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2021-04-12comms: better COMMENTnia1-2/+2
2021-03-26comms/asterisk16: Update to 16.17.0gdt2-20/+19
This is a micro update that is mostly security fixes and bug fixes with very small improvements. In addition to this being a security fix, asterisk16 is a leaf package. Upstream changes: Security bugs fixed in this release: ----------------------------------- * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash (Reported by Gregory Massel) * ASTERISK-29260 - sRTP Replay Protection ignored; even tears down long calls (Reported by Alexander Traud) * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash (Reported by Ivan Poddubny) Bugs fixed in this release: ----------------------------------- * ASTERISK-29215 - res_pjsip_session: NULL active_media_state topology caused asterisk crash (Reported by sungtae kim) * ASTERISK-29035 - chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) * ASTERISK-29071 - app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs (Reported by Stefan Ruf) * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events (Reported by N A) * ASTERISK-24434 - Fix differing usage of assignment operators in modules.conf (Reported by Rusty Newton) * ASTERISK-29306 - strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition (Reported by Vitezslav Novy) * ASTERISK-29300 - res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent (Reported by Sebastian Damm) * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address (Reported by Brian Paboojian) * ASTERISK-29266 - ICE Role conflict with an unauthorized session (Reported by Salah Ahmed) * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed into progress (Reported by Sebastian Damm) * ASTERISK-29297 - say: Y2021 problem – Asterisk cannot say year 2021 in Dutch (Reported by Jacek Konieczny) * ASTERISK-29315 - res_pjsip: re-registration gets stuck if setting initial auth credentials fails (Reported by Nick French) * ASTERISK-29312 - res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters (Reported by Alexei Gradinari) * ASTERISK-16799 - Callee declined when 'beep' audio file does not exist (Reported by IAMJames_) * ASTERISK-29313 - res_pjsip_refer: Segfault in progress notify (Reported by George Joseph) * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to return one (no more) record (Reported by Boris P. Korzun) * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't (Reported by Benjamin Keith Ford) * ASTERISK-29311 - res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit (Reported by Jaco Kroon) * ASTERISK-28452 - pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer (Reported by Michael Maier) * ASTERISK-29287 - app.h: C++ compatibility broken (Reported by Jean Aunis - Prescom) * ASTERISK-28369 - app_queue: Member device state "invalid" when second call is ringing and hint is used (Reported by Boolah ) * ASTERISK-29203 - res_pjsip_t38: Crash when changing state (Reported by Gregory Massel) * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client (Reported by Edvin Vidmar) * ASTERISK-29196 - res_pjsip: Segmentation fault (Reported by Mauri de Souza Meneguzzo (3CPlus)) * ASTERISK-29280 - chan_sip: Allow peers without audio (text+video). (Reported by Alexander Traud) * ASTERISK-29265 - chan_sip: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29261 - res_pjsip: user=phone validation fail for isup numbers containing *# (Reported by Mark Petersen) * ASTERISK-29259 - channel: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29258 - chan_sip: Audio stream rejected, Other stream present: Invalid SDP. (Reported by Alexander Traud) * ASTERISK-29220 - After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used (Reported by Robert Cripps) * ASTERISK-29248 - res_pjsip_session: res sometimes uninitialized reported by compiler Clang. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- * ASTERISK-29321 - sorcery: Add support for more intelligent reloading. (Reported by Joshua C. Colp) * ASTERISK-29325 - res_pjsip_registrar: Include source IP address and port in log messages (Reported by Joshua C. Colp) * ASTERISK-29326 - asterisk: Update copyright/company (Reported by Joshua C. Colp) * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI events (Reported by Sébastien Duthil) * ASTERISK-29275 - Support of MIME-type for wav16 (Reported by Boris P. Korzun) * ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code (Reported by Dan Cropp) * ASTERISK-29262 - Support of various URL-schemes by MoH (Reported by Boris P. Korzun)
2021-03-25Fix autoconf fallout.nia2-9/+17
2021-03-18Switch HOMEPAGE to HTTPS.fcambus1-2/+2
2021-03-16qodem: update to 1.0.1.fcambus2-7/+13
qodem (1.0.1-1) unstable; urgency=low * Bug fixes * Linux console GPM mouse support
2021-02-28asterisk14 was deletedjnemeth1-2/+1
2021-02-28asterisk14: Delete this package as discussed on pkgsrc-users on Dec. 26th.jnemeth57-6685/+0
2021-02-28asterisk13: Update to Asterisk 13.38.2:jnemeth2-12/+12
The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2021-001: Remote crash in res_pjsip_diversion If a registered user is tricked into dialing a * AST-2021-002: Remote crash possible when negotiating T.38 When * AST-2021-003: Remote attacker could prematurely tear down SRTP calls An unauthenticated remote attacker could replay SRTP packets which could cause an Asterisk instance configured without strict RTP validation to tear down calls prematurely. * AST-2021-004: An unsuspecting user could crash Asterisk with multiple hold/unhold requests Due to a signedness comparison mismatch, an authenticated WebRTC client could cause a stack overflow and Asterisk crash by sending multiple hold/unhold requests in quick succession. * AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver Given a scenario where an outgoing call is placed from Asterisk to a remote SIP server it is possible for a crash to occur. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2021-001.pdf https://downloads.asterisk.org/pub/security/AST-2021-002.pdf https://downloads.asterisk.org/pub/security/AST-2021-003.pdf https://downloads.asterisk.org/pub/security/AST-2021-004.pdf https://downloads.asterisk.org/pub/security/AST-2021-005.pdf Thank you for your continued support of Asterisk!
2021-02-28hylafax: fix builds with tiff 4.2gutteridge2-4/+5
2021-02-17Add tio.fcambus1-1/+2
2021-02-17comms/tio: import tio-1.32.fcambus4-0/+29
"tio" is a simple TTY terminal application which features a straightforward commandline interface to easily connect to TTY devices for basic input/output.
2021-02-11asterisk16: Add forgotten patchesryoon3-0/+105
2021-02-11asterisk16: Fix segfaut under NetBSD/aarch64 9.99.80. Bump PKGREVISIONryoon2-2/+6
The problem is reported by Markus Kilbinger on port-arm mailing list.
2021-02-11asterisk16: Update to 16.16.0ryoon3-37/+37
Changelog: The following issues are resolved in this release: Security bugs fixed in this release: * [ASTERISK-29219] res_pjsip_diversion: Crash if Tel URI contains History-Info (Reported by Torrey Searle) Bugs fixed in this release: * [ASTERISK-29229] Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription (Reported by Jean Aunis Prescom) * [ASTERISK-29238] chan_sip: SDP: Offers without any enabled stream are accepted. (Reported by Alexander Traud) * [ASTERISK-29237] chan_sip: SDP: m=video is parsed even when disabled. (Reported by Alexander Traud) * [ASTERISK-29222] chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. (Reported by Alexander Traud) * [ASTERISK-29240] chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable (Reported by Ivan Poddubny) * [ASTERISK-27902] chan_pjsip isnt updating hangupcause on 4XX responses (Reported by George Joseph) * [ASTERISK-28016] PJSIP sends duplicate 183 Progress responses (Reported by Alex Hermann) * [ASTERISK-28185] chan_pjsip: Subsequent same responses are not stopped (Reported by Julien) * [ASTERISK-29230] pjsip: Asterisk goes crazy and massively spams logfile if registration cant be send (Reported by Michael Maier) * [ASTERISK-29231] pjsip: SIGSEGV in CLI if no trunk is registered (Reported by Michael Maier) * [ASTERISK-29217] LOCK() can grant the same lock to multiple channels spuriously (Reported by Jaco Kroon) * [ASTERISK-29201] Crash occurs when Transfer and execute Hangup before the Transfer result (Reported by Dan Cropp) * [ASTERISK-28947] Segmentation fault in mixmonitor_ds_destroy (Reported by Robert Sutton) * [ASTERISK-29191] tel: URI in Diversion header causes crash (Reported by Mikhail Ivanov) * [ASTERISK-28883] Spyee information ist missing in ChanSpyStop AMI Event (Reported by Hendrik Wedhorn) * [ASTERISK-29188] null media causing the Asterisk crash (Reported by sungtae kim) * [ASTERISK-29209] Debug messages printed by scope trace might be missing newlines (Reported by Alexander Traud) * [ASTERISK-29024] pjsip: Route Header in Cancel request incorrectly set (Reported by Flole Systems) * [ASTERISK-29211] res_musiconhold: Segfault on realtime music on hold without entries (Reported by Nathan Bruning) * [ASTERISK-29022] Crash when manipulating PJSIP invite dlg ref counts (Reported by Sean Bright) * [ASTERISK-29173] Media cache URL requests allow infinite redirects (Reported by Sean Bright) * [ASTERISK-29175] res_pjsip_stir_shaken: Fix module description (Reported by Stanislav Abramenkov) * [ASTERISK-29148] AST_MODULE_INFO no, MODULEINFO depend (Reported by Alexander Traud) * [ASTERISK-28798] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) * [ASTERISK-29165] res_pjsip: malformed header Accept-Encoding in OPTIONS response (Reported by Alexander Greiner-Baer) * [ASTERISK-29161] Incorrect setup of recall channels (Reported by Boris P. Korzun) * [ASTERISK-29155] app_queue: Deadlock between queues container and individual queues (Reported by George Joseph) Improvements made in this release: * [ASTERISK-28549] Two repeated 183 (Reported by Gant Liu) * [ASTERISK-29216] contrib: systemd asterisk service for centos8 or other newer linux versions (Reported by Mark Petersen) * [ASTERISK-29143] res_http_media_cache: HTTP media cache stored hardcoded in /tmp (Reported by laszlovl) * [ASTERISK-29118] VoiceMail() should have an option to play greetings as Early Media (Reported by Juan Carlos Castro y Castro)
2021-01-17asterisk15: Update to asterisk 15.7.4.jnemeth18-223/+127
