[asterisk-users] Asterisk 13.1.0 Now Available

[Asterisk Development Team]

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:

New Features made in this release:
-----------------------------------
 * ASTERISK-24554 - AMI/ARI: Generate events on connected line
      changes (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
      Corey Farrell)
 * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
      leak (Reported by Corey Farrell)
 * ASTERISK-24430 - missing letter "p" in word response in
      OriginateResponse event documentation (Reported by Dafi Ni)
 * ASTERISK-24437 - Review implementation of ast_bridge_impart for
      leaks and document proper usage (Reported by Scott Griepentrog)
 * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
      Corey Farrell)
 * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
      Corey Farrell)
 * ASTERISK-24458 - chan_phone fails to build on big endian systems
      (Reported by Tzafrir Cohen)
 * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
      (Reported by Olle Johansson)
 * ASTERISK-24304 - asterisk crashing randomly because of unistim
      channel (Reported by dhanapathy sathya)
 * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
      Nick Adams)
 * ASTERISK-24462 - res_pjsip: Stale qualify statistics after
      disablementation (Reported by Kevin Harwell)
 * ASTERISK-24465 - audiohooks list leaks reference to formats
      (Reported by Corey Farrell)
 * ASTERISK-24466 - app_queue: fix a couple leaks to struct
      call_queue (Reported by Corey Farrell)
 * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
      (Reported by Corey Farrell)
 * ASTERISK-24411 - [patch] Status of outbound registration is not
      changed upon unregistering. (Reported by John Bigelow)
 * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)
 * ASTERISK-24480 - res_http_websockets: Module reference decrease
      below zero (Reported by Corey Farrell)
 * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
      audiohook callback (Reported by Corey Farrell)
 * ASTERISK-24487 - configuration: sections should be loadable as
      template even when not marked (Reported by Scott Griepentrog)
 * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
      unescapes semicolons ("\;" -> ";") turning them into comments
      (corruption) on rewrite of a config file (Reported by George
      Joseph)
 * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
      when DNS settings invalid (Reported by Melissa Shepherd)
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)
 * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
      Conkle)
 * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
      extra calls to ast_module_unref (Reported by Corey Farrell)
 * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
      waiting for more matching digits. (Reported by Richard Mudgett)
 * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
      queue caller (Reported by Steve Pitts)
 * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
      (Reported by Corey Farrell)
 * ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
      header fix (Reported by abelbeck)
 * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
      length exceeds 50 (roughly) national symbols (Reported by
      Dmitriy Bubnov)
 * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
      revision r227276 (Reported by Xavier Hienne)
 * ASTERISK-24505 - manager: http connections leak references
      (Reported by Corey Farrell)
 * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
      coverage are enabled (Reported by Corey Farrell)
 * ASTERISK-24444 - PBX: Crash when generating extension for
      pattern matching hint (Reported by Leandro Dardini)
 * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
      packet to JSON for res_hep_rtcp and report blocks are greater
      than 1 (Reported by Gregory Malsack)
 * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
      transfer (Reported by Beppo Mazzucato)
 * 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 (Reported by Matt
      Jordan)
 * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
      (Reported by Leon Rowland)
 * ASTERISK-23651 - Reloading some modules that are loaded already,
      results in 'No such module' before a successful reload (Reported
      by Rusty Newton)
 * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
      endmarked users when last marked user leaves (Reported by Matt
      Jordan)
 * ASTERISK-15242 - transmit_refer leaks sip_refer structures
      (Reported by David Woolley)
 * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
      with "400 bad request" - DEBUG shows "Received a REFER without a
      parseable Refer-To" (Reported by Beppo Mazzucato)
 * ASTERISK-24535 - stringfields: Fix regression from fix for
      unintentional memory retention and another issue exposed by the
      fix (Reported by Corey Farrell)
 * ASTERISK-24471 - Crash - assert_fail in libc in
      pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
      (Reported by yaron nahum)
 * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
      in-dialog with invalid target causes crash (Reported by Joshua
      Colp)
 * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
      module load (Reported by Matt Jordan)
 * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
      allow blocked addresses through (Reported by Matt Jordan)
 * ASTERISK-24542 - [patch]Failure showing codecs via 'core show
      channeltype <tech>' (Reported by snuffy)
 * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
      by xrobau)
 * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
      voicemail under high concurrency with an IMAP backend (Reported
      by David Duncan Ross Palmer)
 * ASTERISK-24572 - [patch]App_meetme is loaded without its
      defaults when the configuration file is missing (Reported by
      Nuno Borges)
 * ASTERISK-24573 - [patch]Out of sync conversation recording when
      divided in multiple recordings (Reported by Nuno Borges)
 * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
      reliably transmitted during transfers (Reported by Matt Jordan)
 * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
      extension to another pjsip extension  (Reported by Abhay Gupta)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
      property 'unanswered' (Reported by Matt Jordan)
 * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
      column in the cel_odbc module (Reported by Etienne Lessard)
 * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
      recordings (Reported by Ben Smithurst)
 * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
      lookups (Reported by Birger "WIMPy" Harzenetter)

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!