The Asterisk Development Team has announced the release of Asterisk 13.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.10.0 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: Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts (Reported by Alexei Gradinari) * ASTERISK-25994 - [patch]res_pjsip: module load priority (Reported by Alexei Gradinari) * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported by Alexei Gradinari) * ASTERISK-25835 - Authentication using 'Username' field from Digest (Reported by Ross Beer) * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime performace (Reported by Alexei Gradinari) Bugs fixed in this release: ----------------------------------- * ASTERISK-26160 - pjsip: Updated->Reachable during qualify (Reported by Matt Jordan) * ASTERISK-26177 - func_odbc: Database handle is kept when it should be released (Reported by Leandro Dardini) * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer) * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details (Reported by Corey Farrell) * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph) * ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim generates a compile error (Reported by George Joseph) * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark Michelson) * ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location (Reported by George Joseph) * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by Alexander Traud) * ASTERISK-26127 - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer (Reported by Joshua Colp) * ASTERISK-26083 - ARI: Announcer channels staying around after playback to a bridge is finished (Reported by Per Jensen) * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported by Alexander Traud) * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '>' (Reported by Vasil Kolev) * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by Alexander Traud) * ASTERISK-25262 - Memory leak when a caller channel does multiple dials and CEL is enabled (Reported by Etienne Lessard) * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels (Reported by Niklas Larsson) * ASTERISK-26096 - res_hep: Crash when configuration file is missing (Reported by Niklas Larsson) * ASTERISK-26089 - Invalid security events during boot using PJSIP Realtime (Reported by Scott Griepentrog) * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by Ross Beer) * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B. Davis) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported by Alexander Traud) * ASTERISK-26070 - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities (Reported by George Joseph) * ASTERISK-26078 - core: Memory leak in logging (Reported by Etienne Lessard) * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered properly (Reported by Ross Beer) * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible (Reported by Private Name) * ASTERISK-25777 - data race in threadpool (Reported by Badalian Vyacheslav) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26029 - parking: ast_parking_park_call should return parking_space instead of parking_exten (Reported by Diederik de Groot) * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero. (Reported by Edwin Vandamme) * ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final response (Reported by Javier Riveros ) * ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown fields (Reported by Joshua Colp) * ASTERISK-24986 - keepalive INFO packages ignored by asterisk (Reported by Ilya Trikoz) * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph) * ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph) * ASTERISK-25964 - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight (Reported by Matt Jordan) * ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined into 1 TCP packet (Reported by Ross Beer) * ASTERISK-25352 - res_hep_rtcp correlation_id is different then res_hep (Reported by Kevin Scott Adams) * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported by Tzafrir Cohen) * ASTERISK-26007 - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon) * ASTERISK-25990 - PJSIP TLS registration should respect client_uri scheme when generating Contact URI (Reported by Sebastian Damm) * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use source port in nonce verification (Reported by Mark Michelson) * ASTERISK-25993 - pjproject: Allow bundling to not require everything it does (Reported by Joshua Colp) * ASTERISK-25956 - Compilation error in conditionally compiled code in config_options.c (Reported by Chris Trobridge) * ASTERISK-25998 - file: Crash when using nativeformats (Reported by Joshua Colp) * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime (Reported by Ross Beer) * ASTERISK-25968 - pjproject_bundled: Configure and make need to be re-tested (Reported by George Joseph) * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload (Reported by John Campbell) * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by Dmitriy Serov) * ASTERISK-25963 - func_odbc requires reconnect checks for stale connections (Reported by Ross Beer) * ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash when running test (Reported by Joshua Colp) * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0 Thank you for your continued support of Asterisk! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-announce mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-announce
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