----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1, 15.7.4 and 16.5.1. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2019-004: Crash when negotiating for T.38 with a declined stream When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint responds with a declined media stream a crash will then occur in Asterisk. * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio frames are given to the audio transcoding support in Asterisk the number of samples are examined and as part of this a message is output to indicate that no samples are present. A change was done to suppress this message for a particular scenario in which the message was not relevant. This change assumed that information about the origin of a frame will always exist when in reality it may not. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2019-004.pdf https://downloads.asterisk.org/pub/security/AST-2019-005.pdf ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2019-002: Remote crash vulnerability with MESSAGE messages A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash. * AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an endpoint to switch it to T.38. If the endpoint responds with an improperly formatted SDP answer including both a T.38 UDPTL stream and an audio or video stream containing only codecs not allowed on the SIP peer or user a crash will occur. The code incorrectly assumes that there will be at least one common codec when T.38 is also in the SDP answer. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2019-002.pdf https://downloads.asterisk.org/pub/security/AST-2019-003.pdf ----- The Asterisk Development Team would like to announce security releases for Asterisk 15 and 16. The available releases are released as versions 15.7.2 and 16.2.1. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2019-001: Remote crash vulnerability with SDP protocol violation When Asterisk makes an outgoing call, a very specific SDP protocol violation by the remote party can cause Asterisk to crash. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2019-001.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.7.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-28222 - Regression: MWI polling no longer works (Reported by abelbeck) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1 ----- The Asterisk Development Team would like to announce the release of Asterisk 15.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.7.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan Hoffmann) * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP Upgrade requests (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on CLI 'pjsip show contacts' (Reported by Alexei Gradinari) * ASTERISK-27920 - app_queue: Queue member considered inuse after immediately hanging up during dialing. (Reported by Cao Minh Hiep) * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does not work (Reported by Cameron) * ASTERISK-28065 - res_odbc: missing SQL error diagnostic (Reported by Alexei Gradinari) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28045 - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp) * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) * ASTERISK-28047 - chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs (Reported by Will) * ASTERISK-28049 - res_pjproject build failure (Reported by Jaco Kroon) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by David Hajek) * ASTERISK-28029 - [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file (Reported by Frederic LE FOLL) * ASTERISK-28005 - channel.c: ARI ring only once (Reported by Hajek Michal) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) Improvements made in this release: ----------------------------------- * ASTERISK-28046 - Remove stale nonoptreq references (Reported by Walter Doekes) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 15 and 16. The available releases are released as versions 15.6.2 and 16.0.1. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS SRV or NAPTR response. The attacker???s request causes Asterisk to segfault and crash. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2018-010.pdf ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade There is a stack overflow vulnerability in the res_http_websocket.so module of Asterisk that allows an attacker to crash Asterisk via a specially crafted HTTP request to upgrade the connection to a websocket. The attacker???s request causes Asterisk to run out of stack space and crash. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2018-009.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.6.0 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-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE (Reported by Joshua Elson) * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27398 - No joint capabilities with video and audio-only streams (Reported by Benjamin Keith Ford) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-28001 - res_pjsip_registrar: Improve performance of inbound handling (Reported by Joshua Colp) * ASTERISK-27966 - pjsip: Race condition in 183 re transmission can result in a deadlock (Reported by Torrey Searle) * ASTERISK-15331 - make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o (Reported by Majdi Bsoul) * ASTERISK-14935 - [regression] menuselect compilation failure on Solaris 10 (Reported by Samuel Owens) * ASTERISK-12382 - menuselect compilation failure on Solaris 10 / gcc 3.4.3 (Reported by rleasure) * ASTERISK-9107 - menuselect compilation failure on Solaris 10/gcc-4.1.1 (Reported by Bob Atkins) * ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only matches against "generic string" headers (Reported by George Joseph) * ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in Developer Mode. (Reported by Alexander Traud) * ASTERISK-27591 - Frack errors in stasis.c and memory leakage (Reported by Siruja Maharjan) * ASTERISK-27978 - res_pjsip: Change default transport keepalive to preserve behavior (Reported by Joshua Colp) * ASTERISK-27968 - systemd: asterisk.service (Reported by seanchann.zhou) * ASTERISK-27880 - [patch] pjproject_bundled: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27810 - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith Ford) * ASTERISK-27972 - res_sorcery_config: Allow object name based matching (Reported by Joshua Colp) * ASTERISK-25548 - stasis: Improve message type "Use of before init/after destruction" error (Reported by Joshua Colp) * ASTERISK-27967 - srtp: rejecting short sdes lifetimes incompatible with obihai ATAs (Reported by Nick French) * ASTERISK-27961 - res_pjsip: Spurious ERROR logging when printing headers in sip_msg (Reported by Nick French) * ASTERISK-27563 - pjsip modules always get -O2 even when DONT_OPTIMIZE is set (Reported by George Joseph) * ASTERISK-27957 - PJSIP proposes ICE candidates on answer even if not in offer (Reported by Torrey Searle) * ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives. (Reported by Alexander Traud) * ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared. (Reported by Alexander Traud) * ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE attribute truncated (Reported by Kevin Harwell) * ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) * ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers (Reported by Ross Beer) * ASTERISK-27763 - res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream (Reported by Thiago Coutinho) * ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) (Reported by Jared Hull) * ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled (Reported by Torrey Searle) * ASTERISK-26686 - res_pjsip: Lock inversion in transport management (Reported by Ross Beer) * ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- * ASTERISK-28006 - PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID (Reported by Eric Dantie) * ASTERISK-27995 - pjproject_bundled: Find shared libraries in root --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27993 - pjsip_wizard example gives wrong info about unsupported SRV records (Reported by Jonathan Harris) * ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break (Reported by Emmanuel BUU) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.5.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-27818 - Username bruteforce is possible when using ACL with PJSIP (Reported by John) * ASTERISK-27807 - iostreams: Potential DoS when client connection closed prematurely (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) * ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) * ASTERISK-27943 - AMI: Action SendText needs to use the correct thread. (Reported by Richard Mudgett) * ASTERISK-27942 - res_pjsip_messaging doesn't accept application/* content-types. (Reported by George Joseph) * ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) * ASTERISK-27936 - res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 (Reported by George Joseph) * ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman) * ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective (Reported by Corey Farrell) * ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not POSIX compatible. (Reported by Alexander Traud) * ASTERISK-27903 - menuselect: GCC 8: restrict-qualified parameter passed and aliased. (Reported by Alexander Traud) * ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27705 - chan_iax2: Stops listening for traffic (Reported by Kirsty Tyerman) * ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27888 - SQL fetch error on query which return 0 columns (Reported by Alexei Gradinari) * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses (Reported by George Joseph) * ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated before terminating nul. (Reported by Alexander Traud) * ASTERISK-27872 - res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload (Reported by Alexei Gradinari) * ASTERISK-27094 - res_fax: Deadlock when using Local channels and fax gateway (Reported by David Brillert) * ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000 (Reported by Dominic) * ASTERISK-25261 - Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User' (Reported by Francois Blackburn) * ASTERISK-27878 - [patch] tcptls.h: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh. (Reported by Alexander Traud) * ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. (Reported by Alexander Traud) * ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause marker bit error (Reported by Torrey Searle) * ASTERISK-27831 - res_rtp_asterisk: Add support for abs-send-time RTP extension (Reported by Joshua Colp) * ASTERISK-27863 - config/ast_destroy_realtime_fields: successful DELETE is treated as failed (Reported by Alexei Gradinari) * ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. (Reported by Nic Colledge) * ASTERISK-27853 - Incorrect error reported when leaving/retrieving a ODBC voicemail (Reported by Nic Colledge) * ASTERISK-27726 - chan_mobile: presents incorrect inbound Caller-ID names (Reported by Brian) * ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers. (Reported by Alexander Traud) * ASTERISK-27860 - [patch] res_pjsip: Register pjsip_transport_management not externally but internally. (Reported by Alexander Traud) * ASTERISK-27852 - cli: "manager show settings" mislabels HTTP timeout as being minutes. (Reported by Corey Farrell) * ASTERISK-27824 - Fix issues exposed by GCC 8 (Reported by George Joseph) * ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. (Reported by Alexander Traud) * ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility. (Reported by Alexander Traud) * ASTERISK-27841 - digest over for manager (ami) over http fails on too long uris (Reported by Jaco Kroon) * ASTERISK-26570 - Macro allows an infinite loop of dialplan inclusion resulting in a crash (Reported by Tzafrir Cohen) * ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on" (Reported by shaurya jain) * ASTERISK-27795 - chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) * ASTERISK-27800 - One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP (Reported by Artur Pires) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. (Reported by Dimos) * ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module (Reported by Marco Giordani) * ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. (Reported by Alexander Traud) * ASTERISK-27804 - bridge_softmix / app_confbridge: Add support for combining REMB reports (Reported by Joshua Colp) * ASTERISK-27418 - app_confbridge: "core show profile bridge" does not output "sfu" when video_mode is sfu (Reported by Carlos Chavez) * ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field designator extension. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- * ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27752 - Ten seconds of silence after mp3 playback (Reported by Sam Wierema) * ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27877 - app_confbridge: Add talking indicator for ConfBridgeList AMI response (Reported by William McCall) * ASTERISK-27873 - documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event (Reported by Alessandro Polidori) * ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory (Reported by Ted G) * ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27796 - res_hep: Allow create_address to resolve a provided hostname (Reported by Sebastian Gutierrez) * ASTERISK-27820 - [patch] Add DragonFly BSD. (Reported by Alexander Traud) * ASTERISK-27793 - cppcheck identifies redundant "if" (Reported by Ilya Shipitsin) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
2021-01-10comms/ruby-termios: update to 1.1.0taca2-7/+7
1.1.0 (2020-12-25) * Fix build problem on Ruby 3.
