The Asterisk Development Team has announced the release of Asterisk 13.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a channel (Reported by Matt Jordan) * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid string copy (Reported by Yura Kocyuba) * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur (Reported by Joshua Colp) * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked (Reported by Matt Jordan) * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-24479 - Enable REF_DEBUG for module references (Reported by Corey Farrell) * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information (Reported by Matt Jordan) * ASTERISK-24772 - ODBC error in realtime sippeers when device unregisters under MariaDB (Reported by Richard Miller) * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown (Reported by Richard Mudgett) * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported by Zane Conkle) * ASTERISK-24015 - app_transfer fails with PJSIP channels (Reported by Private Name) * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk transfer scenario. (Reported by Mark Michelson) * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by Niklas Larsson) * ASTERISK-24716 - Improve pjsip log messages for presence subscription failure (Reported by Rusty Newton) * ASTERISK-24612 - res_pjsip: No information if a required sorcery wizard is not loaded (Reported by Joshua Colp) * ASTERISK-24768 - res_timing_pthread: file descriptor leak (Reported by Matthias Urlichs) * ASTERISK-24685 - "pjsip show version" CLI command (Reported by Joshua Colp) * ASTERISK-24632 - install_prereq script installs pjproject without IPv6 support (Reported by Rusty Newton) * ASTERISK-24085 - Documentation - We should remove or further document the 'contact' section in pjsip.conf (Reported by Rusty Newton) * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by JoshE) * ASTERISK-24700 - CRASH: NULL channel is being passed to ast_bridge_transfer_attended() (Reported by Zane Conkle) * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove (Reported by Corey Farrell) * ASTERISK-24799 - [patch] make fails with undefined reference to SSLv3_client_method (Reported by Alexander Traud) * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC Events (Reported by klaus3000) * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn call (Reported by Marcel Manz) * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event (Reported by Panos Gkikakis) * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime (Reported by Graham Barnett) * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers (Reported by Corey Farrell) * ASTERISK-24796 - Codecs and bucket schema's prevent module unload (Reported by Corey Farrell) * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML (Reported by Ashley Sanders) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-24785 - 'Expires' header missing from 200 OK on REGISTER (Reported by Ross Beer) * ASTERISK-24677 - ARI GET variable on channel provides unhelpful response on non-existent variable (Reported by Joshua Colp) * ASTERISK-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * ASTERISK-24812 - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding (Reported by Matt Jordan) * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill (Reported by JoshE) * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime fail (Reported by Terry Wilson) * ASTERISK-23214 - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases (Reported by Rusty Newton) * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller (Reported by Makoto Dei) * ASTERISK-18105 - most of asterisk modules are unbuildable in cygwin environment (Reported by feyfre) * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) * ASTERISK-24751 - Integer values in json payload to ARI cause asterisk to crash (Reported by jeffrey putnam) * ASTERISK-24838 - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling (Reported by Richard Mudgett) * ASTERISK-24825 - Caller ID not recognized using Centrex/Distinctive dialing (Reported by Richard Mudgett) * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT (Reported by Stefan Engstr??m) * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers (Reported by Kevin Harwell) * ASTERISK-24755 - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge (Reported by John Bigelow) * ASTERISK-24739 - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules (Reported by Ed Hynan) * ASTERISK-23390 - NewExten Event with application AGI shows up before and after AGI runs (Reported by Benjamin Keith Ford) * ASTERISK-24786 - [patch] - Asterisk terminates when playing a voicemail stored in LDAP (Reported by Graham Barnett) * ASTERISK-24808 - res_config_odbc: Improper escaping of backslashes occurs with MySQL (Reported by Javier Acosta) * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported by Anatoli) * ASTERISK-20850 - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. (Reported by Diederik de Groot) * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI connection on error (Reported by Dmitriy Serov) * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported by Frank DiGennaro) * ASTERISK-21038 - Bad command completion of "core set debug channel" (Reported by Richard Kenner) * ASTERISK-18708 - func_curl hangs channel under load (Reported by Dave Cabot) * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by Atis Lezdins) * ASTERISK-24876 - Investigate reference leaks from tests/channels/local/local_optimize_away (Reported by Corey Farrell) * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported by Corey Farrell) * ASTERISK-24817 - init_logger_chain: unreachable code block (Reported by Corey Farrell) * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by snuffy) * ASTERISK-24879 - [patch]Compilation fails due to 64bit time under OpenBSD (Reported by snuffy) Improvements made in this release: ----------------------------------- * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes (Reported by Ben Merrills) * ASTERISK-24811 - asterisk-publication sorcery object does not use realtime (Reported by Matt Hoskins) * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context (Reported by Graham Barnett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0 Thank you for your continued support of Asterisk!
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