The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.5.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-26982 <https://issues.asterisk.org/jira/browse/ASTERISK-26982>] - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) - [ASTERISK-26979 <https://issues.asterisk.org/jira/browse/ASTERISK-26979>] - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros ) - [ASTERISK-25665 <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) - [ASTERISK-26998 <https://issues.asterisk.org/jira/browse/ASTERISK-26998>] - res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) - [ASTERISK-26143 <https://issues.asterisk.org/jira/browse/ASTERISK-26143>] - res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) - [ASTERISK-26606 <https://issues.asterisk.org/jira/browse/ASTERISK-26606>] - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) - [ASTERISK-26983 <https://issues.asterisk.org/jira/browse/ASTERISK-26983>] - Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) - [ASTERISK-25032 <https://issues.asterisk.org/jira/browse/ASTERISK-25032>] - [patch]cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) - [ASTERISK-26173 <https://issues.asterisk.org/jira/browse/ASTERISK-26173>] - func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) - [ASTERISK-25506 <https://issues.asterisk.org/jira/browse/ASTERISK-25506>] - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) - [ASTERISK-24529 <https://issues.asterisk.org/jira/browse/ASTERISK-24529>] - Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) - [ASTERISK-26860 <https://issues.asterisk.org/jira/browse/ASTERISK-26860>] - Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) - [ASTERISK-26922 <https://issues.asterisk.org/jira/browse/ASTERISK-26922>] - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) - [ASTERISK-26974 <https://issues.asterisk.org/jira/browse/ASTERISK-26974>] - res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) - [ASTERISK-26908 <https://issues.asterisk.org/jira/browse/ASTERISK-26908>] - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) - [ASTERISK-25823 <https://issues.asterisk.org/jira/browse/ASTERISK-25823>] - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) - [ASTERISK-26926 <https://issues.asterisk.org/jira/browse/ASTERISK-26926>] - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) - [ASTERISK-26951 <https://issues.asterisk.org/jira/browse/ASTERISK-26951>] - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) - [ASTERISK-26930 <https://issues.asterisk.org/jira/browse/ASTERISK-26930>] - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) - [ASTERISK-26929 <https://issues.asterisk.org/jira/browse/ASTERISK-26929>] - pjsip: Add database tables for RLS (Reported by Joshua Colp) - [ASTERISK-26953 <https://issues.asterisk.org/jira/browse/ASTERISK-26953>] - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) - [ASTERISK-26890 <https://issues.asterisk.org/jira/browse/ASTERISK-26890>] - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) - [ASTERISK-26692 <https://issues.asterisk.org/jira/browse/ASTERISK-26692>] - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) - [ASTERISK-26835 <https://issues.asterisk.org/jira/browse/ASTERISK-26835>] - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) - [ASTERISK-26853 <https://issues.asterisk.org/jira/browse/ASTERISK-26853>] - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) - [ASTERISK-26613 <https://issues.asterisk.org/jira/browse/ASTERISK-26613>] - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) - [ASTERISK-26169 <https://issues.asterisk.org/jira/browse/ASTERISK-26169>] - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) - [ASTERISK-21856 <https://issues.asterisk.org/jira/browse/ASTERISK-21856>] - STUN never works when asterisk started without internet access (Reported by Jeremy Kister) - [ASTERISK-20984 <https://issues.asterisk.org/jira/browse/ASTERISK-20984>] - Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) - [ASTERISK-26528 <https://issues.asterisk.org/jira/browse/ASTERISK-26528>] - [UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) - [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>] - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) - [ASTERISK-26903 <https://issues.asterisk.org/jira/browse/ASTERISK-26903>] - Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) - [ASTERISK-26928 <https://issues.asterisk.org/jira/browse/ASTERISK-26928>] - pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) - [ASTERISK-26927 <https://issues.asterisk.org/jira/browse/ASTERISK-26927>] - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) - [ASTERISK-26905 <https://issues.asterisk.org/jira/browse/ASTERISK-26905>] - pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) - [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>] - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) - [ASTERISK-25974 <https://issues.asterisk.org/jira/browse/ASTERISK-25974>] - Unused realtime MOH classes not purged on 'moh reload' (Reported by Sébastien Couture) - [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>] - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) - [ASTERISK-21721 <https://issues.asterisk.org/jira/browse/ASTERISK-21721>] - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) - [ASTERISK-26915 <https://issues.asterisk.org/jira/browse/ASTERISK-26915>] - chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) - [ASTERISK-26363 <https://issues.asterisk.org/jira/browse/ASTERISK-26363>] - res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) - [ASTERISK-26896 <https://issues.asterisk.org/jira/browse/ASTERISK-26896>] - Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) - [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>] - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) - [ASTERISK-21009 <https://issues.asterisk.org/jira/browse/ASTERISK-21009>] - xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) - [ASTERISK-25490 <https://issues.asterisk.org/jira/browse/ASTERISK-25490>] - [patch]SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) - [ASTERISK-26086 <https://issues.asterisk.org/jira/browse/ASTERISK-26086>] - res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) - [ASTERISK-23996 <https://issues.asterisk.org/jira/browse/ASTERISK-23996>] - No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) - [ASTERISK-24712 <https://issues.asterisk.org/jira/browse/ASTERISK-24712>] - xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) - [ASTERISK-26814 <https://issues.asterisk.org/jira/browse/ASTERISK-26814>] - pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) - [ASTERISK-23510 <https://issues.asterisk.org/jira/browse/ASTERISK-23510>] - JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) - [ASTERISK-21855 <https://issues.asterisk.org/jira/browse/ASTERISK-21855>] - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available (Reported by Jeremy Kister) - [ASTERISK-25622 <https://issues.asterisk.org/jira/browse/ASTERISK-25622>] - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) - [ASTERISK-26515 <https://issues.asterisk.org/jira/browse/ASTERISK-26515>] - rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) - [ASTERISK-26818 <https://issues.asterisk.org/jira/browse/ASTERISK-26818>] - cdr: Problem setting variables in h exten (Reported by scgm11) - [ASTERISK-26875 <https://issues.asterisk.org/jira/browse/ASTERISK-26875>] - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26088 <https://issues.asterisk.org/jira/browse/ASTERISK-26088>] - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) - [ASTERISK-26427 <https://issues.asterisk.org/jira/browse/ASTERISK-26427>] - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0 *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
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