2021-01-09(comms/py-colorama) PKGREVISION++, other than py36 need re-pkgmef1-1/+2
2021-01-09(comms/py-colorama) Add patch for py36, can't decode byte 0xc2mef2-1/+36
2021-01-03py-colorama: updated to 0.4.4adam2-7/+7
0.4.4 Re-org of README, to put the most insteresting parts near the top. Added Linux makefile targets and Windows powershell scripts to automate bootstrapping a development environment, and automate the process of testing wheels before they are uploaded to PyPI. Use stdlib unittest.mock where available Travis CI now also builds on arm64 Demo06 demonstrates existing cursor positioning feature Fix OSC regex & handling to prevent hang or crash Document enterprise support by Tidelift
2021-01-03Disable -march=native default.jnemeth2-3/+4
2021-01-03Disable -march=native default.jnemeth2-3/+4
2021-01-03asterisk16: Update to 16.15.1gdt2-20/+19
upstream changes: security fixes and bug fixes
2021-01-02Update to Asterisk 13.38.1jnemeth18-263/+264
----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18. The available releases are released as versions 13.38.1, 16.15.1, 17.9.1 and 18.1.1. The following security vulnerabilities were resolved in these versions: * AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk when a SIP message is received that has a History-Info header, which contains a tel-uri. * AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk when a SIP 181 response is received that has a Diversion header, which contains a tel-uri. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2020-003.pdf https://downloads.asterisk.org/pub/security/AST-2020-004.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.38.0. The release of Asterisk 13.38.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-29057 - pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) Improvements made in this release: ----------------------------------- * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) * ASTERISK-29108 - resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis - Prescom) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format (Reported by ?????????) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-29081 - res_stasis: Add compare function for bridges moh container (Reported by Hajek Michal) * ASTERISK-29085 - func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT (Reported by P??ter Juh??sz) * ASTERISK-28416 - Unable to get rtp codec payload code for slin (Reported by Brian J. Murrell) New Features made in this release: ----------------------------------- * ASTERISK-29027 - Implement support for History-Info (Reported by Torrey Searle) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.38.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5. The following security vulnerabilities were resolved in these versions: * AST-2020-001: Remote crash in res_pjsip_session Upon receiving a new SIP Invite, Asterisk did not return the created dialog locked or referenced. * AST-2020-002: Outbound INVITE loop on challenge with different nonce. If Asterisk is challenged on an outbound INVITE and the nonce is changed in each response, Asterisk will continually send INVITEs in a loop. This causes Asterisk to consume more and more memory since the transaction will never terminate (even if the call is hung up), ultimately leading to a restart or shutdown of Asterisk. Outbound authentication must be configured on the endpoint for this to occur. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2020-001.pdf https://downloads.asterisk.org/pub/security/AST-2020-002.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.37.0. The release of Asterisk 13.37.0 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-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) Improvements made in this release: ----------------------------------- * ASTERISK-29010 - Allow disabling of FollowMe prompt (Reported by Dennis) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.37.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.36.0. The release of Asterisk 13.36.0 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-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-29029 - Voicemail "pollmailboxes"-option not working, bug in function handle_subscribe (Reported by Karsten Wemheuer) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTERISK-28995 - res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) * ASTERISK-28978 - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) * ASTERISK-28975 - res_http_websocket: Text payload data doesn't necessary include trailing zero (Reported by Nickolay V. Shmyrev) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.36.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.35.0. The release of Asterisk 13.35.0 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-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28957 - chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) * ASTERISK-28888 - res_corosync: causes asterisk crash in huge distributed environment. (Reported by Universit?? di Bologna - CESIA VoIP) * ASTERISK-28955 - "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin) * ASTERISK-28942 - res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) * ASTERISK-28952 - Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) * ASTERISK-28950 - Stale code in app_queue to check untouched channel (Reported by Walter Doekes) * ASTERISK-28644 - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) * ASTERISK-28900 - res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) * ASTERISK-28929 - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28885 - res_rtp_asterisk: Simultaneous termination and ICE complete can cause crash (Reported by Josep B) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) Improvements made in this release: ----------------------------------- * ASTERISK-28959 - res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) * ASTERISK-28958 - Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) * ASTERISK-28945 - AMI SendText - add Content-Type parameter (Reported by Kevin Harwell) * ASTERISK-28949 - res_http_websocket: Add masking to websocket client (Reported by Moises Silva) * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.35.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.34.0. The release of Asterisk 13.34.0 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-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-28776 - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) * ASTERISK-28829 - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) * ASTERISK-25844 - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) * ASTERISK-22920 - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) * ASTERISK-28859 - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) Improvements made in this release: ----------------------------------- * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) * ASTERISK-28879 - pjproject has race conditions in it's build system (Reported by Guido Falsi) * ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) * ASTERISK-28832 - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.34.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. The release of Asterisk 13.33.0 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-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) * ASTERISK-28847 - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander Traud) * ASTERISK-28837 - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) * ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) * ASTERISK-28812 - First DTMF is not get (Reported by Bernard Merindol) * ASTERISK-28758 - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) * ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not setup yet (Reported by Kevin Harwell) * ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. (Reported by Alexander Traud) * ASTERISK-28818 - [patch] BuildSystem: Allow space in path. (Reported by Alexander Traud) * ASTERISK-28801 - [patch] stasis: Avoid always true warnings with clang. (Reported by Alexander Traud) * ASTERISK-28796 - func_channel: cannot read fields exten, context, userfield, channame from dialplan (Reported by S??bastien Duthil) * ASTERISK-28803 - [patch] chan_unistim: Avoid tautological warnings with clang. (Reported by Alexander Traud) * ASTERISK-28808 - [patch] test_stasis: Avoid always true warning with clang. (Reported by Alexander Traud) * ASTERISK-28056 - res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR (Reported by Jason Hord) * ASTERISK-28789 - test_utils: incorrectly printing error 'declined to load' (Reported by Alexander Traud) * ASTERISK-28788 - func_aes: incorrectly printing error 'declined to load' (Reported by Alexander Traud) * ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd unless asterisk is running as root (Reported by Jaco Kroon) * ASTERISK-21205 - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28759 - A non negotiated rtp frame causes call disconnection when there is a SSRC change (Reported by Paulo Vicentini) * ASTERISK-26711 - func_enum: ENUM code wrong case (Reported by Vitold) * ASTERISK-23407 - Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.33.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.32.0. The release of Asterisk 13.32.0 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-28766 - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting to SLIN (Reported by Ross Beer) * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected (Reported by Peter Sokolov) * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) * ASTERISK-28719 - Cannot remove defaultrule from queue using realtime queues (Reported by EDV O-TON) * ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) * ASTERISK-26082 - res_pjsip_messaging: MessageSend Content-Type can't be changed (Reported by Alex) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) Improvements made in this release: ----------------------------------- * ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh) * ASTERISK-24798 - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) * ASTERISK-28726 - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.32.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.31.0. The release of Asterisk 13.31.0 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-17491 - CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) * ASTERISK-28677 - CDR billsec is always 0 for transferred calls (Reported by Maciej Michno) * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show translation' output (Reported by Sean Bright) * ASTERISK-24484 - Update documentation for statsd module - usage requirements unclear (Reported by Dan Jenkins) * ASTERISK-28702 - chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas) * ASTERISK-28695 - core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn) * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano) * ASTERISK-23739 - [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) * ASTERISK-28682 - app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell) * ASTERISK-28507 - Wiki docs missing for MessageWaiting (Reported by David M. Lee) * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard Kenner) * ASTERISK-28497 - func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun) * ASTERISK-28647 - chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis - Prescom) * ASTERISK-28667 - Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann) * ASTERISK-28664 - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) * ASTERISK-28663 - jansson: Support old versions (Reported by Joshua C. Colp) * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL) * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) * ASTERISK-28628 - Debian 10.2: Warning when app_voicemail is compiling (Reported by Stanislav Abramenkov) * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud) * ASTERISK-28651 - chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon) * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer) Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) ----- The Asterisk Development Team would like to announce the release of Asterisk 13.30.0. The release of Asterisk 13.30.