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2018-02-27Revbump for packages depending on devel/libusb{,compat}khorben3-5/+6
2018-02-10Import global switch for libusb's implementation [2/2]khorben7-14/+14
This switch is meant to be used by packages requiring an implementation of the former libusb (as in devel/libusb). The original implementation can be chosen by setting LIBUSB_TYPE to "native". The alternative implementation libusb-compat (as in devel/libusb-compat) wraps libusb1 (in devel/libusb1). This implementation can be chosen by setting LIBUSB_TYPE to "compat". On NetBSD, it has the advantage of not requiring root privileges to locate and use USB devices without a kernel driver. This second part switches packages using libusb to this framework. It does not change compilation options or dependencies at this point. Compile-tested on most packages affected and available on NetBSD/amd64.
2018-01-28Bump PKGREVISION for gdbm shlib major bumpwiz2-4/+4
2018-01-24update Asterisk to 14.7.5 -- this is a bug fix and security update,jnemeth4-48/+58
it fixes AST-2017-005, AST-2017-006, AST-2017-006, AST-2017-008, AST-2017-009, AST-2017-010, AST-2017-011, AST-2017-012, AST-2017-013, and AST-2017-014. Note that several of these are related to PJSIP which pkgsrc doesn't use. ----- 14.7.5 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2. The following security vulnerabilities were resolved in these versions: * AST-2017-014: Crash in PJSIP resource when missing a contact header A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and using the PJSIP channel driver, it would cause Asterisk to crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled a user would have to first be authorized before reaching the crash point. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2017-014.pdf Thank you for your continued support of Asterisk! ----- 14.7.4 ----- The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert9, 13.18.4, 14.7.4 and 15.1.4. The release of these versions resolves the following security vulnerabilities: * AST-2017-012: Remote Crash Vulnerability in RTCP Stack If a compound RTCP packet is received containing more than one report (for example a Receiver Report and a Sender Report) the RTCP stack will incorrectly store report information outside of allocated memory potentially causing a crash. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4 The security advisories are available at: http://downloads.asterisk.org/pub/security/AST-2017-012.html http://downloads.asterisk.org/pub/security/AST-2017-012.pdf Thank you for your continued support of Asterisk! ----- 14.7.3 ----- The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert8, 13.18.3, 14.7.3 and 15.1.3. The release of these versions resolves the following security vulnerabilities: * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny If the chan_skinny (AKA SCCP protocol) channel driver is flooded with certain requests it can cause the asterisk process to use excessive amounts of virtual memory eventually causing asterisk to stop processing requests of any kind. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3 The security advisories are available at: http://downloads.asterisk.org/pub/security/AST-2017-013.pdf Thank you for your continued support of Asterisk! ----- 14.7.2 ----- The Asterisk Development Team would like to announce the release of Asterisk 14.7.2. The release of Asterisk 14.7.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more (Reported by Michael Maier) * ASTERISK-27391 - Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.2 Thank you for your continued support of Asterisk! ----- 14.7.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 14.7.0. The release of Asterisk 14.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-27278 - [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE (Reported by David J. Pryke) * ASTERISK-27255 - alembic: Add support for Microsoft SQL server (Reported by Florian Floimair) * ASTERISK-27253 - [patch] libsrtp-2.1.x support (Reported by Alexander Traud) * ASTERISK-27220 - Enable CHANNEL function to get from and to tag from SIP Headers (Reported by Andre Nazario) * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif (Reported by Andrey) * ASTERISK-27173 - Support for GMIME 3.0 (Reported by Tzafrir Cohen) * ASTERISK-27092 - [patch] app_queue: Add Priority to AMI QueueStatus (Reported by Niklas Larsson) * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) * ASTERISK-27372 - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. (Reported by dtryba) * ASTERISK-27270 - cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured (Reported by Tzafrir Cohen) * ASTERISK-25266 - Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate (Reported by Allen Ford) * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing unavailable endpoints (Reported by Richard Mudgett) * ASTERISK-27305 - res_ari: Memory leaks in ARI when using Content-Type: application/json (Reported by David Hajek) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS (Reported by Alexander Traud) * ASTERISK-27317 - vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell) * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from ast_strings_match (Reported by Corey Farrell) * ASTERISK-27284 - Status of RFC 3323 and PJSIP (Reported by dtryba) * ASTERISK-27296 - [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved (Reported by Benoît Dereck-Tricot) * ASTERISK-27216 - app_queue: does its check-makeannouncement-logic twice each head-caller-loop (Reported by Stefan Engström) * ASTERISK-27298 - Problem with expires on pjsip / outbound-publish (Reported by Cyrille Demaret) * ASTERISK-27295 - Contact is improperly translated after d178f497 (Reported by Sean Bright) * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) (Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-16898 - SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 (Reported by Marcello Ceschia) * ASTERISK-27274 - RTCP needs better packet validation to resist port scans. (Reported by Richard Mudgett) * ASTERISK-27252 - RTP: One way audio with direct media and strictrtp=yes. (Reported by Richard Mudgett) * ASTERISK-25524 - module reload res_calendar.so does not reload everything in calendar.conf (Reported by Jesper) * ASTERISK-24588 - res_calendar does not process CalDAV from Owncloud [fix included] (Reported by Stefan Gofferje) * ASTERISK-25523 - res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. (Reported by Jesper) * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and wireshark disagree (Reported by Tzafrir Cohen) * ASTERISK-27248 - [patch]external_media_address and external_signaling_address don't always honor localnet (Reported by Walter Doekes) * ASTERISK-27217 - chan_sip: Asterisk crashing when subscription doesn't get set (Reported by Bryan Walters) * ASTERISK-24066 - res_smdi: convert to astobj2 (Reported by Corey Farrell) * ASTERISK-17540 - SDP origin attribute modified when