Rilasciato Asterisk 13.17.0

Il giorno 12 giugno 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.17.0.

Dal post originale:

The following issues are resolved in this release:

Bugs fixed in this release:
* ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer)
* ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet)
* ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan)
* 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 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield)
* ASTERISK-25101 - DTLS configuration can not be specified in the general section - documentation (Reported by Ben Langfeld)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard)
* 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 - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros )
* 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 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without locks (Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder)

Improvements made in this release:
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton)
* ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex)
* 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:

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