpt.c
team/group/edv-appleraisin-trunk/apps/rpt_flow.pdf
team/group/edv-appleraisin-trunk/build_tools/make_version_h
team/group/edv-appleraisin-trunk/channels/chan_usbradio.c
team/group/edv-appleraisin-trunk/channels/xpmr/
Modified:
team/group/edv-appleraisin-trunk/ (props changed)
team/group/edv-appleraisin-trunk/BUGS
team/group/edv-appleraisin-trunk/CHANGES
team/group/edv-appleraisin-trunk/CREDITS
team/group/edv-appleraisin-trunk/Makefile
team/group/edv-appleraisin-trunk/UPGRADE.txt
team/group/edv-appleraisin-trunk/addons/ (props changed)
team/group/edv-appleraisin-trunk/addons/app_mysql.c
team/group/edv-appleraisin-trunk/addons/app_saycountpl.c
team/group/edv-appleraisin-trunk/addons/cdr_mysql.c
team/group/edv-appleraisin-trunk/addons/chan_mobile.c
team/group/edv-appleraisin-trunk/addons/chan_ooh323.c
team/group/edv-appleraisin-trunk/addons/format_mp3.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/dlist.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/dlist.h
team/group/edv-appleraisin-trunk/addons/ooh323c/src/memheap.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooCalls.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooCalls.h
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooCmdChannel.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooGkClient.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooLogChan.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooTimer.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/oochannels.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooh245.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooh323.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooq931.c
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ooq931.h
team/group/edv-appleraisin-trunk/addons/ooh323c/src/ootypes.h
team/group/edv-appleraisin-trunk/addons/ooh323c/src/printHandler.c
team/group/edv-appleraisin-trunk/addons/ooh323cDriver.c
team/group/edv-appleraisin-trunk/addons/res_config_mysql.c
team/group/edv-appleraisin-trunk/agi/Makefile
team/group/edv-appleraisin-trunk/agi/agi.xml
team/group/edv-appleraisin-trunk/agi/eagi-test.c
team/group/edv-appleraisin-trunk/apps/ (props changed)
team/group/edv-appleraisin-trunk/apps/app_adsiprog.c
team/group/edv-appleraisin-trunk/apps/app_alarmreceiver.c
team/group/edv-appleraisin-trunk/apps/app_amd.c
team/group/edv-appleraisin-trunk/apps/app_authenticate.c
team/group/edv-appleraisin-trunk/apps/app_cdr.c
team/group/edv-appleraisin-trunk/apps/app_chanisavail.c
team/group/edv-appleraisin-trunk/apps/app_channelredirect.c
team/group/edv-appleraisin-trunk/apps/app_chanspy.c
team/group/edv-appleraisin-trunk/apps/app_confbridge.c
team/group/edv-appleraisin-trunk/apps/app_controlplayback.c
team/group/edv-appleraisin-trunk/apps/app_dahdibarge.c
team/group/edv-appleraisin-trunk/apps/app_dahdiras.c
team/group/edv-appleraisin-trunk/apps/app_db.c
team/group/edv-appleraisin-trunk/apps/app_dial.c
team/group/edv-appleraisin-trunk/apps/app_dictate.c
team/group/edv-appleraisin-trunk/apps/app_directed_pickup.c
team/group/edv-appleraisin-trunk/apps/app_directory.c
team/group/edv-appleraisin-trunk/apps/app_disa.c
team/group/edv-appleraisin-trunk/apps/app_dumpchan.c
team/group/edv-appleraisin-trunk/apps/app_echo.c
team/group/edv-appleraisin-trunk/apps/app_exec.c
team/group/edv-appleraisin-trunk/apps/app_externalivr.c
team/group/edv-appleraisin-trunk/apps/app_fax.c
team/group/edv-appleraisin-trunk/apps/app_festival.c
team/group/edv-appleraisin-trunk/apps/app_flash.c
team/group/edv-appleraisin-trunk/apps/app_followme.c
team/group/edv-appleraisin-trunk/apps/app_forkcdr.c
team/group/edv-appleraisin-trunk/apps/app_getcpeid.c
team/group/edv-appleraisin-trunk/apps/app_ices.c
