The Asterisk Development Team would like to announce the release of Asterisk 17.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.1.0 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:
Security bugs fixed in this release:
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chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) |
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Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons) |
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res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) |
Improvements made in this release:
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res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) |
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Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) |
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Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) |
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Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) |
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add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle) |
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Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair) |
Bugs fixed in this release:
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Memory Leak in res_rtp_asterisk.c (Reported by Ted G) |
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app_meetme, chan_ooh323 and cdr_mysql don’t build on 17.0.0 (Reported by George Joseph) |
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res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) |
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res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer) |
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Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) |
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chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL) |
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res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt) |
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res_parking: Doesn’t park when parkee and parker are the same (Reported by Ross Beer) |
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Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed) |
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res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell) |
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app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) |
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chan_dahdi: PRI span status may stay “Down, Active” after a short alarm (Reported by Frederic LE FOLL) |
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res_rtp_asterisk: ICE Completion Crash when sent packet length doesn’t match (Reported by Joshua Elson) |
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FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) |
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bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) |
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parking: Deadlock when multi call parking (Reported by Joshua C. Colp) |
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ARI causes STASIS Deadlock (Reported by Ross Beer) |
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Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha) |
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ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell) |
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utils.c throws repeated warnings; “pthread_attr_setstacksize: Invalid argument” (Reported by Speed Dial Dave) |
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race condition on pjsip channelstats command (Reported by Salah Ahmed) |
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cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder) |
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MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) |
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pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) |
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Asterisk Deadlocks (Reported by Aheliotech) |
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chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) |
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res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) |
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CDR backend unload problem during active call(s) (Reported by Marian Piater) |
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stasis.c: Crash during unload (Reported by Kevin Harwell) |
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Wrong contact representation in ipv6 mode (Reported by Jørgen H) |
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Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) |
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res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) |
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pjsip: Memory Leak (Reported by Mark) |
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Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) |
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chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) |
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Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) |
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setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) |
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ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) |
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chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) |
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codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) |
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translate: Crash when frame does not have a “src” field set (Reported by Gregory Massel) |
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chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) |
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pjsip mwi: n+1 sip notify’s sent on re-register (Reported by Chris Savinovich) |
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PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) |
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app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) |
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compile menuselect on gentoo (Reported by Kilburn) |
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Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) |
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cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) |
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json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) |
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res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) |
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packet lost on UDPTL wrap around (Reported by Torrey Searle) |
New Features made in this release:
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app_senddtmf: Allow “receiving” DTMF with PlayDTMF instead of only “sending” (Reported by lvl) |
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func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) |
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func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) |
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Unregister a realtime moh class (Reported by Byron Clark) |
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Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
Thank you for your continued support of Asterisk!
