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.
The following are the issues resolved in this release candidate:
- [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
- [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
- [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:
Thank you for your continued support of Asterisk!