team/group/edv-appleraisin-trunk/apps/app_image.c
team/group/edv-appleraisin-trunk/apps/app_ivrdemo.c
team/group/edv-appleraisin-trunk/apps/app_jack.c
team/group/edv-appleraisin-trunk/apps/app_macro.c
team/group/edv-appleraisin-trunk/apps/app_meetme.c
team/group/edv-appleraisin-trunk/apps/app_milliwatt.c
team/group/edv-appleraisin-trunk/apps/app_minivm.c
team/group/edv-appleraisin-trunk/apps/app_mixmonitor.c
team/group/edv-appleraisin-trunk/apps/app_morsecode.c
team/group/edv-appleraisin-trunk/apps/app_mp3.c
team/group/edv-appleraisin-trunk/apps/app_nbscat.c
team/group/edv-appleraisin-trunk/apps/app_originate.c
team/group/edv-appleraisin-trunk/apps/app_osplookup.c
team/group/edv-appleraisin-trunk/apps/app_page.c
team/group/edv-appleraisin-trunk/apps/app_parkandannounce.c
team/group/edv-appleraisin-trunk/apps/app_playback.c
team/group/edv-appleraisin-trunk/apps/app_playtones.c
team/group/edv-appleraisin-trunk/apps/app_privacy.c
team/group/edv-appleraisin-trunk/apps/app_queue.c
team/group/edv-appleraisin-trunk/apps/app_read.c
team/group/edv-appleraisin-trunk/apps/app_readexten.c
team/group/edv-appleraisin-trunk/apps/app_readfile.c
team/group/edv-appleraisin-trunk/apps/app_record.c
team/group/edv-appleraisin-trunk/apps/app_saycounted.c
team/group/edv-appleraisin-trunk/apps/app_sayunixtime.c
team/group/edv-appleraisin-trunk/apps/app_senddtmf.c
team/group/edv-appleraisin-trunk/apps/app_sendtext.c
team/group/edv-appleraisin-trunk/apps/app_setcallerid.c
team/group/edv-appleraisin-trunk/apps/app_skel.c
team/group/edv-appleraisin-trunk/apps/app_sms.c
team/group/edv-appleraisin-trunk/apps/app_softhangup.c
team/group/edv-appleraisin-trunk/apps/app_speech_utils.c
team/group/edv-appleraisin-trunk/apps/app_stack.c
team/group/edv-appleraisin-trunk/apps/app_talkdetect.c
team/group/edv-appleraisin-trunk/apps/app_test.c
team/group/edv-appleraisin-trunk/apps/app_transfer.c
team/group/edv-appleraisin-trunk/apps/app_url.c
team/group/edv-appleraisin-trunk/apps/app_userevent.c
team/group/edv-appleraisin-trunk/apps/app_verbose.c
team/group/edv-appleraisin-trunk/apps/app_voicemail.c
team/group/edv-appleraisin-trunk/apps/app_waitforring.c
team/group/edv-appleraisin-trunk/apps/app_waitforsilence.c
team/group/edv-appleraisin-trunk/apps/app_waituntil.c
team/group/edv-appleraisin-trunk/apps/app_while.c
team/group/edv-appleraisin-trunk/apps/app_zapateller.c
team/group/edv-appleraisin-trunk/apps/confbridge/ (props changed)
team/group/edv-appleraisin-trunk/apps/confbridge/conf_config_parser.c
team/group/edv-appleraisin-trunk/apps/confbridge/include/confbridge.h
team/group/edv-appleraisin-trunk/bootstrap.sh
team/group/edv-appleraisin-trunk/bridges/ (props changed)
team/group/edv-appleraisin-trunk/bridges/bridge_builtin_features.c (contents, props changed)
team/group/edv-appleraisin-trunk/bridges/bridge_multiplexed.c (contents, props changed)
team/group/edv-appleraisin-trunk/bridges/bridge_simple.c
team/group/edv-appleraisin-trunk/bridges/bridge_softmix.c
team/group/edv-appleraisin-trunk/build_tools/cflags-devmode.xml
team/group/edv-appleraisin-trunk/build_tools/cflags.xml
team/group/edv-appleraisin-trunk/build_tools/embed_modules.xml
team/group/edv-appleraisin-trunk/build_tools/make_defaults_h
team/group/edv-appleraisin-trunk/build_tools/menuselect-deps.in
team/group/edv-appleraisin-trunk/build_tools/mkpkgconfig
team/group/edv-appleraisin-trunk/build_tools/prep_tarball
team/group/edv-appleraisin-trunk/cel/ (props changed)
team/group/edv-appleraisin-trunk/cel/cel_custom.c
team/group/edv-appleraisin-trunk/cel/cel_manager.c
team/group/edv-appleraisin-trunk/cel/cel_odbc.c (contents, props changed)
team/group/edv-appleraisin-trunk/cel/cel_pgsql.c
