The Asterisk Development Team has announced the first release candidate of Asterisk 12.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.8.0-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-15242] – transmit_refer leaks sip_refer structures
- [ASTERISK-20127] – [Regression] Config.c config_text_file_load() unescapes semicolons (“\;” -> “;”) turning them into comments (corruption) on rewrite of a config file
- [ASTERISK-23651] – Reloading some modules that are loaded already, results in ‘No such module’ before a successful reload
- [ASTERISK-24257] – agent must dial acceptdtmf twice to bridge to queue caller
- [ASTERISK-24307] – Unintentional memory retention in stringfields
- [ASTERISK-24336] – PJSIP timer_min_se value under 90 causes crash
- [ASTERISK-24438] – res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
- [ASTERISK-24444] – PBX: Crash when generating extension for pattern matching hint
- [ASTERISK-24447] – Bridge DTMF hooks: Audio doesn’t pass when waiting for more matching digits.
- [ASTERISK-24468] – Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
- [ASTERISK-24469] – Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
- [ASTERISK-24471] – Crash – assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
- [ASTERISK-24480] – res_http_websockets: Module reference decrease below zero
- [ASTERISK-24482] – func_talkdetect: Fix stasis message leak in audiohook callback
- [ASTERISK-24487] – configuration: sections should be loadable as template even when not marked
- [ASTERISK-24489] – Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
- [ASTERISK-24491] – Memory leak in res_hep
- [ASTERISK-24492] – main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
- [ASTERISK-24498] – Segmentation fault in res_hep_rtcp on attended transfer
- [ASTERISK-24499] – Need more explicit debug when PJSIP dialstring is invalid
- [ASTERISK-24500] – Regression introduced in chan_mgcp by SVN revision r227276
- [ASTERISK-24501] – ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd
- [ASTERISK-24502] – Build fails when dev-mode, dont optimize and coverage are enabled
- [ASTERISK-24504] – chan_console: Fix reference leaks to pvt
- [ASTERISK-24505] – manager: http connections leak references
- [ASTERISK-24508] – pjsip – REFER request from SNOM is rejected with “400 bad request” – DEBUG shows “Received a REFER without a parseable Refer-To”
- [ASTERISK-24516] – Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
- [ASTERISK-24522] – ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
- [ASTERISK-24528] – res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash
- [ASTERISK-24531] – res_pjsip_acl: ACLs not applied on initial module load
- [ASTERISK-24533] – 2 threads created per chan_sip entry
- [ASTERISK-24535] – stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix
- [ASTERISK-24537] – Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers
- [ASTERISK-24572] – App_meetme is loaded without its defaults when the configuration file is missing
- [ASTERISK-24573] – Out of sync conversation recording when divided in multiple recordings
Improvement
- [ASTERISK-24279] – Documentation: Clarify the behaviour of the CDR property ‘unanswered’
- [ASTERISK-24283] – Microseconds precision in the eventtime column in the cel_odbc module
- [ASTERISK-24530] – app_record stripping 1/4 second from recordings
- [ASTERISK-24577] – Speed up loopback switches by avoiding unneeded lookups
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.8.0-rc1
Thank you for your continued support of Asterisk!