issuing re-INVITE because of directmedia=yes (Reported by saghul) * ASTERISK-27254 - alembic: prune_on_boot fix erroneous (Reported by Florian Floimair) * ASTERISK-27232 - When in queue on g722 with interruptions, music on hold can get stuck and no longer play (Reported by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27165 - CDR: CDR(start,u) function won't work in cdr_custom config (Reported by Jacek Konieczny) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen) * ASTERISK-27241 - libc segfault upon entry into app_directory (Reported by David Moore) * ASTERISK-27152 - Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash (Reported by Ross Beer) * ASTERISK-27103 - core: ast_safe_system command injection possible. (Reported by Corey Farrell) * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) * ASTERISK-26994 - Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name (Reported by James Terhune) * ASTERISK-20858 - app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) * ASTERISK-16777 - several filename bugs in Record() application (Reported by klaus3000) * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) * ASTERISK-27168 - alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-19103 - When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. (Reported by Jim Van Meggelen) * ASTERISK-21241 - When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored (Reported by Eelco Brolman) * ASTERISK-27204 - [patch] app_queue: Wrong queue stat calculation (Reported by sungtae kim) * ASTERISK-27207 - XMPP OAuth not working due to inverted logic (Reported by Michael Kuron) * ASTERISK-27174 - res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical (Reported by Mark Thompson) * ASTERISK-27202 - If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed (Reported by Seán C. McCord) * ASTERISK-27147 - Either asterisk or pjproject isn't re-using tcp connections (again) (Reported by George Joseph) * ASTERISK-27193 - IPv6 receive address in message doesn't include brackets (Reported by Scott Griepentrog) * ASTERISK-26745 - Asymmetric codecs when asymmetric_rtp_codec=no (Reported by Jesse Ross) * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used (Reported by Torrey Searle) * ASTERISK-27110 - RTP session is not fully destroyed on channel hangup (Reported by Matt Jordan) * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira Emus) * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces around the status element in XML (Reported by Abraham Liebsch) * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with devmode enabled. (Reported by Corey Farrell) * ASTERISK-27130 - Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly (Reported by Sergej Kasumovic) * ASTERISK-25810 - say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. (Reported by Nicolas Riendeau) * ASTERISK-27142 - sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 (Reported by Corey Farrell) * ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use (Reported by Joshua Colp) * ASTERISK-27123 - confbridge: Name recordings are left on filesystem (Reported by Sergej Kasumovic) * ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep adding up (Reported by Sergej Kasumovic) * ASTERISK-26807 - sounds: New 3-D Binaural audio features require new sound prompts (Reported by Rusty Newton) * ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts differ in content from the English files (Reported by Benoit Duverger) * ASTERISK-26274 - Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) (Reported by Rusty Newton) * ASTERISK-27128 - [patch]res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media (Reported by Dan Jenkins) * ASTERISK-27124 - app_playback.c:say_date_generic use timezonename parameter (Reported by Holger Hans Peter Freyther) * ASTERISK-27127 - configs: Erroneous load directive in sample configuration results in "Error loading module 'res_pjsip_multihomed.so'" (Reported by HZMI8gkCvPpom0tM) * ASTERISK-27105 - [patch]core: when setting 'maxfiles' in asterisk.conf, a message is printed, even in rasterisk -x (Reported by Tzafrir Cohen) * ASTERISK-27036 - res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@' (Reported by Maxim Vasilev) * ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session in use (Reported by Jatin Jain) * ASTERISK-27093 - ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting (Reported by James Terhune) New Features made in this release: ----------------------------------- * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action (Reported by Thomas Sevestre) * ASTERISK-27117 - core: Add support for timelen parsing to ast_parse_arg and ACO. (Reported by Corey Farrell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.0 Thank you for your continued support of Asterisk! ----- 14.6.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka) * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) * ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell) * ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) * ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson) * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) * ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) * ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex) * ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) * ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier) * ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) * ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) * ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot) * ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp) * ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) * ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen) * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton) * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou) * ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield) * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) * ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters) * ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard) * ASTERISK-26715 - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert) * ASTERISK-26975 - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz) * ASTERISK-27012 - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) * ASTERISK-26789 - Audit manipulation of channel flags without locks (Reported by Joshua Colp) * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder) Improvements made in this release: ----------------------------------- * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari) * ASTERISK-27043 - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi) * ASTERISK-27042 - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi) * ASTERISK-26419 - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton) * ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex) * ASTERISK-26124 - res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0 Thank you for your continued support of Asterisk!