team/group/edv-appleraisin-trunk/cel/cel_radius.c
team/group/edv-appleraisin-trunk/cel/cel_sqlite3_custom.c
team/group/edv-appleraisin-trunk/cel/cel_tds.c
team/group/edv-appleraisin-trunk/channels/ (props changed)
team/group/edv-appleraisin-trunk/channels/chan_agent.c
team/group/edv-appleraisin-trunk/channels/chan_alsa.c
team/group/edv-appleraisin-trunk/channels/chan_bridge.c
team/group/edv-appleraisin-trunk/channels/chan_console.c
team/group/edv-appleraisin-trunk/channels/chan_dahdi.c
team/group/edv-appleraisin-trunk/channels/chan_gtalk.c
team/group/edv-appleraisin-trunk/channels/chan_h323.c
team/group/edv-appleraisin-trunk/channels/chan_iax2.c
team/group/edv-appleraisin-trunk/channels/chan_jingle.c
team/group/edv-appleraisin-trunk/channels/chan_local.c
team/group/edv-appleraisin-trunk/channels/chan_mgcp.c
team/group/edv-appleraisin-trunk/channels/chan_misdn.c
team/group/edv-appleraisin-trunk/channels/chan_multicast_rtp.c
team/group/edv-appleraisin-trunk/channels/chan_nbs.c
team/group/edv-appleraisin-trunk/channels/chan_oss.c
team/group/edv-appleraisin-trunk/channels/chan_phone.c
team/group/edv-appleraisin-trunk/channels/chan_sip.c
team/group/edv-appleraisin-trunk/channels/chan_skinny.c
team/group/edv-appleraisin-trunk/channels/chan_unistim.c
team/group/edv-appleraisin-trunk/channels/chan_vpb.cc
team/group/edv-appleraisin-trunk/channels/console_gui.c
team/group/edv-appleraisin-trunk/channels/console_video.c
team/group/edv-appleraisin-trunk/channels/iax2-provision.c
team/group/edv-appleraisin-trunk/channels/misdn/isdn_lib.c
team/group/edv-appleraisin-trunk/channels/misdn/isdn_msg_parser.c
team/group/edv-appleraisin-trunk/channels/sig_analog.c
team/group/edv-appleraisin-trunk/channels/sig_analog.h
team/group/edv-appleraisin-trunk/channels/sig_pri.c
team/group/edv-appleraisin-trunk/channels/sig_pri.h
team/group/edv-appleraisin-trunk/channels/sig_ss7.c
team/group/edv-appleraisin-trunk/channels/sig_ss7.h
team/group/edv-appleraisin-trunk/channels/sip/ (props changed)
team/group/edv-appleraisin-trunk/channels/sip/config_parser.c
team/group/edv-appleraisin-trunk/channels/sip/dialplan_functions.c
team/group/edv-appleraisin-trunk/channels/sip/include/config_parser.h
team/group/edv-appleraisin-trunk/channels/sip/include/dialog.h
team/group/edv-appleraisin-trunk/channels/sip/include/reqresp_parser.h
team/group/edv-appleraisin-trunk/channels/sip/include/sdp_crypto.h
team/group/edv-appleraisin-trunk/channels/sip/include/sip.h
team/group/edv-appleraisin-trunk/channels/sip/include/sip_utils.h
team/group/edv-appleraisin-trunk/channels/sip/include/srtp.h
team/group/edv-appleraisin-trunk/channels/sip/reqresp_parser.c
team/group/edv-appleraisin-trunk/channels/sip/sdp_crypto.c
team/group/edv-appleraisin-trunk/channels/vcodecs.c
team/group/edv-appleraisin-trunk/configure
team/group/edv-appleraisin-trunk/configure.ac
team/group/edv-appleraisin-trunk/doc/appdocsxml.dtd
team/group/edv-appleraisin-trunk/doc/asterisk.8
team/group/edv-appleraisin-trunk/doc/asterisk.sgml
team/group/edv-appleraisin-trunk/funcs/func_base64.c
team/group/edv-appleraisin-trunk/funcs/func_rand.c
team/group/edv-appleraisin-trunk/pbx/ (props changed)
team/group/edv-appleraisin-trunk/pbx/pbx_ael.c
team/group/edv-appleraisin-trunk/pbx/pbx_config.c
team/group/edv-appleraisin-trunk/pbx/pbx_dundi.c
team/group/edv-appleraisin-trunk/pbx/pbx_loopback.c
team/group/edv-appleraisin-trunk/pbx/pbx_lua.c
team/group/edv-appleraisin-trunk/pbx/pbx_realtime.c
team/group/edv-appleraisin-trunk/pbx/pbx_spool.c
team/group/edv-appleraisin-trunk/sounds/Makefile (contents, props changed)
team/group/edv-appleraisin-trunk/tests/test_acl.c
team/group/edv-appleraisin-trunk/tests/test_amihooks.c
