head 1.28; access; symbols pkgsrc-2023Q4:1.28.0.8 pkgsrc-2023Q4-base:1.28 pkgsrc-2023Q3:1.28.0.6 pkgsrc-2023Q3-base:1.28 pkgsrc-2023Q2:1.28.0.4 pkgsrc-2023Q2-base:1.28 pkgsrc-2023Q1:1.28.0.2 pkgsrc-2023Q1-base:1.28 pkgsrc-2022Q4:1.27.0.8 pkgsrc-2022Q4-base:1.27 pkgsrc-2022Q3:1.27.0.6 pkgsrc-2022Q3-base:1.27 pkgsrc-2022Q2:1.27.0.4 pkgsrc-2022Q2-base:1.27 pkgsrc-2022Q1:1.27.0.2 pkgsrc-2022Q1-base:1.27 pkgsrc-2021Q4:1.26.0.10 pkgsrc-2021Q4-base:1.26 pkgsrc-2021Q3:1.26.0.8 pkgsrc-2021Q3-base:1.26 pkgsrc-2021Q2:1.26.0.6 pkgsrc-2021Q2-base:1.26 pkgsrc-2021Q1:1.26.0.4 pkgsrc-2021Q1-base:1.26 pkgsrc-2020Q4:1.26.0.2 pkgsrc-2020Q4-base:1.26 pkgsrc-2020Q3:1.25.0.4 pkgsrc-2020Q3-base:1.25 pkgsrc-2020Q2:1.25.0.2 pkgsrc-2020Q2-base:1.25 pkgsrc-2020Q1:1.24.0.2 pkgsrc-2020Q1-base:1.24 pkgsrc-2019Q4:1.22.0.6 pkgsrc-2019Q4-base:1.22 pkgsrc-2019Q3:1.22.0.2 pkgsrc-2019Q3-base:1.22 pkgsrc-2013Q2:1.21.0.4 pkgsrc-2013Q2-base:1.21 pkgsrc-2012Q4:1.21.0.2 pkgsrc-2012Q4-base:1.21 pkgsrc-2012Q2:1.20.0.4 pkgsrc-2012Q2-base:1.20 pkgsrc-2012Q1:1.20.0.2 pkgsrc-2012Q1-base:1.20 pkgsrc-2011Q4:1.19.0.2 pkgsrc-2011Q4-base:1.19 pkgsrc-2011Q3:1.18.0.4 pkgsrc-2011Q3-base:1.18 pkgsrc-2011Q2:1.18.0.2 pkgsrc-2011Q2-base:1.18 pkgsrc-2011Q1:1.17.0.4 pkgsrc-2011Q1-base:1.17 pkgsrc-2010Q4:1.17.0.2 pkgsrc-2010Q4-base:1.17 pkgsrc-2010Q3:1.15.0.2 pkgsrc-2010Q3-base:1.15 pkgsrc-2010Q2:1.14.0.2 pkgsrc-2010Q2-base:1.14 pkgsrc-2010Q1:1.9.0.2 pkgsrc-2010Q1-base:1.9 pkgsrc-2009Q4:1.8.0.2 pkgsrc-2009Q4-base:1.8 pkgsrc-2009Q3:1.6.0.2 pkgsrc-2009Q3-base:1.6 pkgsrc-2009Q2:1.3.0.2 pkgsrc-2009Q2-base:1.3 pkgsrc-base:1.1.1.1 TNF:1.1.1; locks; strict; comment @# @; 1.28 date 2023.01.03.16.53.17; author ryoon; state Exp; branches; next 1.27; commitid d5difSUJEltNi78E; 1.27 date 2022.03.04.12.22.31; author ryoon; state Exp; branches; next 1.26; commitid sUIW3xx3ljtYBTuD; 1.26 date 2020.12.10.13.52.30; author gdt; state Exp; branches; next 1.25; commitid 9BppbQKlNgsHxczC; 1.25 date 2020.05.01.07.57.36; author ryoon; state Exp; branches; next 1.24; commitid 4rc6iTRiN0KkKv6C; 1.24 date 2020.03.22.23.09.24; author tnn; state Exp; branches; next 1.23; commitid BXSezkXANC3O4s1C; 1.23 date 2020.01.11.08.36.13; author ryoon; state Exp; branches; next 1.22; commitid uJEndhKPn1VNwfSB; 1.22 date 2019.08.20.13.47.42; author ryoon; state Exp; branches; next 1.21; commitid WMmdTIUqUPcFSLzB; 1.21 date 2012.09.14.02.41.05; author jnemeth; state dead; branches; next 1.20; 1.20 date 2012.01.17.02.12.52; author jnemeth; state Exp; branches; next 1.19; 1.19 date 2011.12.05.04.18.32; author jnemeth; state Exp; branches; next 1.18; 1.18 date 2011.06.06.06.25.06; author jnemeth; state Exp; branches; next 1.17; 1.17 date 2010.12.12.10.19.44; author jnemeth; state Exp; branches; next 1.16; 1.16 date 2010.11.15.05.18.16; author jnemeth; state Exp; branches; next 1.15; 1.15 date 2010.09.23.23.30.38; author jnemeth; state Exp; branches; next 1.14; 1.14 date 2010.05.22.22.21.45; author jnemeth; state Exp; branches; next 1.13; 1.13 date 2010.05.20.17.14.45; author jnemeth; state Exp; branches; next 1.12; 1.12 date 2010.05.07.23.57.56; author jnemeth; state Exp; branches; next 1.11; 1.11 date 2010.05.07.07.20.55; author jnemeth; state Exp; branches; next 1.10; 1.10 date 2010.05.06.20.10.17; author jnemeth; state Exp; branches; next 1.9; 1.9 date 2010.03.01.07.06.48; author jnemeth; state Exp; branches; next 1.8; 1.8 date 2009.12.30.04.11.02; author jnemeth; state Exp; branches; next 1.7; 1.7 date 2009.11.20.04.30.08; author jnemeth; state Exp; branches; next 1.6; 1.6 date 2009.09.14.08.44.51; author jnemeth; state Exp; branches; next 1.5; 1.5 date 2009.08.20.20.37.06; author jnemeth; state Exp; branches; next 1.4; 1.4 date 2009.07.07.08.14.42; author jnemeth; state Exp; branches; next 1.3; 1.3 date 2009.06.14.18.31.59; author joerg; state Exp; branches 1.3.2.1; next 1.2; 1.2 date 2009.06.14.17.38.38; author joerg; state Exp; branches; next 1.1; 1.1 date 2009.06.12.09.04.56; author jnemeth; state Exp; branches 1.1.1.1; next ; 1.3.2.1 date 2009.09.13.14.18.23; author tron; state Exp; branches; next ; 1.1.1.1 date 2009.06.12.09.04.56; author jnemeth; state Exp; branches; next ; desc @@ 1.28 log @asterisk16: Update to 16.29.1 * Use bash for configure script. It uses bash-specific syntax. * Use menuselect command to adjust options instead of manually crafted makeopts file. Manually crafted file does not work properly for me and 16.29.1 now. * I have no idea about x11 option's status. It seems that gtk2 config UI is not available in this release at least, if I understand correctly. Changelog: 16.29.1 Bugs fixed in this release: [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE (Reported By: Michael Bradeen) [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By: shawty) [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported By: nappsoft) [ASTERISK-30338] Backport 2.13 security fixes from pjproject 16.29.0 New Features made in this release: * [ASTERISK-30037] Add test support to calling external processes (Reported by Philip Prindeville) * [ASTERISK-30161] locks: add AMI event for deadlock (Reported by N A) * [ASTERISK-30211] app_confbridge: Add end_marked_any option (Reported by N A) * [ASTERISK-30186] res_pjsip: Add support for reloading TLS certificate and key information (Reported by Joshua C. Colp) * [ASTERISK-29899] features: Add advanced transfer initiation options (Reported by N A) Bugs fixed in this release: * [ASTERISK-30235] res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * [ASTERISK-30234] res_geolocation: may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * [ASTERISK-30215] Inbound SIP INVITE with Geo Location causing a Segmentation Fault (Reported by Dan Cropp) * [ASTERISK-30135] [res_musiconhold] Allows the moh only for the answered call (Reported by sungtae kim) * [ASTERISK-26894] pjsip should support tel uri scheme (Reported by Gergely D?ms?di) * [ASTERISK-30210] func_frame_trace: Channel masquerade triggers assertion (Reported by N A) * [ASTERISK-30190] res_geolocation: GEOLOC_PROFILE isn t returning correct values on incoming channel (Reported by George Joseph) * [ASTERISK-29185] chan_pjsip: Endpoint: allow = all is broken. (Reported by Alexander Traud) * [ASTERISK-30192] res_tonedetect: fix typo for frametype (Reported by N A) * [ASTERISK-29453] alembic: incoming_call_offer_pref and outgoing_call_offer_pref missing in ps_endpoints table (Reported by Daniel Th men) * [ASTERISK-26826] testsuite: Add support for Python 3 (Reported by Joshua C. Colp) * [ASTERISK-30167] res_geolocation: Refactor for issues found by users (Reported by George Joseph) * [ASTERISK-28422] Memory Leak in Confbridge menu (Reported by Ted G) * [ASTERISK-29917] ami: FilterList action doesn t exist (Reported by N A) * [ASTERISK-30020] ConfbridgeListRooms Event Not Documented (Reported by Michael Cargile) * [ASTERISK-30018] app_meetme: MeetmeList AMI event not documented (Reported by Michael Cargile) * [ASTERISK-30151] Documentation doesn t include info about field , a 3rd required parameter. (Reported by Chris Young) Improvements made in this release: * [ASTERISK-30241] res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * [ASTERISK-30178] extend user_eq_phone behavior to local uri s (Reported by Michael Bradeen) * [ASTERISK-30046] Reimplement res/res_crypto.c internals with EVP_PKEY interface to Openssl API s (Reported by Philip Prindeville) * [ASTERISK-30045] Add test coverage to res/res_crypto.c functionality (Reported by Philip Prindeville) * [ASTERISK-30185] res_geolocation: Allow location parameters to be specified in profiles (Reported by George Joseph) * [ASTERISK-30177] res_geolocation: Add option to suppress empty elements (Reported by George Joseph) * [ASTERISK-30182] res_geolocation: Add built-in profiles to use in fully dynamic configurations (Reported by George Joseph) * [ASTERISK-29906] update RLS to reflect the changes to the lists (Reported by Alexei Gradinari) * [ASTERISK-30163] general: fix minor formatting issues (Reported by N A) * [ASTERISK-30164] chan_iax2: Add missing option documentation (Reported by N A) * [ASTERISK-30160] cdr.conf: Remove obsolete app_mysql reference (Reported by N A) * [ASTERISK-30159] general: Remove obsolete SVN references (Reported by N A) * [ASTERISK-30153] logger: Improve log levels (Reported by N A) 16.28.0 The following issues are resolved in this release: Improvements made in this release: * [ASTERISK-30128] Create PJSIP interface module for Geolocation (Reported by George Joseph) * [ASTERISK-30127] Create core Geolocation capability for Asterisk (Reported by George Joseph) * [ASTERISK-30089] general: fix typos (Reported by N A) * [ASTERISK-30050] Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by Stanislav Abramenkov) Bugs fixed in this release: * [ASTERISK-30167] res_geolocation: Refactor for issues found by users (Reported by George Joseph) * [ASTERISK-29966] pbx_variables: ast_str_strlen can be wrong (Reported by N A) * [ASTERISK-29905] OSX: bininstall launchd issue on cross-platfrom build (Reported by Sergey V. Lobanov) * [ASTERISK-30137] manager: Global disabled event filtered is incomplete (Reported by N A) * [ASTERISK-30109] res_pjsip: no contact-status AMI event on register of prune-on-boot contact that uses the same URI as before Asterisk restart (Reported by Michael Neuhauser) * [ASTERISK-30126] Spelling mistake in configs/samples/queues.conf. sample (Reported by Sam Banks) * [ASTERISK-29991] chan_dahdi, callerid: Caller ID does not honor presentation (Reported by N A) * [ASTERISK-29907] res_pjsip, app_confbridge: Video call through ConfBridge with normal endpoints causes infinite loop/crash (Reported by N A) * [ASTERISK-30029] build: Git security vulnerability fix is sad with our accessing git as root during make install (Reported by Joshua C. Colp) * [ASTERISK-30138] Compile failure in res_geolocation/geoloc_ eprofile.c when optimization is enabled (Reported by George Joseph) * [ASTERISK-30096] cel_odbc: Column type 9 (field cdr:cel:eventtime ) is unsupported at this time (Reported by Morvai Szabolcs) * [ASTERISK-30083] chan_iax2: Optional dependency on openssl/ res_crypto is now mandatory (Reported by Dmitry Melekhov) * [ASTERISK-30123] features: Update automixmon documentation to reflect reality (Reported by Trevor Peirce) * [ASTERISK-30117] pbx_lua: Remove compiler warnings (Reported by Boris P. Korzun) * [ASTERISK-30001] db: Removing nonexistent entries shows Database entry removed (Reported by N A) * [ASTERISK-29822] cli: Typing \? freezes the CLI permanently with remote console (Reported by N A) * [ASTERISK-30106] res_calendar_icalendar: Microsoft online ICS calendars no longer work (Reported by N A) * [ASTERISK-30115] app_dial: Allow hook flashes to propogate on outbound dials (Reported by N A) * [ASTERISK-29989] app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy (Reported by N A) * [ASTERISK-30072] res_pjsip: allow TLS verification of wildcard cert-bearing servers (Reported by Kevin Harwell) * [ASTERISK-30075] say: Abort if channel hangs up during playback (Reported by N A) New Features made in this release: * [ASTERISK-30136] db: Add AMI action to retrieve all keys beginning with a prefix (Reported by N A) * [ASTERISK-30000] chan_dahdi: Add POLARITY function (Reported by N A) * [ASTERISK-30062] cli: Add CLI command to execute a dialplan app (Reported by N A) * [ASTERISK-29999] pjsip: Get information from 200 OK INVITE reply headers (Reported by Jos Lopes) * [ASTERISK-30061] pbx: Add pbx helper application (Reported by N A) 16.27.0 Improvements made in this release: * [ASTERISK-30090] xmldocs: Use example tags for examples (Reported by N A) * [ASTERISK-29906] update RLS to reflect the changes to the lists (Reported by Alexei Gradinari) * [ASTERISK-29891] provide a display name for RLS subscriptions (Reported by Alexei Gradinari) * [ASTERISK-30086] res_parking: Warn when invalid parking space requested (Reported by N A) * [ASTERISK-30058] Evaluate dialplan functions and variables in agi exec (Reported by Shloime Rosenblum) * [ASTERISK-30027] ari: expose channel driver s unique id (i.e. Call-ID for chan_sip/chan_pjsip) in ARI channel resource (Reported by Moritz Fain) * [ASTERISK-29845] res_pjsip_outbound_registration: Show time remaining until registration lapses (Reported by N A) Bugs fixed in this release: * [ASTERISK-30097] console: Recent documentation changes for connecting to remote console are inconsistent (Reported by Matthias Hensler) * [ASTERISK-30043] Wrong party is disconnected when hook-flashing on 3-way bridge (Reported by Josh Alberts) * [ASTERISK-29603] res_pjsip: UPDATE/re-INVITE not sent when timers =always is specified in pjsip.conf (Reported by Ray Crumrine) * [ASTERISK-30092] DateTime application: wrong inflection for one o clock in German (Reported by Christof Efkemann) * [ASTERISK-30064] pbx: iax2 switch causes crash due to deadlock and assertion (Reported by N A) * [ASTERISK-29981] res_calendar: Asterisk crashes when starting, and will not run (Reported by N A) * [ASTERISK-30039] cli: Targeted debug on startup deadlocks and creates unstable system (Reported by N A) * [ASTERISK-30051] res_pjsip: No video after un-hold with moh_passthrough=yes (Reported by Maximilian Fridrich) * [ASTERISK-24601] Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body (Reported by Marco Paland) * [ASTERISK-30060] loader: format warnings in dev mode (Reported by N A) * [ASTERISK-30059] menuselect: libxml include fails under Gentoo (Reported by waltermoeller) * [ASTERISK-30065] pjsip: Open Websocket connection is not reused for outgoing requests (Reported by LA) * [ASTERISK-30042] res_pjsip_transport_websocket: Registration over websocket returns a rewritten contact (Reported by Thomas Guebels) * [ASTERISK-29993] chan_dahdi: Operator control option borks both lines involved on callee disconnect (Reported by N A) * [ASTERISK-30044] GCC 12 issues (Reported by George Joseph) New Features made in this release: * [ASTERISK-30063] app_voicemail: Add option to prevent deletion of messages (Reported by N A) * [ASTERISK-30087] res_parking: Add music on hold override option (Reported by N A) * [ASTERISK-29965] res_pjsip_outbound_registration: Make max registration delay configurable (Reported by N A) * [ASTERISK-30036] app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported by N A) 16.26.1 Bugs fixed in this release: * [ASTERISK-30065] pjsip: Open Websocket connection is not reused for outgoing requests (Reported by LA) 16.26.0 Security bugs fixed in this release: * [ASTERISK-29476] res_stir_shaken: Blind SSRF vulnerabilities (Reported by Clint Ruoho) * [ASTERISK-29838] ${SQL_ESC()} not correctly escaping a terminating \ (Reported by Leandro Dardini) * [ASTERISK-29872] res_stir_shaken: Resource exhaustion with large files (Reported by Benjamin Keith Ford) New Features made in this release: * [ASTERISK-29931] Option to allow a user to not hear the join sound on enter but everyone else can (Reported by Michael Cargile) * [ASTERISK-29968] func_db: Add a function to return cardinality of keys at prefix (Reported by N A) * [ASTERISK-29486] Hint-like extension value lookup function without device state (Reported by N A) * [ASTERISK-29941] chan_pjsip: Add ability to send flash events (Reported by N A) * [ASTERISK-29820] cli: Add command to evaluate a function (Reported by N A) * [ASTERISK-29876] app_queue: Add music on hold option (Reported by N A) Bugs fixed in this release: * [ASTERISK-28518] chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold (Reported by Josh Alberts) * [ASTERISK-29990] chan_dahdi: adding ring cadences is not idempotent on dahdi restart (Reported by N A) * [ASTERISK-30007] chan_iax2: Prevent crashes due to attempted encryption with missing secrets (Reported by N A) * [ASTERISK-29728] menuselect: Disabled by default modules that are enabled are always recompiled (Reported by N A) * [ASTERISK-30002] app_meetme: Don t erroneously set global variables when channel is NULL (Reported by N A) * [ASTERISK-29994] chan_dahdi: Round robin array size is too small for max number of groups (Reported by N A) * [ASTERISK-22246] Asterisk s T flag is ignored when used with r or R flags. (documentation bug) (Reported by Rusty Newton) * [ASTERISK-26582] Asterisk seems to ignore the n parameter for disable console colorization (Reported by Sebastian Gutierrez) * [ASTERISK-29843] Session timers get removed on UPDATE (Reported by Mark Petersen) * [ASTERISK-29943] file.c: seeking to negative file offset is not prevented (Reported by N A) * [ASTERISK-29955] chan_sip: SIP route header is missing on UPDATE (Reported by Mark Petersen) * [ASTERISK-29842] Do not change 180 Ringing to 183 Progress even if early_media already enabled (Reported by Mark Petersen) * [ASTERISK-29948] iostream: Infinite TCP timeout writing data (Reported by N A) * [ASTERISK-29253] Incorrect bridging on transfer (Reported by Yury Kirsanov) * [ASTERISK-30024] Failed to sign STIR/SHAKEN payload with functionality not enabled (Reported by Claude Diderich) * [ASTERISK-30006] res_pjsip: UDP transport does not work when async_operations is greater than 1 (Reported by Ross Beer) * [ASTERISK-29655] res_pjsip_session: No video to caller if no camera available (Reported by Michael Auracher) * [ASTERISK-29638] res_pjsip_session: No video after early media (Reported by Michael Auracher) * [ASTERISK-30015] pjsip / WebRTC: Chrome creating large number of SDP attributes (Reported by Josh Hogan) * [ASTERISK-30021] ast_variable_list_replace_variable uses variable with new keyword (Reported by Jasper Hafkenscheid) * [ASTERISK-30023] cdr_adaptive_odbc: does not support DATETIME database columns (Reported by Gregory Massel) * [ASTERISK-29411] Crash in pjsip_msg_find_hdr_by_name (Reported by LA) * [ASTERISK-29535] Segmentation fault in libasteriskpj.so.2 (Reported by Daniel Bonazzi) * [ASTERISK-26719] pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK 1) (Reported by Tzafrir Cohen) * [ASTERISK-29988] REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn t (Reported by George Joseph) * [ASTERISK-29986] build: Asterisk 18.11.0 doesn t compile when wget isn t available (Reported by Stefan Ruijsenaars) * [ASTERISK-29895] chan_iax2: Fix misaligned spacing in iax2 show netstats printout (Reported by N A) * [ASTERISK-29939] agi: Fix xmldoc bug with set music (Reported by N A) * [ASTERISK-28891] documentation: AGICommand_set+music documentation arguments displayed incorreclty (Reported by Jonathan Harris) * [ASTERISK-29048] chan_iax2: iax2 show registry shows host for perceived (Reported by David Herselman) * [ASTERISK-26689] res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity (Reported by Dmitriy Serov) * [ASTERISK-29929] res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions (Reported by Boris P. Korzun) * [ASTERISK-29674] Adjust for 64bit time_t (Reported by Andre Heider) * [ASTERISK-29961] RLS: domain part of uri list attribute mismatch with SUBSCRIBE request (Reported by Alexei Gradinari) * [ASTERISK-29950] SayNumber can handle 01 to 07 , but not 08 or 09 (Reported by Jim Van Meggelen) * [ASTERISK-29928] logging messages truncated when using MUSL runtime (Reported by Philip Prindeville) * [ASTERISK-29960] ari: Retrieving stored recording can returns wrong file (Reported by Arix) Improvements made in this release: * [ASTERISK-24827] Missing documentation for chan_dahdi dial string ring cadences (Reported by Scott Griepentrog) * [ASTERISK-29940] general: Add since tags to xmldocs (Reported by N A) * [ASTERISK-29951] app_mf, app_sf: Return -1 on hangup (Reported by N A) * [ASTERISK-29954] app_meetme: Emit warning if conference not found (Reported by N A) * [ASTERISK-29351] Qualify pjproject 2.12 for Asterisk (Reported by George Joseph) * [ASTERISK-29877] app_mf: Allow reading a maximum number of digits (Reported by N A) * [ASTERISK-29976] Should Readme include information about install_prereq script? (Reported by Marcel Wagner) * [ASTERISK-29970] Use pkg-config to find libxml2 headers and libraries (Reported by Hugh McMaster) * [ASTERISK-25716] Documentation: Document explanations and examples for possible values of DIALSTATUS (Reported by Rusty Newton) * [ASTERISK-29980] build: External binary modules don t use https (Reported by INVADE International Ltd.) * [ASTERISK-29967] pbx_builtins: Add missing documentation (Reported by N A) 16.25.3 Bugs fixed in this release: * [ASTERISK-30024] Failed to sign STIR/SHAKEN payload with functionality not enabled (Reported by Claude Diderich) 16.25.2 The following security vulnerabilities were resolved in 16.25.2: * AST-2022-001: res_stir_shaken: resource exhaustion with large files When using STIR/SHAKEN, it's possible to download files that are not certificates. These files could be much larger than what you would expect to download. * AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When using STIR/SHAKEN, it's possible to send arbitrary requests like GET to interfaces such as localhost using the Identity header. * AST-2022-003: func_odbc: Possible SQL Injection Some databases can use backslashes to escape certain characters, such as backticks. If input is provided to func_odbc which includes backslashes it is possible for func_odbc to construct a broken SQL query and the SQL query to fail. 16.25.1 Bugs fixed in this release: * [ASTERISK-29988] REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn??t (Reported by George Joseph) * [ASTERISK-29986] build: Asterisk 18.11.0 doesn??t compile when wget isn??t available (Reported by Stefan Ruijsenaars) 15.25.0 Security bugs fixed in this release: * [ASTERISK-29945] pjproject: Security fixes for things (Reported by Kevin Harwell) New Features made in this release: * [ASTERISK-29853] ami: Allow events to be globally disabled (Reported by N A) * [ASTERISK-29840] func_channel: Add LASTCONTEXT and LASTEXTEN fields (Reported by N A) Bugs fixed in this release: * [ASTERISK-29924] res_config_pgsql: omit unsupported column type text' error (Reported by Boris P. Korzun) * [ASTERISK-29923] docs, LICENSE: pbx.digium.com no longer exists (Reported by N A) * [ASTERISK-29904] RLS: Batched Notifications stop working (Reported by Alexei Gradinari) * [ASTERISK-29365] taskprocessor: Can cause assert at shutdown (Reported by Joshua C. Colp) * [ASTERISK-29873] Queue Realtime load (Reported by Alexei Gradinari) * [ASTERISK-18416] Realtime queue agents unavailable via AMI before a call event. (Reported by kwk) * [ASTERISK-27597] AMI Queuestatus not working (with realtime queue) (Reported by cagdas kopuz) * [ASTERISK-29886] Asterisk AMI sends not-valid XML (Reported by Napadailo Yaroslav) Improvements made in this release: * [ASTERISK-29906] update RLS to reflect the changes to the lists (Reported by Alexei Gradinari) * [ASTERISK-29909] app_queue: Add support for withdrawing a call (Reported by Kfir Itzhak) * [ASTERISK-29353] Qualify jansson 2.14 for asterisk (Reported by George Joseph) * [ASTERISK-29897] channels: Increase core debug levels for chatty debugs (Reported by N A) * [ASTERISK-29896] xmldocs: Add since tag (Reported by N A) * [ASTERISK-29861] asterisk.h: add macro for curl user agent (Reported by N A) * [ASTERISK-29920] app_voicemail: Warn if trying to manage nonexistent mailbox (Reported by N A) * [ASTERISK-29925] func_db: Warn about malformed key names (Reported by N A) * [ASTERISK-29809] curl, stir_shaken: refactor curl code (Reported by N A) * [ASTERISK-29891] provide a display name for RLS subscriptions (Reported by Alexei Gradinari) * [ASTERISK-29866] cli: add core dump information to core show settings (Reported by N A) * [ASTERISK-29898] documentation: Add default attributes to documentation (Reported by N A) * [ASTERISK-29900] app_mp3: Document and warn about https incompatibility (Reported by N A) 16.24.1 The following security vulnerabilities were resolved in 16.24.1: * AST-2022-004: pjproject: integer underflow on STUN message The header length on incoming STUN messages that contain an ERROR-CODE attribute is not properly checked. This can result in an integer underflow. Note, this requires ICE or WebRTC support to be in use with a malicious remote party. * AST-2022-005: pjproject: undefined behavior after freeing a dialog set When acting as a UAC, and when placing an outgoing call to a target that then forks Asterisk may experience undefined behavior (crashes, hangs, etc??) after a dialog set is prematurely freed. * AST-2022-006: pjproject: unconstrained malformed multipart SIP message If an incoming SIP message contains a malformed multi-part body an out of bounds read access may occur, which can result in undefined behavior. Note, it??s currently uncertain if there is any externally exploitable vector within Asterisk for this issue, but providing this as a security issue out of caution.[cleardot] @ text @@@comment $NetBSD: PLIST,v 1.27 2022/03/04 12:22:31 ryoon Exp $ include/asterisk.h include/asterisk/_private.h include/asterisk/abstract_jb.h include/asterisk/acl.h include/asterisk/adsi.h include/asterisk/ael_structs.h include/asterisk/agi.h include/asterisk/alaw.h include/asterisk/alertpipe.h include/asterisk/aoc.h include/asterisk/app.h include/asterisk/ari.h include/asterisk/ast_expr.h include/asterisk/ast_version.h include/asterisk/astdb.h include/asterisk/astmm.h include/asterisk/astobj2.h include/asterisk/audiohook.h include/asterisk/autochan.h include/asterisk/autoconfig.h include/asterisk/backtrace.h include/asterisk/beep.h include/asterisk/bridge.h include/asterisk/bridge_after.h include/asterisk/bridge_basic.h include/asterisk/bridge_channel.h include/asterisk/bridge_channel_internal.h include/asterisk/bridge_features.h include/asterisk/bridge_internal.h include/asterisk/bridge_roles.h include/asterisk/bridge_technology.h include/asterisk/bucket.h include/asterisk/build.h include/asterisk/buildinfo.h include/asterisk/buildopts.h include/asterisk/calendar.h include/asterisk/callerid.h include/asterisk/causes.h include/asterisk/ccss.h include/asterisk/cdr.h include/asterisk/cel.h include/asterisk/celt.h include/asterisk/channel.h include/asterisk/channel_internal.h include/asterisk/channelstate.h include/asterisk/chanvars.h include/asterisk/cli.h include/asterisk/codec.h include/asterisk/compat.h include/asterisk/compiler.h include/asterisk/config.h include/asterisk/config_options.h include/asterisk/conversions.h include/asterisk/core_local.h include/asterisk/core_unreal.h include/asterisk/crypto.h include/asterisk/data_buffer.h include/asterisk/datastore.h include/asterisk/devicestate.h include/asterisk/dial.h include/asterisk/dlinkedlists.h include/asterisk/dns.h include/asterisk/dns_core.h include/asterisk/dns_internal.h include/asterisk/dns_naptr.h include/asterisk/dns_query_set.h include/asterisk/dns_recurring.h include/asterisk/dns_resolver.h include/asterisk/dns_srv.h include/asterisk/dns_test.h include/asterisk/dns_tlsa.h include/asterisk/dns_txt.h include/asterisk/dnsmgr.h include/asterisk/doxygen/architecture.h include/asterisk/doxygen/licensing.h include/asterisk/doxyref.h include/asterisk/dsp.h include/asterisk/dundi.h include/asterisk/endian.h include/asterisk/endpoints.h include/asterisk/enum.h include/asterisk/event.h include/asterisk/event_defs.h include/asterisk/extconf.h include/asterisk/features.h include/asterisk/features_config.h include/asterisk/file.h include/asterisk/format.h include/asterisk/format_cache.h include/asterisk/format_cap.h include/asterisk/format_compatibility.h include/asterisk/frame.h include/asterisk/framehook.h include/asterisk/fskmodem.h include/asterisk/fskmodem_float.h include/asterisk/fskmodem_int.h include/asterisk/global_datastores.h include/asterisk/hashtab.h include/asterisk/heap.h include/asterisk/http.h include/asterisk/http_websocket.h include/asterisk/ilbc.h include/asterisk/image.h include/asterisk/indications.h include/asterisk/inline_api.h include/asterisk/io.h include/asterisk/iostream.h include/asterisk/json.h include/asterisk/linkedlists.h include/asterisk/localtime.h include/asterisk/lock.h include/asterisk/logger.h include/asterisk/logger_category.h include/asterisk/manager.h include/asterisk/max_forwards.h include/asterisk/md5.h include/asterisk/media_cache.h include/asterisk/media_index.h include/asterisk/message.h include/asterisk/mixmonitor.h include/asterisk/mod_format.h include/asterisk/module.h include/asterisk/monitor.h include/asterisk/multicast_rtp.h include/asterisk/musiconhold.h include/asterisk/mwi.h include/asterisk/named_locks.h include/asterisk/netsock2.h include/asterisk/network.h include/asterisk/optional_api.h include/asterisk/options.h include/asterisk/opus.h include/asterisk/parking.h include/asterisk/paths.h include/asterisk/pbx.h include/asterisk/phoneprov.h include/asterisk/pickup.h include/asterisk/pktccops.h include/asterisk/plc.h include/asterisk/poll-compat.h include/asterisk/presencestate.h include/asterisk/privacy.h include/asterisk/pval.h include/asterisk/res_fax.h include/asterisk/res_geolocation.h include/asterisk/res_hep.h include/asterisk/res_mwi_external.h include/asterisk/res_odbc.h include/asterisk/res_odbc_transaction.h include/asterisk/res_pjproject.h include/asterisk/res_pjsip.h include/asterisk/res_pjsip_body_generator_types.h include/asterisk/res_pjsip_cli.h include/asterisk/res_pjsip_outbound_publish.h include/asterisk/res_pjsip_presence_xml.h include/asterisk/res_pjsip_pubsub.h include/asterisk/res_pjsip_session.h include/asterisk/res_srtp.h include/asterisk/res_stir_shaken.h include/asterisk/rtp_engine.h include/asterisk/say.h include/asterisk/sched.h include/asterisk/sdp_srtp.h include/asterisk/security_events.h include/asterisk/security_events_defs.h include/asterisk/select.h include/asterisk/sem.h include/asterisk/serializer.h include/asterisk/sha1.h include/asterisk/silk.h include/asterisk/sip_api.h include/asterisk/slin.h include/asterisk/slinfactory.h include/asterisk/smdi.h include/asterisk/smoother.h include/asterisk/sorcery.h include/asterisk/sounds_index.h include/asterisk/speech.h include/asterisk/spinlock.h include/asterisk/srv.h include/asterisk/stasis.h include/asterisk/stasis_app.h include/asterisk/stasis_app_device_state.h include/asterisk/stasis_app_impl.h include/asterisk/stasis_app_mailbox.h include/asterisk/stasis_app_playback.h include/asterisk/stasis_app_recording.h include/asterisk/stasis_app_snoop.h include/asterisk/stasis_bridges.h include/asterisk/stasis_cache_pattern.h include/asterisk/stasis_channels.h include/asterisk/stasis_endpoints.h include/asterisk/stasis_internal.h include/asterisk/stasis_message_router.h include/asterisk/stasis_system.h include/asterisk/stasis_test.h include/asterisk/statsd.h include/asterisk/stream.h include/asterisk/stringfields.h include/asterisk/strings.h include/asterisk/stun.h include/asterisk/syslog.h include/asterisk/taskprocessor.h include/asterisk/tcptls.h include/asterisk/tdd.h include/asterisk/term.h include/asterisk/test.h include/asterisk/threadpool.h include/asterisk/threadstorage.h include/asterisk/time.h include/asterisk/timing.h include/asterisk/transcap.h include/asterisk/translate.h include/asterisk/udptl.h include/asterisk/ulaw.h include/asterisk/unaligned.h include/asterisk/uri.h include/asterisk/utf8.h include/asterisk/utils.h include/asterisk/uuid.h include/asterisk/vector.h include/asterisk/version.h include/asterisk/xml.h include/asterisk/xmldoc.h include/asterisk/xmpp.h lib/asterisk/libasteriskpj.so lib/asterisk/libasteriskpj.so.2 lib/asterisk/modules/app_adsiprog.so lib/asterisk/modules/app_agent_pool.so lib/asterisk/modules/app_alarmreceiver.so lib/asterisk/modules/app_amd.so lib/asterisk/modules/app_attended_transfer.so lib/asterisk/modules/app_authenticate.so lib/asterisk/modules/app_blind_transfer.so lib/asterisk/modules/app_bridgeaddchan.so lib/asterisk/modules/app_bridgewait.so lib/asterisk/modules/app_cdr.so lib/asterisk/modules/app_celgenuserevent.so lib/asterisk/modules/app_chanisavail.so lib/asterisk/modules/app_channelredirect.so lib/asterisk/modules/app_chanspy.so lib/asterisk/modules/app_confbridge.so lib/asterisk/modules/app_controlplayback.so lib/asterisk/modules/app_db.so lib/asterisk/modules/app_dial.so lib/asterisk/modules/app_dictate.so lib/asterisk/modules/app_directed_pickup.so lib/asterisk/modules/app_directory.so lib/asterisk/modules/app_disa.so lib/asterisk/modules/app_dtmfstore.so lib/asterisk/modules/app_dumpchan.so lib/asterisk/modules/app_echo.so lib/asterisk/modules/app_exec.so lib/asterisk/modules/app_externalivr.so lib/asterisk/modules/app_festival.so lib/asterisk/modules/app_followme.so lib/asterisk/modules/app_forkcdr.so lib/asterisk/modules/app_getcpeid.so lib/asterisk/modules/app_ices.so lib/asterisk/modules/app_image.so lib/asterisk/modules/app_mf.so lib/asterisk/modules/app_milliwatt.so lib/asterisk/modules/app_minivm.so lib/asterisk/modules/app_mixmonitor.so lib/asterisk/modules/app_morsecode.so lib/asterisk/modules/app_mp3.so lib/asterisk/modules/app_nbscat.so lib/asterisk/modules/app_originate.so lib/asterisk/modules/app_page.so lib/asterisk/modules/app_playback.so lib/asterisk/modules/app_playtones.so lib/asterisk/modules/app_privacy.so lib/asterisk/modules/app_queue.so lib/asterisk/modules/app_read.so lib/asterisk/modules/app_readexten.so lib/asterisk/modules/app_record.so lib/asterisk/modules/app_reload.so lib/asterisk/modules/app_sayunixtime.so lib/asterisk/modules/app_senddtmf.so lib/asterisk/modules/app_sendtext.so lib/asterisk/modules/app_sf.so lib/asterisk/modules/app_sms.so lib/asterisk/modules/app_softhangup.so lib/asterisk/modules/app_speech_utils.so lib/asterisk/modules/app_stack.so lib/asterisk/modules/app_stasis.so lib/asterisk/modules/app_stream_echo.so lib/asterisk/modules/app_system.so lib/asterisk/modules/app_talkdetect.so lib/asterisk/modules/app_test.so lib/asterisk/modules/app_transfer.so lib/asterisk/modules/app_url.so lib/asterisk/modules/app_userevent.so lib/asterisk/modules/app_verbose.so lib/asterisk/modules/app_voicemail.so lib/asterisk/modules/app_waitforcond.so lib/asterisk/modules/app_waitforring.so lib/asterisk/modules/app_waitforsilence.so lib/asterisk/modules/app_waituntil.so lib/asterisk/modules/app_while.so lib/asterisk/modules/app_zapateller.so lib/asterisk/modules/bridge_builtin_features.so lib/asterisk/modules/bridge_builtin_interval_features.so lib/asterisk/modules/bridge_holding.so lib/asterisk/modules/bridge_native_rtp.so lib/asterisk/modules/bridge_simple.so lib/asterisk/modules/bridge_softmix.so ${PLIST.unixodbc}lib/asterisk/modules/cdr_adaptive_odbc.so lib/asterisk/modules/cdr_csv.so lib/asterisk/modules/cdr_custom.so lib/asterisk/modules/cdr_manager.so ${PLIST.unixodbc}lib/asterisk/modules/cdr_odbc.so ${PLIST.pgsql}lib/asterisk/modules/cdr_pgsql.so lib/asterisk/modules/cdr_sqlite3_custom.so lib/asterisk/modules/cel_custom.so lib/asterisk/modules/cel_manager.so ${PLIST.unixodbc}lib/asterisk/modules/cel_odbc.so ${PLIST.pgsql}lib/asterisk/modules/cel_pgsql.so lib/asterisk/modules/cel_sqlite3_custom.so lib/asterisk/modules/chan_bridge_media.so lib/asterisk/modules/chan_iax2.so ${PLIST.mgcp}lib/asterisk/modules/chan_mgcp.so ${PLIST.jabber}lib/asterisk/modules/chan_motif.so lib/asterisk/modules/chan_oss.so lib/asterisk/modules/chan_pjsip.so lib/asterisk/modules/chan_rtp.so lib/asterisk/modules/chan_sip.so lib/asterisk/modules/chan_skinny.so lib/asterisk/modules/chan_unistim.so lib/asterisk/modules/codec_a_mu.so lib/asterisk/modules/codec_adpcm.so lib/asterisk/modules/codec_alaw.so lib/asterisk/modules/codec_g722.so lib/asterisk/modules/codec_g726.so lib/asterisk/modules/codec_gsm.so lib/asterisk/modules/codec_ilbc.so lib/asterisk/modules/codec_lpc10.so lib/asterisk/modules/codec_resample.so ${PLIST.speex}lib/asterisk/modules/codec_speex.so lib/asterisk/modules/codec_ulaw.so lib/asterisk/modules/format_g719.so lib/asterisk/modules/format_g723.so lib/asterisk/modules/format_g726.so lib/asterisk/modules/format_g729.so lib/asterisk/modules/format_gsm.so lib/asterisk/modules/format_h263.so lib/asterisk/modules/format_h264.so lib/asterisk/modules/format_ilbc.so ${PLIST.speex}lib/asterisk/modules/format_ogg_speex.so lib/asterisk/modules/format_pcm.so lib/asterisk/modules/format_siren14.so lib/asterisk/modules/format_siren7.so lib/asterisk/modules/format_sln.so lib/asterisk/modules/format_vox.so lib/asterisk/modules/format_wav.so lib/asterisk/modules/format_wav_gsm.so lib/asterisk/modules/func_aes.so lib/asterisk/modules/func_base64.so lib/asterisk/modules/func_blacklist.so lib/asterisk/modules/func_callcompletion.so lib/asterisk/modules/func_callerid.so lib/asterisk/modules/func_cdr.so lib/asterisk/modules/func_channel.so lib/asterisk/modules/func_config.so lib/asterisk/modules/func_curl.so lib/asterisk/modules/func_cut.so lib/asterisk/modules/func_db.so lib/asterisk/modules/func_devstate.so lib/asterisk/modules/func_dialgroup.so lib/asterisk/modules/func_dialplan.so lib/asterisk/modules/func_enum.so lib/asterisk/modules/func_env.so lib/asterisk/modules/func_evalexten.so lib/asterisk/modules/func_extstate.so lib/asterisk/modules/func_frame_drop.so lib/asterisk/modules/func_frame_trace.so lib/asterisk/modules/func_global.so lib/asterisk/modules/func_groupcount.so lib/asterisk/modules/func_hangupcause.so lib/asterisk/modules/func_holdintercept.so lib/asterisk/modules/func_iconv.so lib/asterisk/modules/func_jitterbuffer.so lib/asterisk/modules/func_json.so lib/asterisk/modules/func_lock.so lib/asterisk/modules/func_logic.so lib/asterisk/modules/func_math.so lib/asterisk/modules/func_md5.so lib/asterisk/modules/func_module.so ${PLIST.unixodbc}lib/asterisk/modules/func_odbc.so lib/asterisk/modules/func_periodic_hook.so lib/asterisk/modules/func_pitchshift.so lib/asterisk/modules/func_pjsip_aor.so lib/asterisk/modules/func_pjsip_contact.so lib/asterisk/modules/func_pjsip_endpoint.so lib/asterisk/modules/func_presencestate.so lib/asterisk/modules/func_rand.so lib/asterisk/modules/func_realtime.so lib/asterisk/modules/func_sayfiles.so lib/asterisk/modules/func_scramble.so lib/asterisk/modules/func_sha1.so lib/asterisk/modules/func_shell.so lib/asterisk/modules/func_sorcery.so ${PLIST.speex}lib/asterisk/modules/func_speex.so lib/asterisk/modules/func_sprintf.so lib/asterisk/modules/func_srv.so lib/asterisk/modules/func_strings.so lib/asterisk/modules/func_sysinfo.so lib/asterisk/modules/func_talkdetect.so lib/asterisk/modules/func_timeout.so lib/asterisk/modules/func_uri.so lib/asterisk/modules/func_version.so lib/asterisk/modules/func_vmcount.so lib/asterisk/modules/func_volume.so lib/asterisk/modules/pbx_ael.so lib/asterisk/modules/pbx_config.so lib/asterisk/modules/pbx_dundi.so lib/asterisk/modules/pbx_loopback.so lib/asterisk/modules/pbx_realtime.so lib/asterisk/modules/pbx_spool.so lib/asterisk/modules/res_adsi.so lib/asterisk/modules/res_ael_share.so lib/asterisk/modules/res_agi.so lib/asterisk/modules/res_ari.so lib/asterisk/modules/res_ari_applications.so lib/asterisk/modules/res_ari_asterisk.so lib/asterisk/modules/res_ari_bridges.so lib/asterisk/modules/res_ari_channels.so lib/asterisk/modules/res_ari_device_states.so lib/asterisk/modules/res_ari_endpoints.so lib/asterisk/modules/res_ari_events.so lib/asterisk/modules/res_ari_model.so lib/asterisk/modules/res_ari_playbacks.so lib/asterisk/modules/res_ari_recordings.so lib/asterisk/modules/res_ari_sounds.so lib/asterisk/modules/res_calendar.so lib/asterisk/modules/res_clialiases.so lib/asterisk/modules/res_clioriginate.so lib/asterisk/modules/res_config_curl.so ${PLIST.ldap}lib/asterisk/modules/res_config_ldap.so ${PLIST.unixodbc}lib/asterisk/modules/res_config_odbc.so ${PLIST.pgsql}lib/asterisk/modules/res_config_pgsql.so lib/asterisk/modules/res_config_sqlite3.so lib/asterisk/modules/res_convert.so lib/asterisk/modules/res_crypto.so lib/asterisk/modules/res_curl.so lib/asterisk/modules/res_fax.so ${PLIST.spandsp}lib/asterisk/modules/res_fax_spandsp.so lib/asterisk/modules/res_format_attr_celt.so lib/asterisk/modules/res_format_attr_g729.so lib/asterisk/modules/res_format_attr_h263.so lib/asterisk/modules/res_format_attr_h264.so lib/asterisk/modules/res_format_attr_ilbc.so lib/asterisk/modules/res_format_attr_opus.so lib/asterisk/modules/res_format_attr_silk.so lib/asterisk/modules/res_format_attr_siren14.so lib/asterisk/modules/res_format_attr_siren7.so lib/asterisk/modules/res_format_attr_vp8.so lib/asterisk/modules/res_hep.so lib/asterisk/modules/res_hep_pjsip.so lib/asterisk/modules/res_hep_rtcp.so lib/asterisk/modules/res_http_media_cache.so lib/asterisk/modules/res_http_websocket.so lib/asterisk/modules/res_limit.so lib/asterisk/modules/res_manager_devicestate.so lib/asterisk/modules/res_manager_presencestate.so