2018-01-23update to Asterisk 13.19.0 -- this contains both security fixesjnemeth5-80/+86
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007, AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12, AST-2017-13, and AST-2017-14 (note that a number of these only pertain to PJSIP which isn't used in pkgsrc) ----- 13.19.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. (Reported by Richard Mudgett) * ASTERISK-27413 - Add cache_media_frames debugging option. (Reported by Richard Mudgett) * ASTERISK-27206 - res_pjsip: No mechanism exists to limit endpoint identification to IP only (Reported by Ben Merrills) Bugs fixed in this release: ----------------------------------- * ASTERISK-27531 - Compiler optimizations can break module load sequence. (Reported by abelbeck) * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on read() (Reported by Abhay Gupta) * ASTERISK-25079 - AMI bridge of channels results in MOH not destroyed and robotic audio on one channel (Reported by Zane Conkle) * ASTERISK-27490 - chan_console: 'set active' fails to work (Reported by Tzafrir Cohen) * ASTERISK-24756 - ConfBridge sound_muted does not work from CLI or AMI (Reported by Thomas Frederiksen) * ASTERISK-25649 - Transfer application does not work with Local channels - documentation misleading (Reported by Ivan Ullmann) * ASTERISK-25869 - chan_sip: "rejected because extension not found" should be logged as a security event (Reported by Brian J. Murrell) * ASTERISK-27440 - Strictrtp has issues to qualify video rtp streams (Reported by Wim De Vlaminck) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-19657 - Coverity Report: Fix issues for error type CHAR_IO (Reported by Matt Jordan) * ASTERISK-27175 - iax.conf demo peer is invalid (Reported by Tzafrir Cohen) * ASTERISK-27430 - README refers to security documents that do not exist. (Reported by Corey Farrell) * ASTERISK-20281 - "core set verbose" behaves strangely, can't alias it, cli.conf example broken (Reported by Tim Ringenbach at Asteria Solutions Group) * ASTERISK-27382 - crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) * ASTERISK-27408 - Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett) * ASTERISK-27453 - RTP: Blind transfer direct media scenario results in one way audio. (Reported by Richard Mudgett) * ASTERISK-20643 - SIP ICE support - remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message (Reported by Roy) * ASTERISK-26980 - pjsip: Clean up WebRTC disables (Reported by abelbeck) * ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests (Reported by George Joseph) * ASTERISK-27454 - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported by Joshua Colp) * ASTERISK-23735 - Transcoding makes bad choice in high-rate translations (Reported by Richard Kenner) * ASTERISK-27445 - ARI: Updating a bridge gives wrong error message. (Reported by Frank Durden) * ASTERISK-24662 - [patch] column and row headers for Signed Linear format variants in output of 'core show translation' are ambiguous (Reported by Rusty Newton) * ASTERISK-27353 - H323 audio starts with a delay of 2 seconds. (Reported by Marco Giordani) * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate media (Reported by Kevin Harwell) * ASTERISK-27437 - [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. (Reported by Alexander Traud) * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around IPv6 addresses. (Reported by Alexander Traud) * ASTERISK-27435 - [patch] configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) * ASTERISK-27431 - Asterisk fails to build when openssl headers are not installed. (Reported by Corey Farrell) * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra (Reported by Ivan Larionov) * ASTERISK-27421 - RTP source learning not working with devices that have some clock issues (Reported by nappsoft) * ASTERISK-27361 - Attended transfer crashes in Asterisk 13.17.2 (Reported by Alessandro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by Max Norba) * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) (Reported by Vasilii Rogin) * ASTERISK-27415 - asterisk.conf: Setting astctl without setting astrundir is ineffective. (Reported by Corey Farrell) * ASTERISK-27411 - pjsip: TCP connections may not be destroyed (Reported by Joshua Colp) * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488 responses. (Reported by Corey Farrell) * ASTERISK-27337 - chan_sip: Security vulnerability with client code header (revisited) (Reported by Richard Mudgett) * ASTERISK-27319 - (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines (Reported by Kim youngsung) * ASTERISK-27391 - Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) * ASTERISK-27393 - res_pjsip: Crash occurs when an empty contact read from astdb or database (Reported by Aaron An) * ASTERISK-27290 - res_pjsip: PIDF contact field has malformed/invalid XML (Reported by basildane) * ASTERISK-27032 - res_pjsip: TLS options do not handle empty values (Reported by seanchann.zhou) * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-27378 - Modules: Fix issues with CLI completion. (Reported by Corey Farrell) * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more (Reported by Michael Maier) * ASTERISK-27390 - Audit menuselect module dependencies (Reported by Corey Farrell) * ASTERISK-27389 - Optional API modules should not allow unload. (Reported by Corey Farrell) * ASTERISK-27369 - Bridge() dialplan application fails without setting BRIDGERESULT channel variable (Reported by James Terhune) * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' (Reported by Anthony Messina) * ASTERISK-27194 - jitterbuffer: Does not handle case where translator returns null frame. (Reported by Joshua Elson) * ASTERISK-26639 - core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. (Reported by Mr Dini) * ASTERISK-27372 - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. (Reported by Jonathan Cloots) * ASTERISK-25960 - The config_hook unit test causes Asterisk to crash if run a second time (Reported by George Joseph) * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes (Reported by Martin Cisárik) * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last but first on SDP media level. (Reported by Alexander Traud) * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 (Reported by Tzafrir Cohen) * ASTERISK-23462 - Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't (Reported by Rusty Newton) * ASTERISK-27328 - Missing openssl dependencies in res_rtp_asterisk and tcptls (Reported by Tzafrir Cohen) * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address. (Reported by Alexander Traud) * ASTERISK-27343 - Fails to build in FreeBSD due to sys/sysmacros.h not existing there (Reported by Guido Falsi) * ASTERISK-27340 - backtrace.c: Crash due to double-free. (Reported by Corey Farrell) * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when stopping. (Reported by Alexander Traud) * ASTERISK-27333 - sip_to_pjsip not correctly handling disallow=all directive (Reported by Torrey Searle) Improvements made in this release: ----------------------------------- * ASTERISK-24297 - cdr.c: Minor code optimizations. (Reported by Richard Mudgett) * ASTERISK-27449 - [PATCH] When failing to acquire target during attended transfer, display wanted extension (Reported by Niklas Larsson) * ASTERISK-27456 - app_voicemail: Add new object for VoicemailUserEntry (Reported by sungtae kim) * ASTERISK-27380 - ast_coredumper: allow pointing out the asterisk binary explicitly (Reported by Tzafrir Cohen) * ASTERISK-23556 - Compilation warning for invert.c (array subscript is above array bounds) (Reported by Marcello Ceschia) * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard Mudgett) * ASTERISK-27335 - CDR performance needs improvement. (Reported by Richard Mudgett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0 Thank you for your continued support of Asterisk! ----- 13.18.5 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2. The following security vulnerabilities were resolved in these versions: * AST-2017-014: Crash in PJSIP resource when missing a contact header A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and using the PJSIP channel driver, it would cause Asterisk to crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled a user would have to first be authorized before reaching the crash point. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2017-014.pdf Thank you for your continued support of Asterisk! ----- 13.18.4 ----- The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert9, 13.18.4, 14.7.4 and 15.1.4. The release of these versions resolves the following security vulnerabilities: * AST-2017-012: Remote Crash Vulnerability in RTCP Stack If a compound RTCP packet is received containing more than one report (for example a Receiver Report and a Sender Report) the RTCP stack will incorrectly store report information outside of allocated memory potentially causing a crash. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4 The security advisories are available at: http://downloads.asterisk.org/pub/security/AST-2017-012.html http://downloads.asterisk.org/pub/security/AST-2017-012.pdf Thank you for your continued support of Asterisk! ----- 13.18.3 ----- The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert8, 13.18.3, 14.7.3 and 15.1.3. The release of these versions resolves the following security vulnerabilities: * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny If the chan_skinny (AKA SCCP protocol) channel driver is flooded with certain requests it can cause the asterisk process to use excessive amounts of virtual memory eventually causing asterisk to stop processing requests of any kind. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3 The security advisories are available at: http://downloads.asterisk.org/pub/security/AST-2017-013.pdf Thank you for your continued support of Asterisk! ----- 13.18.2 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.18.2. The release of Asterisk 13.18.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more (Reported by Michael Maier) * ASTERISK-27391 - Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2 Thank you for your continued support of Asterisk! ----- 13.18.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.18.0. The release of Asterisk 13.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-27278 - [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE (Reported by David J. Pryke) * ASTERISK-27255 - alembic: Add support for Microsoft SQL server (Reported by Florian Floimair) * ASTERISK-27253 - [patch] libsrtp-2.1.x support (Reported by Alexander Traud) * ASTERISK-27220 - Enable CHANNEL function to get from and to tag from SIP Headers (Reported by Andre Nazario) * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif (Reported by Andrey) * ASTERISK-27173 - Support for GMIME 3.0 (Reported by Tzafrir Cohen) * ASTERISK-27092 - [patch] app_queue: Add Priority to AMI QueueStatus (Reported by Niklas Larsson) * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) * ASTERISK-27372 - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. (Reported by dtryba) * ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs (Reported by dtryba) * ASTERISK-27270 - cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured (Reported by Tzafrir Cohen) * ASTERISK-27301 - [patch] app_queue: Music On Hold for real-time queues is not reset to default (Reported by Nathan Bruning) * ASTERISK-25266 - Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate (Reported by Allen Ford) * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing unavailable endpoints (Reported by Richard Mudgett) * ASTERISK-27305 - res_ari: Memory leaks in ARI when using Content-Type: application/json (Reported by David Hajek) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS (Reported by Alexander Traud) * ASTERISK-27317 - vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell) * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from ast_strings_match (Reported by Corey Farrell) * ASTERISK-27296 - [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved (Reported by Benoît Dereck-Tricot) * ASTERISK-27284 - Status of RFC 3323 and PJSIP (Reported by dtryba) * ASTERISK-27216 - app_queue: does its check-makeannouncement-logic twice each head-caller-loop (Reported by Stefan Engström) * ASTERISK-27295 - Contact is improperly translated after d178f497 (Reported by Sean Bright) * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) (Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-16898 - SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 (Reported by Marcello Ceschia) * ASTERISK-27274 - RTCP needs better packet validation to resist port scans. (Reported by Richard Mudgett) * ASTERISK-27252 - RTP: One way audio with direct media and strictrtp=yes. (Reported by Richard Mudgett) * ASTERISK-25524 - module reload res_calendar.so does not reload everything in calendar.conf (Reported by Jesper) * ASTERISK-24588 - res_calendar does not process CalDAV from Owncloud [fix included] (Reported by Stefan Gofferje) * ASTERISK-25523 - res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. (Reported by Jesper) * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and wireshark disagree (Reported by Tzafrir Cohen) * ASTERISK-27248 - [patch]external_media_address and external_signaling_address don't always honor localnet (Reported by Walter Doekes) * ASTERISK-27165 - CDR: CDR(start,u) function won't work in cdr_custom config (Reported by Jacek Konieczny) * ASTERISK-27217 - chan_sip: Asterisk crashing when subscription doesn't get set (Reported by Bryan Walters) * ASTERISK-24066 - res_smdi: convert to astobj2 (Reported by Corey Farrell) * ASTERISK-17540 - SDP origin attribute modified when issuing re-INVITE because of directmedia=yes (Reported by saghul) * ASTERISK-27254 - alembic: prune_on_boot fix erroneous (Reported by Florian Floimair) * ASTERISK-27232 - When in queue on g722 with interruptions, music on hold can get stuck and no longer play (Reported by Jens T.) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) * ASTERISK-27177 - ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen) * ASTERISK-27241 - libc segfault upon entry into app_directory (Reported by David Moore) * ASTERISK-27152 - Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash (Reported by Ross Beer) * ASTERISK-27103 - core: ast_safe_system command injection possible. (Reported by Corey Farrell) * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) * ASTERISK-26994 - Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name (Reported by James Terhune) * ASTERISK-20858 - app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) * ASTERISK-16777 - several filename bugs in Record() application (Reported by klaus3000) * ASTERISK-27168 - alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-23608 - ControlPlayback fails to play files with names containing certain non-alpha characters (Reported by Jonathan White) * ASTERISK-19103 - When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. (Reported by Jim Van Meggelen) * ASTERISK-21241 - When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored (Reported by Eelco Brolman) * ASTERISK-27204 - [patch] app_queue: Wrong queue stat calculation (Reported by sungtae kim) * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) * ASTERISK-27207 - XMPP OAuth not working due to