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-28589 - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) * ASTERISK-28580 - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sarda½½ons) Improvements made in this release: ----------------------------------- * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) * ASTERISK-28542 - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28663 - jansson: Support old versions (Reported by Joshua C. Colp) * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) * ASTERISK-28641 - res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer) * ASTERISK-28644 - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL) * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt) * ASTERISK-28631 - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer) * ASTERISK-28612 - res_pjsip_t38: crash on reinvite with zero port and no c= line (Reported by Salah Ahmed) * ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed) * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL) * ASTERISK-28616 - parking: Deadlock when multi call parking (Reported by Joshua C. Colp) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28608 - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson) * ASTERISK-28618 - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) * ASTERISK-26481 - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) * ASTERISK-28572 - Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha) * ASTERISK-28585 - ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell) * ASTERISK-28590 - utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" (Reported by Speed Dial Dave) * ASTERISK-28578 - race condition on pjsip channelstats command (Reported by Salah Ahmed) * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder) * ASTERISK-28575 - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28561 - Asterisk Deadlocks (Reported by Aheliotech) * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) * ASTERISK-28566 - CDR backend unload problem during active call(s) (Reported by Marian Piater) * ASTERISK-28544 - Wrong contact representation in ipv6 mode (Reported by J½½rgen H) * ASTERISK-28534 - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) * ASTERISK-28521 - pjsip: Memory Leak (Reported by Mark) * ASTERISK-28523 - Asterisk 16.5.0 Memory leak (Reported by Cyril Rami½½re) * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) New Features made in this release: ----------------------------------- * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.30.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5. The following security vulnerabilities were resolved in these versions: * AST-2019-006: SIP request can change address of a SIP peer. A SIP request can be sent to Asterisk that can change a SIP peer½½½s IP address. A REGISTER does not need to occur, and calls can be hijacked as a result. The only thing that needs to be known is the peer½½½s name; authentication details such as passwords do not need to be known. This vulnerability is only exploitable when the ½½½nat½½½ option is set to the default, or ½½½auto_force_rport½½½. * AST-2019-007: AMI user could execute system commands. A remote authenticated Asterisk Manager Interface (AMI) user without ½½½system½½½ authorization could use a specially crafted ½½½Originate½½½ AMI request to execute arbitrary system commands. * AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a crash will occur. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2019-006.pdf https://downloads.asterisk.org/pub/security/AST-2019-007.pdf https://downloads.asterisk.org/pub/security/AST-2019-008.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.29.1. The release of Asterisk 13.29.1 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-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28575 - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.1 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.29.0. The release of Asterisk 13.29.0 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-28521 - pjsip: Memory Leak (Reported by Mark) * ASTERISK-28523 - Asterisk 16.5.0 Memory leak (Reported by Cyril Rami½½re) * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich) * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) * ASTERISK-28505 - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) * ASTERISK-28487 - compile menuselect on gentoo (Reported by Kilburn) * ASTERISK-28472 - Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) * ASTERISK-28498 - cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) * ASTERISK-28483 - packet lost on UDPTL wrap around (Reported by Torrey Searle) * ASTERISK-28480 - json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) * ASTERISK-28477 - Crash when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-28478 - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) New Features made in this release: ----------------------------------- * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1, 15.7.4 and 16.5.1. The following security vulnerabilities were resolved in these versions: * AST-2019-004: Crash when negotiating for T.38 with a declined stream When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint responds with a declined media stream a crash will then occur in Asterisk. * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio frames are given to the audio transcoding support in Asterisk the number of samples are examined and as part of this a message is output to indicate that no samples are present. A change was done to suppress this message for a particular scenario in which the message was not relevant. This change assumed that information about the origin of a frame will always exist when in reality it may not. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2019-004.pdf https://downloads.asterisk.org/pub/security/AST-2019-005.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.28.0 The release of Asterisk 13.28.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with no body causes crash (Reported by Gil Richard) * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 reINVITE (Reported by Francesco Castellano) Bugs fixed in this release: ----------------------------------- * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 (Reported by abelbeck) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak with specific usage (Reported by Joshua C. Colp) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate (Reported by vijay kumar) * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event (Reported by Abhay Gupta) * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry Svyatogorov) * ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying (Reported by pasandev) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-27994 - PJSIP: Early media ringback not indicated after Progress() (Reported by Gregory Massel) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4. The following security vulnerabilities were resolved in these versions: * AST-2019-002: Remote crash vulnerability with MESSAGE messages A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash. * AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an endpoint to switch it to T.38. If the endpoint responds with an improperly formatted SDP answer including both a T.38 UDPTL stream and an audio or video stream containing only codecs not allowed on the SIP peer or user a crash will occur. The code incorrectly assumes that there will be at least one common codec when T.38 is also in the SDP answer. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2019-002.pdf https://downloads.asterisk.org/pub/security/AST-2019-003.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.27.0. The release of Asterisk 13.27.0 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-28375 - res_pjsip: New configuration setting to allow disabling norefersub (Reported by Dan Cropp) * ASTERISK-28320 - Added ARI resource /ari/channels/{channelid}/rtp_statistics (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-28412 - GCC 9 catches more string formatting issues (Reported by George Joseph) * ASTERISK-28392 - The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds (Reported by George Joseph) * ASTERISK-28402 - res_pjsip_registrar: SEGV in registrar_find_contact (Reported by Ross Beer) * ASTERISK-28143 - app_amd: Infinite loop on silent calls (Reported by Abhay Gupta) * ASTERISK-28353 - stasis: Crash at shutdown when statistics enabled (Reported by Joshua C. Colp) * ASTERISK-28374 - latest asterisk unconditionally launch gcc --version, even if the compiler is different (Reported by Guido Falsi) * ASTERISK-28391 - res_indications: Crash requesting autocomplete on indications cli command (Reported by Lucas Mendes) * ASTERISK-27935 - app_voicemail: emailbody per user can't contain commas (Reported by S½½bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIfTime application causes unexpected behavior (Reported by Janu) * ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323 Disabled (Reported by Dmitry Shubin) * ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info (Reported by Salah Ahmed) * ASTERISK-28362 - strtok_r() makes gcc compile warning (Reported by sungtae kim) Improvements made in this release: ----------------------------------- * ASTERISK-28363 - Millisecond-resolution call stats including PDD in channel variables (Reported by Antoni Goldstein) * ASTERISK-20207 - Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Steven Wheeler) * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to work with. (Reported by Corey Farrell) * ASTERISK-28343 - Added app_name, app_data to channel type (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.27.