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