inverted logic (Reported by Michael Kuron) * ASTERISK-27174 - res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical (Reported by Mark Thompson) * ASTERISK-27202 - If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed (Reported by Seán C. McCord) * ASTERISK-27147 - Either asterisk or pjproject isn't re-using tcp connections (again) (Reported by George Joseph) * ASTERISK-27193 - IPv6 receive address in message doesn't include brackets (Reported by Scott Griepentrog) * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used (Reported by Torrey Searle) * ASTERISK-27110 - RTP session is not fully destroyed on channel hangup (Reported by Matt Jordan) * ASTERISK-26745 - Asymmetric codecs when asymmetric_rtp_codec=no (Reported by Jesse Ross) * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira Emus) * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces around the status element in XML (Reported by Abraham Liebsch) * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with devmode enabled. (Reported by Corey Farrell) * ASTERISK-27130 - Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly (Reported by Sergej Kasumovic) * ASTERISK-25810 - say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. (Reported by Nicolas Riendeau) * ASTERISK-27142 - sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 (Reported by Corey Farrell) * ASTERISK-27124 - app_playback.c:say_date_generic use timezonename parameter (Reported by Holger Hans Peter Freyther) * ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use (Reported by Joshua Colp) * ASTERISK-27128 - [patch]res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media (Reported by Dan Jenkins) * ASTERISK-27123 - confbridge: Name recordings are left on filesystem (Reported by Sergej Kasumovic) * ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep adding up (Reported by Sergej Kasumovic) * ASTERISK-26807 - sounds: New 3-D Binaural audio features require new sound prompts (Reported by Rusty Newton) * ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts differ in content from the English files (Reported by Benoit Duverger) * ASTERISK-26274 - Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) (Reported by Rusty Newton) * ASTERISK-27127 - configs: Erroneous load directive in sample configuration results in "Error loading module 'res_pjsip_multihomed.so'" (Reported by HZMI8gkCvPpom0tM) * ASTERISK-27105 - [patch]core: when setting 'maxfiles' in asterisk.conf, a message is printed, even in rasterisk -x (Reported by Tzafrir Cohen) * ASTERISK-27036 - res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@' (Reported by Maxim Vasilev) * ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session in use (Reported by Jatin Jain) * ASTERISK-27093 - ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting (Reported by James Terhune) New Features made in this release: ----------------------------------- * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action (Reported by Thomas Sevestre) * ASTERISK-27117 - core: Add support for timelen parsing to ast_parse_arg and ACO. (Reported by Corey Farrell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0 Thank you for your continued support of Asterisk! ----- 13.17.0 ---- The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka) * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) * ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell) * ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) * ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson) * ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) * ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex) * ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) * ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier) * ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) * ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) * ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot) * ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp) * ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) * ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen) * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton) * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou) * ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield) * ASTERISK-25101 - DTLS configuration can not be specified in the general section - documentation (Reported by Ben Langfeld) * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) * ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters) * ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard) * ASTERISK-26715 - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert) * ASTERISK-26975 - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz) * ASTERISK-27012 - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) * ASTERISK-26789 - Audit manipulation of channel flags without locks (Reported by Joshua Colp) * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder) Improvements made in this release: ----------------------------------- * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari) * ASTERISK-27043 - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi) * ASTERISK-27042 - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi) * ASTERISK-26419 - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton) * ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex) * ASTERISK-26124 - res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0 Thank you for your continued support of Asterisk!
2018-01-07Fix indentation in buildlink3.mk files.rillig7-15/+15
The actual fix as been done by "pkglint -F */*/buildlink3.mk", and was reviewed manually. There are some .include lines that still are indented with zero spaces although the surrounding .if is indented. This is existing practice.
2018-01-01Sort PLIST files.rillig4-15/+15
Unsorted entries in PLIST files have generated a pkglint warning for at least 12 years. Somewhat more recently, pkglint has learned to sort PLIST files automatically. Since pkglint 5.4.23, the sorting is only done in obvious, simple cases. These have been applied by running: pkglint -Cnone,PLIST -Wnone,plist-sort -r -F
2018-01-01Revbump after boost updateadam5-9/+10
2017-12-24fidogate: fix HOMEPAGEwiz1-3/+2
2017-12-09gammu: Do not set a LIB_SUFFIX in CMakeLists.txtleot1-1/+7
On some platforms (strictly speaking the ones that have libm somewhere in a path with /lib64/) LIB_SUFFIX is set to `64' leading to install phase/PLIST errors due libraries and pkg-config `.pc' files are tried to be installed in `lib64/'. Add a `cmakelists' SUBST_CLASS to avoid that. This should fix problems noticed on Joyent CentOS 7.2/x86_64 bulk builds.
2017-11-30Revbump after textproc/icu updateadam13-26/+26
2017-11-23recursive bump for libxkbcommon removal from at-spi2-corewiz4-8/+8
2017-11-08py-gammu: Update comms/py-gammu to 2.10leot2-7/+7
Changes: 2.10 ==== * Testsuite compatibility with Gammu 1.38.5.
2017-10-22gammu: Update comms/gammu to 1.38.5leot2-8/+7
Changes: 20171018 - 1.38.5 [+] * Added SMS_1_REFERENCE to SMSD run on receive environment [-] * Improved logging of run scripts in SMSD [-] * Improved support for Huawei E1780 and E1552. [-] * Allow 0 for setuid/setgid in SMSD. [+] * Added RunOnIncomingCall to SMSD. [-] * Fixed SQL error when retry of multipart message [*] * Added status code column [-] * Fixed some SQL queries for Access and Oracle databases. [-] * Add option to prefer GSM charset for USSD. [-] * Sanitize international numbers stored in the database to always start with +. [-] * Improved support for Telit devices. [+] * Added USSD support to SMSD. [-] * Fixed call hangup in SMSD with some modems. [-] * Fixed decoding USSD response with some modems.
2017-09-26*: remove qt3 and the packages using it, including KDE3wiz6-49/+2
Announced in https://mail-index.netbsd.org/pkgsrc-users/2017/09/10/msg025556.html
2017-09-18revbump for requiring ICU 59.xmaya14-28/+28
2017-09-17p5-Asterisk: update to 1.08.wiz2-11/+11
1.08 Package asterisk::perl to resolve pause index upload. 1.07 Replace Config with Conf namespace to resolve conflict with Asterisk::config distro 1.06 New upload with original asterisk-perl distro name More test script updates to increase code coverage. 1.05 Fix Asterisk::Manager undefined response RT#115789 ( Thanks Chris Hemmerly) Fix MakeFile.PL and Asterisk::Perl for Pause Indexing (Thanks Jim Keenan) minor updates on the test scripts 1.04 Asterisk-Perl distribution now on Github. Added simple test scripts Travis and CoverAll integration with new Github repository Asterisk-Perl distribution now ready for Pull Request Challenge (http://cpan-prc.org/)
2017-09-16Reset maintainerwiz6-12/+12
2017-09-11Lose the debug options, after they've served their purpose.hauke1-6/+2
2017-09-11Heed a pkglint warning wrt. VARBASE.hauke1-2/+2
2017-09-11Built with gcc 5.4 on netbsd-8, conserver terminates because of ahauke3-3/+25
buffer overflow in StrTime(), when it tries to stuff a 25 char string into a 25 byte buffer.
2017-09-06Comment out dead sites.wiz1-2/+2
2017-09-06Follow some redirects.wiz1-3/+3
2017-09-04Follow some redirects.wiz1-3/+3
2017-09-04Comment out dead sites.wiz1-2/+2
2017-09-03Follow some redirects.wiz1-3/+3
2017-09-03Comment out dead MASTER_SITES/HOMEPAGEs.wiz5-12/+12
2017-08-24Revbump for boost updateadam5-7/+10
2017-08-19comms/modemd: Install manpages into ${PKGMANDIR}.jlam2-5/+5
Set MANDIR in Makefile.inc to point to ${PKGMANDIR} so that the BSD makefiles that include Makefile.inc will install manpages into the correct location.
2017-08-16Comment out dead sites.wiz11-30/+30
2017-08-01Comment out some dead HOMEPAGEs.wiz2-5/+5
2017-08-01Follow some http -> https redirects.wiz4-9/+9
2017-08-01Added ALTERNATIVESadam1-0/+1
2017-07-31Version 3.4:adam3-21/+45
Improvements: * miniterm: suspend function (temporarily release port, Ctrl-T s) * context manager automatically opens port on __enter__ * list_ports: add interface number to location string * protocol_socket: Retry if BlockingIOError occurs in reset_input_buffer. Bugfixes: * list_ports: option to include symlinked devices * list_ports: workaround for special characters in port names Bugfixes (posix): * allow calling cancel functions w/o error if port is closed * protocol_socket: sync error handling with posix version * posix: ignore more blocking errors and EINTR, timeout only applies to blocking I/O * fix: port_publisher typo
2017-07-30Use https for www.gnome.org HOMEPAGEs.wiz1-2/+2
2017-07-30Switch github HOMEPAGEs to https.wiz1-2/+2
2017-07-28Update comms/py-gammu to 2.9.leot2-7/+7
Changes: 2.9 === * Fixed compilation under Windows. 2.8 === * Make parameters to CancelCall and AnswerCall optional. * Added support for UTF-16 Unicode chars (emojis).
2017-07-28Update comms/gammu to 1.38.4leot2-7/+7
Changes: 20170618 - 1.38.4 [-] * Improved support for Huawei E3531 and E1756. [-] * Fixed several issues with using library on Windows. 20170523 - 1.38.3 [-] * Improved support for ZTE MF626. [-] * Fixed USSD handling with longer codes. [-] * Increased default value for StatusFrequency. [-] * Improved SMSD response on signals. [-] * Improved SMSD throughput on big queue. [-] * Improved SMSD compatibility with Microsoft SQL Server.
2017-07-20Renamed comms/py-python-termstyle to comms/py-termstyleadam6-14/+14
2017-07-200.3.9adam3-8/+9
* Revert fix for issue 103 which causes problems for dependent applications 0.3.8 * Fix issue 121: "invalid escape sequence" deprecation fixes on Python 3.6+ * Fix issue 110: fix "set console title" when working with unicode strings * Fix issue 103: enable color when using "input" function on Python 3.5+ * Fix issue 95: enable color when stderr is a tty but stdout is not
2017-06-21Update to Asterisk 14.5.0: this is mostly a bug fix releases withjnemeth8-107/+88
patches for a number of security issues, several of which do not apply to this package because they relate to PJSIP: AST-2016-009, AST-2016-010, AST-2017-001, AST-2017-002, AST-2017-003, and AST-2017-004. ----- 14.5.0 The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. The release of Asterisk 14.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) * ASTERISK-26143 - res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-26983 - Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) * ASTERISK-26173 - func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) * ASTERISK-25506 - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) * ASTERISK-24529 - Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) * ASTERISK-26860 - Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported by Jeremy Kister) * ASTERISK-20984 - Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) * ASTERISK-26903 - Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) * ASTERISK-26928 - pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) * ASTERISK-26927 - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) * ASTERISK-26905 - pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) * ASTERISK-26897 - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) * ASTERISK-25974 - Unused realtime MOH classes not purged on 'moh reload' (Reported by Sébastien Couture) * ASTERISK-26916 - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-26915 - chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) * ASTERISK-26705 - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) * ASTERISK-25490 - [patch]SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) * ASTERISK-26086 - res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) * ASTERISK-23996 - No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) * ASTERISK-24712 - xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) * ASTERISK-26814 - pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) * ASTERISK-21855 - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available * ASTERISK-25622 - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) * ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0 Thank you for your continued support of Asterisk! ----- 14.4.0 The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. The release of Asterisk 14.4.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: *New Features made in this release:* ----------------------------------- - [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>] - func_channel: Add ability to get the callid so dialplan has access to it. (Reported by Richard Mudgett) - [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>] - res_pjsip: Add endpoint identification scheme based on a configured SIP header/value (Reported by Matt Jordan) - [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>] - [patch] Allow "Comedian Mail" branding to be removed (Reported by John Covert) *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>] - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) - [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>] - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) - [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>] - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) - [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>] - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) - [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>] - res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets (Reported by Max Norba) - [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>] - res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. (Reported by Vinod Dharashive) - [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>] - res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) - [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>] - Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) - [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>] - app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) - [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>] - res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome (Reported by Dan Jenkins) - [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>] - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) - [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>] - autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) - [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/ASTERISK-26869>] - res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension (Reported by Torrey Searle) - [ASTERISK-26668 