team/group/edv-appleraisin-trunk/tests/test_aoc.c
team/group/edv-appleraisin-trunk/tests/test_app.c
team/group/edv-appleraisin-trunk/tests/test_ast_format_str_reduce.c
team/group/edv-appleraisin-trunk/tests/test_astobj2.c
team/group/edv-appleraisin-trunk/tests/test_db.c
team/group/edv-appleraisin-trunk/tests/test_devicestate.c
team/group/edv-appleraisin-trunk/tests/test_dlinklists.c
team/group/edv-appleraisin-trunk/tests/test_event.c
team/group/edv-appleraisin-trunk/tests/test_expr.c
team/group/edv-appleraisin-trunk/tests/test_format_api.c
team/group/edv-appleraisin-trunk/tests/test_func_file.c
team/group/edv-appleraisin-trunk/tests/test_gosub.c
team/group/edv-appleraisin-trunk/tests/test_heap.c
team/group/edv-appleraisin-trunk/tests/test_locale.c
team/group/edv-appleraisin-trunk/tests/test_logger.c
team/group/edv-appleraisin-trunk/tests/test_pbx.c
team/group/edv-appleraisin-trunk/tests/test_poll.c
team/group/edv-appleraisin-trunk/tests/test_sched.c
team/group/edv-appleraisin-trunk/tests/test_security_events.c
team/group/edv-appleraisin-trunk/tests/test_skel.c
team/group/edv-appleraisin-trunk/tests/test_stringfields.c
team/group/edv-appleraisin-trunk/tests/test_strings.c
team/group/edv-appleraisin-trunk/tests/test_substitution.c
team/group/edv-appleraisin-trunk/tests/test_time.c
team/group/edv-appleraisin-trunk/tests/test_utils.c
Propchange: team/group/edv-appleraisin-trunk/
('svnmerge-integrated' removed)
Modified: team/group/edv-appleraisin-trunk/BUGS
URL: http://svnview.digium.com/svn/asterisk/team/group/edv-appleraisin-trunk/BUGS?view=diff&rev=364041&r1=364040&r2=364041
==============================================================================
--- team/group/edv-appleraisin-trunk/BUGS (original)
+++ team/group/edv-appleraisin-trunk/BUGS Thu Apr 26 10:06:55 2012
@@ -4,7 +4,7 @@
To learn about and report Asterisk bugs, please visit
the official Asterisk Bug Tracker at:
- https://issues.asterisk.org
+ https://issues.asterisk.org/jira
For more information on using the bug tracker, or to
learn how you can contribute by acting as a bug marshal
Modified: team/group/edv-appleraisin-trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/group/edv-appleraisin-trunk/CHANGES?view=diff&rev=364041&r1=364040&r2=364041
==============================================================================
--- team/group/edv-appleraisin-trunk/CHANGES (original)
+++ team/group/edv-appleraisin-trunk/CHANGES Thu Apr 26 10:06:55 2012
@@ -9,7 +9,250 @@
==============================================================================
------------------------------------------------------------------------------
---- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
+--- 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.
+ * 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
+ that Asterisk can ensure that these functions do *not* get called by any
+ 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.
+
+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.
+
+Voicemail
+------------------
+ * Addition of the VM_INFO function - see Dialplan 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
+ 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
+ * 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.
+ * Adds an option send_diversion which can be disabled to prevent
+ diversion headers from automatically being added to invites.
+
+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
+--------------------
+ * Added ability to use multiple lines on phone, so for one device in
+ configuration multiple lines can be defined, it allows to have multiple calls
+ on one phone, callwaiting and switching between calls.