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.26.0. The release of Asterisk 13.26.0 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-28267 - res_stasis: Add ability to switch applications (Reported by Benjamin Keith Ford) Bugs fixed in this release: ----------------------------------- * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate (Reported by Olivier Krief) * ASTERISK-28350 - manager: Stasis backed up due to locking (Reported by Joshua C. Colp) * ASTERISK-25792 - chan_sip: qualifygap bounds checking (Reported by Paul Sandys) * ASTERISK-28341 - res_config_odbc eliminates empty custom (½½½@½½½ prefix) variables (Reported by Alexei Gradinari) * ASTERISK-28333 - StasisEnd event makes wrong timestamp value (Reported by sungtae kim) * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent (Reported by Jared Hull) * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats without channel lock or reference (Reported by Francisco Seratti) * ASTERISK-28314 - ARI: API changed but "apiVersion" in rest-api\resources.json did not (Reported by Stefan Repke) * ASTERISK-28335 - stasis: Make topic and maybe subscription names unique and more useful (Reported by Joshua C. Colp) * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation (Reported by sungtae kim) * ASTERISK-28332 - Variable ALTCONF ignored when service is used in Debian (Reported by Cirillo Ferreira) * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of 183 without SDP (Reported by Torrey Searle) * ASTERISK-28328 - MeetMe global non-admin mute is muting admins that subsequently join (Reported by Philip Mott) * ASTERISK-28168 - app_queue: Adding a blank entry into sql queue_members crashes asterisk. (Reported by Michael) * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion script fails (Reported by Guido Weckwerth) * ASTERISK-28272 - The basic-pbx config samples don't produce a running asterisk (Reported by George Joseph) * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect (Reported by Alex Odrov) * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no license header (Reported by Jeremy Lain½½) * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC (Reported by Michael) * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces (Reported by Nikolay shakin) * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash (Reported by Jonathan Harris) * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked (Reported by Ross Beer) * ASTERISK-28301 - Allow voicemail boxes to be subscribed to with a presence event package (Reported by George Joseph) * ASTERISK-28303 - res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps (Reported by Torrey Searle) * ASTERISK-28302 - ARI: "Error destroying mutex" when listing all ARI applications (Reported by Stefan Repke) * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some applications (Reported by George Joseph) * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI events when GETting causes overload of events (Reported by George Joseph) * ASTERISK-28284 - switching between native_bridge and simple_bridge can cause one way audio (Reported by Torrey Searle) * ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls (Reported by Paulo Vicentini) * ASTERISK-28251 - CI: Fix CI so it reverifies commit message changes (Reported by George Joseph) * ASTERISK-28277 - database: Add some basic logging (Reported by Joshua C. Colp) * ASTERISK-28181 - ari: Originating overwrites channel start time (Reported by sungtae kim) Improvements made in this release: ----------------------------------- * ASTERISK-28326 - ari: Added timestamp for some ari events. (Reported by sungtae kim) * ASTERISK-28317 - Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function (Reported by Cirillo Ferreira) * ASTERISK-28279 - Added creation timestamp for bridge (Reported by sungtae kim) * ASTERISK-28292 - Changed to show all channel stats including wrong media (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.26.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.25.0. The release of Asterisk 13.25.0 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-28288 - Resources (udptl fd) leaking for T.38 calls (Reported by Paulo Vicentini) * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked (Reported by Ross Beer) * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile (Reported by David Wilcox) * ASTERISK-28104 - AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps (Reported by George Joseph) * ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI (Reported by Ray) * ASTERISK-28173 - Deadlock in chan_sip handling subscribe request during res_parking reload (Reported by Giuseppe Sucameli) * ASTERISK-28263 - codec_opus: errors setting max_playback_rate and bitrate to "sdp" (Reported by Gianluca Merlo) * ASTERISK-28250 - build: Cross-compilation fails for target arm-linux-gnueabihf (Reported by Jean Aunis - Prescom) * ASTERISK-28156 - Race condition involving session->media (res_pjsip_session) leads to crash. (Reported by Paulo Vicentini) * ASTERISK-28257 - res_http_websocket: PING / PONG opcodes break data reception (Reported by Jeremy Lain½½) * ASTERISK-28252 - HangupHandler manager events are never thrown (Reported by Gerald Schnabel) * ASTERISK-28231 - res_http_websocket: Not responding to Connection Close Frame (opcode 8) (Reported by Jeremy Lain½½) * ASTERISK-28249 - res_monitor: Segfault with Monitor(wav,file,i) (Reported by Valentin Vidi½½) * ASTERISK-28244 - stasis: Filter messages at publishing to AMI/ARI (Reported by Joshua C. Colp) * ASTERISK-28197 - stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases (Reported by Mohit Dhiman) * ASTERISK-28232 - core: RAII using clang use-after-scope issue (Reported by Diederik de Groot) * ASTERISK-28225 - app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent" (Reported by boatright) * ASTERISK-28212 - stasis: Statistics broke ABI under developer mode (Reported by Joshua C. Colp) * ASTERISK-28222 - Regression: MWI polling no longer works (Reported by abelbeck) * ASTERISK-28221 - Bug in ast_coredumper (Reported by Andrew Nagy) * ASTERISK-28162 - [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation (Reported by Alexei Gradinari) * ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs (Reported by George Joseph) * ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem (Reported by David Kuehling) * ASTERISK-28117 - stasis: Add statistics for usage when in developer mode (Reported by Joshua C. Colp) * ASTERISK-28201 - [patch] confbridge: no announce to the marked users when they join an empty conference (Reported by Alexei Gradinari) * ASTERISK-28194 - chan_sip: Leak using contact ACL (Reported by Giuseppe Sucameli) * ASTERISK-28186 - stasis: Filter messages at publishing based on to_* presence (Reported by Joshua C. Colp) * ASTERISK-27095 - chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE (Reported by George Joseph) * ASTERISK-28182 - chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE (Reported by nappsoft) Improvements made in this release: ----------------------------------- * ASTERISK-28246 - Support skipping on the g726 format (Reported by Eyal Hasson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.24.1. The release of Asterisk 13.24.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-28222 - Regression: MWI polling no longer works (Reported by abelbeck) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.1 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.24.0. The release of Asterisk 13.24.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP Upgrade requests (Reported by Sean Bright) New Features made in this release: ----------------------------------- * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28125 - app_queue: Revert broken queue channel reference patch (Reported by lvl) * ASTERISK-28151 - app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default (Reported by Ronald Raikes) * ASTERISK-28157 - Asterisk crashes when the res_pjsip_* modules unload (Reported by sungtae kim) * ASTERISK-28159 - SIGABRT caused by stack corruption in hashkeys_read when no matching keys present (Reported by Michael Walton) * ASTERISK-28140 - repeated segmentation faults (Reported by Eyal Hasson) * ASTERISK-28103 - stasis: Filter messages at publishing to reduce work done (Reported by Joshua C. Colp) * ASTERISK-28129 - Incorrect Behavior for rewrite_contact when Re-Invite omits routset (Reported by Torrey Searle) * ASTERISK-28158 - Some conditions prevent running of el_end, break the terminal. (Reported by Corey Farrell) * ASTERISK-28162 - [patch] need to reset DTMF last sequence number and timestamp on voice packet with marker bit (Reported by Alexei Gradinari) * ASTERISK-28110 - rtp: Incorrect Packetization (Reported by Robert Cripps) * ASTERISK-28146 - pbx_config: Only the first [globals] section is processed. (Reported by Corey Farrell) * ASTERISK-28150 - Formatting error in documentation (Reported by Scott Griepentrog) * ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit van Drongelen) * ASTERISK-28137 - res_pjsip_notify: improve realtime performance on CLI completion on the endpoint (Reported by Alexei Gradinari) * ASTERISK-27980 - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on CLI 'pjsip show contacts' (Reported by Alexei Gradinari) * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does not work (Reported by Cameron) * ASTERISK-27920 - app_queue: Queue member considered inuse after immediately hanging up during dialing. (Reported by Cao Minh Hiep) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-28065 - res_odbc: missing SQL error diagnostic (Reported by Alexei Gradinari) * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28049 - res_pjproject build failure (Reported by Jaco Kroon) * ASTERISK-28029 - [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file (Reported by Frederic LE FOLL) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) Improvements made in this release: ----------------------------------- * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI (Reported by Alexei Gradinari) * ASTERISK-28136 - Allow the sip_to_pjsip script to be used in a pipe (Reported by Pascal Cadotte Michaud) * ASTERISK-28046 - Remove stale nonoptreq references (Reported by Walter Doekes) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3. These releases are available for immediate download at The following security vulnerabilities were resolved in these versions: * AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade There is a stack overflow vulnerability in the res_http_websocket.so module of Asterisk that allows an attacker to crash Asterisk via a specially crafted HTTP request to upgrade the connection to a websocket. The attacker½½½s request causes Asterisk to run out of stack space and crash. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1 The security advisory is available at: https://downloads.asterisk.org/pub/security/AST-2018-009.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.23.0. The release of Asterisk 13.23.0 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-27881 - PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. (Reported by Alexander Traud) * ASTERISK-28001 - res_pjsip_registrar: Improve performance of inbound handling (Reported by Joshua Colp) * ASTERISK-27999 - Wrong SRTP use status report (Reported by Salah Ahmed) * ASTERISK-27966 - pjsip: Race condition in 183 re transmission can result in a deadlock (Reported by Torrey Searle) * ASTERISK-15331 - make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o (Reported by Majdi Bsoul) * ASTERISK-14935 - [regression] menuselect compilation failure on Solaris 10 (Reported by Samuel Owens) * ASTERISK-12382 - menuselect compilation failure on Solaris 10 / gcc 3.4.3 (Reported by rleasure) * ASTERISK-9107 - menuselect compilation failure on Solaris 10/gcc-4.1.1 (Reported by Bob Atkins) * ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only matches against "generic string" headers (Reported by George Joseph) * ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in Developer Mode. (Reported by Alexander Traud) * ASTERISK-27591 - Frack errors in stasis.c and memory leakage (Reported by Siruja Maharjan) * ASTERISK-27978 - res_pjsip: Change default transport keepalive to preserve behavior (Reported by Joshua Colp) * ASTERISK-27957 - PJSIP proposes ICE candidates on answer even if not in offer (Reported by Torrey Searle) * ASTERISK-27880 - [patch] pjproject_bundled: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-25548 - stasis: Improve message type "Use of before init/after destruction" error (Reported by Joshua Colp) * ASTERISK-27972 - res_sorcery_config: Allow object name based matching (Reported by Joshua Colp) * ASTERISK-27967 - srtp: rejecting short sdes lifetimes incompatible with obihai ATAs (Reported by Nick French) * ASTERISK-27961 - res_pjsip: Spurious ERROR logging when printing headers in sip_msg (Reported by Nick French) * ASTERISK-27563 - pjsip modules always get -O2 even when DONT_OPTIMIZE is set (Reported by George Joseph) * ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives. (Reported by Alexander Traud) * ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared. (Reported by Alexander Traud) * ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) * ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers (Reported by Ross Beer) * ASTERISK-27763 - res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream (Reported by Thiago Coutinho) * ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) (Reported by Jared Hull) * ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled (Reported by Torrey Searle) * ASTERISK-26686 - res_pjsip: Lock inversion in transport management (Reported by Ross Beer) * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE (Reported by Joshua Elson) Improvements made in this release: ----------------------------------- * ASTERISK-28006 - PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID (Reported by Eric Dantie) * ASTERISK-27995 - pjproject_bundled: Find shared libraries in root --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27993 - pjsip_wizard example gives wrong info about unsupported SRV records (Reported by Jonathan Harris) * ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break (Reported by Emmanuel BUU) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.23.0 ----- The Asterisk Development Team would like to announce the release of Asterisk 13.22.0. The release of Asterisk 13.22.