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>] - core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) - [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>] - chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) - [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/ASTERISK-26872>] - Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) - [ASTERISK-26717 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>] - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) - [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>] - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) - [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>] - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) - [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>] - chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) - [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>] - chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis - Prescom) - [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>] - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) - [ASTERISK-26353 <https://issues.asterisk.org/jira/browse/ASTERISK-26353>] - res_musiconhold: musiconhold seems to think that the general section is a class and issues warning (Reported by Jonathan Harris) - [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>] - res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) - [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>] - app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) - [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>] - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by Jørgen H) - [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) - [ASTERISK-26774 <https://issues.asterisk.org/jira/browse/ASTERISK-26774>] - core: Playback URL fails after some time (Reported by Igor Gamayunov) - [ASTERISK-26825 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>] - pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) - [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>] - PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) - [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>] - res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by Jørgen H) - [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>] - res_pjsip_outbound_registration doesn't know about network change events (Reported by George Joseph) - [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>] - bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) - [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>] - res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) - [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>] - [patch] Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) - [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>] - Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) - [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>] - PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) - [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>] - [patch] DUNDi weight parameter not processed correctly (Reported by Peter Racz) - [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>] - res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) - [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/ASTERISK-26738>] - Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c (Reported by Michael Maier) - [ASTERISK-25893 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>] - Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) - [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>] - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) - [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>] - [patch] Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) - [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>] - [patch] Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) - [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>] - res_config_sqlite3 uses incorrect query - unnecessary escape (Reported by Stepan) - [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>] - SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) - [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>] - http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) - [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>] - Phone default have not ringing on ARM (Reported by Igor Goncharovsky) - [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>] - pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) - [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>] - res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) - [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>] - Asterisk fails building with OpenSSL 1.1.0 (Reported by Tzafrir Cohen) - [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>] - VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) - [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>] - [patch] 'Silence' is truncated in Record() (Reported by var) - [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>] - chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) - [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>] - core: Protect flags during ast_waitfor (Reported by Joshua Colp) - [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>] - pbx: AMI Originate ignore "failed" extension on call failure (Reported by Nasir Iqbal) - [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>] - configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) - [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>] - Crash in srv.c on startup with pjsip (Reported by nappsoft) - [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>] - res_stasis_device_state: Duplicate subscriptions when multiple received at same time (Reported by Joshua Colp) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>] - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) - [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>] - chan_sip: Add rtcp-mux support (Reported by Sean Bright) *Thank you for your continued support of Asterisk!* ----- 14.3.0 The Asterisk Development Team has announced the release of Asterisk 14.3.0. The release of Asterisk 14.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * ASTERISK-26767 - ARI channelvars cause memory leak (Reported by Sébastien Duthil) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref count trap tripped. (Reported by Richard Mudgett) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26647 - Support older DNS style for OpenBSD (Reported by snuffy) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0 Thank you for your continued support of Asterisk!
2017-06-07Fix build with Perl 5.26.0ryoon4-2/+28
2017-06-05Recursive revbump from lang/perl5 5.26.0ryoon8-14/+16
2017-06-04Update to Asterisk 13.16.0: this is mostly a bugfix release.jnemeth6-54/+55
The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. The release of Asterisk 13.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) * ASTERISK-26143 - res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-26983 - Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) * ASTERISK-26173 - func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) * ASTERISK-25506 - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) * ASTERISK-24529 - Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) * ASTERISK-26860 - Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported by Jeremy Kister) * ASTERISK-20984 - Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) * ASTERISK-26903 - Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) * ASTERISK-26928 - pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) * ASTERISK-26927 - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) * ASTERISK-26905 - pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) * ASTERISK-26897 - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) * ASTERISK-25974 - Unused realtime MOH classes not purged on 'moh reload' (Reported by Sébastien Couture) * ASTERISK-26916 - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-26915 - chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) * ASTERISK-26705 - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) * ASTERISK-25490 - [patch]SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) * ASTERISK-24712 - xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) * ASTERISK-26086 - res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) * ASTERISK-23996 - No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) * ASTERISK-26814 - pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) * ASTERISK-21855 - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available (Reported by Jeremy Kister) * ASTERISK-25622 - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0 Thank you for your continued support of Asterisk!
