The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.6.0 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-27108 <https://issues.asterisk.org/jira/browse/ASTERISK-27108>] - Crash using 'data get' CLI command (Reported by Sean Bright) - [ASTERISK-27106 <https://issues.asterisk.org/jira/browse/ASTERISK-27106>] - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) - [ASTERISK-27100 <https://issues.asterisk.org/jira/browse/ASTERISK-27100>] - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) - [ASTERISK-27090 <https://issues.asterisk.org/jira/browse/ASTERISK-27090>] - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) - [ASTERISK-25665 <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) - [ASTERISK-27065 <https://issues.asterisk.org/jira/browse/ASTERISK-27065>] - call hangup after leaving app_queue (Reported by Marek Cervenka) - [ASTERISK-26978 <https://issues.asterisk.org/jira/browse/ASTERISK-26978>] - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) - [ASTERISK-24052 <https://issues.asterisk.org/jira/browse/ASTERISK-24052>] - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) - [ASTERISK-27074 <https://issues.asterisk.org/jira/browse/ASTERISK-27074>] - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell) - [ASTERISK-27075 <https://issues.asterisk.org/jira/browse/ASTERISK-27075>] - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) - [ASTERISK-27060 <https://issues.asterisk.org/jira/browse/ASTERISK-27060>] - Comment typo format_g729.c (Reported by Matthew Fredrickson) - [ASTERISK-27041 <https://issues.asterisk.org/jira/browse/ASTERISK-27041>] - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) - [ASTERISK-27026 <https://issues.asterisk.org/jira/browse/ASTERISK-27026>] - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) - [ASTERISK-27057 <https://issues.asterisk.org/jira/browse/ASTERISK-27057>] - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) - [ASTERISK-27024 <https://issues.asterisk.org/jira/browse/ASTERISK-27024>] - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) - [ASTERISK-27046 <https://issues.asterisk.org/jira/browse/ASTERISK-27046>] - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) - [ASTERISK-27022 <https://issues.asterisk.org/jira/browse/ASTERISK-27022>] - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) - [ASTERISK-26923 <https://issues.asterisk.org/jira/browse/ASTERISK-26923>] - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) - [ASTERISK-27053 <https://issues.asterisk.org/jira/browse/ASTERISK-27053>] - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) - [ASTERISK-27052 <https://issues.asterisk.org/jira/browse/ASTERISK-27052>] - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex) - [ASTERISK-27039 <https://issues.asterisk.org/jira/browse/ASTERISK-27039>] - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) - [ASTERISK-26996 <https://issues.asterisk.org/jira/browse/ASTERISK-26996>] - chan_pjsip: Flipping between codecs (Reported by Michael Maier) - [ASTERISK-26281 <https://issues.asterisk.org/jira/browse/ASTERISK-26281>] - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) - [ASTERISK-26973 <https://issues.asterisk.org/jira/browse/ASTERISK-26973>] - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) - [ASTERISK-19291 <https://issues.asterisk.org/jira/browse/ASTERISK-19291>] - Background in realtime (Reported by Andrew Nowrot) - [ASTERISK-27025 <https://issues.asterisk.org/jira/browse/ASTERISK-27025>] - channel / meetme: Fix missing parentheses (Reported by Joshua Colp) - [ASTERISK-27021 <https://issues.asterisk.org/jira/browse/ASTERISK-27021>] - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) - [ASTERISK-24858 <https://issues.asterisk.org/jira/browse/ASTERISK-24858>] - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) - [ASTERISK-23951 <https://issues.asterisk.org/jira/browse/ASTERISK-23951>] - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) - [ASTERISK-25294 <https://issues.asterisk.org/jira/browse/ASTERISK-25294>] - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen) - [ASTERISK-23839 <https://issues.asterisk.org/jira/browse/ASTERISK-23839>] - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton) - [ASTERISK-22432 <https://issues.asterisk.org/jira/browse/ASTERISK-22432>] - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou) - [ASTERISK-25662 <https://issues.asterisk.org/jira/browse/ASTERISK-25662>] - Malformed AGI 520 Usage response (Reported by Tony Mountifield) - [ASTERISK-27008 <https://issues.asterisk.org/jira/browse/ASTERISK-27008>] - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) - [ASTERISK-26399 <https://issues.asterisk.org/jira/browse/ASTERISK-26399>] - app_queue: Agent not called when caller is parked (Reported by wushumasters) - [ASTERISK-26400 <https://issues.asterisk.org/jira/browse/ASTERISK-26400>] - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard) - [ASTERISK-26715 <https://issues.asterisk.org/jira/browse/ASTERISK-26715>] - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert) - [ASTERISK-26975 <https://issues.asterisk.org/jira/browse/ASTERISK-26975>] - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz) - [ASTERISK-27012 <https://issues.asterisk.org/jira/browse/ASTERISK-27012>] - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec) - [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-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-26964 <https://issues.asterisk.org/jira/browse/ASTERISK-26964>] - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) - [ASTERISK-26789 <https://issues.asterisk.org/jira/browse/ASTERISK-26789>] - Audit manipulation of channel flags without locks (Reported by Joshua Colp) - [ASTERISK-26333 <https://issues.asterisk.org/jira/browse/ASTERISK-26333>] - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26230 <https://issues.asterisk.org/jira/browse/ASTERISK-26230>] - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari) - [ASTERISK-27043 <https://issues.asterisk.org/jira/browse/ASTERISK-27043>] - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi) - [ASTERISK-27042 <https://issues.asterisk.org/jira/browse/ASTERISK-27042>] - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi) - [ASTERISK-26419 <https://issues.asterisk.org/jira/browse/ASTERISK-26419>] - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton) - [ASTERISK-26976 <https://issues.asterisk.org/jira/browse/ASTERISK-26976>] - libsrtp-2.x.x support (Reported by Alex) - [ASTERISK-26124 <https://issues.asterisk.org/jira/browse/ASTERISK-26124>] - res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0 *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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