lib/asterisk/modules/res_monitor.so lib/asterisk/modules/res_musiconhold.so lib/asterisk/modules/res_mutestream.so lib/asterisk/modules/res_mwi_devstate.so ${PLIST.unixodbc}lib/asterisk/modules/res_odbc.so ${PLIST.unixodbc}lib/asterisk/modules/res_odbc_transaction.so lib/asterisk/modules/res_parking.so lib/asterisk/modules/res_phoneprov.so lib/asterisk/modules/res_pjproject.so lib/asterisk/modules/res_pjsip.so lib/asterisk/modules/res_pjsip_acl.so lib/asterisk/modules/res_pjsip_authenticator_digest.so lib/asterisk/modules/res_pjsip_caller_id.so lib/asterisk/modules/res_pjsip_config_wizard.so lib/asterisk/modules/res_pjsip_dialog_info_body_generator.so lib/asterisk/modules/res_pjsip_diversion.so lib/asterisk/modules/res_pjsip_dlg_options.so lib/asterisk/modules/res_pjsip_dtmf_info.so lib/asterisk/modules/res_pjsip_empty_info.so lib/asterisk/modules/res_pjsip_endpoint_identifier_anonymous.so lib/asterisk/modules/res_pjsip_endpoint_identifier_ip.so lib/asterisk/modules/res_pjsip_endpoint_identifier_user.so lib/asterisk/modules/res_pjsip_exten_state.so lib/asterisk/modules/res_pjsip_header_funcs.so lib/asterisk/modules/res_pjsip_history.so lib/asterisk/modules/res_pjsip_logger.so lib/asterisk/modules/res_pjsip_messaging.so lib/asterisk/modules/res_pjsip_mwi.so lib/asterisk/modules/res_pjsip_mwi_body_generator.so lib/asterisk/modules/res_pjsip_nat.so lib/asterisk/modules/res_pjsip_notify.so lib/asterisk/modules/res_pjsip_one_touch_record_info.so lib/asterisk/modules/res_pjsip_outbound_authenticator_digest.so lib/asterisk/modules/res_pjsip_outbound_publish.so lib/asterisk/modules/res_pjsip_outbound_registration.so lib/asterisk/modules/res_pjsip_path.so lib/asterisk/modules/res_pjsip_phoneprov_provider.so lib/asterisk/modules/res_pjsip_pidf_body_generator.so lib/asterisk/modules/res_pjsip_pidf_digium_body_supplement.so lib/asterisk/modules/res_pjsip_pidf_eyebeam_body_supplement.so lib/asterisk/modules/res_pjsip_publish_asterisk.so lib/asterisk/modules/res_pjsip_pubsub.so lib/asterisk/modules/res_pjsip_refer.so lib/asterisk/modules/res_pjsip_registrar.so lib/asterisk/modules/res_pjsip_rfc3326.so lib/asterisk/modules/res_pjsip_sdp_rtp.so lib/asterisk/modules/res_pjsip_send_to_voicemail.so lib/asterisk/modules/res_pjsip_session.so lib/asterisk/modules/res_pjsip_sips_contact.so lib/asterisk/modules/res_pjsip_stir_shaken.so lib/asterisk/modules/res_pjsip_t38.so lib/asterisk/modules/res_pjsip_transport_websocket.so lib/asterisk/modules/res_pjsip_xpidf_body_generator.so ${PLIST.mgcp}lib/asterisk/modules/res_pktccops.so lib/asterisk/modules/res_realtime.so ${PLIST.unbound}lib/asterisk/modules/res_resolver_unbound.so lib/asterisk/modules/res_rtp_asterisk.so 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libdata/asterisk/sounds/en/wx/around.gsm libdata/asterisk/sounds/en/wx/barometer.gsm libdata/asterisk/sounds/en/wx/ceiling.gsm libdata/asterisk/sounds/en/wx/dew-point.gsm libdata/asterisk/sounds/en/wx/falling.gsm libdata/asterisk/sounds/en/wx/feet.gsm libdata/asterisk/sounds/en/wx/foot.gsm libdata/asterisk/sounds/en/wx/gust.gsm libdata/asterisk/sounds/en/wx/gusting-to.gsm libdata/asterisk/sounds/en/wx/gusts.gsm libdata/asterisk/sounds/en/wx/gusty.gsm libdata/asterisk/sounds/en/wx/heat-index.gsm libdata/asterisk/sounds/en/wx/humidity.gsm libdata/asterisk/sounds/en/wx/inch.gsm libdata/asterisk/sounds/en/wx/inches.gsm libdata/asterisk/sounds/en/wx/kilometer.gsm libdata/asterisk/sounds/en/wx/large.gsm libdata/asterisk/sounds/en/wx/meter.gsm libdata/asterisk/sounds/en/wx/mist.gsm libdata/asterisk/sounds/en/wx/northeast.gsm libdata/asterisk/sounds/en/wx/northwest.gsm libdata/asterisk/sounds/en/wx/percent.gsm libdata/asterisk/sounds/en/wx/point.gsm libdata/asterisk/sounds/en/wx/rising.gsm libdata/asterisk/sounds/en/wx/southeast.gsm libdata/asterisk/sounds/en/wx/southwest.gsm libdata/asterisk/sounds/en/wx/steady.gsm libdata/asterisk/sounds/en/wx/temperature.gsm libdata/asterisk/sounds/en/wx/wind-chill.gsm libdata/asterisk/sounds/en/wx/winds.gsm libdata/asterisk/sounds/en/wyoming.gsm libdata/asterisk/sounds/en/yeah.gsm libdata/asterisk/sounds/en/year.gsm libdata/asterisk/sounds/en/years.gsm libdata/asterisk/sounds/en/yes-dear.gsm libdata/asterisk/sounds/en/yes-dear2.gsm libdata/asterisk/sounds/en/you-are-caller-num.gsm libdata/asterisk/sounds/en/you-are-curr-call-num.gsm libdata/asterisk/sounds/en/you-can-press.gsm libdata/asterisk/sounds/en/you-dialed-wrong-number.gsm libdata/asterisk/sounds/en/you-entered.gsm libdata/asterisk/sounds/en/you-have-dialed.gsm libdata/asterisk/sounds/en/you-have-reached-a-test-number.gsm libdata/asterisk/sounds/en/you-have-these-options.gsm libdata/asterisk/sounds/en/you-must-first-dial.gsm libdata/asterisk/sounds/en/you-seem-impatient.gsm libdata/asterisk/sounds/en/you-sound-cute.gsm libdata/asterisk/sounds/en/you-wish-to-join.gsm libdata/asterisk/sounds/en/you_say_yes.gsm libdata/asterisk/sounds/en/your-account.gsm libdata/asterisk/sounds/en/your-msg-has-been-saved.gsm libdata/asterisk/sounds/en/your-msg-is-too-short.gsm libdata/asterisk/sounds/en/your-notifications.gsm libdata/asterisk/sounds/en/your-req-notification.gsm libdata/asterisk/sounds/en/your-temp-greeting.gsm libdata/asterisk/sounds/en/your.gsm libdata/asterisk/sounds/en/yourcallisimportant.gsm libdata/asterisk/sounds/en/yourcallisimpotent.gsm libdata/asterisk/sounds/en/zip-code.gsm libdata/asterisk/sounds/en/zombies.gsm libdata/asterisk/static-http/ajamdemo.html libdata/asterisk/static-http/appdocsxml.xslt libdata/asterisk/static-http/astman.css libdata/asterisk/static-http/astman.js libdata/asterisk/static-http/core-en_US.xml libdata/asterisk/static-http/mantest.html libdata/asterisk/static-http/prototype.js libexec/agi-bin/agi-test.agi libexec/agi-bin/eagi-sphinx-test libexec/agi-bin/eagi-test libexec/agi-bin/jukebox.agi ${PLIST.webvmail}libexec/cgi-bin/vmail man/man8/astdb2bdb.8 man/man8/astdb2sqlite3.8 man/man8/asterisk.8 man/man8/astgenkey.8 man/man8/autosupport.8 man/man8/safe_asterisk.8 sbin/astcanary sbin/astdb2bdb sbin/astdb2sqlite3 sbin/asterisk sbin/astgenkey sbin/astversion sbin/autosupport sbin/rasterisk sbin/safe_asterisk share/doc/asterisk/BUGS share/doc/asterisk/CHANGES share/doc/asterisk/COPYING share/doc/asterisk/CREDITS share/doc/asterisk/ChangeLog share/doc/asterisk/IAX2-security.pdf share/doc/asterisk/IAX2-security.txt share/doc/asterisk/LICENSE share/doc/asterisk/README-SERIOUSLY.bestpractices.md share/doc/asterisk/README.md share/doc/asterisk/README.txt share/doc/asterisk/Zaptel-to-DAHDI.txt share/examples/asterisk/acl.conf share/examples/asterisk/adsi.conf share/examples/asterisk/agents.conf share/examples/asterisk/alarmreceiver.conf share/examples/asterisk/alsa.conf share/examples/asterisk/amd.conf share/examples/asterisk/app_mysql.conf share/examples/asterisk/app_skel.conf share/examples/asterisk/ari.conf share/examples/asterisk/ast_debug_tools.conf share/examples/asterisk/asterisk.adsi share/examples/asterisk/asterisk.conf share/examples/asterisk/calendar.conf share/examples/asterisk/ccss.conf share/examples/asterisk/cdr.conf share/examples/asterisk/cdr_adaptive_odbc.conf share/examples/asterisk/cdr_beanstalkd.conf share/examples/asterisk/cdr_custom.conf share/examples/asterisk/cdr_manager.conf share/examples/asterisk/cdr_mysql.conf share/examples/asterisk/cdr_odbc.conf share/examples/asterisk/cdr_pgsql.conf share/examples/asterisk/cdr_sqlite3_custom.conf share/examples/asterisk/cdr_syslog.conf share/examples/asterisk/cdr_tds.conf share/examples/asterisk/cel.conf share/examples/asterisk/cel_beanstalkd.conf share/examples/asterisk/cel_custom.conf share/examples/asterisk/cel_odbc.conf share/examples/asterisk/cel_pgsql.conf share/examples/asterisk/cel_sqlite3_custom.conf share/examples/asterisk/cel_tds.conf share/examples/asterisk/chan_dahdi.conf share/examples/asterisk/chan_mobile.conf share/examples/asterisk/cli.conf share/examples/asterisk/cli_aliases.conf share/examples/asterisk/cli_permissions.conf share/examples/asterisk/codecs.conf share/examples/asterisk/confbridge.conf share/examples/asterisk/config_test.conf share/examples/asterisk/console.conf share/examples/asterisk/dbsep.conf share/examples/asterisk/dnsmgr.conf share/examples/asterisk/dsp.conf share/examples/asterisk/dundi.conf share/examples/asterisk/enum.conf share/examples/asterisk/extconfig.conf share/examples/asterisk/extensions.ael share/examples/asterisk/extensions.conf share/examples/asterisk/extensions.lua share/examples/asterisk/extensions_minivm.conf share/examples/asterisk/features.conf share/examples/asterisk/festival.conf share/examples/asterisk/followme.conf share/examples/asterisk/func_odbc.conf share/examples/asterisk/geolocation.conf share/examples/asterisk/hep.conf share/examples/asterisk/http.conf share/examples/asterisk/iax.conf share/examples/asterisk/iaxprov.conf share/examples/asterisk/indications.conf share/examples/asterisk/logger.conf share/examples/asterisk/manager.conf share/examples/asterisk/meetme.conf share/examples/asterisk/mgcp.conf share/examples/asterisk/minivm.conf share/examples/asterisk/misdn.conf share/examples/asterisk/modules.conf share/examples/asterisk/motif.conf share/examples/asterisk/musiconhold.conf share/examples/asterisk/muted.conf share/examples/asterisk/ooh323.conf share/examples/asterisk/osp.conf share/examples/asterisk/oss.conf share/examples/asterisk/phone.conf share/examples/asterisk/phoneprov.conf share/examples/asterisk/pjproject.conf share/examples/asterisk/pjsip.conf share/examples/asterisk/pjsip_notify.conf share/examples/asterisk/pjsip_wizard.conf share/examples/asterisk/queuerules.conf share/examples/asterisk/queues.conf share/examples/asterisk/res_config_mysql.conf share/examples/asterisk/res_config_sqlite.conf share/examples/asterisk/res_config_sqlite3.conf share/examples/asterisk/res_corosync.conf share/examples/asterisk/res_curl.conf share/examples/asterisk/res_fax.conf share/examples/asterisk/res_ldap.conf share/examples/asterisk/res_odbc.conf share/examples/asterisk/res_parking.conf share/examples/asterisk/res_pgsql.conf share/examples/asterisk/res_pktccops.conf share/examples/asterisk/res_snmp.conf share/examples/asterisk/res_stun_monitor.conf share/examples/asterisk/resolver_unbound.conf share/examples/asterisk/rtp.conf share/examples/asterisk/say.conf share/examples/asterisk/sip.conf share/examples/asterisk/sip_notify.conf share/examples/asterisk/skinny.conf share/examples/asterisk/sla.conf share/examples/asterisk/smdi.conf share/examples/asterisk/sorcery.conf share/examples/asterisk/ss7.timers share/examples/asterisk/stasis.conf share/examples/asterisk/statsd.conf share/examples/asterisk/stir_shaken.conf share/examples/asterisk/telcordia-1.adsi share/examples/asterisk/test_sorcery.conf share/examples/asterisk/udptl.conf share/examples/asterisk/unistim.conf share/examples/asterisk/users.conf share/examples/asterisk/voicemail.conf share/examples/asterisk/vpb.conf share/examples/asterisk/xmpp.conf ${PLIST.webvmail}share/httpd/htdocs/_asterisk/animlogo.gif ${PLIST.webvmail}share/httpd/htdocs/_asterisk/play.gif @ 1.27 log @asterisk16: Update to 16.24.0 Changelog: Asterisk 16.24.0 Now Available The following issues are resolved in this release: New Features made in this release: * [ASTERISK-29808] cdr: allow disabling CDR by default (Reported by N A) * [ASTERISK-29830] ami: Add AMI event for Wink (Reported by N A) * [ASTERISK-29802] app_sf: Add full tech-agnostic SF support (Reported by N A) * [ASTERISK-29759] app_sendtext: Add ReceiveText application (Reported by N A) * [ASTERISK-29706] func_json: Add JSON parsing function (Reported by N A) Bugs fixed in this release: * [ASTERISK-29888] res_pjsip_outbound_authenticator_digest: ABRT attempting to clean up auth_sess (Reported by George Joseph) * [ASTERISK-29854] func_frame_drop: fix buffer usage typo (Reported by N A) * [ASTERISK-29857] res_tonedetect: fix logic errors in code (Reported by N A) * [ASTERISK-29869] rtp sequence number can skip after DTMF under certain bridges (Reported by Torrey Searle) * [ASTERISK-29817] gethostbyname_r is misdetected on NetBSD and causes a build failure (Reported by Micha G rny) * [ASTERISK-29698] Segfault if sorcery object_lifetime_maximum and qualify_frequency the same value (Reported by Alexei Gradinari) * [ASTERISK-29851] rdtsc is not enabled (stubbed out) on NetBSD (Reported by Micha G rny) * [ASTERISK-29852] make_version uses GNU-ism that break git-svn-id parsing on NetBSD (Reported by Micha G rny) * [ASTERISK-29850] ast_get_tid() not implemented for NetBSD (Reported by Micha G rny) * [ASTERISK-29818] Build failure on NetBSD due to hmac function collision (Reported by Micha G rny) * [ASTERISK-29867] configure fails if libsrtp dev files are not installed (Reported by Sean Bright) * [ASTERISK-29856] res_rtp_asterisk: Invalid comparison creates unreachable code (Reported by N A) * [ASTERISK-29813] res_pjsip_session doesn t support multipart message bodies (Reported by George Joseph) * [ASTERISK-29858] Regression: Using external pjproject not working after hack commit (Reported by George Joseph) * [ASTERISK-29859] VoiceMailMain() fails when encountering non-numeric CALLERID(num) (Reported by Mark Murawski) * [ASTERISK-29847] pbx_variables: ASTSBINDIR is missing (Reported by N A) * [ASTERISK-29824] It s hard to make changes to bundled pjproject (Reported by George Joseph) * [ASTERISK-29695] SAY.CONF wrong logic when converting 24hour time to say 12 hour am/pm (Reported by Vincent Dubois) * [ASTERISK-29664] PJSIP processing token with % incorrectly (Reported by Dan Cropp) * [ASTERISK-29827] Support for Nordic language syntax in Queues (Reported by Mark Petersen) * [ASTERISK-29515] app_queue: QueueSummary and QueueStatus events don t exist in documentation (Reported by Luke Escude) * [ASTERISK-29746] tcptls.c: TCP client connect fails due to interrupt (Reported by Kevin Harwell) * [ASTERISK-29806] app_queue: extension state incorrect (Reported by Steve Davies) * [ASTERISK-29816] SAY_DTMF_INTERRUPT channel variable is not honored (Reported by Sean Bright) * [ASTERISK-29821] Deadlock in bridge_channel_internal_join() on local channels. (Reported by Krzysztof Trempala) * [ASTERISK-29722] test_timezone_watch breaks during DST to ST transition (Reported by Josh Soref) * [ASTERISK-29804] bundled_pjproject: sip_inv is missing multipart support in some cases (Reported by George Joseph) * [ASTERISK-29794] ast_coredumper does not delete results when requested and a specific output dir is set (Reported by Frederic Van Espen) * [ASTERISK-29803] pbx_variables: cp4 variables is used uninitialized (Reported by N A) * [ASTERISK-29766] pbx_variables: MSet truncates sets after 24 variables (Reported by N A) * [ASTERISK-29772] chan_sip: ${CHANNEL(ruri)} in Dial/Queue b (test,s,1) cause a coredump (Reported by Mark Petersen) * [ASTERISK-29790] xmldoc: Dump invalid to XML DTD: XSLT (Reported by Alexander Traud) * [ASTERISK-29791] xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported by Alexander Traud) * [ASTERISK-26991] documentation: Doxygen site is no longer being updated (Reported by Joshua C. Colp) * [ASTERISK-20259] Update Doxygen Configuration for make progdocs (Reported by Andrew Latham) * [ASTERISK-29785] res_pjsip_sdp_rtp: Warns on every offered crypto suite (Reported by Alexander Traud) * [ASTERISK-27406] Infinite loop when out of ports and rtpstart value is odd (Reported by Thomas Guebels) * [ASTERISK-28053] chan_pjsip: Wrong or missing Q.850 reason in CANCEL (Reported by Simone Lazzaris) * [ASTERISK-29761] res: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29763] main: Fix for Doxygen (Reported by Alexander Traud) Improvements made in this release: * [ASTERISK-29832] Enable pickup on channel after having received 183 Progress (Reported by Mark Petersen) * [ASTERISK-28890] res_pjsip_sdp_rtp: Keepalive not supported for video streams (Reported by Luke Escude) * [ASTERISK-29831] Queue don t play thank-you when here is no hold time announcements (Reported by Mark Petersen) * [ASTERISK-29855] frame.h: fix CNG documentation typo (Reported by N A) * [ASTERISK-29848] documentation: Document special system and channel variables (Reported by N A) * [ASTERISK-29819] utils.c: Remove all usages of ast_gethostbyname() (Reported by Sean Bright) * [ASTERISK-29815] dsp: Define magic number as macro (Reported by N A) * [ASTERISK-29807] cli: add module refresh command (Reported by N A) * [ASTERISK-29829] app_mp3: Throw warning if attempting to play a nonexistent stream (Reported by N A) * [ASTERISK-24427] Documentation is missing for a few AMI Events Including CDR and events triggered after the QueueStatus action (Reported by Dafi Ni) * [ASTERISK-29795] DIALEDPEERNUMBER not set on destination channel for Queue calls (Reported by Mark Petersen) * [ASTERISK-29801] app.c: Throw warnings for nonexistent options (Reported by N A) * [ASTERISK-29797] Support for Danish language syntax in VM (Reported by Mark Petersen) * [ASTERISK-29758] configs: Minor updates to sample configs (Reported by N A) * [ASTERISK-29800] strings: Fix misusage in comment examples (Reported by N A) * [ASTERISK-29745] pbx: Add public API for more elegant variable substitution with extensions (Reported by N A) * [ASTERISK-29729] Incompatibility with newer spandsp releases (3.0.0+) (Reported by Dustin Marquess) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0 Asterisk 16.23.0 Now Available The following issues are resolved in this release: New Features made in this release: * [ASTERISK-29720] res_tonedetect: Add call progress tone detection (Reported by N A) * [ASTERISK-18069] app_queue Add Login Time and Last Paused Times to Queue Members (Reported by Jamuel Starkey) Bugs fixed in this release: * [ASTERISK-29779] progdocs: Hidden code sections with syntax errors. (Reported by Alexander Traud) * [ASTERISK-29732] progdocs: Fix grouping for latest Doxygen (Reported by Alexander Traud) * [ASTERISK-29771] Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning (Reported by Mario Ban) * [ASTERISK-29776] stir/shaken: Requires GNU designator (Reported by Alexander Traud) * [ASTERISK-29773] progdocs: doxyref.h outdated (Reported by Alexander Traud) * [ASTERISK-29765] xmldoc: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29764] chan_misdn: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29762] channels: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29730] Segfault in __ao2_ref if refdebug = yes (Reported by Alexei Gradinari) * [ASTERISK-29748] bridging: Infinite loop when both Local channel halves in same bridge (Reported by Joshua C. Colp) * [ASTERISK-29753] parking: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29754] odbc: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29756] res_ari: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29755] frame: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29751] channel: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29750] stasis: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29752] app: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29749] res_xmpp: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29737] chan_iax2: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29747] res_pjsip: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29743] bridges: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29742] addons: Fix for Doxygen. (Reported by Alexander Traud) * [ASTERISK-29741] tests: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29740] apps: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29736] bridge_channel: Fix for Doxygen (Reported by Alexander Traud) * [ASTERISK-29733] progdocs: Avoid name with Doxygen \file (Reported by Alexander Traud) * [ASTERISK-29734] progdocs: Use Doxygen \example correctly (Reported by Alexander Traud) * [ASTERISK-29735] progdocs: Avoid multiple use of section labels (Reported by Alexander Traud) * [ASTERISK-29744] app_morsecode: Fix deadlock (Reported by N A) * [ASTERISK-29705] app_read: Fix custom terminator functionality regression (Reported by N A) * [ASTERISK-29703] res_pjsip_callerid: Fix OLI parsing (Reported by N A) * [ASTERISK-29702] sig_analog: Fix truncated buffer copy (Reported by N A) * [ASTERISK-29724] BuildSystem: In POSIX sh, == in place of = is undefined. (Reported by Alexander Traud) * [ASTERISK-28040] pbx: dialplan reload is removing