Asterisk Development Team
2019-Dec-23 22:55 UTC
[asterisk-users] Asterisk 17.1.0 Now Available
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:
-----------------------------------
* ASTERISK-28589 - chan_sip: Depending on configuration an
INVITE can alter Addr of a peer
(Reported by Andrey V.
T.)
* ASTERISK-28580 - Bypass SYSTEM write permission in manager
action allows system commands execution
(Reported by Eliel
Sarda��ons)
* ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
declined stream causes crash
(Reported by Alexei
Gradinari)
Improvements made in this release:
-----------------------------------
* ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
retries reached
(Reported by Daniel)
* ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
(Reported by Sam Banks)
* ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
backend when format differs from attachfmt column
(Reported by cmaj)
* ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
clear out any .lock files in the voice mail directory on
startup.
(Reported by Michael)
* ASTERISK-28542 - [patch] add the ability for asterisk to
generate on-hold re-invites
(Reported by Torrey Searle)
* ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
codec
(Reported by Florian Floimair)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
non-existent media stream if codecs create additional streams
and offer does not have them
(Reported by nappsoft)
* ASTERISK-28641 - res_pjsip Segfaults when realtime
configuration to an AOR points to a not existent AOR
(Reported by Ross Beer)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry
exception
(Reported by Walter Doekes)
* ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
compatibility check failure when negociated ptime is not default
ptime.
(Reported by Frederic LE FOLL)
* ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
UTF-8 string on hangup when TEST_FRAMEWORK enabled
(Reported by Bernhard Schmidt)
* ASTERISK-28631 - res_parking: Doesn't park when parkee and
parker are the same
(Reported by Ross Beer)
* ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok
received
(Reported by Salah Ahmed)
* ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
failover
(Reported by Kevin Harwell)
* ASTERISK-28608 - app_amd: Use time calculation to calculate
timeout
(Reported by Michael Cargile)
* ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
Active" after a short alarm
(Reported by Frederic LE FOLL)
* ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
sent packet length doesn't match
(Reported by Joshua
Elson)
* ASTERISK-26481 - FILE function grabs garbage along with read
data when target line has no newline
(Reported by Jonathan
Harris)
* ASTERISK-28618 - bridge_softmix: hold not cleared when
joining a softmix bridge
(Reported by Kevin Harwell)
* ASTERISK-28616 - parking: Deadlock when multi call parking
(Reported by Joshua C. Colp)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported
by Ross Beer)
* ASTERISK-28572 - Memory leaks in res_calendar_exchange and
res_calendar_icalendar
(Reported by Yoooooo Ha)
* ASTERISK-28585 - ari/resource_events: Crash in event session
cleanup
(Reported by Kevin Harwell)
* ASTERISK-28590 - utils.c throws repeated warnings;
"pthread_attr_setstacksize: Invalid argument"
(Reported by
Speed Dial Dave)
* ASTERISK-28578 - race condition on pjsip channelstats
command
(Reported by Salah Ahmed)
* ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
removed) column
(Reported by Christoph Moench-Tegeder)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
16.5
(Reported by Niklas Larsson)
* ASTERISK-28561 - Asterisk Deadlocks
(Reported by
Aheliotech)
* ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
over AMI
(Reported by Jeremiah Gadd)
* ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
unsolicited_mwi container
(Reported by Kevin Harwell)
* ASTERISK-28566 - CDR backend unload problem during active
call(s)
(Reported by Marian Piater)
* ASTERISK-28553 - stasis.c: Crash during unload
(Reported by Kevin Harwell)
* ASTERISK-28544 - Wrong contact representation in ipv6 mode
(Reported by J��rgen H)
* ASTERISK-28534 - Segmentation fault when there is no priority
for an extension
(Reported by Timothy Vanderaerden)
* ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
is configured
(Reported by Juan Martin)
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported
by Cyril Rami��re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on
FreeBSD
(Reported by Guido Falsi)
* ASTERISK-23756 - setvar directive when used in template and a
child of said template, results in duplicate variable names
(Reported by Michael Goryainov)
* ASTERISK-28527 - ChanIsAvail() creates a CDR if
unanswered=yes is set in cdr.conf
(Reported by Frederic LE
FOLL)
* ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
PRI channel hangs up
(Reported by Frederic LE FOLL)
* ASTERISK-28511 - codec_resample: Bad sound quality when up
sampling from SLIN16 to SLIN32
(Reported by Ruddy G)
* ASTERISK-28499 - translate: Crash when frame does not have a
"src" field set
(Reported by Gregory Massel)
* ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
type not at end of a struct
(Reported by Alexander Traud)
* ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
re-register
(Reported by Chris Savinovich)
* ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
characters, NEC only supports up to 32 characters
(Reported by Dan Cropp)
* ASTERISK-28505 - app_voicemail/IMAP: segfault in
leave_voicemail because not checking mailstream
(Reported
by Alexei Gradinari)
* ASTERISK-28487 - compile menuselect on gentoo
(Reported
by Kilburn)
* ASTERISK-28472 - Asterisk occasionally passes a NULL as
srtp->session to srtp_protect/unprotect causing SEGV
(Reported by Jonas Swiatek)
* ASTERISK-28498 - cel / cdr: Event times may be incorrect
(Reported by Joshua C. Colp)
* ASTERISK-28480 - json integer overflow in ssrc and timestamp
(Reported by Salah Ahmed)
* ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
entries
(Reported by Ian Jones)
* ASTERISK-28483 - packet lost on UDPTL wrap around
(Reported by Torrey Searle)
New Features made in this release:
-----------------------------------
* ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
PlayDTMF instead of only "sending"
(Reported by lvl)
* ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
header
(Reported by Martin Tomec)
* ASTERISK-28533 - func_jitterbuffer: Add support for video
synchronization
(Reported by Joshua C. Colp)
* ASTERISK-17808 - [patch] Unregister a realtime moh class
(Reported by Byron Clark)
* ASTERISK-28489 - 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!
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