+ * Added option 'sharpdial' allowing end dialing by pressing # key
+ * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
+ * Added global 'debug' option, that enables debug in channel driver
+ * Added ability for translation on-screen menu to multiple languages. Tested on
+ Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
+ ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
+ menu of phone
+ * Reworked dialing number input: added dialing by timeout, immediate dial on
+ on dialplan compare, phone number length now not limited by screen size
+ * Added ability for pickup a call using fetures.conf defined value and
+ on-screen key
+
+Codec changes
+-------------
+ * Codec lists may now be modified by the '!' character, to allow succinct
+ specification of a list of codecs allowed and disallowed, without the
+ requirement to use two different keywords. For example, to specify all
+ codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
+
+Music On Hold Changes
+---------------------
+ * Added 'announcement' option which will play at the start of MOH and between
+ songs in modes of MOH that can detect transitions between songs (eg.
+ files, mp3, etc).
+
+Queue changes
+-------------
+ * Added queue options autopausebusy and autopauseunavail for automatically
+ pausing a queue member when their device reports busy or congestion.
+
+Voicemail changes
+-----------------
+ * 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/
+
+Applications
+------------
+ * 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 beneftis eg. format can
+ now be used without specifying time zone as well.
+ * Added 'F()' option to Queue and Bridge. Similar to the dial option, these 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/Bridge
+ will be used.
+
+Parking
+------------
+ * New per parking lot options: comebackcontext and comebackdialtime. See
+ configs/features.conf.sample for more details.
+
+ * Channel variable PARKER is now set when comebacktoorigin is disabled in
+ a parking lot.
+
+ * 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.
+
+CDR postgresql driver changes
+-----------------------------
+ * Added command "cdr show pgsql status" to check connection status
+
+AMI (Asterisk Manager Interface) changes
+----------------------------------------
+ * Originate now generates an error response if the extension given
+ is not found in the dialplan
+
+ * MixMonitor will now show IDs associated with the mixmonitor upon creating them
+ if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
+ on option to close specific MixMonitors.
+
+ * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
+ to include information about peers configured with nat=auto_force_rport by
+ returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
+ set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
+ is not enabled.
+
+ * Hangup now can take a regular expression as the Channel option. If you want
+ to hangup multiple channels, use /regex/ as the Channel option. Existing
+ behavior to hanging up a single channel is unchanged, but if you pass a regex,
+ the manager will send you a list of channels back that were hung up.
+
+FAX changes
+-----------
+ * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
+ control of faxdetect.
+
+DUNDi changes
+-------------
+ * Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
+ used within the dynamic weight attribute when specifying a mapping.
+
+Core changes
+------------
+ * Each logging destination and console now have an independent notion of the
+ current verbosity level. Logger.conf now allows an optional argument to
+ the 'verbose' specifier, indicating the level of verbosity sent to that
+ particular logging destination. Additionally, remote consoles now each
+ have their own verbosity level. The command 'core set verbose' will now set
+ a separate level for each remote console without affecting any other
+ console.
+
+Dialplan functions
+------------------
+ * 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.
+
+Followme changes
+-------------
+ * 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.
+
+RTP changes
+-------------
+ * A new option, 'probation' has been added to rtp.conf
+ RTP in strictrtp mode can now require more than 1 packet to exit learning
+ mode with a new source (and by default requires 4). The probation option
+ allows the user to change the required number of packets in sequence to any
+ desired value. Use a value of 1 to essentially restore the old behavior.
+ Also, with strictrtp on, Asterisk will now drop all packets until learning
+ mode has successfully exited. These changes are based on how pjmedia handles
+ media sources and source changes.
+
+Text Messaging
+--------------
+ * 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.
+
+res_corosync
+------------
+ * A new module, res_corosync, has been introduced. This module uses the
+ Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
+ of Asterisk servers to both Message Waiting Indication (MWI) and/or
+ Device State (presence) information. This module is very similar to, and
+ is a replacement for the res_ais module that was in previous releases of
+ Asterisk.
+
+AGI
+---
+ * A new channel variable, AGIEXITONHANGUP, has been added which allows
+ Asterisk to behave like it did in Asterisk 1.4 and earlier where the
+ AGI application would exit immediately after a channel hangup is detected.
+ * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
+ are resolved and each address is attempted in turn until one succeeds or
+ all fail.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
------------------------------------------------------------------------------
Text Messaging
@@ -19,16 +262,19 @@
SIP MESSAGE and XMPP are currently supported. There are options in
jabber.conf and sip.conf to allow enabling these features.
-> jabber.conf: see the "sendtodialplan" and "context" options.
- -> sip.conf: see the "accept_outofcall_message" and "auth_message_requests"
- options.
+ -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
+ and "outofcall_message_context" options.
The MESSAGE() dialplan function and MessageSend() application have been
added to go along with this functionality. More detailed usage information
can be found on the Asterisk wiki (http://wiki.asterisk.org/).
+ * If real-time text support (T.140) is negotiated, it will be preferred for
+ sending text via the SendText application. For example, via SIP, messages
+ that were once sent via the SIP MESSAGE request would be sent via RTP if
+ T.140 text is negotiated for a call.
Parking
-------
* parkedmusicclass can now be set for non-default parking lots.
- * ParkedCall application can now specify a specific parkinglot.
Asterisk Manager Interface
--------------------------
@@ -41,6 +287,13 @@
Description field that is set by 'description' in the channel configuration
file.
* Added Uniqueid header to UserEvent.
+ * Added new action FilterAdd to control event filters for the current session.
+ This requires the system permission and uses the same filter syntax as
+ filters that can be defined in manager.conf
+ * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
+ versions had some instances of the event converted, but others were left
+ as-is. All Unlink events should now be converted to Bridge events. The AMI
+ protocol version number was incremented to 1.2 as a result of this change.
Asterisk HTTP Server
--------------------------
@@ -67,11 +320,24 @@
--------------------------
* The filter option in cdr_adaptive_odbc now supports negating the argument,
thus allowing records which do NOT match the specified filter.
+ * Added ability to log CONGESTION calls to CDR
CODECS
--------------------------
* Ability to define custom SILK formats in codecs.conf.
* Addition of speex32 audio format with translation.
+ * CELT codec pass-through support and ability to define
+ custom CELT formats in codecs.conf.
+ * Ability to read raw signed linear files with sample rates
+ ranging from 8khz - 192khz. The new file extensions introduced
+ are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
+ * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
+ Skinny, H.323, etc) can still only support the following codecs:
+ Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
+ siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
+ Video: h261, h263, h263p, h264, mpeg4
+ Image: jpeg, png
+ Text: red, t140
ConfBridge
--------------------------
@@ -79,6 +345,14 @@
mixing audio at sample rates ranging from 8khz-96khz.
* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
and bridge profiles on a channel.
+ * CONFBRIDGE_INFO dialplan function capable of retrieving information
+ about a conference such as locked status and number of parties, admins,
+ and marked users.
+ * Addition of video_mode option in confbridge.conf for adding video support
+ into a bridge profile.
+ * Addition of the follow_talker video_mode in confbridge.conf. This video
+ mode dynamically switches the video feed to always display the loudest talker
+ supplying video in the conference.
Dialplan Variables
------------------
@@ -98,6 +372,8 @@
for a given string to replace with another string as many times as the
user specifies or just throughout the whole string.
* Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
+ * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
+ * Added extensions to chan_ooh323 in function CHANNEL()
libpri channel driver (chan_dahdi) DAHDI changes
--------------------------
@@ -129,6 +405,11 @@
compatability for a FollowMe call with certain dialplan apps, options, and
functions.
+Meetme
+--------------------------
+ * Added option "k" that will automatically close the conference when there's
+ only one person left when a user exits the conference.
+
CEL
--------------------------
* cel_pgsql now supports the 'extra' column for data added using the
@@ -140,19 +421,100 @@
in the sample extensions.lua file for syntax details.
* Applications that perform jumps in the dialplan such as Goto will now
execute properly. When pbx_lua detects that the context, extension, or
- priority we are executing on has changed it will immediatly return control
+ priority we are executing on has changed it will immediately return control
to the asterisk PBX engine. Currently the engine cannot detect a Goto to
the priority after the currently executing priority.
* An autoservice is now started by default for pbx_lua channels. It can be
stopped and restarted using the autoservice_stop() and autoservice_start()
functions.
+res_fax
+--------------------------
+ * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
+ into a FAXStatus event with an 'Operation' header that will be either
+ 'send', 'receive', and 'gateway'.
+ * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
+ Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
+ feature will handle converting a fax call between an audio T.30 fax terminal
+ and an IFP T.38 fax terminal.
