Asterisk 13.3.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.3.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0-rc1 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 candidate:

Bug

  • [ASTERISK-15434] – When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] – Cannot disallow unknown format ”
  • [ASTERISK-17721] – Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] – most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] – func_curl hangs channel under load
  • [ASTERISK-19470] – Documentation on app_amd is incorrect
  • [ASTERISK-20850] – Nested functions aren’t portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] – Bad command completion of “core set debug channel”
  • [ASTERISK-22670] – Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-23214] – chan_sip WARNING message ‘We are requesting SRTP for audio, but they responded without it’ is ambiguous and wrong in some cases
  • [ASTERISK-23390] – NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24015] – app_transfer fails with PJSIP channels
  • [ASTERISK-24085] – Documentation – We should remove or further document the ‘contact’ section in pjsip.conf
  • [ASTERISK-24451] – chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] – Enable REF_DEBUG for module references
  • [ASTERISK-24499] – Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24612] – res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24616] – Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24632] – install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24677] – ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24685] – “pjsip show version” CLI command
  • [ASTERISK-24689] – Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24700] – CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] – Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24716] – Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24724] – ‘httpstatus’ Web Page Produces Incomplete HTML
  • [ASTERISK-24727] – PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24739] – – Out of files — call fails — numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] – Segmentation fault on aoc-e event
  • [ASTERISK-24741] – dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] – Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] – res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24751] – Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] – Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] – Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] – res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] – res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] – ${CHANNEL(pjsip)} – segfault
  • [ASTERISK-24772] – ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24785] – ‘Expires’ header missing from 200 OK on REGISTER
  • [ASTERISK-24786] – – Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] – – Microsoft exchange incompatibility for playing back messages stored in IMAP – play_message: No origtime
  • [ASTERISK-24791] – Crash in ast_rtcp_write_report
  • [ASTERISK-24796] – Codecs and bucket schema’s prevent module unload
  • [ASTERISK-24797] – bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] – make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] – Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24807] – Missing mandatory field Max-Forwards
  • [ASTERISK-24808] – res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] – ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] – asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] – init_logger_chain: unreachable code block
  • [ASTERISK-24825] – Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] – Fix Frame Leaks
  • [ASTERISK-24830] – res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24838] – chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] – res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24872] – AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] – Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] – Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] – Compilation under OpenBSD
  • [ASTERISK-24882] – chan_sip: Improve usage of REF_DEBUG

Improvement

  • [ASTERISK-24745] – Add no_answer to ARI hangup causes
  • [ASTERISK-24790] – Reduce spurious noise in logs from voicemail – Couldn’t find mailbox %s in context
  • [ASTERISK-24811] – asterisk-publication sorcery object does not use realtime

New Feature

  • [ASTERISK-17899] – Adds a ‘ignorecryptolifetime’ (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional ‘lifetime’
  • [ASTERISK-24703] – ARI: Add the ability to “transfer” (redirect) a channel

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0-rc1


Thank you for your continued support of Asterisk!

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