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-27818 - Username bruteforce is possible when using ACL with PJSIP (Reported by John) Bugs fixed in this release: ----------------------------------- * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) * ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) * ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) * ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman) * ASTERISK-27943 - AMI: Action SendText needs to use the correct thread. (Reported by Richard Mudgett) * ASTERISK-27942 - res_pjsip_messaging doesn't accept application/* content-types. (Reported by George Joseph) * ASTERISK-27936 - res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 (Reported by George Joseph) * ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective (Reported by Corey Farrell) * ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not POSIX compatible. (Reported by Alexander Traud) * ASTERISK-27903 - menuselect: GCC 8: restrict-qualified parameter passed and aliased. (Reported by Alexander Traud) * ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27705 - chan_iax2: Stops listening for traffic (Reported by Kirsty Tyerman) * ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27888 - SQL fetch error on query which return 0 columns (Reported by Alexei Gradinari) * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses (Reported by George Joseph) * ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated before terminating nul. (Reported by Alexander Traud) * ASTERISK-27094 - res_fax: Deadlock when using Local channels and fax gateway (Reported by David Brillert) * ASTERISK-25261 - Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User' (Reported by Francois Blackburn) * ASTERISK-27878 - [patch] tcptls.h: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27872 - res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload (Reported by Alexei Gradinari) * ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh. (Reported by Alexander Traud) * ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. (Reported by Alexander Traud) * ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause marker bit error (Reported by Torrey Searle) * ASTERISK-27863 - config/ast_destroy_realtime_fields: successful DELETE is treated as failed (Reported by Alexei Gradinari) * ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH. (Reported by Alexander Traud) * ASTERISK-27853 - Incorrect error reported when leaving/retrieving a ODBC voicemail (Reported by Nic Colledge) * ASTERISK-27726 - chan_mobile: presents incorrect inbound Caller-ID names (Reported by Brian) * ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers. (Reported by Alexander Traud) * ASTERISK-27860 - [patch] res_pjsip: Register pjsip_transport_management not externally but internally. (Reported by Alexander Traud) * ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. (Reported by Nic Colledge) * ASTERISK-27852 - cli: "manager show settings" mislabels HTTP timeout as being minutes. (Reported by Corey Farrell) * ASTERISK-27824 - Fix issues exposed by GCC 8 (Reported by George Joseph) * ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility. (Reported by Alexander Traud) * ASTERISK-27841 - digest over for manager (ami) over http fails on too long uris (Reported by Jaco Kroon) * ASTERISK-26570 - Macro allows an infinite loop of dialplan inclusion resulting in a crash (Reported by Tzafrir Cohen) * ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on" (Reported by shaurya jain) * ASTERISK-26806 - pjsip_options: rework to make more efficient (Reported by Kevin Harwell) * ASTERISK-27814 - translate: interpolated frames are not passed through (Reported by Kevin Harwell) * ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. (Reported by Dimos) * ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module (Reported by Marco Giordani) * ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. (Reported by Alexander Traud) * ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field designator extension. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- * ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27752 - Ten seconds of silence after mp3 playback (Reported by Sam Wierema) * ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27877 - app_confbridge: Add talking indicator for ConfBridgeList AMI response (Reported by William McCall) * ASTERISK-27873 - documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event (Reported by Alessandro Polidori) * ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory (Reported by Ted G) * ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. (Reported by Alexander Traud) * ASTERISK-27796 - res_hep: Allow create_address to resolve a provided hostname (Reported by Sebastian Gutierrez) * ASTERISK-27820 - [patch] Add DragonFly BSD. (Reported by Alexander Traud) * ASTERISK-27793 - cppcheck identifies redundant "if" (Reported by Ilya Shipitsin) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.22.0 ----- The Asterisk Development Team would like to announce security releases for Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2. The following security vulnerabilities were resolved in these versions: * AST-2018-007: Infinite loop when reading iostreams When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or sends a specially crafted message then Asterisk gets caught in an infinite loop while trying to read the data stream. Thus rendering the system as unusable. * AST-2018-008: PJSIP endpoint presence disclosure when using ACL When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2018-007.pdf https://downloads.asterisk.org/pub/security/AST-2018-008.pdf ----- The Asterisk Development Team would like to announce the release of Asterisk 13.21.0. The release of Asterisk 13.21.0 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-27704 - Add cache_pools debug option to pjproject.conf (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally. (Reported by Alexander Traud) * ASTERISK-27774 - res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) * ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro César Arruda) * ASTERISK-27614 - res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) * ASTERISK-27740 - chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed (Reported by Shannon Price) * ASTERISK-27706 - PJSIP: Deadlock shutting down subscription TCP connection and sending subscription message. (Reported by Ross Beer) * ASTERISK-27435 - [patch] configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) * ASTERISK-27761 - [patch] BuildSystem: With external editline, do not require libs for internal editline. (Reported by Alexander Traud) * ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status (Reported by Kevin Harwell) * ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported by Ross Beer) * ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on a channel are equal (Reported by George Joseph) * ASTERISK-27745 - [patch] BuildSystem: Remove unused dependency on libltdl. (Reported by Alexander Traud) * ASTERISK-12841 - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by Michiel van Baak) * ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. (Reported by Alexander Traud) * ASTERISK-27741 - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't account for more than one 'Reason' header (Reported by Ross Beer) * ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux. (Reported by Alexander Traud) * ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x on openSUSE. (Reported by Alexander Traud) * ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt Sampson) * ASTERISK-27641 - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported by Alexander Traud) * ASTERISK-25586 - uuid_generate_random detection failure (Reported by John Nemeth) * ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported by Alexander Traud) * ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to colon character when not found. (Reported by Alexander Traud) * ASTERISK-27703 - AMI Action VoicemailUsersList returns 0 MessageCount (Reported by Sébastien Duthil) * ASTERISK-27674 - chan_sip: RTP framing issues on outgoing calls (Reported by Jean Aunis - Prescom) * ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints (Reported by Ross Beer) * ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by Alexander Traud) * ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly but explicitly on external libraries. (Reported by Alexander Traud) * ASTERISK-27719 - [patch] res_http_post: Enable GMime in NetBSD. (Reported by Alexander Traud) * ASTERISK-27716 - [patch] BuildSystem: Enable autotools in NetBSD. (Reported by Alexander Traud) * ASTERISK-27714 - [patch] chan_unistim: NetBSD has an incompatible struct in_pktinfo. (Reported by Alexander Traud) * ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t explicitly to its proposed type. (Reported by Alexander Traud) * ASTERISK-27712 - [patch] BuildSystem: Detect whether uselocale(.) is available. (Reported by Alexander Traud) * ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of pthread_* on NetBSD. (Reported by Alexander Traud) * ASTERISK-27710 - [patch] BuildSystem: Install init scripts on openSUSE Tumbleweed. (Reported by Alexander Traud) * ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison in ./configure. (Reported by Alexander Traud) * ASTERISK-27610 - app_amd.so returning TOOLONG before reaching the timeout (Reported by Michael Cargile) * ASTERISK-26688 - Documentation: voicemail.conf.sample shows 512 limit for emailbody field, however this is only true if compiled with LOW_MEMORY option (Reported by Fran Vicente) * ASTERISK-27568 - PJSIP: Crash during SIP attended transfer. (Reported by Bryan Walters) * ASTERISK-27686 - [patch] install_prereq: Update FreeBSD libraries. (Reported by Alexander Traud) * ASTERISK-24488 - Wrong remote identity and target in dialog package XML in NOTIFY (Reported by Alejandro Padilla) * ASTERISK-27646 - ICE fails with no candidate nominated (Reported by Thomas Guebels) * ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- * ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by Nathan Bruning) * ASTERISK-26540 - cdr_radius: use radcli instead of freeradius-client (Reported by Tzafrir Cohen) * ASTERISK-27770 - [patch] install_prereq: Add Slackware (somehow). (Reported by Alexander Traud) * ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux. (Reported by Alexander Traud) * ASTERISK-27738 - [patch] install_prereq: Add Arch Linux. (Reported by Alexander Traud) * ASTERISK-27736 - [patch] install_prereq: Add SUSE. (Reported by Alexander Traud) * ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex) * ASTERISK-27728 - [patch] BuildSystem: Add NetBSD. (Reported by Alexander Traud) * ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported by Richard Mudgett) * ASTERISK-27729 - [patch] install_prereq: Add NetBSD. (Reported by Alexander Traud) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.21.0 ----- The release of Asterisk 13.20.0 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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-27583 - Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute (Reported by Sandro Gauci) * ASTERISK-27582 - Segmentation fault occurs in Asterisk with an invalid SDP media format description (Reported by Sandro Gauci) * ASTERISK-27618 - Crash occurs when sending a repeated number of INVITE messages over TCP or TLS transport (Reported by Sandro Gauci) * ASTERISK-27640 - SUBSCRIBE message with a large Accept value causes stack corruption (Reported by Sandro Gauci) New Features made in this release: ----------------------------------- * ASTERISK-27117 - core: Add support for timelen parsing to ast_parse_arg and ACO. (Reported by Corey Farrell) Bugs fixed in this release: ----------------------------------- * ASTERISK-27703 - AMI Action VoicemailUsersList returns 0 MessageCount (Reported by Sébastien Duthil) * ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt" needs update or removal. (Reported by Rusty Newton) * ASTERISK-27689 - [patch] rtp_engine: Load format name / mime type in uppercase again. (Reported by Alexander Traud) * ASTERISK-27679 - res_pjsip: Endpoint destruction does not free DTLS configuration (Reported by Mak Dee) * ASTERISK-27684 - [patch] install_prereq: Update OpenBSD libraries. (Reported by Alexander Traud) * ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on OpenBSD. (Reported by Alexander Traud) * ASTERISK-27680 - [patch] res_calendar: Specialized calendars depend on symbols of general calendar. (Reported by Alexander Traud) * ASTERISK-27677 - [patch] BuildSystem: Enable system provided libedit on OpenBSD. (Reported by Alexander Traud) * ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported by Alexander Traud) * ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with previous paths. (Reported by Alexander Traud) * ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash is not installed. (Reported by Alexander Traud) * ASTERISK-27666 - chan_sip: Crash processing CANCEL request (Reported by Leandro Dardini) * ASTERISK-27584 - Internal pjproject build doesn't disable bcg729 (Reported by Stuart Henderson) * ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported by Alexander Traud) * ASTERISK-27642 - [patch] backtrace: Avoid -Wlogical-not-parentheses. (Reported by Alexander Traud) * ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu libraries. (Reported by Alexander Traud) * ASTERISK-27656 - CDR: Leaking channel snapshots allocated by stasis_channel.c (Reported by Kristijan Vrban) * ASTERISK-27426 - chan_console: cannot read and write at the same time with alsa backend (Reported by Tzafrir Cohen) * ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI event (Reported by sungtae kim) * ASTERISK-27652 - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson) * ASTERISK-27612 - Subscriptions Persist After Expiration and TCP/TLS Disconnect (Reported by Ross Beer) * ASTERISK-27571 - res_pjsip: If SIP response is received during shutdown a crash may occur (Reported by Joshua Colp) * ASTERISK-27637 - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported by Alexander Traud) * ASTERISK-27635 - [patch] app_voicemail: Avoid always true warnings with clang. (Reported by Alexander Traud) * ASTERISK-27599 - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries. (Reported by Alexander Traud) * ASTERISK-26563 - core: macOS devmode build fails: variable 'freeswap' set but not used (Reported by David M. Lee) * ASTERISK-27630 - [patch] editline: Avoid shifting a negative signed value. (Reported by Alexander Traud) * ASTERISK-16172 - Problems with siren14 codec; problems with siren7 sound files. (Reported by Steve Murphy) * ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with autoconf 2.60 (Reported by Stéphan Kochen) * ASTERISK-27603 - [patch] install_prereq: Download latest Jansson. (Reported by Alexander Traud) * ASTERISK-27607 - [patch] res_config_mysql: Avoid the header mysql_version.h. (Reported by Alexander Traud) * ASTERISK-24598 - When running ./contrib/scripts/install_prereq install-unpackaged pjproject is installed in wrong place (Reported by PowerPBX) * ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory. (Reported by Alexander Traud) * ASTERISK-27600 - [patch] BuildSystem: Allow make clean all again. (Reported by Alexander Traud) * ASTERISK-27598 - [patch] install_prereq: Support package manager DNF. (Reported by Alexander Traud) * ASTERISK-26596 - Placing call on hold temporarily locks up set (Reported by Igor Goncharovsky) * ASTERISK-27596 - [patch] BuildSystem: Use the detected name for MD5 everywhere. (Reported by Alexander Traud) * ASTERISK-27594 - [patch] BuildSystem: Invoke install not in GNU but POSIX style. (Reported by Alexander Traud) * ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml. (Reported by Alexander Traud) * ASTERISK-27592 - [patch] BuildSystem: Detect external library Lua in version 5.3. (Reported by Alexander Traud) * ASTERISK-26832 - res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581 (Reported by Ross Beer) * ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id -u instead. (Reported by Alexander Traud) * ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES compiler flag (Reported by Jean Aunis - Prescom) * ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting an int to unsigned char. (Reported by Alexander Traud) * ASTERISK-27560 - [patch] clang 5 does not know -Wno-format-truncation (Reported by Alexander Traud) * ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an int to unsigned short. (Reported by Alexander Traud) * ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only matches against header if match by ip fails (Reported by George Joseph) * ASTERISK-27549 - [patch] translate: Avoid absolute value on unsigned substraction. (Reported by Alexander Traud) * ASTERISK-27553 - [patch] res_curl: Avoid error message on unload. (Reported by Alexander Traud) * ASTERISK-27557 - [patch] clang 5.0: implicit conversion to char changes value to negative. (Reported by Alexander Traud) * ASTERISK-27559 - [patch] editline: Avoid comparison between pointer and zero character constant. (Reported by Alexander Traud) * ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative signed value. (Reported by Alexander Traud) * ASTERISK-25329 - Asterisk configure fails on 'cannot find ptlib-config', despite ptlib-config existing (Reported by Rusty Newton) * ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to positive values as intended. (Reported by Alexander Traud) * ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header guard. (Reported by Alexander Traud) * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) * ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled. (Reported by Tzafrir Cohen) * ASTERISK-27498 - ICE candidate parser - ICE foundation parsing too short (Reported by Michele Prà ) * ASTERISK-27366 - Asterisk Turkish Language Set Problem (Reported by Halil Ä°brahim YILDIZ) * ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) * 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-24198 - Typo's (Reported by Walter Doekes) * ASTERISK-27229 - bridge: Old channel video source not set to NULL after unref (Reported by Richard Kenner) Improvements made in this release: ----------------------------------- * ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools on OpenBSD. (Reported by Alexander Traud) * ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to ConfbridgeList AMI events (Reported by Richard Mudgett) * ASTERISK-27647 - app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking. (Reported by Richard Mudgett) * ASTERISK-27084 - Reduce verbosity while loading PBX extensions. (Reported by Ludovic Gasc (Eyepea)) * ASTERISK-24372 - [patch] Add config option to play a prompt to the "winner" in app_followme (Reported by Graham Mainwaring) * ASTERISK-27461 - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata) * ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate from chan_sip to chan_pjsip realtime (Reported by Torrey Searle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.20.0 ----- 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 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3. The following security vulnerabilities were resolved in these versions: * AST-2018-001: Crash when receiving unnegotiated dynamic payload The RTP support in Asterisk maintains its own registry of dynamic codecs and desired payload numbers. While an SDP negotiation may result in a codec using a different payload number these desired ones are still stored internally. When an RTP packet was received this registry would be consulted if the payload number was not found in the negotiated SDP. This registry was incorrectly consulted for all packets, even those which are dynamic. If the payload number resulted in a codec of a different type than the RTP stream (for example the payload number resulted in a video codec but the stream carried audio) a crash could occur if no stream of that type had been negotiated. This was due to the code incorrectly assuming that a stream of the type would always exist. * AST-2018-002: Crash when given an invalid SDP media format description By crafting an SDP message with an invalid media format description Asterisk crashes when using the pjsip channel driver because pjproject's sdp parsing algorithm fails to catch the invalid media format description. * AST-2018-003: Crash with an invalid SDP fmtp attribute By crafting an SDP message body with an invalid fmtp attribute Asterisk crashes when using the pjsip channel driver because pjproject's fmtp retrieval function fails to check if fmtp value is empty (set empty if previously parsed as invalid). * AST-2018-004: Crash when receiving SUBSCRIBE request When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed despite having a fixed limit of 32. If more than 32 Accept headers were present the code would write outside of its memory and cause a crash. * AST-2018-005: Crash when large numbers of TCP connections are closed suddenly A crash occurs when a number of authenticated INVITE messages are sent over TCP or TLS and then the connection is suddenly closed. This issue leads to a segmentation fault. * AST-2018-006: WebSocket frames with 0 sized payload causes DoS When reading a websocket, the length was not being checked. If a payload of length 0 was read, it would result in a busy loop that waited for the underlying connection to close. For a full list of changes in the current releases, please see the ChangeLogs: https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2 The security advisories are available at: https://downloads.asterisk.org/pub/security/AST-2018-001.pdf https://downloads.asterisk.org/pub/security/AST-2018-002.pdf https://downloads.asterisk.org/pub/security/AST-2018-003.pdf https://downloads.asterisk.org/pub/security/AST-2018-004.pdf https://downloads.asterisk.org/pub/security/AST-2018-005.pdf https://downloads.asterisk.org/pub/security/AST-2018-006.pdf ----- The release of Asterisk 13.19.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27656 - CDR: Leaking channel snapshots allocated by stasis_channel.c (Reported by Kristijan Vrban) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.1
2020-12-31asterisk16: Avoid using -march=native, it breaks binary packages.nia2-3/+6
Also avoid passing crazy optimization and debug flags in general, just honor the user's CFLAGS.