minus symbol from dynamic hints (Reported by Daniel Zanutti) * [ASTERISK-29391] VoiceMail does not cancel recording on rerecord hangup (Reported by N A) * [ASTERISK-29709] res_snmp: Not build on recent Debian distributions. (Reported by Alexander Traud) * [ASTERISK-29710] stasis: Clang 13 warns about the unused but set variable dispatched. (Reported by Alexander Traud) * [ASTERISK-29711] aelparse: GCC 11.2 found two maybe uninitialized (Reported by Alexander Traud) * [ASTERISK-29713] GCC 11.2: two stringop-overread (Reported by Alexander Traud) * [ASTERISK-29682] Squash compiler issues generated by gcc 11 (Reported by George Joseph) * [ASTERISK-29693] Using with-crypto and with-ssl fails on a recompile (Reported by George Joseph) * [ASTERISK-27816] func_talkdetect s logic is completely broken (Reported by Moritz Fain) * [ASTERISK-29691] stun: Not all users provide a dst to ast_stun_request (Reported by Dennis Haney) * [ASTERISK-26497] make install downloads x86_32 variants of external modules on non Intel architectures (Reported by Corey Farrell) Improvements made in this release: * [ASTERISK-29777] documentation: Standardize example syntax (Reported by N A) * [ASTERISK-29715] app_voicemail: Refactor email generation functions (Reported by N A) * [ASTERISK-29727] Add type for JSON stasis message RTCP Report Received/Sent (Reported by Boris P. Korzun) * [ASTERISK-29714] Spelling errors (Reported by Josh Soref) * [ASTERISK-29707] chan_iax2: Allow both key and secret to be specified at dial time (Reported by N A) * [ASTERISK-29662] Add mix option to Playback application for say and filename (Reported by Shloime Rosenblum) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0 Asterisk 16.22.0 Now Available The following issues are resolved in this release: New Features made in this release: * [ASTERISK-29656] Add CHANNEL_EXISTS function (Reported by N A) Bugs fixed in this release: * [ASTERISK-20219] IAX2 Call Encryption Fails with RSA authentication (Reported by Michael Munger) * [ASTERISK-29402] res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it (Reported by Matthew Kern) * [ASTERISK-29673] app_read: Fix null pointer crash regression (Reported by N A) * [ASTERISK-29671] res_rtp_asterisk: memory leak (Reported by Jean Aunis Prescom) * [ASTERISK-29663] messaging: AMI MessageSend does not support same parameters as dialplan application (Reported by Brian J. Murrell) * [ASTERISK-29578] app_queue: Custom device state using included hints do not update (Reported by N A) * [ASTERISK-29660] Build failure when disabling PJSIP support (Reported by Guido Falsi) Improvements made in this release: * [ASTERISK-29637] Add support for future dates in Say.c (Reported by Shloime Rosenblum) * [ASTERISK-29525] PJSIP remove_existing unavailable contacts (Reported by Joseph Nadiv) * [ASTERISK-29661] func_vmcount: Add support for multiple mailboxes (Reported by N A) * [ASTERISK-29275] Support of MIME-type for wav16 (Reported by Boris P. Korzun) * [ASTERISK-29529] Add custom logging level (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0 Asterisk 16.21.1 Now Available The following issue is resolved in this release: Bugs fixed in this release: * [ASTERISK-29685] pbx_ael: Infinite loop on reload (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1 Asterisk 16.21.0 Now Available The following issues are resolved in this release: Deprecations made in this release: * [ASTERISK-29548] app_meetme: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29549] app_osploop: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29550] chan_alsa: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29551] chan_mgcp: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29552] chan_skinny: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29553] res_pktccops: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29554] cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29555] app_mysql: Deprecated in 1.8, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29557] app_ices: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29558] app_macro: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29559] app_fax: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29560] app_url: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29561] app_image: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29562] app_nbscat: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29563] app_dahdiras: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29564] cdr_syslog: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29565] chan_oss: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29566] chan_phone: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29567] chan_sip: Deprecated in 17, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29568] chan_nbs: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29569] chan_misdn: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29570] chan_vpb: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29571] res_config_sqlite: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29572] res_monitor: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) * [ASTERISK-29573] conf2ael: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) * [ASTERISK-29574] muted: Deprecated in 16, to be removed in 19 (Reported by Joshua C. Colp) Improvements made in this release: * [ASTERISK-29472] res_pjsip: OLI/ANI2 support missing (Reported by N A) * [ASTERISK-29626] app_stack: Include calling location if attempting to branch to nonexistent location (Reported by N A) * [ASTERISK-29632] Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present (Reported by Charlie Smurthwaite) * [ASTERISK-29605] chan_iax2: Add ANI2 (Reported by N A) * [ASTERISK-29508] STUN server address refresh (Reported by S bastien Duthil) * [ASTERISK-29612] bridge_basic: Don t throw warning if attended transfer is cancelled (Reported by N A) * [ASTERISK-29544] Media Cache Delayed remote sound file retrieve delays all playbacks (Reported by Andre Barbosa) * [ASTERISK-29541] app_morsecode: Add American Morse code (Reported by N A) * [ASTERISK-29495] Return integer instead of float if response is a whole number (Reported by N A) * [ASTERISK-29543] app_originate: Allow specifying codec(s) to use (Reported by N A) Bugs fixed in this release: * [ASTERISK-29673] app_read: Fix null pointer crash regression (Reported by N A) * [ASTERISK-29660] Build failure when disabling PJSIP support (Reported by Guido Falsi) * [ASTERISK-29654] pjproject includes trailing whitespace in sdp format attributes (Reported by George Joseph) * [ASTERISK-29635] MP3Player don t work with actual mpg123 versions (Reported by Carlos Oliva) * [ASTERISK-27176] test_abstract_jb: frames leak (Reported by Corey Farrell) * [ASTERISK-29634] res_snmp: gcc 11 needs -fPIC to compile correctly (Reported by George Joseph) * [ASTERISK-29630] Asterisk is unable to read extended number format terminfo files (Reported by Sean Bright) * [ASTERISK-28004] dns: Core ast_dns_get_nameservers does not support configured IPv6 servers (Reported by Isaac McDonald) * [ASTERISK-29618] ConfBridge errors on creation conference room (Reported by Alexander Zharov) * [ASTERISK-29622] ARI: external media create doesn t use body parameter (Reported by sungtae kim) * [ASTERISK-29614] app_agent_pool: XML Doc: unterminated entity reference (Reported by Alexander Traud) * [ASTERISK-29609] Subsequent ael reload will cause a lock up (Reported by Mark Murawski) * [ASTERISK-28701] app_queue: Core reload resets queue stats, even when keepstats=yes (Reported by Luke Escude) * [ASTERISK-29616] res_rtp_asterisk: sqrt(.) requires the header math.h. (Reported by Alexander Traud) * [ASTERISK-29518] sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling (Reported by Sarah Autumn) * [ASTERISK-29582] res_pjproject: Can t map pjproject log messages to Asterisk TRACE (Reported by George Joseph) * [ASTERISK-29575] app_milliwatt: Milliwatt application doesn t use the proper timings (Reported by N A) * [ASTERISK-20339] chan_mgcp, resp_pktccops ast_debug support (Reported by Tomas Maldonado) * [ASTERISK-29540] aelparse: include of context with timings fails (Reported by Alexander Traud) * [ASTERISK-29539] Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) (Reported by Ernani Jos Camargo Azevedo) New Features made in this release: * [ASTERISK-29496] Add SendMF application (Reported by N A) * [ASTERISK-29627] Add STRBETWEEN function (Reported by N A) * [ASTERISK-29628] Add file and directory functions (Reported by N A) * [ASTERISK-29531] Add SAYFILES function (Reported by N A) * [ASTERISK-29546] Add tone detection module (Reported by N A) * [ASTERISK-18454] Option for Read to be able to accept # (Reported by Sta Retji) * [ASTERISK-29542] Add audio scrambler (Reported by N A) * [ASTERISK-29478] Function to drop frames in the TX or RX directions (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0 Asterisk 16.20.0 Now Available The following issues are resolved in this release: Security bugs fixed in this release: * [ASTERISK-29415] Crash in PJSIP TLS transport (Reported by Andrew Yager) * [ASTERISK-29381] chan_pjsip: Remote denial of service by an authenticated user (Reported by Ivan Poddubny) New Features made in this release: * [ASTERISK-29389] Add PJSIP_HEADERS() and ability to read header by pattern (Reported by Igor Goncharovsky) * [ASTERISK-29477] Function to asynchronously store digits dialed (Reported by N A) * [ASTERISK-29454] New application to reload modules (Reported by N A) * [ASTERISK-29444] Add application to wait for condition (Reported by N A) * [ASTERISK-29442] app_dial: Expand A option to allow announcement playback to caller (Reported by N A) Bugs fixed in this release: * [ASTERISK-29494] cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used (Reported by N A) * [ASTERISK-29513] statsd: Remove non-standard metric type Meter (Reported by Rijnhard Hessel) * [ASTERISK-29526] G729 audio gets corrupted by Asterisk due to smoother (Reported by under) * [ASTERISK-29392] chan_iax2: Asterisk crashes when queueing video with format (Reported by Michael Welk) * [ASTERISK-29507] STUN timeout is silently delaying calls (Reported by S bastien Duthil) * [ASTERISK-27871] Remote URL in playback must end with file extension (Reported by Caesar) * [ASTERISK-29503] Updated identify/match syntax not supported by config wizard (Reported by Sean Bright) * [ASTERISK-29480] fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew (Reported by Dan Cropp) * [ASTERISK-29485] core: Inband generation of tones for Busy() and Congestion() may not occur (Reported by Joshua C. Colp) * [ASTERISK-29479] Channels are not put on hold for Session Progress with inactive audio (Reported by Bernd Zobl) Improvements made in this release: * [ASTERISK-29528] Add support for multiple files for agent announcements (Reported by N A) * [ASTERISK-29501] ARI Stasis Playback doesn t hangup call when processing a list of invalid files (Reported by Andre Barbosa) * [ASTERISK-29464] ARI PlaybackFinish skip error events (Reported by Andre Barbosa) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0 @ text @d1 1 a1 1 @@comment $NetBSD$ d146 1 d374 1 d2605 1 @ 1.26 log @asterisk16: Update to 16.15.0 Upstream changes: bugfixes minor improvements STIR/SHAKEN support @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.25 2020/05/01 07:57:36 ryoon Exp $ d250 1 d261 1 d277 1 d281 1 d296 1 d374 1 d382 1 d397 2 d544 1 @ 1.25 log @asterisk16: Update to 16.10.0 Changelog: 16.10.0: New Features made in this release: ----------------------------------- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) Bugs fixed in this release: ----------------------------------- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C. Colp) [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander Traud) [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for a user (Reported by Joshua C. Colp) [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK (Reported by nappsoft) [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK (Reported by nappsoft) [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol) [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet (Reported by Kevin Harwell) [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect. (Reported by Alexander Traud) [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. (Reported by Alexander Traud) [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by Alexander Traud) [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield, channame from dialplan (Reported by Sébastien Duthil) [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. (Reported by Alexander Traud) [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with clang. (Reported by Alexander Traud) [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang. (Reported by Alexander Traud) [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR (Reported by Jason Hord) [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported by Kevin Harwell) [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load' (Reported by Alexander Traud) [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load' (Reported by Alexander Traud) [ASTERISK-28790] - Crash during conference call using confbridge and video (Reported by Pascal Cadotte Michaud) [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is running as root (Reported by Jaco Kroon) [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream (Reported by Joshua C. Colp) [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have reflected state (Reported by Joshua C. Colp) [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when there is a SSRC change (Reported by Paulo Vicentini) [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold) [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) Improvements made in this release: ----------------------------------- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi) [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) [ASTERISK-28782] - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshua C. Colp) 16.9.0: Bugs fixed in this release: ----------------------------------- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell) [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN (Reported by Ross Beer) [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp) [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues (Reported by EDV O-TON) [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by Sébastien Duthil) [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be changed (Reported by Alex) [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer) [ASTERISK-28679] - stasis application is destroyed after its creation (Reported by Francois Blackburn) [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected (Reported by Peter Sokolov) Improvements made in this release: ----------------------------------- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh) [ASTERISK-28733] - stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp) [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) [ASTERISK-28726] - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) 16.8.0: New Features made in this release: ----------------------------------- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything (Reported by candrews) [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported by Francois Blackburn) [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer) [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by Maciej Michno) [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas) [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output (Reported by Sean Bright) [ASTERISK-24484] - Update documentation for statsd module - usage requirements unclear (Reported by Dan Jenkins) [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn) [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano) [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar) [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell) [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M. Lee) [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve XML version number (Reported by Bryan Nelson) [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens) [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) [ASTERISK-28562] - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard Kenner) [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun) [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis - Prescom) [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann) [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL) [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud) [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G) [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon) [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer) [ASTERISK-28625] - Playback of local files impacted by large media cache (Reported by Kevin Reeves) Improvements made in this release: ----------------------------------- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.24 2020/03/22 23:09:24 tnn Exp $ a62 1 include/asterisk/dns_txt.h d73 1 d114 1 d159 1 d218 1 d505 1 d531 1 d2644 1 @ 1.24 log @asterisk16: fix L§inux packaging issues @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.23 2020/01/11 08:36:13 ryoon Exp $ d63 1 @ 1.23 log @asterisk16: Update to 16.7.0 Changelog: 16.7.0 Security bugs fixed in this release: ----------------------------------- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons) Improvements made in this release: ----------------------------------- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle) [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair) Bugs fixed in this release: ----------------------------------- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G) [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer) [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt) [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL) [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer) [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed) [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell) [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL) [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson) [ASTERISK-26481] - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by Joshua C. Colp) [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer) [ASTERISK-28572] - Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha) [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell) [ASTERISK-28590] - utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" (Reported by Speed Dial Dave) [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by Salah Ahmed) [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder) [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech) [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported by Marian Piater) [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell) [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H) [ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark) [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) [ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) New Features made in this release: ----------------------------------- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell) [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H) [ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark) [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) [ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) New Features made in this release: ----------------------------------- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) 16.6.0 Security bugs fixed in this release: ----------------------------------- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) Bugs fixed in this release: ----------------------------------- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark) [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich) [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn) [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle) [ASTERISK-28477] - Crash when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis) [ASTERISK-28478] - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis) [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp) [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) New Features made in this release: ----------------------------------- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron Clark) [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.22 2019/08/20 13:47:42 ryoon Exp $ d529 1 @ 1.22 log @comms/asterisk16: import asterisk-16.5.0 Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). @ text @d1 1 a1 1 @@comment $NetBSD$ d165 1 @ 1.21 log @Remove Asterisk 1.6. This version series went end-of-line on April 21st, 2012. It most likely has multiple security issues. By this point, all users of this package should have migrated to comms/asterisk18 or comms/asterisk10 as this version has been marked as being deprecated for some time now. Note that this directory is likely to re-appear in late 2017 when Asterisk 16 comes out, assuming the current schedule is followed. However that will be a vastly different version as Asterisk 11 is only in the RC stage now (i.e. it will be five major versions after the one that is expected to be released later this year). @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.20 2012/01/17 02:12:52 jnemeth Exp $ a7 2 include/asterisk/aes.h include/asterisk/aes_internal.h d10 2 d13 1 a17 1 include/asterisk/astobj.h a18 1 include/asterisk/astosp.h d20 1 d22 12 a33 3 include/asterisk/bridging.h include/asterisk/bridging_features.h include/asterisk/bridging_technology.h d37 1 d40 1 d42 2 d45 2 d49 1 d53 4 d58 1 d64 9 d74 2 d80 1 d86 1 d88 4 d93 1 d101 2 d107 2 a108 2 include/asterisk/jabber.h include/asterisk/jingle.h d114 1 d116 4 d123 1 d125 3 a127 1 include/asterisk/netsock.h d129 1 d131 2 d135 3 d140 1 d143 3 d147 11 a157 1 include/asterisk/rtp.h d160 3 d164 1 d166 2 d171 3 d175 1 d177 18 d197 2 d204 1 d213 1 d215 2 d220 3 d224 1 d227 1 d229 3 d233 1 a254 1 lib/asterisk/modules/app_macro.so d262 1 a262 1 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share/doc/asterisk/voicemail_odbc_postgresql.txt a2531 1 share/examples/asterisk/adtranvofr.conf a2532 1 share/examples/asterisk/ais.conf d2536 4 d2542 2 d2546 1 d2549 1 d2553 1 d2555 7 d2563 1 d2568 2 d2585 1 a2585 2 share/examples/asterisk/gtalk.conf share/examples/asterisk/h323.conf a2589 2 share/examples/asterisk/jabber.conf share/examples/asterisk/jingle.conf d2597 1 d2600 1 d2605 4 d2611 1 d2613 4 d2619 1 d2621 1 d2623 2 a2624 1 share/examples/asterisk/rpt.conf d2632 4 d2637 1 a2639 1 share/examples/asterisk/usbradio.conf d2643 1 a2643 1 share/examples/rc.d/asterisk a2645 2 @@pkgdir libdata/asterisk/firmware/iax @@pkgdir libdata/asterisk/documentation/thirdparty @ 1.20 log @PR/35369 -- David Wetzel -- add support for speex codec (enabled by default) @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.19 2011/12/05 04:18:32 jnemeth Exp $ @ 1.19 log @Now that -current has sqlite3 included in base, enable it here. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.18 2011/06/06 06:25:06 jnemeth Exp $ d208 1 d256 1 @ 1.18 log @Upgrade to 1.6.2.18. This fixes several security issues including: AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, and AST-2011-006. =========================================================================== 1.6.2.18: The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Only offer codecs both sides support for directmedia. * Resolution of several DTMF based attended transfer issues. NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip * Fix channel redirect out of MeetMe() and other issues with channel softhangup * Fix voicemail sequencing for file based storage. * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip. In addition to the changes listed above, commits to resolve security issues AST-2011-005 and AST-2011-006 have been merged into this release. More information about AST-2011-005 and AST-2011-006 can be found at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18 =========================================================================== 1.6.2.17.3 The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues: * File Descriptor Resource Exhaustion (AST-2011-005) * Asterisk Manager User Shell Access (AST-2011-006) The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3 Security advisory AST-2011-005 and AST-2011-006 are available at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf =========================================================================== 1.6.2.17.2: The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which contained a bug which caused duplicate manager entries (issue #18987). The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.6.2.17.1: The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.6.2.16.2: The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.2 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.17 2010/12/12 10:19:44 jnemeth Exp $ d190 1 @ 1.17 log @Update to 1.6.2.15. This is primarily a bugfix release. - disable automatic Lua detection for now until lang/lua/builtin.mk exists The release of Asterisk 1.6.2.15 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * When using chan_skinny, don't crash when parking a non-bridged call. (Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA) * Add ability for Asterisk to try both the encoded and unencoded subscription URI for a match in hints. (Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman) * Set the caller id on CDRs when it is set on the parent channel. (Closes issue #17569. Reported, patched by tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Multiple fixes related to Local channels. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15 @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.16 2010/11/15 05:18:16 jnemeth Exp $ d309 1 @ 1.16 log @Update to 1.6.2.14 The release of Asterisk 1.6.2.14 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk. (Closes issue #17005. Reported by alexcarey. Patched by dvossel) * Parse all "Accept" headers for SIP SUBSCRIBE requests. (Closes issue #17758. Reported by ibc. Patched by dvossel) * Fix issue where queue stats would be reset on reload. (Closes issue #17535. Reported by raarts. Patched by tilghman) * Fix issue where MoH files were no longer rescanned on during a reload. (Closes issue #16744. Reported by pj. Patched by Qwell) * Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent. (Closes issue #16903. Reported, patched by Nick_Lewis) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14 @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.15 2010/09/23 23:30:38 jnemeth Exp $ d311 1 a311 1 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.19 d313 2 a314 2 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.19 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.19 @ 1.15 log @ Update to the 1.6.2 series (specifically 1.6.2.13). This is a feature update, so users that are upgrading should read UPDATE.txt. pkgsrc changes: - update to 1.6.2.13 - bury the asterisk-sounds-extra inside this one to keep it in sync - handle sound tarballs directly (upstream had changed this to do a download during the install phase and dump files in $HOME) - add new documentation files: - asterisk.txt - building_queues.txt - database_transactions.txt - followme.txt ======== 1.6.2.13 ======== This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13 ======== 1.6.2.12 ======== The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue where DNID does not get cleared on a new call when using immediate=yes with ISDN signaling. (Closes issue #17568. Reported by wuwu. Patched by rmudgett) * Several updates to res_config_ldap. (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer. Tested by suretec) * Prevent loss of Caller ID information set on local channel after masquerade. (Closes issue #17138. Reported by kobaz, patched by jpeeler) * Fix SIP peers memory leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set, but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (Closes issue #17633. Reported, patched by urosh. Patched by dvossel. Tested by: dvossel, urosh, okrief, alecdavis) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12 ======== 1.6.2.11 ======== The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it. (Closes issue #17504. Reported, patched by rrb3942) * Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (Closes issue #16029. Reported, patched by Guggemand) * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (Closes issue #17469. Reported, patched by wdoekes) * Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. (Closes issue #15871. Reported, patched by Ivan) * Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers. (Closes issue #16102. Reported, patched by Delvar) * cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist. (Closes issue #17478. Reported, patched by kobaz) * Avoid crashing when installing a duplicate translation path with a lower cost. (Closes issue #17092. Reported, patched by moy) * Add missing handling for ringing state for use with queue empty options. (Closes issue #17471. Reported, patched by jazzy) * Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (Closes issue #17498. Reported, patched by corruptor) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11 ======== 1.6.2.10 ======== The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow users to specify a port for DUNDI peers. (Closes issue #17056. Reported, patched by klaus3000) * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. (Closes issue #16815. Reported, patched by rain) * If there is realtime configuration, it does not get re-read on reload unless the config file also changes. (Closes issue #16982. Reported, patched by dmitri) * Send AgentComplete manager event for attended transfers. (Closes issue #16819. Reported, patched by elbriga) * Correct manager variable 'EventList' case. (Closes issue #17520. Reported, patched by kobaz) In addition, changes to res_timing_pthread that should make it more stable have also been implemented. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10 ======= 1.6.2.9 ======= The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix the PickupChan() application (Closes issue #16863. Reported, patched by schern. Patched by cjacobsen. Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) * Improve logging by displaying line number (Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by dant, pabelanger, lmadsen) * Notify CLI when modules are loaded/unloaded (Closes issue #17308. Reported, patched by pabelanger. Tested by russell) * Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems (Closes issue #17028. Reported by pabelanger. Patched by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 ======= 1.6.2.8 ======= The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Enable auto complete for CLI command 'logger set level'. (Closes issue #17152. Reported, patched by pabelanger) * Make the mixmonitor thread process audio frames faster. (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) * Add missing 'useragent' field to sip-friends.sql file. (Closes issue #17171. Reported, patched by thehar) * Add example dialplan for dialing ISN numbers (http://www.freenum.org) (Closes issue #17058. Reported, patched by pprindeville) * Fix issue with double "sip:" in header field. (Closes issue #15847. Reported, patched by ebroad) * Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'. (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) * When StopMonitor() is called, ensure that it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Small error in the T.140 RTP port verbose log. (Closes issue #16998. Reported, patched by frawd. Tested by russell) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8 ======= 1.6.2.7 ======= The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Include an extra newline after "Aliased CLI command" to get back the prompt. (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) * Prevent segfault if bad magic number is encountered. (Issue #17037. Reported, patched by alecdavis) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 ======= 1.6.2.6 ======= The release of Asterisk 1.6.2.6 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Make sure to clear red alarm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * Fix problem with duplicate TXREQ packets in chan_iax2 (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) * Fix crash in app_voicemail related to message counting. (Closes issue #16921. Reported, tested by whardier. Patched by seanbright) * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts (Reported, Patched, and Tested by alecdavis) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) * Fix ConfBridge crash when no timing module is loaded. (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6 ======= 1.6.2.5 ======= The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.6.2.5 The releases of Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 resolve an issue with invalid parsing of ACL (Access Control List) rules leading to a possible compromise in security. The issue and resolution are described in the AST-2010-003 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2010-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.5 Security advisory AST-2010-003 is available at: http://downloads.asterisk.org/pub/security/AST-2010-003.pdf ======= 1.6.2.4 ======= The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.6.2.4 The releases of Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 include documention describing a possible dialplan string injection with common usage of the ${EXTEN} (and other expansion variables). The issue and resolution are described in the AST-2010-002 security advisory. If you have a channel technology which can accept characters other than numbers and letters (such as SIP) it may be possible to craft an INVITE which sends data such as 300&Zap/g1/4165551212 which would create an additional outgoing channel leg that was not originally intended by the dialplan programmer. Please note that this is not limited to an specific protocol or the Dial() application. The expansion of variables into programmatically-interpreted strings is a common behavior in many script or script-like languages, Asterisk included. The ability for a variable to directly replace components of a command is a feature, not a bug - that is the entire point of string expansion. However, it is often the case due to expediency or design misunderstanding that a developer will not examine and filter string data from external sources before passing it into potentially harmful areas of their dialplan. With the flexibility of the design of Asterisk come these risks if the dialplan designer is not suitably cautious as to how foreign data is allowed to enter the system unchecked. This security release is intended to raise awareness of how it is possible to insert malicious strings into dialplans, and to advise developers to read the best practices documents so that they may easily avoid these dangers. For more information about the details of this vulnerability, please read the security advisory AST-2010-002, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.4 Security advisory AST-2010-002 is available at: http://downloads.asterisk.org/pub/security/AST-2010-002.pdf The README-SERIOUSLY.bestpractices.txt document is available in the top-level directory of your Asterisk sources, or available in all Asterisk branches from 1.2 and up. http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt ======= 1.6.2.3 ======= Was never released. ======= 1.6.2.2 ======= The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.6.2.2 The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001. The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well. For more information about the details of this vulnerability, please read the security advisory AST-2009-009, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.2 Security advisory AST-2010-001 is available at: http://downloads.asterisk.org/pub/security/AST-2010-001.pdf ======= 1.6.2.1 ======= The release of Asterisk 1.6.2.1 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * CLI 'queue show' formatting fix. (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by ppyy.) * Fix misreverting from 177158. (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.) * Fixes subscriptions being lost after 'module reload'. (Closes issue #16093. Reported by jlaroff. Patched by dvossel.) * app_queue segfaults if realtime field uniqueid is NULL (Closes issue #16385. Reported, Tested, Patched by haakon.) * Fix to Monitor which previously assumed the file to write to did not contain pathing. (Closes issue #16377, #16376. Reported by bcnit. Patched by dant. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.1-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.1 ======= 1.6.2.0 ======= The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a complete list of changes, please see the CHANGES file. For those upgrading from a previous release, please see UPGRADE.txt It should be explicitly stated that Asterisk 1.6.2.0 is a major upgrade over any previous release, and special care should be taken when upgrading existing systems. Please see the UPGRADE.txt file for more information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/UPGRADE.txt A detailed overview to the new features available in Asterisk 1.6.2.0 are forthcoming within the next few days. Please watch http://blogs.asterisk.org for further information! Below is a summary of several new features available in this release: * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with support for LibOpenR2. http://www.libopenr2.org/ * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this option is enabled, Asterisk will watch for a CNG tone in the incoming audio for a received call. If it is detected, the channel will jump to the 'fax' extension in the dialplan. * A new application, Originate, has been introduced, that allows asynchronous call origination from the dialplan. * Added ConfBridge dialplan application which does conference bridges without DAHDI. For information on its use, please see the output of "core show application ConfBridge" from the CLI. * extensions.conf now allows you to use keyword "same" to define an extension without actually specifying an extension. It uses exactly the same pattern as previously used on the last "exten" line. For example: exten => 123,1,NoOp(something) same => n,SomethingElse() * Asterisk now provides the ability to define custom CLI aliases. For example, if you would like to define short form aliases for frequently used commands, such as "sh ch" for "core show channels", that is now possible. See the cli_aliases.conf configuration file for more information. * Asterisk now has support for subscribing to the state of remote voice mailboxes via SIP. * Asterisk now includes expanded HD codec support. G.722.1 and G.722.1C (Siren7/Siren14) passthrough, recording, and playback is now supported. Transcoding will be made available via add-on modules soon for this version of Asterisk. This is just a subset of the changes available in this release. Please see the CHANGES file for additional information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/CHANGES A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.0-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.0 @ text @d1 1 a1 1 @@comment $NetBSD$ d91 1 d2420 1 @ 1.14 log @Update to 1.6.1.20. Apparently they decided to do one final bug fix release: The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to security maintenance only. The releases of Asterisk 1.6.0.28 and 1.6.1.20 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix issue where MixMonitor() recordings would be shorter than total duration . (Closes issue #17078. Reported,tested by geoff2010. Patched by dhubbard) * When StopMonitor() is called, ensure it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Allow hidecalleridname feature to work. (Closes issue #17143. Reported, patched by djensen99) * Resolve deadlocks in chan_local. (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) * Ensure channel state is not incorrectly set in the case of a very early answer by chan_dahdi. (Closes issue #17067. Reported, patched by tzafrir) * Registration fix for SIP realtime. Make sure realtime fields are not empty. (Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney) Information about the Asterisk maintenance schedule is available at: http://www.asterisk.org/asterisk-versions For a full list of changes in the current release candidates, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.20 @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.13 2010/05/20 17:14:45 jnemeth Exp $ d22 3 d26 1 d92 1 d103 1 d114 2 d124 1 d149 1 d152 1 d180 4 d190 1 d216 2 d223 1 d253 1 d271 1 a278 1 lib/asterisk/modules/res_indications.so d289 2 a294 1 libdata/asterisk/moh/.asterisk-moh-opsound-wav-2.03 d308 39 a346 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.18 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.18 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.18 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.18 d354 49 d405 15 d422 88 d512 1 d516 1 d520 1 d537 1 d539 1 d542 3 d553 13 d567 21 d598 10 d620 2 d731 76 d813 195 d1009 9 d1019 57 d1077 42 d1180 60 d1241 111 d1356 12 d1395 76 d1475 5 d1481 4 d1486 4 d1491 6 d1498 16 d1517 1 d1527 46 d1574 2 d1577 17 d1604 37 d1659 138 d1798 3 d1802 3 d1807 1 d1810 29 d1952 106 d2285 7 d2293 9 d2420 1 d2422 1 d2426 1 d2431 1 d2476 2 d2517 1 a2539 2 @@pkgdir libdata/asterisk/sounds/fr @@pkgdir libdata/asterisk/sounds/es d2541 1 @ 1.13 log @Update to Asterisk 1.6.1.19. 