Asterisk 13.1.0 Now Available

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

The release of Asterisk 13.1.0 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-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-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, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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