Asterisk 14.2.0-rc1 Now Available

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

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

The release of Asterisk 14.2.0-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-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-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-26444] – ‘features show’ command in CLI does not return prompt.
  • [ASTERISK-26457] – force_rport,auto_comedia: No NAT detection triggered.
  • [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

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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0-rc1

Thank you for your continued support of Asterisk!

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