2017-05-29Add fixes for AST-2017-002, AST-2017-003, and AST-2017-004. Notejnemeth2-11/+11
that the first two don't affect pkgsrc as we are using chan_sip not PJSIP. The last only affects users of SCCP, which is Cisco's proprietary protocol. ----- AST-2017-002 A remote crash can be triggered by sending a SIP packet to Asterisk with a specially crafted CSeq header and a Via header with no branch parameter. The issue is that the PJSIP RFC 2543 transaction key generation algorithm does not allocate a large enough buffer. By overrunning the buffer, the memory allocation table becomes corrupted, leading to an eventual crash. This issue is in PJSIP, and so the issue can be fixed without performing an upgrade of Asterisk at all. However, we are releasing a new version of Asterisk with the bundled PJProject updated to include the fix. If you are running Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-003 The multi-part body parser in PJSIP contains a logical error that can make certain multi-part body parts attempt to read memory from outside the allowed boundaries. A specially-crafted packet can trigger these invalid reads and potentially induce a crash. The issue is within the PJSIP project and not in Asterisk. Therefore, the problem can be fixed without upgrading Asterisk. However, we will be releasing a new version of Asterisk where the bundled version of PJSIP has been updated to have the bug patched. If you are using Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-004 A remote memory exhaustion can be triggered by sending an SCCP packet to Asterisk system with chan_skinny enabled that is larger than the length of the SCCP header but smaller than the packet length specified in the header. The loop that reads the rest of the packet doesn't detect that the call to read() returned end-of-file before the expected number of bytes and continues infinitely. The partial data message logging in that tight loop causes Asterisk to exhaust all available memory.
2017-05-13Update to Asterisk 13.15.0. This is mostly a bug fix release with a fewjnemeth5-42/+24
minor enhancements. 13.14.1 was released to fix AST-2017-001. ----- 13.15.0 The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. The release of Asterisk 13.15.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: *New Features made in this release:* ----------------------------------- - [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>] - func_channel: Add ability to get the callid so dialplan has access to it. (Reported by Richard Mudgett) - [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>] - res_pjsip: Add endpoint identification scheme based on a configured SIP header/value (Reported by Matt Jordan) - [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>] - [patch] Allow "Comedian Mail" branding to be removed (Reported by John Covert) *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>] - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) - [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>] - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) - [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>] - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) - [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>] - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) - [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>] - res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets (Reported by Max Norba) - [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>] - res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. (Reported by Vinod Dharashive) - [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>] - res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) - [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>] - Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) - [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>] - app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) - [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>] - res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome (Reported by Dan Jenkins) - [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>] - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) - [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>] - autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) - [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/ASTERISK-26869>] - res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension (Reported by Torrey Searle) - [ASTERISK-26668 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>] - core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) - [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>] - chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) - [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/ASTERISK-26872>] - Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) - [ASTERISK-26717 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>] - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) - [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>] - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) - [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>] - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) - [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>] - chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) - [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>] - chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis - Prescom) - [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>] - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) - [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>] - res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) - [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>] - app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) - [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>] - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by Jørgen H) - [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) - [ASTERISK-26825 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>] - pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) - [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>] - PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) - [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>] - res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by Jørgen H) - [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>] - res_pjsip_outbound_registration doesn't know about network change events (Reported by George Joseph) - [ASTERISK-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>] - chan_sip : Asterisk restart seems to be required for changing encryption option (Reported by benasse) - [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>] - bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) - [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>] - res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) - [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>] - [patch] Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) - [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>] - Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) - [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>] - PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) - [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>] - [patch] DUNDi weight parameter not processed correctly (Reported by Peter Racz) - [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>] - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) - [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>] - res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) - [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/ASTERISK-26738>] - Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c (Reported by Michael Maier) - [ASTERISK-25893 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>] - Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) - [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>] - [patch] Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) - [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>] - [patch] Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) - [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>] - res_config_sqlite3 uses incorrect query - unnecessary escape (Reported by Stepan) - [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>] - SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) - [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>] - http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) - [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>] - Phone default have not ringing on ARM (Reported by Igor Goncharovsky) - [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>] - pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) - [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>] - res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) - [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>] - Asterisk fails building with OpenSSL 1.1.0 (Reported by Tzafrir Cohen) - [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>] - VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) - [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>] - [patch] 'Silence' is truncated in Record() (Reported by var) - [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>] - chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) - [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>] - core: Protect flags during ast_waitfor (Reported by Joshua Colp) - [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>] - pbx: AMI Originate ignore "failed" extension on call failure (Reported by Nasir Iqbal) - [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>] - configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) - [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>] - Crash in srv.c on startup with pjsip (Reported by nappsoft) - [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>] - res_stasis_device_state: Duplicate subscriptions when multiple received at same time (Reported by Joshua Colp) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>] - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) - [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>] - chan_sip: Add rtcp-mux support (Reported by Sean Bright) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0 *Thank you for your continued support of Asterisk!* ----- 13.14.0 The Asterisk Development Team has announced the release of Asterisk 13.14.0. The release of Asterisk 13.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0 Thank you for your continued support of Asterisk! -----
2017-05-11Update comms/gammu to 1.38.2leot3-9/+9
Changes: 20170328 - 1.38.2 [-] * Improved support for Huawei K3765, E150 and E372. [-] * Fixed decoding of unicode surrogates at message boundary. [+] * Environment variable PHONE_ID for external program. [-] * SMS compatibility with devices following old version of GSM 03.38. [-] * Unicode is now preferred when handling USSD. [+] * Improved decoding of MMS indication SMS. 20170105 - 1.38.1 [-] * Fixed sending SMS to numbers starting with 000. [-] * Fixed parsing of vcalendar files with VALUE=DATE-TIME. [-] * Fixed compatibility with D-Link dwm-157. [-] * Updated list of GSM countries and networks. 20161212 - 1.38.0 [-] * MySQL script for SMSD is compatible with strict mode. [-] * Fixed USSD responses for some AT modems. [-] * Fixed parsing network status for some modems (eg. Quectel UC15). [-] * Fixed handling of emojis and other Unicode chars from supplementary plan. [-] * Fixed compilation with C90 compiler.
2017-05-09Requires termcap.jperkin1-1/+2
2017-05-07Remove patch that has no effect.wiz2-11/+1
2017-04-30Recursive revbump from boost updateryoon5-10/+10
generated by cgit v1.2.3 (git 2.39.1) at 2025-01-11 03:30:03 +0000