2020-12-10asterisk16: Update to 16.15.0gdt4-27/+32
Upstream changes: bugfixes minor improvements STIR/SHAKEN support
2020-12-04Revbump packages with a runtime Python dep but no version prefix.nia2-4/+4
For the Python 3.8 default switch.
2020-11-26py-serial: updated to 3.5adam4-15/+24
Version 3.5 Bugfixes: - spy: ensure bytes in write() Bugfixes (posix): - serialposix: Fix inconsistent state after exception in open() Bugfixes (win32): - win32: Fix exception for composite serial number search on Windows Bugfixes (MacOS): - list_ports_osx: kIOMasterPortDefault no longer exported on Big Sur - list_ports_osx: getting USB info on BigSur/AppleSilicon
2020-11-05*: Recursive revbump from textproc/icu-68.1ryoon12-24/+24
2020-11-05*: Recursive revbump from textproc/icu-68.1ryoon2-4/+4
2020-10-06(comms/openobex) Updated 1.7.1 to 1.7.2. ChangeLog unknown. Adapt to Doxygen ↵mef3-11/+10
1.8.20
2020-10-04kermit: Take MAINTAINERshipgdt1-2/+2
I am talking to upstream about integrating patches, and about to package an alpha in wip. This should be viewed as a soft MAINTAINERship, but please ask me if you want to do anything signficant to avoid duplicated effort.
2020-09-08*: use MASTER_SITE_PERL_CPANwiz2-4/+4
2020-09-06p5-Device-Modem: update to 1.59.wiz2-8/+7
1.59 Mon Jun 15 08:17:54 CEST 2020 - Merged pull request #6 from ghciv6/fix_multi_instance_log fixed log handling with multi instances and typo in close(). Thanks to @ghciv6 ! 1.58 - Updated test suite a bit. - Added the tests to the manifest. - Got rid of indirect object syntax. - Moved test.pl to the actual test suite. - Updated $VERSION declarations according to: http://www.dagolden.com/index.php/369/version-numbers-should-be-boring/ - Added some extra tests (xt/author, xt/release). - Fixed some spelling.
2020-08-31*: bump PKGREVISION for perl-5.32.wiz19-38/+38
2020-08-18*: revbump for libsndfileleot1-2/+2
2020-08-18synce-libsynce: fix buildtnn2-1/+22
2020-08-17*: revbump after fontconfig bl3 changes (libuuid removal)leot13-24/+26
2020-08-13asterisk16: Update to 16.12.0ryoon4-35/+53
Changelog: Bugs fixed in this release: ----------------------------------- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco Kroon) [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 (Reported by Jared Smith) [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna - CESIA VoIP) [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin) [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported by Walter Doekes) [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) [ASTERISK-28940] - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) [ASTERISK-28900] - res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by nappsoft) [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) Improvements made in this release: ----------------------------------- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) [ASTERISK-28958] - Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin Harwell) [ASTERISK-28949] - res_http_websocket: Add masking to websocket client (Reported by Moises Silva) [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell)
2020-07-30kermit: add a more detailed patch commentgutteridge2-3/+7
2020-07-30kermit: fix compilation on Linux with glibc >= 2.28gutteridge2-6/+29
Fix taken from the upstream project's 9.0.305 Alpha.01 release, noted to be a temporary workaround. (Separately, from how I read the change log, there has been no stable 9.0 release since 9.0.302.) Tested on Debian 9.13 (which has an older version of glibc which wouldn't reproduce the issue) and Fedora 31 & 32. (This issue was reported on pkgsrc-users back in July 2019 by Pierre Dupond, and I'd provided a workaround for it in that email chain, but I'd never actually committed anything to pkgsrc.)
2020-07-21py-esptool: updated to 2.8adam3-22/+24
Version 2.8 Features esptool.py image_info now prints a summary of segment memory types (IRAM, DRAM, etc) based on the address range. esptool.py write_flash will warn if it looks like a bootloader binary is built for ESP32-S2 or another newer chip (support for flashing ESP32-S2 will be added in a future version.) Bug Fixes Removed ESP8266 SDK & ESP-IDF dependencies when building the flasher stub binaries. Previously the SDKs were used to include some register address macros, only. This removes any uncertainty about whether the flasher stub binary is a derived work of either SDK. The flasher stub binary itself is the same as the binary in v2.7. Fixed minor issues running esptool automated tests on macOS. Minor flake8 fixes including compatibility with newer flake8 versions. ESP32 Only Features Support detection of new ESP32 silicon revisions New esptool.py elf2image --min-rev X option allows creating a .bin file which only supports a minimum ESP32 silicon revision. Bugfixes Fix burning custom MAC with espefuse.py when 3/4 Coding Scheme is set
2020-06-12asterisk16: Update to 16.11.0ryoon2-20/+19
Changelog: Bugs fixed in this release: ----------------------------------- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by nappsoft) [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim) [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) [ASTERISK-28870] - streams: One memory leak and one issue cloning streams (Reported by George Joseph) [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud) Improvements made in this release: ----------------------------------- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) [ASTERISK-28896] - ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported by Guido Falsi) [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi) [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak)
2020-06-11efax-gtk: Update to 3.2.15ryoon2-8/+7
Changelog: Version 3.2.15 (3rd June 2020) -------------- Fix build for gcc-10 (efax/efaxlib.h, efax/efaxlib.c, efax/Makefile.am, efax/Makefile.in). Version 3.2.14 (6th March 2020) -------------- Remove X11 specific code to allow the program to run better against wayland compositors (acinclude.m4, configure.ac; dialogs.cpp, helpfile.cpp, logger.cpp, main.cpp, mainwindow.cpp, prog_defs.h; src/Makefile.am). Fix label layout in settings dialog (settings.cpp). Apply SO_REUSEADDR option when constructing sockets (socket_server.cpp). Deal with strict aliasing warning (efax/efaxos.c).
2020-06-02Revbump for icuadam14-28/+28
2020-05-31comms/asterisk16: remove unknow configure optionrillig1-3/+1
2020-05-22revbump after updating security/nettleadam5-10/+10
2020-05-21comms/asterisk15: remove unknown configure option --with-ltdlrillig1-3/+1
This option has been removed in 2018, see ChangeLog.
2020-05-21(comms/obexapp) Build fix: Remove obexapp.1 obexapp.h from SUBST_FILES.pathsmef1-2/+2
2020-05-20mark packages that fail with -Werror=char-subscriptsrillig2-2/+16
These packages are susceptible to bugs when confronted with non-ASCII characters. See https://gcc.gnu.org/bugzilla/show_bug.cgi?id=94182. It takes some time to analyze and fix these individually, therefore they are only marked as "needs work".
2020-05-17g/c references to openjdk7tnn1-2/+2
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