Asterisk 13.1.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.1.0-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-21721] – SIP Failed to parse multiple Supported: headers
• [ASTERISK-23651] – Reloading some modules that are loaded already, results in ‘No such module’ before a successful reload
• [ASTERISK-24190] – IMAP voicemail causes segfault
• [ASTERISK-24250] – Voicemail with multi-recipients To: header fix
• [ASTERISK-24257] – agent must dial acceptdtmf twice to bridge to queue caller
• [ASTERISK-24304] – asterisk crashing randomly because of unistim channel
• [ASTERISK-24307] – Unintentional memory retention in stringfields
• [ASTERISK-24336] – PJSIP timer_min_se value under 90 causes crash
• [ASTERISK-24411] – Status of outbound registration is not changed upon unregistering.
• [ASTERISK-24430] – missing letter “p” in word response in OriginateResponse event documentation
• [ASTERISK-24432] – Install refcounter.py when REF_DEBUG is enabled
• [ASTERISK-24436] – Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
• [ASTERISK-24437] – Review implementation of ast_bridge_impart for leaks and document proper usage
• [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-24453] – manager: acl_change_sub leaks
• [ASTERISK-24454] – app_queue: ao2_iterator not destroyed, causing leak
• [ASTERISK-24455] – func_cdr: CDR_PROP leaks payload
• [ASTERISK-24457] – res_fax: fax gateway frames leak
• [ASTERISK-24458] – chan_phone fails to build on big endian systems
• [ASTERISK-24462] – res_pjsip: Stale qualify statistics after disablementation
• [ASTERISK-24465] – audiohooks list leaks reference to formats
• [ASTERISK-24466] – app_queue: fix a couple leaks to struct call_queue
• [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-24476] – main/app.c / app_voicemail: ast_writestream leaks
• [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-24542] – Failure showing codecs via ‘core show channeltype <tech>’
• [ASTERISK-24556] – Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension
• [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

New Feature

• [ASTERISK-24554] – AMI/ARI: Generate events on connected line changes

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0-rc1

Thank you for your continued support of Asterisk!