Asterisk 14.2.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 14.2.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.2.0 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:

Bug

  • [ASTERISK-22480] – Embedded pjproject: build.mak contains hardcoded full path to version.mak
  • [ASTERISK-24274] – Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
  • [ASTERISK-24400] – ooh323 sends wrong hangup code
  • [ASTERISK-25070] – Fix FTBFS on Hurd
  • [ASTERISK-26307] – res_pjsip_caller_id: Crash on outgoing change
  • [ASTERISK-26309] – res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
  • [ASTERISK-26311] – rtp_engine: Allow more than 32 dynamic payload types.
  • [ASTERISK-26343] – ASTERISK-25951 causes issues for callerid manipulation through agi
  • [ASTERISK-26344] – Asterisk 13.11.0 + PJSIP crash
  • [ASTERISK-26356] – menuselect: invalid test for GTK2
  • [ASTERISK-26387] – Asterisk segfaults shortly after starting even with no active calls.
  • [ASTERISK-26412] – build: Prepare for gcc 6.2
  • [ASTERISK-26421] – Segmentation Fault with ARI originate into mixing bridge with 43 clients
  • [ASTERISK-26423] – res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
  • [ASTERISK-26439] – chan_rtp: Crash when originating
  • [ASTERISK-26444] – ‘features show’ command in CLI does not return prompt.
  • [ASTERISK-26457] – force_rport,auto_comedia: No NAT detection triggered.
  • [ASTERISK-26462] – app_queue: While using queues with realtime, setting back to an empty context doesn’t stop the exit key usage
  • [ASTERISK-26468] – ari: Bridge events stop working after this sequence of ARI calls
  • [ASTERISK-26476] – chan_sip: Incorrect display option “Outbound reg. retry 403” in “sip show settings”
  • [ASTERISK-26480] – CLI: core set debug: Auto-completes File not Module
  • [ASTERISK-26482] – chan_pjsip: segfault on already disconnected session
  • [ASTERISK-26503] – app_voicemail: Asterisk crashes when MailboxExists is used
  • [ASTERISK-26506] – res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
  • [ASTERISK-26509] – A few non-critical deprecation warnings when building on Ubuntu 16.10
  • [ASTERISK-26510] – pjproject_bundled uses the –strip-components option of tar which isn’t supported in older versions
  • [ASTERISK-26513] – tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
  • [ASTERISK-26514] – Super Awesome Company: Don’t specify transport in pjsip.conf
  • [ASTERISK-26516] – pjsip: Memory corruption with possible memory leak.
  • [ASTERISK-26520] – codec_opus: Generated fmtp line has no content
  • [ASTERISK-26523] – chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes – rtptimeout behaving badly – regression
  • [ASTERISK-26524] – astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
  • [ASTERISK-26526] – [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
  • [ASTERISK-26537] – AMI: NewConnectedLine event is not documented
  • [ASTERISK-26541] – res_pjsip_sdp_rtp: Restrict number of formats to maximum
  • [ASTERISK-26549] – app_dial: When PickupChan() is used some channels may have incorrect device state
  • [ASTERISK-26555] – Multi-party Video: Fix some post Asterisk-11 regressions
  • [ASTERISK-26556] – manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
  • [ASTERISK-26565] – chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
  • [ASTERISK-26571] – res_pjsip: Resolution incorrect when explicit IPv6 transport configured
  • [ASTERISK-26575] – testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
  • [ASTERISK-26592] – Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
  • [ASTERISK-26605] – codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
  • [ASTERISK-26608] – Compile and link failures on OpenBSD
  • [ASTERISK-26618] – build: Backport addition of librt check to configure.ac

Improvement

  • [ASTERISK-26176] – chan_sip: Add AccountCode to AMI PeerEntry
  • [ASTERISK-26418] – res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
  • [ASTERISK-26488] – ARI: Add ‘ari show app’, ‘ari show apps’, and ‘ari set debug’ CLI commands
  • [ASTERISK-26538] – codec_opus: Add sample to configs/samples/codecs.conf.sample
  • [ASTERISK-26558] – app_queue: add variable to know if the call is not answered after a queue

New Feature

  • [ASTERISK-26470] – ARI: Add an ‘asterisk_id’ field to outgoing events
  • [ASTERISK-26492] – ARI: Add ability to specify channel variables on websocket events
  • [ASTERISK-26595] – ARI: Add the ability to control the source of video in a multi-party mixing bridge

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0

Thank you for your continued support of Asterisk!

Scroll to Top