[asterisk-users] Asterisk 13.13.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Nov 23 10:59:51 CST 2016

The Asterisk Development Team has announced the release of Asterisk 13.13.0.
This release is available for immediate download at

The release of Asterisk 13.13.0 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-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      events (Reported by Matt Jordan)

Bugs fixed in this release:
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no content
      (Reported by scgm11)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded. (Reported by Richard
 * ASTERISK-26516 - pjsip: Memory corruption with possible memory
      leak. (Reported by Richard Mudgett)
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage (Reported by George Joseph)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
      temporarily locks up set (Reported by Jason)
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded. (Reported by Joshua Colp)
 * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
      Dmitry Melekhov)
 * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
      regressions (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2 (Reported by George
 * ASTERISK-26509 - A few non-critical deprecation warnings when
      building on Ubuntu 16.10 (Reported by Jonathan Harris)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)
 * ASTERISK-26468 - ari: Bridge events stop working after this
      sequence of ARI calls (Reported by Daniele Pallastrelli)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state (Reported by Joshua
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings" (Reported by Sergey
 * ASTERISK-26537 - AMI: NewConnectedLine event is not documented
      (Reported by Etienne Lessard)
 * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
      argument 2 to memcpy (Reported by Badalian Vyacheslav)
 * ASTERISK-26524 - astobj2: data_size variable is wasted space
      when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
 * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
 * ASTERISK-26387 - Asterisk segfaults shortly after starting even
      with no active calls.  (Reported by Harley Peters)
 * ASTERISK-26514 - Super Awesome Company: Don't specify transport
      in pjsip.conf (Reported by Rusty Newton)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance (Reported by Joshua Colp)
 * ASTERISK-26510 - pjproject_bundled uses the --strip-components
      option of tar which isn't supported in older versions (Reported
      by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak (Reported by Matt Jordan)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
      (Reported by Bill Brigden)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)
 * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
      cause audio loss and wonkiness (Reported by Andreas Wetzel)
 * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
      installations. (Reported by Alexander Traud)
 * ASTERISK-26421 - Segmentation Fault with ARI originate into
      mixing bridge with 43 clients (Reported by Andrew Nagy)
 * ASTERISK-26444 - 'features show' command in CLI does not return
      prompt. (Reported by John Kiniston)
 * ASTERISK-26482 - [patch] chan_pjsip: segfault on already
      disconnected session (Reported by Alexei Gradinari)
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage (Reported by Leandro Dardini)
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)
 * ASTERISK-26618 - build: Backport addition of librt check to
      configure.ac (Reported by Kevin Harwell)

Improvements made in this release:
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
      support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * ASTERISK-26558 - app_queue: add variable to know if the call is
      not answered after a queue (Reported by scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by scgm11)
 * ASTERISK-26538 - codec_opus: Add sample to
      configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
      'ari set debug' CLI commands (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
      blacklisting host subnets that are not involved in RTP (Reported
      by Michael Walton)

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


Thank you for your continued support of Asterisk!

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