1.6.1.18 and 1.6.1.19 are primarily bug fix releases. At this point the 1.6.1 series is going to security fixes only. That means this package will be moving to the 1.6.2 series in the near future. ----- 1.6.1.18: The following are a few of the issues resolved by community developers: * Make sure to clear red alarm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * Fix problem with duplicate TXREQ packets in chan_iax2. (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) * Update documentation to not imply we support overriding options. (Closes issue #16855. Reported by davidw) * Modify queued frames from Local channels to not set the other side to up. (Closes issue #16816. Reported, tested by jamhed) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.18 ----- 1.6.1.19: The following are a few of the issues resolved by community developers: * Fix building CDR and CEL SQLite3 modules. (Closes issue #17017. Reported by alephlg. Patched by seanbright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Pass the PID of the Asterisk process, not the PID of the canary. (Closes issue #17065. Reported by globalnetinc. Patched by makoto. Tested by frawd, globalnetinc) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.19 @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.12 2010/05/07 23:57:56 jnemeth Exp $ d273 1 a273 1 libdata/asterisk/moh/.asterisk-moh-opsound-wav d287 4 a290 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.17 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.17 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.17 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.17 d321 3 d328 1 d390 1 d415 1 d417 1 d419 1 d421 1 d423 1 d425 5 d542 1 d1073 1 @ 1.12 log @Add an "ldap" option which defaults to enabled, since most modern systems come with LDAP support built-in. This has no effect on such systems. However, on older systems, it will pull in openldap-client. But, a builder may still disable the option if they wish. This fixes: PR pkg/41987 - Robert Elz -- comms/asterisk16 PLIST problem @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.11 2010/05/07 07:20:55 jnemeth Exp $ d948 3 a1067 3 share/doc/asterisk/asterisk-1.6.1.14-summary.txt share/doc/asterisk/asterisk-1.6.1.16-summary.txt share/doc/asterisk/${PKGNAME}-summary.txt @ 1.11 log @Install various docs found in the tarball. README-SERIOUSLY.bestpractices.txt is the new README from 1.6.1.16 and AST-2010-002. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.10 2010/05/06 20:10:17 jnemeth Exp $ d253 1 a253 1 lib/asterisk/modules/res_config_ldap.so @ 1.10 log @Add a webvmail option which installs the vmail.cgi script accessing voicemail using a browser. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.9 2010/03/01 07:06:48 jnemeth Exp $ d1049 53 @ 1.9 log @ Update to Asterisk 1.6.1.17. This fixes AST-2010-001 and AST-2010-003. AST-2010-002 was just a warning about dialplan scripting errors that could lead to security issues. Asterisk 1.6.1.13: general bug fixes Asterisk 1.6.1.14: fix AST-2010-001 Asterisk 1.6.1.15: not released, skipped for security releases Asterisk 1.6.1.16: fix AST-2010-002 Asterisk 1.6.1.17: fix AST-2010-003 Note that the only change in Asterisk 1.6.1.16 was the addtion of a README file. However, the package doesn't install random docs. That is planned for a future update seperate from the upstream updates. ----- Asterisk 1.6.1.13: The release of Asterisk 1.6.1.13 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Restarts busydetector (if enabled) when DTMF is received after call is bridged (Closes issue #16389. Reported, Tested, Patched by alecdavis.) * Send parking lot announcement to the channel which parked the call, not the park-ee. (Closes issue #16234. Reported, Tested by yeshuawatso. Patched by tilghman.) * When the field is blank, don't warn about the field being unable to be coerced just skip the column. (Closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list.) * Don't queue frames to channels that have no means to process them. (Closes issue #15609. Reported, Tested by aragon. Patched by tilghman.) * Fixes holdtime playback issue in app_queue. (Closes issue #16168. Reported, Patched by nickilo. Tested by wonderg, nickilo.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.13-summary.t xt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.13 ----- Asterisk 1.6.1.14: The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001. The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well. For more information about the details of this vulnerability, please read the security advisory AST-2009-009, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.14 Security advisory AST-2010-001 is available at: http://downloads.asterisk.org/pub/security/AST-2010-001.pdf ----- Asterisk 1.6.1.16: The releases of Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 include documention describing a possible dialplan string injection with common usage of the ${EXTEN} (and other expansion variables). The issue and resolution are described in the AST-2010-002 security advisory. If you have a channel technology which can accept characters other than numbers and letters (such as SIP) it may be possible to craft an INVITE which sends data such as 300&Zap/g1/4165551212 which would create an additional outgoing channel leg that was not originally intended by the dialplan programmer. Please note that this is not limited to an specific protocol or the Dial() application. The expansion of variables into programmatically-interpreted strings is a common behavior in many script or script-like languages, Asterisk included. The ability for a variable to directly replace components of a command is a feature, not a bug - that is the entire point of string expansion. However, it is often the case due to expediency or design misunderstanding that a developer will not examine and filter string data from external sources before passing it into potentially harmful areas of their dialplan. With the flexibility of the design of Asterisk come these risks if the dialplan designer is not suitably cautious as to how foreign data is allowed to enter the system unchecked. This security release is intended to raise awareness of how it is possible to insert malicious strings into dialplans, and to advise developers to read the best practices documents so that they may easily avoid these dangers. For more information about the details of this vulnerability, please read the security advisory AST-2010-002, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.16 Security advisory AST-2010-002 is available at: http://downloads.asterisk.org/pub/security/AST-2010-002.pdf The README-SERIOUSLY.bestpractices.txt document is available in the top-level directory of your Asterisk sources, or available in all Asterisk branches from 1.2 and up. http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt ----- Asterisk 1.6.1.17: The releases of Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 resolve an issue with invalid parsing of ACL (Access Control List) rules leading to a possible compromise in security. The issue and resolution are described in the AST-2010-003 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2010-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.17 Security advisory AST-2010-003 is available at: http://downloads.asterisk.org/pub/security/AST-2010-003.pdf ----- @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.8 2009/12/30 04:11:02 jnemeth Exp $ d1031 1 d1127 2 @ 1.8 log @ Update to 1.6.1.12. 1.6.1.10 and 1.6.1.12 are general bug fix releases. For more information see: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-1.6.1.10-summary.html or http://tinyurl.com/yzyr9tt and http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-1.6.1.12-summary.html or http://tinyurl.com/yfxlyjp . 1.6.1.11 fixes AST-2009-010 which allows people to remotely crash the server. The description of the issue is: An attacker sending a valid RTP comfort noise payload containing a data length of 24 bytes or greater can remotely crash Asterisk. Commit during freeze approved by wiz@@. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.7 2009/11/20 04:30:08 jnemeth Exp $ d287 4 a290 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.16 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.16 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.16 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.16 @ 1.7 log @ Fix three security advisories by updating to Asterisk 1.6.1.9 and update PLIST for new Music On Hold files. 1.6.1.8 fixes AST-2009-007. ----- A missing ACL check for handling SIP INVITEs allows a device to make calls on networks intended to be prohibited as defined by the "deny" and "permit" lines in sip.conf. The ACL check for handling SIP registrations was not affected. ----- 1.6.1.9 fixes AST-2009-008 and AST-2009-009. ----- It is possible to determine if a peer with a specific name is configured in Asterisk by sending a specially crafted REGISTER message twice. The username that is to be checked is put in the user portion of the URI in the To header. A bogus non-matching value is put into the username portion of the Digest in the Authorization header. If the peer does exist the second REGISTER will receive a response of 403 Authentication user name does not match account name. If the peer does not exist the response will be 404 Not Found if alwaysauthreject is disabled and 401 Unauthorized if alwaysauthreject is enabled. ----- Asterisk includes a demonstration AJAX based manager interface, ajamdemo.html which uses the prototype.js framework. An issue was uncovered in this framework which could allow someone to execute a cross-site AJAX request exploit. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.6 2009/09/14 08:44:51 jnemeth Exp $ a268 1 libdata/asterisk/firmware/iax/iaxy.bin d287 4 a290 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.15 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.15 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.15 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.15 d678 1 d1128 1 @ 1.6 log @ Update to Asterisk 1.6.1.6 - 1.6.1.6 fixes AST-2009-006 which is an IAX2 DOS vulnerability - 1.6.1.5 contains a variety of bug fixes: Category: Applications/app_chanspy #15660: ChanSpy "whisper" is broken in 1.4.26 Category: Applications/app_fax #15606: app_fax.c is not compiling under OpenBSD #15610: T.38 re-INVITE received after T.38 already negotiated fails Category: Applications/app_milliwatt #15386: [patch] Milliwatt() is off by -11dbm Category: Applications/app_mixmonitor #15699: [patch] using ast_free instead of mixmonitor_free Category: Applications/app_queue #14536: [patch] After a caller is processed by app_queue the queue_log logs the hangup as TRANSFER #15664: [patch] QUEUE_MEMBER_LIST() returns member names instead of Category: Applications/app_stack #15557: [patch] Gosub() dequotes once more than Macro() #15617: [patch] crash in LOCAL() if Gosub stack is allocated but empty Category: Applications/app_voicemail #15717: MWI is not sent to a SIP phone upon registration, but is after the mailbox is updated/checked #15720: opendir() return code is not checked in last_message_index() Category: Applications/app_voicemail/IMAP #14496: [patch] IMAP crash multiple callers / callers hangup at beep #14597: greetings can not be retrieved from IMAP #14950: [patch] Greetings are stored as IMAP messages even when imapgreetings=no #15729: IMAP greetings not stored in dovecot Category: CDR/General #15751: [patch] Core dump in ast_bridge_call features.c line 2772 Category: Channels/chan_agent #15668: AGENTACCEPTDTMF is incorrectly spelled as AGENTACCEPTDMTF in code to recognize channel variables. Category: Channels/chan_dahdi #15655: [patch] Dialplan starts execution before call is accepted #15727: [patch] Message Waiting Indication(MWI) is randomly generated when FXO is set to DTMF Caller ID Category: Channels/chan_misdn #12113: [patch] asterisk crash at reload chan_misdn.so Category: Channels/chan_sip/General #12869: [patch] 'context' doesn't change when 'sip reload' issued when driven from realtime #15362: [patch] log message output is truncated #15596: [patch] all codecs allowed, but textsupport=no crashes on T140RED enabled call Category: Channels/chan_sip/Registration #14366: [patch] Registration expiry not compatible with some ITSP #15539: [patch] Register request line contains wrong address when domain and registrar host differ Category: Channels/chan_sip/T.38 #15182: [patch] T.38 invite does not always comply with RFC 2327 Category: Channels/chan_sip/Video #15121: [patch] Video support in SIP channel driver appears to be totally broken Category: Core/BuildSystem #15697: most cleaner alaw don't compile #15698: [patch] If enable DEBUG_FD_LEAKS - h323 can't start. #15714: [patch] Asterisk won't build with curl unless curl_config is present Category: Core/General #14730: [patch] Fix runlevels in Debian rc files #15273: [patch] german time (20:01:00 oh clock) is announced wrong #15649: T38 Faxing failing on 1.6.1 svn #15667: LOGGER WARNING : error executing after rotate Category: Core/ManagerInterface #15397: [patch] segfault in action_coreshowchannels() at manager.c #15730: [patch] manager keeps creating /tmp/ast-ami-XXXXXX files (without deleting) when a single manager client remains logged in Category: Core/PBX #15242: [patch] log does not indicate which function is missing closing parenthesis Category: Documentation #15755: Description in queues.conf on call recording is slightly misleading Category: Functions/func_iconv #15169: When building with uClibc, configure script mistakenly assumes iconv is always available Category: General #15571: [patch] 'received' typos in trunk, in 6 files #15595: [patch] fix spelling for typos, mainly in comments. Category: PBX/pbx_dundi #15322: [patch] DUNDILOOKUP() does not accept comma as argument separator Category: Resources/General #15624: res_ais, communication ok, but wrong state send and receive. Category: Resources/res_config_ldap #13725: [patch] ERROR[7387]: res_config_ldap.c:1292 update_ldap: Couldn't modify dn:cn=1001,dc=xxx,dc=xxx because Invalid syntax #15710: Typo in LDAP schema files on line 598 Category: Resources/res_musiconhold #15051: [patch] Moh class set in the dialplan is ignored with realtime moh ---------------------------------------------------------------------- Commits Not Associated with an Issue [Back to Top] This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well. +------------------------------------------------------------------------+ | Revision | Author | Summary | Issues | | | | | Referenced | |----------+------------+-----------------------------------+------------| | | | Restore explicit export of | | | 209058 | kpfleming | ASTCFLAGS/ASTLDFLAGS and | | | | | underscore-variants to sub-makes. | | |----------+------------+-----------------------------------+------------| | 209237 | mmichelson | Gracefully handle malformed RTP | | | | | text packets. | | |----------+------------+-----------------------------------+------------| | 209262 | kpfleming | Make T.38 switchover in | | | | | ReceiveFAX synchronous. | | |----------+------------+-----------------------------------+------------| | 209281 | kpfleming | Cleanup T.38 negotiation changes. | | |----------+------------+-----------------------------------+------------| | 209327 | tilghman | Publish French extra sounds | | |----------+------------+-----------------------------------+------------| | | | Fix some places where | | | 209714 | russell | ast_event_type was used instead | | | | | of ast_event_ie_type. | | |----------+------------+-----------------------------------+------------| | 209781 | kpfleming | Minor changes inspired by testing | | | | | with latest GCC. | | |----------+------------+-----------------------------------+------------| | 209900 | russell | Resolve a valgrind warning about | #15396 | | | | a read from uninitialized memory. | | |----------+------------+-----------------------------------+------------| | 211115 | russell | Resolve a deadlock involving | | | | | app_chanspy and masquerades. | | |----------+------------+-----------------------------------+------------| | 211277 | tilghman | Small oops. Clear the flags which | | | | | have been checked. | | |----------+------------+-----------------------------------+------------| | 211569 | tilghman | AST-2009-005 | | |----------+------------+-----------------------------------+------------| | 211586 | tilghman | Conversion specifiers, not format | | | | | specifiers | | |----------+------------+-----------------------------------+------------| | | | Check an actual populated | | | 212069 | file | variable when seeing if we need | | | | | to do video or not. | | |----------+------------+-----------------------------------+------------| | | | Ensure that T38FaxVersion is put | | | 212115 | kpfleming | into outgoing SDP in the proper | | | | | case. | | |----------+------------+-----------------------------------+------------| | 212386 | seanbright | Handle slin16 for extra sounds as | | | | | well. | | |----------+------------+-----------------------------------+------------| | 212768 | rmudgett | Removed some deadwood and added | | | | | some doxygen comments. | | |----------+------------+-----------------------------------+------------| | | | Make the default extconfig.conf | | | 212862 | tilghman | match entries with the sample | | | | | res_mysql.conf. | | |----------+------------+-----------------------------------+------------| | 212928 | kpfleming | Convert this branch to Opsound | | | | | music-on-hold. | | |----------+------------+-----------------------------------+------------| | | | Remove some | | | 212942 | kpfleming | accidentally-committed | | | | | properties. | | |----------+------------+-----------------------------------+------------| | 213449 | twilson | Make LOAD_ORDER actually work | | |----------+------------+-----------------------------------+------------| | 213452 | twilson | Oops, committed this first. Make | | | | | the merged property happy | | |----------+------------+-----------------------------------+------------| | | | Make autoheader descriptions | | | 214365 | tilghman | render correctly in our | #14906 | | | | autoconfig.h file. | | |----------+------------+-----------------------------------+------------| | | | One more build system change, to | | | 214496 | tilghman | make the descriptions look | | | | | better, if we have better | | | | | information. | | +------------------------------------------------------------------------+ @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.5 2009/08/20 20:37:06 jnemeth Exp $ d275 3 a277 3 libdata/asterisk/moh/CHANGES-asterisk-moh-opsound-2.01 libdata/asterisk/moh/CREDITS-asterisk-moh-opsound-2.01 libdata/asterisk/moh/LICENSE-asterisk-moh-opsound-2.01 @ 1.5 log @Digium in their infinite wisdom decided to replace the Music-On-Hold sounds files in all release tarballs of Asterisk. This is just an update for the new sound files. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.4 2009/07/07 08:14:42 jnemeth Exp $ d275 3 a277 3 libdata/asterisk/moh/CHANGES-asterisk-moh-opsound-wav libdata/asterisk/moh/CREDITS-asterisk-moh-opsound-wav libdata/asterisk/moh/LICENSE-asterisk-moh-opsound-wav d946 1 d1126 2 @ 1.4 log @Update to Asterisk 1.6.1.1 ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 ------------- ------------------------------------------------------------------------------ Device State Handling --------------------- * The event infrastructure in Asterisk got another big update to help support distributed events. It currently supports distributed device state and distributed Voicemail MWI (Message Waiting Indication). A new module has been merged, res_ais, which facilitates communicating events between servers. It uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM (Cluster Management) and EVT (Event) services to maintain a cluster of Asterisk servers, and to share events between them. For more information on setting this up, see doc/distributed_devstate.txt. Dialplan Functions ------------------ * Added a new dialplan function, AST_CONFIG(), which allows you to access variables from an Asterisk configuration file. * The JACK_HOOK function now has a c() option to supply a custom client name. * Added two new dialplan functions from libspeex for audio gain control and denoise, AGC() and DENOISE(). Both functions can be applied to the tx and rx directions of a channel from the dialplan. * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages based on other parameters. The default is still to search based on the forwarding station ID. However, there are new options that allow you to search based on the message desk terminal ID, or the message desk number. * TIMEOUT() has been modified to be accurate down to the millisecond. * ENUM*() functions now include the following new options: - 'u' returns the full URI and does not strip off the URI-scheme. - 's' triggers ISN specific rewriting - 'i' looks for branches into an Infrastructure ENUM tree - 'd' for a direct DNS lookup without any flipping of digits. * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa') * CHANNEL() now has options for the maximum, minimum, and standard or normal deviation of jitter, rtt, and loss for a call using chan_sip. DAHDI channel driver (chan_dahdi) Changes ---------------------------------------- * Channels can now be configured using named sections in chan_dahdi.conf, just like other channel drivers, including the use of templates. * The default for pridialplan has changed from 'national' to 'unknown'. PBX Changes ----------- * It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated. * Dialplan matching has been extended to allow an extension to return to the PBX core to wait for more digits. This is done by using the new dialplan application called "Incomplete". This will permit a whole new level of extension control, by giving the administrator more control over early matches employing one of the short-circuit pattern match operators. Note that custom applications can trigger this same behavior by returning the special value AST_PBX_INCOMPLETE. The dial() application ---------------------- * Dial has a new option: F(context^extension^pri), which permits a callee to continue in the dialplan, at the specified label, if the caller hangs up. * The Dial() application no longer copies the language used by the caller to the callee's channel. If you desire for the caller's channel's language to be used for file playback to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" . The chanspy() application ------------------------- * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the technology name (e.g. SIP, IAX, etc) of the channel being spied on. * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is like the pre-existing whisper mode, except that the spy can also talk to the participant on the bridged channel as well. * Chanspy has a new option, 'n', which will allow for the spied-on party's name to be spoken instead of the channel name or number. For more information on the use of this option, issue the command "core show application ChanSpy" from the Asterisk CLI. * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between spy modes. Use of this feature overrides the typical use of numeric DTMF. In other words, if using the 'd' option, it is not possible to enter a number to append to the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will change to whisper mode, and pressing 6 will change to barge mode. Other Application Changes ------------------------- * Directory now permits both first and last names to be matched at the same time. In addition, the number of digits to enter of the name can be set in the arguments to Directory; previously, you could enter only 3, regardless of how many names are in your company. For large companies, this should be quite helpful. * Voicemail now permits a mailbox setting to wrap around from first to last messages, if the "messagewrap" option is set to a true value. * Voicemail now permits an external script to be run, for password validation. The script should output "VALID" or "INVALID" on stdout, depending upon the wish to validate or invalidate the password given. Arguments are: "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for more details * The voicemail externnotify script now accepts an additional (last) parameter containing the number of urgent messages in the INBOX. * The Jack application now has a c() option to supply a custom client name. * ExternalIVR now takes several options that affect the way it performs, as well as having several new commands. Please see doc/externalivr.txt for the complete documentation. * Added ability to communicate over a TCP socket instead of forking a child process for the ExternalIVR application. * ChanIsAvail has a new option, 'a', which will return all available channels instead of just the first one if you give the function more then one channel to check. * PrivacyManager now takes an option where you can specify a context where the given number will be matched. This way you have more control over who is allowed and it stops the people who blindly enter 10 digits. * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks answer times, disposition, on orig CDR against updates; 'D' Copies the disposition from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(), obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func. * SendImage() no longer hangs up the channel on error; instead, it sets the status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or 'UNSUPPORTED'. This change makes SendImage() more consistent with other applications. * Park has a new option, 's', which silences the announcement of the parking space number. * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as invalid input and will be assumed to mean that no timeout is desired. SIP Changes ----------- * Added DNS manager support to registrations for peers referencing peer entries. DNS manager runs in the background which allows DNS lookups to be run asynchronously as well as periodically updating the IP address. These properties allow for better performance as well as recovery in the event of an IP change. * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve load/reload of large numbers of peers/users by ~40x (for large lists of peers. Initially, we saw 4x improvement in call setup/destruction, but at the time of merging, this gain has disappeared; further research will be done to try and restore this performance improvement. Astobj2 refcounting is now used for users, peers, and dialogs. Users are encouraged to assist in regression testing and problem reporting! * Added ability to specify registration expiry time on a per registration basis in the register line. * Added support for Realtime Text redundancy - T140 RED - in T.140 to prevent text loss due to lost packets. * Added t38pt_usertpsource option. See sip.conf.sample for details. * Added SIPnotify AMI command, for sending arbitrary SIP notify commands. * 'sip show peers' and 'sip show users' display their entries sorted in alphabetical order, as opposed to the order they were in, in the config file or database. * Videosupport now supports an additional option, "always", which always sets up video RTP ports, even on clients that don't support it. This helps with callfiles and certain transfers to ensure that if two video phones are connected, they will always share video feeds. IAX Changes ----------- * Existing DNS manager lookups extended to check for SRV records. * IAX2 encryption support has been improved to support periodic key rotation within a call for enhanced security. The option "keyrotate" has been provided to disable this functionality to preserve backwards compatibility with older versions of IAX2 that do not support key rotation. CLI Changes ----------- * New CLI command, "config reload " which reloads any module that references that particular configuration file. Also added "config list" which shows which configuration files are in use. * New CLI commands, "pri show version" and "ss7 show version" that will display which version of libpri and libss7 are being used, respectively. A new API call was added so trunk will now have to be compiled against a versions of libpri and libss7 that have them or it will not know that these libraries exist. * The commands "core show globals", "core set global" and "core set chanvar" has been deprecated in favor of the more semanticly correct "dialplan show globals", "dialplan set chanvar" and "dialplan set global". * New CLI command "dialplan show chanvar" to list all variables associated with a given channel. DNS manager changes ------------------- * Addresses managed by DNS manager now can check to see if there is a DNS SRV record for a given domain and will use that hostname/port if present. AMI - The manager (TCP/TLS/HTTP) -------------------------------- * The Status action now takes an optional list of variables to display along with channel status. ODBC Changes ------------ * res_odbc no longer has a limit of 1023 total possible unshared connections, as some people were running into this limit. This limit has been increased to 4.2 billion. Queue changes ------------- * The TRANSFER queue log entry now includes the caller's original position in the transferred-from queue. * A new configuration option, "timeoutpriority" has been added. Please see the section labeled "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option as well as an explanation about timeout options in general Realtime changes ---------------- * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given adaptive capabilities. What this means in practical terms is that if your realtime table lacks critical fields, Asterisk will now emit warnings to that effect. Also, some of the realtime drivers have the ability (if configured) to automatically add those columns to the table with the correct type and length. Miscellaneous ------------- * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using the 'setvar' option to cause a given audio file to be played upon completion of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and Skinny channels only. * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt for more information. * Config file variables may now be appended to, by using the '+=' append operator. This is most helpful when working with long SQL queries in func_odbc.conf, as the queries no longer need to be specified on a single line. @ text @d1 1 a1 1 @@comment $NetBSD$ d274 9 a282 6 libdata/asterisk/moh/.asterisk-moh-freeplay-wav libdata/asterisk/moh/CHANGES-asterisk-moh-freeplay-wav libdata/asterisk/moh/LICENSE-asterisk-moh-freeplay-wav libdata/asterisk/moh/fpm-calm-river.wav libdata/asterisk/moh/fpm-sunshine.wav libdata/asterisk/moh/fpm-world-mix.wav a1124 2 @@pkgdir libdata/asterisk/sounds/fr @@pkgdir libdata/asterisk/sounds/es @ 1.3 log @Convert @@exec/@@unexec to @@pkgdir or drop it. @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.2 2009/06/14 17:38:38 joerg Exp $ d34 1 d37 1 d56 1 d94 1 d100 1 a140 1 lib/asterisk/modules/app_pickupchan.so d211 1 d257 1 d267 1 d284 5 a288 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.14 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.14 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.14 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.14 d737 4 d1040 1 d1047 1 d1065 1 d1067 1 d1122 2 @ 1.3.2.1 log @Pullup ticket 2887 - requested by jnemeth asterisk16: security update --- Apply patch to update comms/asterisk16 to version 1.6.0.15: - 1.6.0.11 was never released - 1.6.0.12 fixes AST-2009-005 which is a remote DOS issue in SIP - 1.6.0.13 fixes a bug in 1.6.0.12 security fix - 1.6.0.14 has additional updates for AST-2009-001 and AST-2009-005 plus SIP Changes ----------- * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed with Microsoft OCS). This option is disabled by default. In addition, this behavior is automatic when the SDP received is in response to a T.38 re-INVITE that Asterisk initiated. In this situation , since the call will fail if Asterisk does not process the incoming SDP, Aster isk will accept the SDP even if the SDP version number is not properly incremente d, but will generate a warning in the log indicating that the SIP peer that sent the SDP should have the 'ignoresdpversion' option set. Closed Issues This is a list of all issues from the issue tracker that were closed by changes that went into this release. Category: Addons/General #15269: [patch] memory leak in asterisk some bug fixing and removing Redundant condition Category: Applications/General #15022: [patch] Language handling for numbers, dates, etc is misbehaving when utilizing sub-regional languages Category: Applications/app_chanspy #15660: ChanSpy "whisper" is broken in 1.4.26 Category: Applications/app_fax #15355: app_fax does not compile with iaxmodem 1.2.0 #15480: [patch] Not all fixes from #14849 are committed #15606: app_fax.c is not compiling under OpenBSD #15610: T.38 re-INVITE received after T.38 already negotiated fails Category: Applications/app_meetme #15493: [patch] contrib/scripts/meetme.sql doesn't contain all fields Category: Applications/app_milliwatt #15386: [patch] Milliwatt() is off by -11dbm Category: Applications/app_mixmonitor #15259: MixMonitor is not releasing the file handle on the recorded file #15699: [patch] using ast_free instead of mixmonitor_free Category: Applications/app_queue #14536: [patch] After a caller is processed by app_queue the queue_log logs the hangup as TRANSFER #14631: [patch] Ghost calls with queues and spa942 and 922 #15664: [patch] QUEUE_MEMBER_LIST() returns member names instead of interfaces Category: Applications/app_stack #15557: [patch] Gosub() dequotes once more than Macro() #15617: [patch] crash in LOCAL() if Gosub stack is allocated but empty Category: Applications/app_voicemail #14554: [patch] # for fastforward goes beyond end of message #14932: [patch] asterisk-1.6.0.9-x86_64 segfaults when leaving a voicemail internally to another extension #15331: [patch] Log message does not match conditional check #15333: [patch] add FILE_STORAGE to Voicemail Build Options #15720: opendir() return code is not checked in last_message_index() Category: Applications/app_voicemail/IMAP #14496: [patch] IMAP crash multiple callers / callers hangup at beep #14597: greetings can not be retrieved from IMAP #14950: [patch] Greetings are stored as IMAP messages even when imapgreetings=no Category: CDR/General #15751: [patch] Core dump in ast_bridge_call features.c line 2772 Category: Channels/General #15330: [patch] Using CHANNEL function from ZOMBIE channel stops Asterisk #15416: No voice on PRI calls with asterisk 1.4.25 & 26 Category: Channels/chan_dahdi #13917: [patch] fxo modules incorrectly believes channel is answered, if telco reverses line polarity at off hook. #14383: priexclusive parameter ignored if pri = pri_cpe ? #14434: [patch] Dahdi does not wait for wink on outbound calls before dialing DTMF with Signalling type = em_w #14434: [patch] Dahdi does not wait for wink on outbound calls before dialing DTMF with Signalling type = em_w #14477: pseudo channel disappears after dahdi restart #14696: reload in console overwrites priindication=outofband setting #14726: Conditional compilation of a diagnostic message needs an L modifier to %d for a 64 bit integer #15248: [patch] Multiple Groups Not working #15389: [patch] no audio with SIP call to ISDN PRI, if neither Progress or Proceeding are received. #15655: [patch] Dialplan starts execution before call is accepted Category: Channels/chan_iax2 #15361: [patch] AST-2009-001 breaks IAX2 RFC5456 compliance - Timestamps in POKE/PONG zero in 2 of 4 Bytes #15404: [patch] Unrequired Debug Message Category: Channels/chan_misdn #11974: external lines connected with message !! Got Busy in Connected State !?! #12113: [patch] asterisk crash at reload chan_misdn.so #14355: [patch] Segfault if you transfer a call into a meetme room #14692: [patch] ISDN-Transfer causes backcall attempt of attendent phone Category: Channels/chan_sip/General #11231: [patch] Many retransmits when chan_sip generates multiple outstanding requests #12434: Handle wrong at offer/answer in sdp in media description(m=) #12869: [patch] 'context' doesn't change when 'sip reload' issued when driven from realtime #13432: [patch] outboundproxy=proxy.mmmydomain.net where domain can not be resolved silently removes the sip section #13623: Asterisk segfaults when using SIP session timers #14239: [patch] 491-request pending is sent out of dialog #14464: [patch] lock during simple call processing #14575: BYE to 408 Request Timeout #14659: [patch] MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port (5060) #15213: [patch] asterisk lock in sipsock_read for several seconds and drop sip packets #15283: [patch] CLI NOTIFY always tries to use UDP, even if the peer is connected via TCP #15345: [patch] SIP deadlock in 1.4 revision 199472 #15349: Deadlock in do_monitor() of chan_sip #15362: [patch] log message output is truncated #15376: SIP option (SIP_OPT_ flag) is not handled correctly #15403: [patch] Session timer is not activated Category: Channels/chan_sip/Interoperability #13958: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv' #14465: [patch] Incorrect From: header information when CALLERPRES=PRES_PROHIB #14584: [patch] Asterisk does not stop retransmission #14725: Asterisk doesn't add Route headers in NOTIFY when the SUBSCRIBE came from a proxy #15158: [patch] Message: "Unable to handle indication 3" Revision: 200362 #15442: [patch] Asterisk cannot handle SIP 183 "Session Progress" if no SDP is contained in it Category: Channels/chan_sip/Registration #14344: [patch] Outbound proxy not used for registrations #14366: [patch] Registration expiry not compatible with some ITSP #15102: [patch] Registration Deadlock between Asterisk and Polycom Soundpoint IP 450 #15539: [patch] Register request line contains wrong address when domain and registrar host differ Category: Channels/chan_sip/T.38 #14849: [patch] SendFax function not working as expected on > 1.6.0.7 #15182: [patch] T.38 invite does not always comply with RFC 2327 Category: Channels/chan_sip/TCP-TLS #13865: [patch] SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist #14452: in "_sip_tcp_helper_thread" Buffer is filled with dirty bytes Category: Channels/chan_sip/Video #15121: [patch] Video support in SIP channel driver appears to be totally broken Category: Core/BuildSystem #15697: most cleaner alaw don't compile #15698: [patch] If enable DEBUG_FD_LEAKS - h323 can't start. #15714: [patch] Asterisk won't build with curl unless curl_config is present Category: Core/Channels #14723: ERROR[5003]: channel.c:2043 __ast_read: ast_read() called with no recorded file descriptor. Category: Core/Configuration #14509: [patch] users.conf (and other .conf files) have incorrect whitespacing Category: Core/General #14730: [patch] Fix runlevels in Debian