The Asterisk Development Team has announced the first release candidate of Asterisk 13.13.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.13.0-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: 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 Mudgett) * 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 Joseph) * 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 Colp) * 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 Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * 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 Gilmour) * 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-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) 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 candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0-rc1 Thank you for your continued support of Asterisk! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev