The Asterisk Development Team would like to announce the first release candidate of Asterisk 14.6.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.6.0-rc1 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 candidate:
Bugs fixed in this release:
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Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) |
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app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) |
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core_local: local channel data not being properly unref’ed and unlocked (Reported by Kevin Harwell) |
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bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) |
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Comment typo format_g729.c (Reported by Matthew Fredrickson) |
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Core/PBX: Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) |
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res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) |
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Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) |
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nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) |
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res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) |
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res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) |
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bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) |
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res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) |
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Asterisk build process fails with flag –with-pjproject-bundled with curl download command and slow network (Reported by alex) |
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chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) |
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chan_pjsip: Flipping between codecs (Reported by Michael Maier) |
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chan_pjsip would send INVITE to ‘Unreachable’ endpoints (Reported by Jacek Konieczny) |
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bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) |
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Background in realtime (Reported by Andrew Nowrot) |
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channel / meetme: Fix missing parentheses (Reported by Joshua Colp) |
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GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) |
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Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) |
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Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) |
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srtp’s crypto_get_random deprecated (Reported by Tzafrir Cohen) |
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AGI – RECORD FILE – documentation doesn’t describe BEEP argument (Reported by Rusty Newton) |
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Async AGI crashes Asterisk when issuing “set variable” command without args (Reported by Antoine Pitrou) |
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Malformed AGI 520 Usage response (Reported by Tony Mountifield) |
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res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) |
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app_queue: Agent not called when caller is parked (Reported by wushumasters) |
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app_queue: Queue member stops being called after AMI “Redirect” action for queues with wrapuptime (Reported by Etienne Lessard) |
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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) |
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app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz) |
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app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec) |
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res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros ) |
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chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) |
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res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) |
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Audit manipulation of channel flags without locks (Reported by Joshua Colp) |
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Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder) |
Information Requests made in this release:
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libsrtp-2.x.x support (Reported by Alex) |
Improvements made in this release:
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res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari) |
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Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi) |
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Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi) |
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audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton) |
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res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0-rc1
Thank you for your continued support of Asterisk!