rc files #15273: [patch] german time (20:01:00 oh clock) is announced wrong #15649: T38 Faxing failing on 1.6.1 svn #15667: LOGGER WARNING : error executing after rotate Category: Core/Internationalization #15346: [patch] TW is not an ISO Language Code Category: Core/ManagerInterface #15397: [patch] segfault in action_coreshowchannels() at manager.c Category: Core/PBX #15057: [patch] hints with 2+ devices that include ONHOLD are often set wrong #15242: [patch] log does not indicate which function is missing closing parenthesis #15303: new_find_extension arguments in wrong order Category: Documentation #15518: iax.conf, IP-based access control #15755: Description in queues.conf on call recording is slightly misleading Category: Functions/func_callerid #15476: callerid(num) is wrong when username is missing Category: Functions/func_devstate #15413: [patch] Mapping of extension state to device state is incorrect Category: Functions/func_iconv #15169: When building with uClibc, configure script mistakenly assumes iconv is always available Category: Functions/func_realtime #15517: [patch] memory leak in func_realtime Category: Functions/func_uri #15439: [patch] URIENCODE() throws a warning when passed an empty string Category: General #15420: [patch] No audio on calls from asterisk sip phones to nortel set until dtmf from sip phone #15571: [patch] 'received' typos in trunk, in 6 files #15595: [patch] fix spelling for typos, mainly in comments. #15595: [patch] fix spelling for typos, mainly in comments. Category: PBX/pbx_dundi #15322: [patch] DUNDILOOKUP() does not accept comma as argument separator Category: Resources/res_config_ldap #13725: [patch] ERROR[7387]: res_config_ldap.c:1292 update_ldap: Couldn't modify dn:cn=1001,dc=xxx,dc=xxx because Invalid syntax #15710: Typo in LDAP schema files on line 598 Category: Resources/res_features #13794: [patch] CDR for picked up parked call gives answer time < start time and no record for parking Category: Resources/res_musiconhold #15051: [patch] Moh class set in the dialplan is ignored with realtime moh ---------------------------------------------------------------------- Commits Not Associated with an Issue This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well. +------------------------------------------------------------------------+ | Revision | Author | Summary | Issues | | | | | Referenced | |----------+------------+-----------------------------------+------------| | 199142 | dvossel | Additional updates to | | | | | AST-2009-001 | | |----------+------------+-----------------------------------+------------| | | | __WORDSIZE is not available on | | | 199858 | seanbright | all platforms, so use sizeof(void | | | | | *) instead. | | |----------+------------+-----------------------------------+------------| | | | The 1.6.0 branch was missing all | | | 199975 | mmichelson | invite_branch logic. It has now | | | | | been added. | | |----------+------------+-----------------------------------+------------| | 200040 | lmadsen | Fix path for .flavor and .version | #14737 | |----------+------------+-----------------------------------+------------| | 200149 | mmichelson | Fix a crash due to a potentially | | | | | NULL p->options. | | |----------+------------+-----------------------------------+------------| | | | Fix all of the parallel build | | | 200228 | seanbright | warnings issued when running make | | | | | -j#. | | |----------+------------+-----------------------------------+------------| | | | Add INFO to our allowed methods | | | 200515 | mmichelson | so that endpoints know they may | | | | | send it to us. | | |----------+------------+-----------------------------------+------------| | 200729 | kpfleming | Document the new automatic | | | | | 'ignoresdpversion' behavior. | | |----------+------------+-----------------------------------+------------| | | | Ensure that configure-script | | | 200767 | kpfleming | testing for compiler attributes | | | | | actually works. | | |----------+------------+-----------------------------------+------------| | | | Fix problems with new compiler | | | 200986 | kpfleming | attribute checking in configure | | | | | script. | | |----------+------------+-----------------------------------+------------| | | | Improve support for media paths | | | 201093 | kpfleming | that can generate multiple frames | | | | | at once. | | |----------+------------+-----------------------------------+------------| | | | fix issue with build_contact | | | 201226 | dvossel | introduced by the "SIP trasnport | | | | | type issues" commit | | |----------+------------+-----------------------------------+------------| | | | Correct AST_LIST_APPEND_LIST | | | 201263 | kpfleming | behavior when list to be appended | | | | | is empty. | | |----------+------------+-----------------------------------+------------| | | | Change the datastore traversal in | | | 201459 | mmichelson | ast_do_masquerade to use a safe | | | | | list traversal. | | |----------+------------+-----------------------------------+------------| | 201463 | mmichelson | Fix problem with no audio due to | | | | | ignoring the SDP. | | |----------+------------+-----------------------------------+------------| | | | Fix memory corruption and leakage | #15109, | | 201612 | russell | related reloads of non files mode | #15123, | | | | MoH classes. | #15195 | |----------+------------+-----------------------------------+------------| | | | One of the changes in 1.6.1 was | | | 201786 | tilghman | to allow app_directory to use | | | | | functionality | | |----------+------------+-----------------------------------+------------| | 201830 | tilghman | If the "h" extension fails, give | | | | | it another chance in main/pbx.c. | | |----------+------------+-----------------------------------+------------| | | | Added deadlock protection to | | | 202006 | mnicholson | try_suggested_sip_codec in | | | | | chan_sip.c. | | |----------+------------+-----------------------------------+------------| | | | Standardize return values of | | | 202259 | russell | load_config() so reload() doesn't | | | | | report an error on success. | | |----------+------------+-----------------------------------+------------| | | | Fix possibility of crashiness | | | 202263 | russell | during reload in custom fields | | | | | handling. | | |----------+------------+-----------------------------------+------------| | 202416 | russell | Make Polycom subscription type | | | | | override check more explicit. | | |----------+------------+-----------------------------------+------------| | | | Fix lock usage in | | | 202471 | seanbright | cdr_sqlite3_custom to avoid | | | | | potential crashes during reload. | | |----------+------------+-----------------------------------+------------| | 202498 | russell | Report CallerID change during a | | | | | masquerade. | | |----------+------------+-----------------------------------+------------| | | | I could have sworn I committed | | | 202763 | mattf | this patch ages ago, but... bug | | | | | fix with setting NAI properly on | | | | | linksets in certain situations. | | |----------+------------+-----------------------------------+------------| | | | Ensure the default settings are | | | 202926 | file | applied for T.38 when we set it | | | | | up for a peer. | | |----------+------------+-----------------------------------+------------| | | | Use the handy UNLINK macro | | | 202968 | mmichelson | instead of hand-coding the same | | | | | thing in-line. | | |----------+------------+-----------------------------------+------------| | 203044 | rmudgett | Improved chan_dahdi.conf pritimer | | | | | error checking. | | |----------+------------+-----------------------------------+------------| | 203117 | russell | Resolve a crash related to a T.38 | | | | | reinvite race condition. | | |----------+------------+-----------------------------------+------------| | 203387 | twilson | I didn't see that Mark already | | | | | fixed the underlying issue! | | |----------+------------+-----------------------------------+------------| | 203447 | dvossel | fixes a few redundant conditions | #15269 | |----------+------------+-----------------------------------+------------| | | | Improve T.38 negotiation by | | | 203701 | file | exchanging session parameters | | | | | between application and channel. | | |----------+------------+-----------------------------------+------------| | 203711 | jpeeler | whitespace fix | | |----------+------------+-----------------------------------+------------| | | | reverse whitespace change 203711 | | | | | that was based on looking at | | | 203717 | jpeeler | sig_analog (which has about a | | | | | 1000 line indentation change that | | | | | is not worth doing here) | | |----------+------------+-----------------------------------+------------| | | | Fix ast_say_counted_noun to | | | 204476 | qwell | correctly handle Polish. Fix a | | | | | comment typo in passing. | | |----------+------------+-----------------------------------+------------| | 204652 | dvossel | removes fake dialog_unref and | | | | | dialog_ref function calls. | | |----------+------------+-----------------------------------+------------| | | | Improve handling of | | | 204949 | kpfleming | AST_CONTROL_T38 and | | | | | AST_CONTROL_T38_PARAMETERS for | | | | | non-T.38-capable channels. | | |----------+------------+-----------------------------------+------------| | 204980 | tilghman | Restore Hungarian (mistakenly | | | | | removed during merge) | | |----------+------------+-----------------------------------+------------| | | | Move OpenSSL initialization to a | | | 205139 | russell | single place, make library usage | | | | | thread-safe. | | |----------+------------+-----------------------------------+------------| | 205152 | russell | Use tabs instead of spaces for | | | | | indentation. | | |----------+------------+-----------------------------------+------------| | | | Add redirection warnings for the | | | 205200 | tilghman | invalid language codes previously | | | | | removed. | | |----------+------------+-----------------------------------+------------| | 205220 | dvossel | ast_samp2tv needs floating point | | | | | for 16khz audio | | |----------+------------+-----------------------------------+------------| | 205224 | tilghman | oops, fixing build | | |----------+------------+-----------------------------------+------------| | | | Update config.guess and | | | 205296 | qwell | config.sub from the | | | | | savannah.gnu.org git repo. | | |----------+------------+-----------------------------------+------------| | 205415 | dvossel | moving ast_devstate_to_extenstate | | | | | to pbx.c from devicestate.c | | |----------+------------+-----------------------------------+------------| | | | pthread_self returns a pthread_t | | | 205533 | mvanbaak | which is not an unsigned int on | | | | | all | | |----------+------------+-----------------------------------+------------| | 205597 | dvossel | Fixes 8khz assumptions | | |----------+------------+-----------------------------------+------------| | 205608 | dvossel | Changing ast_samp2tv to not use | | | | | floating point. | | |----------+------------+-----------------------------------+------------| | 205880 | mmichelson | Fix build. | | |----------+------------+-----------------------------------+------------| | 205940 | kpfleming | Update comments about the level | | | | | of T.38 support in Asterisk. | | |----------+------------+-----------------------------------+------------| | 206369 | rmudgett | Fix some memory leaks in | | | | | chan_misdn. | | |----------+------------+-----------------------------------+------------| | 206387 | russell | Ensure apathetic replies are sent | | | | | out on the proper socket. | | |----------+------------+-----------------------------------+------------| | | | Only print debug info in | | | 206637 | seanbright | codec_dahdi if we are asking for | | | | | it. | | |----------+------------+-----------------------------------+------------| | 206762 | rmudgett | Merged revision 206700 from | | |----------+------------+-----------------------------------+------------| | 206871 | dvossel | avoid segfault caused by user | | | | | error | | |----------+------------+-----------------------------------+------------| | 207097 | jpeeler | Update some missing allowed | | | | | options for overlapdial | | |----------+------------+-----------------------------------+------------| | 207286 | rmudgett | | | |----------+------------+-----------------------------------+------------| | 207683 | kpfleming | Ensure that user-provided CFLAGS | | | | | and LDFLAGS are honored. | | |----------+------------+-----------------------------------+------------| | 207725 | mmichelson | Document default timeout for AMI | | | | | originations. | | |----------+------------+-----------------------------------+------------| | | | Revert r207636, this approach | | | 207783 | jpeeler | could potentially block for an | | | | | unacceptable | | |----------+------------+-----------------------------------+------------| | 208316 | mmichelson | Remove inaccurate XXX comment. | | |----------+------------+-----------------------------------+------------| | | | Rework of T.38 negotiation and | | | 208468 | kpfleming | UDPTL API to address | | | | | interoperability problems | | |----------+------------+-----------------------------------+------------| | 208502 | kpfleming | Use correct formatting for T.38 | | | | | change note in UPGRADE.txt | | |----------+------------+-----------------------------------+------------| | | | Resolve a T.38 negotiation issue | | | 208549 | kpfleming | left over from the udptl-updates | | | | | merge. | | |----------+------------+-----------------------------------+------------| | 208594 | russell | Do not log an ERROR if | | | | | autoservice_stop() returns -1. | | |----------+------------+-----------------------------------+------------| | 208752 | jpeeler | Fix compiling under dev-mode with | | | | | gcc 4.4.0. | | |----------+------------+-----------------------------------+------------| | 208925 | jpeeler | Fix logic errors from 208746 | | |----------+------------+-----------------------------------+------------| | | | Restore explicit export of | | | 209057 | kpfleming | ASTCFLAGS/ASTLDFLAGS and | | | | | underscore-variants to sub-makes. | | |----------+------------+-----------------------------------+------------| | 209061 | dbrooks | Just replacing typos "recieved" | #15360 | | | | with "received". | | |----------+------------+-----------------------------------+------------| | 209259 | kpfleming | Make T.38 switchover in | | | | | ReceiveFAX synchronous. | | |----------+------------+-----------------------------------+------------| | 209280 | kpfleming | Cleanup T.38 negotiation changes. | | |----------+------------+-----------------------------------+------------| | 209325 | tilghman | Publish French extra sounds | | |----------+------------+-----------------------------------+------------| | 209394 | kpfleming | Correct error in backport of | | | | | latest app_fax fixes. | | |----------+------------+-----------------------------------+------------| | | | Fix some places where | | | 209712 | russell | ast_event_type was used instead | | | | | of ast_event_ie_type. | | |----------+------------+-----------------------------------+------------| | 209762 | kpfleming | Minor changes inspired by testing | | | | | with latest GCC. | | |----------+------------+-----------------------------------+------------| | 209896 | russell | Resolve a valgrind warning about | #15396 | | | | a read from uninitialized memory. | | |----------+------------+-----------------------------------+------------| | 211114 | russell | Resolve a deadlock involving | | | | | app_chanspy and masquerades. | | |----------+------------+-----------------------------------+------------| | 211276 | tilghman | Small oops. Clear the flags which | | | | | have been checked. | | |----------+------------+-----------------------------------+------------| | 211551 | tilghman | AST-2009-005 | | |----------+------------+-----------------------------------+------------| | 211587 | tilghman | Conversion specifiers, not format | | | | | specifiers | | |----------+------------+-----------------------------------+------------| | | | Check an actual populated | | | 212068 | file | variable when seeing if we need | | | | | to do video or not. | | |----------+------------+-----------------------------------+------------| | | | Ensure that T38FaxVersion is put | | | 212114 | kpfleming | into outgoing SDP in the proper | | | | | case. | | |----------+------------+-----------------------------------+------------| | 212432 | rmudgett | Fix uninitialized variable. | | |----------+------------+-----------------------------------+------------| | 212765 | rmudgett | Removed some deadwood and added | | | | | some doxygen comments. | | |----------+------------+-----------------------------------+------------| | 212926 | kpfleming | Convert this branch to Opsound | | | | | music-on-hold. | | |----------+------------+-----------------------------------+------------| | | | Remove some | | | 212941 | kpfleming | accidentally-committed | | | | | properties. | | |----------+------------+-----------------------------------+------------| | | | Make autoheader descriptions | | | 214361 | tilghman | render correctly in our | #14906 | | | | autoconfig.h file. | | |----------+------------+-----------------------------------+------------| | | | One more build system change, to | | | 214474 | tilghman | make the descriptions look | | | | | better, if we have better | | | | | information. | | +------------------------------------------------------------------------+ - 1.6.0.15 fixes AST-2009-006 which is a remote DOS issue in IAX2 @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.3 2009/06/14 18:31:59 joerg Exp $ d267 6 a272 9 libdata/asterisk/moh/.asterisk-moh-opsound-wav libdata/asterisk/moh/CHANGES-asterisk-moh-opsound-2.01 libdata/asterisk/moh/CREDITS-asterisk-moh-opsound-2.01 libdata/asterisk/moh/LICENSE-asterisk-moh-opsound-2.01 libdata/asterisk/moh/macroform-cold_day.wav libdata/asterisk/moh/macroform-robot_dity.wav libdata/asterisk/moh/macroform-the_simplicity.wav libdata/asterisk/moh/manolo_camp-morning_coffee.wav libdata/asterisk/moh/reno_project-system.wav d277 4 a280 4 libdata/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.15 libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.4.15 libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.4.15 libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.4.15 a930 1 libdata/asterisk/static-http/docs/node279.html @ 1.2 log @Remove @@dirrm entries from PLISTs @ text @d1 1 a1 1 @@comment $NetBSD: PLIST,v 1.1.1.1 2009/06/12 09:04:56 jnemeth Exp $ a1105 2 @@exec ${MKDIR} %D/libdata/asterisk/sounds/fr @@exec ${MKDIR} %D/libdata/asterisk/sounds/es @ 1.1 log @Initial revision @ text @d1 1 a1 1 @@comment $NetBSD$ a1105 4 @@dirrm share/examples/asterisk @@dirrm libexec/agi-bin @@dirrm libdata/asterisk/static-http/docs @@dirrm libdata/asterisk/static-http a1106 1 @@dirrm libdata/asterisk/sounds/fr a1107 19 @@dirrm libdata/asterisk/sounds/es @@dirrm libdata/asterisk/sounds/en/silence @@dirrm libdata/asterisk/sounds/en/phonetic @@dirrm libdata/asterisk/sounds/en/letters @@dirrm libdata/asterisk/sounds/en/followme @@dirrm libdata/asterisk/sounds/en/digits @@dirrm libdata/asterisk/sounds/en/dictate @@dirrm libdata/asterisk/sounds/en @@dirrm libdata/asterisk/sounds @@dirrm libdata/asterisk/phoneprov @@dirrm libdata/asterisk/moh @@dirrm libdata/asterisk/keys @@dirrm libdata/asterisk/images @@dirrm libdata/asterisk/firmware/iax @@dirrm libdata/asterisk/firmware @@dirrm libdata/asterisk @@dirrm lib/asterisk/modules @@dirrm lib/asterisk @@dirrm include/asterisk @ 1.1.1.1 log @Add Asterisk 1.6.0.10. At the moment, this version doesn't have any hardware support, so it can't replace comms/asterisk. However, apparently there is demand for this version, so wiz@@ suggested it be imported here into comms/asterisk16. The latest version is 1.6.1.1, but I won't have time to update all the patches before the freeze. I'll update to that version sometime after the freeze when I get a chance. @ text @@