+
+SIP Changes
+-----------
+ * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
+ * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
+ * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
+
+Queue changes
+-------------
+ * Added general option negative_penalty_invalid default off. when set
+ members are seen as invalid/logged out when there penalty is negative.
+ for realtime members when set remove from queue will set penalty to -1.
+ * Added queue option autopausedelay when autopause is enabled it will be
+ delayed for this number of seconds since last successful call if there
+ was no prior call the agent will be autopaused immediately.
+ * Added member option ignorebusy this when set and ringinuse is not
+ will allow per member control of multiple calls as ringinuse does for
+ the Queue.
+ * Added global option check_state_unknown to enforce checking of device state
+ when the device state is unknown app_queue will see unknown as available.
+
+Applications
+------------
+ * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
+ a MeetMe conference
+ * Added 'k' option to MeetMe to automatically kill the conference when there's only
+ one participant left (much like a normal call bridge)
+ * Added extra argument to Originate to set timeout.
+
+Asterisk Database
+-----------------
+ * The internal Asterisk database has been switched from Berkeley DB 1.86 to
+ SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
+ utility in the UTILS section of menuselect. If an existing astdb is found and no
+ astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
+ convert an existing astdb to the SQLite3 version automatically at runtime.
+
+Asterisk Modules
+----------------
+ * Modules marked as deprecated are no longer marked as building by default. Enabling
+ these modules is still available via menuselect.
+
+IAX2 Changes
+------------
+ * authdebug is now disabled by default. To enable this functionaility again
+ set authdebug = yes in iax.conf.
+
+RTP Changes
+-----------
+ * The rtp.conf setting "strictrtp" is now enabled by default. In previous
+ releases it was disabled.
+
+PBX Core
+--------
+ * The PBX core previously made a call with a non-existing extension test for
+ extension s at default and jump there if the extension existed.
+ This was a bad default behaviour and violated the principle of least surprise.
+ It has therefore been changed in this release. It may affect some
+ applications and configurations that rely on this behaviour. Most channel
+ drivers have avoided this for many releases by testing whether the extension
+ called exists before starting the PBX and generating a local error.
+ This behaviour still exists and works as before.
+
+ Extension "s" is used when no extension is given in a channel driver,
+ like immediate answer in DAHDI or calling to a domain with no user part
+ in a SIP uri.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
------------------------------------------------------------------------------
SIP Changes
-----------
+ * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
+ now defaults to force_rport. It is very important that phones requiring nat=no be
+ specifically set as such instead of relying on the default setting. If at all
+ possible, all devices should have nat settings configured in the general section as
+ opposed to configuring nat per-device.
* Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
* Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
@@ -182,7 +544,9 @@
and enables symmetric RTP support.
* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
response. This permits the master channel to know how each channel dialled
- in a multi-channel setup resolved in an individual way.
+ in a multi-channel setup resolved in an individual way. This carries a
+ performance penalty and can be disabled in sip.conf using the
+ 'storesipcause' option.
* Added 'externtcpport' and 'externtlsport' options to allow custom port
configuration for the externip and externhost options when tcp or tls is used.
* Added support for message body (stored in content variable) to SIP NOTIFY message
@@ -216,8 +580,13 @@
res_stun_monitor module support in chan_sip.
* Addition of the 'auth_options_requests' option for turning on and off
authentication for OPTIONS requests in chan_sip.
- * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
-
+
+Configuration files
+-------------------
+ * Add #tryinclude statement for config files. This provides the same
+ functionality as the #include statement however an asterisk module will
+ still load if the filename does not exist. Using the #include statement
+ Asterisk will not allow the module to load.
IAX2 Changes
-----------
@@ -321,10 +690,7 @@
notices a change.
* Voicemail now includes rdnis within msgXXXX.txt file.
* Added 'D' command to ExternalIVR full details in doc/externalivr.txt
- * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
- a MeetMe conference
- * Added ability to include '@parkinglot' to ParkedCall extension in order to specify
- a specific parkinglot on which to search the extension.
+ * ParkedCall and Park can now specify the parking lot to use.
Dialplan Functions
------------------
@@ -406,6 +772,8 @@
features.conf that should be the base for dynamic parkinglots.
* Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
parkinglot should have.
+ * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
+ parkinglot should have.
* Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
should have.
@@ -423,7 +791,7 @@
[... 77843 lines stripped ...]
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