The Asterisk Development Team would like to announce the first release candidate of Asterisk 15.2.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.2.0-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:
New Features made in this release:
———————————–
|
PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. (Reported by Richard Mudgett) |
|
|
Add cache_media_frames debugging option. (Reported by Richard Mudgett) |
|
|
res_pjsip: No mechanism exists to limit endpoint identification to IP only (Reported by Ben Merrills) |
Bugs fixed in this release:
———————————–
|
Asterisk Hangs with Bad file descriptor on read() (Reported by Abhay Gupta) |
|
|
AMI bridge of channels results in MOH not destroyed and robotic audio on one channel (Reported by Zane Conkle) |
|
|
DNS: Unexpected rr_type can cause crash (Reported by Corey Farrell) |
|
|
chan_console: ‘set active’ fails to work (Reported by Tzafrir Cohen) |
|
|
ConfBridge sound_muted does not work from CLI or AMI (Reported by Thomas Frederiksen) |
|
|
Transfer application does not work with Local channels – documentation misleading (Reported by Ivan Ullmann) |
|
|
chan_sip: “rejected because extension not found” should be logged as a security event (Reported by Brian J. Murrell) |
|
|
Strictrtp has issues to qualify video rtp streams (Reported by Wim De Vlaminck) |
|
|
Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) |
|
|
Coverity Report: Fix issues for error type CHAR_IO (Reported by Matt Jordan) |
|
|
iax.conf demo peer is invalid (Reported by Tzafrir Cohen) |
|
|
README refers to security documents that do not exist. (Reported by Corey Farrell) |
|
|
“core set verbose” behaves strangely, can’t alias it, cli.conf example broken (Reported by Tim Ringenbach at Asteria Solutions Group) |
|
|
crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) |
|
|
res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) |
|
|
Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) |
|
|
Queue members with hints for state_interface get stuck in “In Use” state. (Reported by Steven T. Wheeler) |
|
|
chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) |
|
|
codec_opus requires libcurl (Reported by Samuel For) |
|
|
pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) |
|
|
CLI Completion Not Working (Reported by Ross Beer) |
|
|
CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=… (Reported by Richard Mudgett) |
|
|
RTP: Blind transfer direct media scenario results in one way audio. (Reported by Richard Mudgett) |
|
|
SIP ICE support – remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message (Reported by Roy) |
|
|
pjsip: Clean up WebRTC disables (Reported by abelbeck) |
|
|
Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests (Reported by George Joseph) |
|
|
res_http_post: Don’t require GMIME_MAJOR_VERSION (Reported by Joshua Colp) |
|
|
Transcoding makes bad choice in high-rate translations (Reported by Richard Kenner) |
|
|
ARI: Updating a bridge gives wrong error message. (Reported by Frank Durden) |
|
|
column and row headers for Signed Linear format variants in output of ‘core show translation’ are ambiguous (Reported by Rusty Newton) |
|
|
H323 audio starts with a delay of 2 seconds. (Reported by Marco Giordani) |
|
|
pjsip: 183 without To tag does not negotiate media (Reported by Kevin Harwell) |
|
|
ICE: server-reflexive candidates (srflx) with Dual-Stack. (Reported by Alexander Traud) |
|
|
chan_sip/ICE: Square brackets around IPv6 addresses. (Reported by Alexander Traud) |
|
|
configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) |
|
|
Asterisk fails to configure on MacOS Sierra (Reported by Ivan Larionov) |
|
|
Asterisk fails to build when openssl headers are not installed. (Reported by Corey Farrell) |
|
|
RTP source learning not working with devices that have some clock issues (Reported by nappsoft) |
|
|
Attended transfer crashes in Asterisk 13.17.2 (Reported by Alessandro Pimenta) |
|
|
Bridging: Crash freeing a frame that’s already been freed (Reported by Richard Kenner) |
|
|
core: Audiohook freeing interpolated frame when it shouldn’t. (Reported by Mikhail) |
|
|
app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) |
|
|
res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in “source ip address” and “destination ip address” fields in HEP packets (Reported by Max Norba) |
|
|
res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) (Reported by Vasilii Rogin) |
|
|
asterisk.conf: Setting astctl without setting astrundir is ineffective. (Reported by Corey Farrell) |
|
|
pjsip: TCP connections may not be destroyed (Reported by Joshua Colp) |
|
|
res_pjsip_session: RTP instances leak on 488 responses. (Reported by Corey Farrell) |
|
|
chan_sip: Security vulnerability with client code header (revisited) (Reported by Richard Mudgett) |
|
|
(Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines (Reported by Kim youngsung) |
|
|
Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) |
|
|
res_pjsip: Crash occurs when an empty contact read from astdb or database (Reported by Aaron An) |
|
|
res_pjsip: PIDF contact field has malformed/invalid XML (Reported by basildane) |
|
|
res_pjsip: TLS options do not handle empty values (Reported by seanchann.zhou) |
|
|
srtp: Add support for ephemeral DTLS certificates (Reported by Sean Bright) |
|
|
format_ogg_opus: remove from source (Reported by Kevin Harwell) |
|
|
tcptls: Print notice when TLS is enabled but not configured. (Reported by Alexander Traud) |
|
|
libsrtp-2.x.x + AES-GCM support (Reported by Alexander Traud) |
|
|
Modules: Fix issues with CLI completion. (Reported by Corey Farrell) |
|
|
Regression: pjsip 13.18.0 – from_user – “+” character isn’t allowed any more (Reported by Michael Maier) |
|
|
channel: Crash when fax gateway is in use with PJSIP (Reported by Jared Hull) |
|
|
Audit menuselect module dependencies (Reported by Corey Farrell) |
|
|
Optional API modules should not allow unload. (Reported by Corey Farrell) |
|
|
Bridge() dialplan application fails without setting BRIDGERESULT channel variable (Reported by James Terhune) |
|
|
res_ari_channels: channel_state_invalid always leaks snapshot reference. (Reported by Marin Odrljin) |
|
|
stream: Allow streams on a topology to be put into groups (Reported by Joshua Colp) |
|
|
alembic: PJSIP scripts are missing column bundle in ps_endpoints table (Reported by Florian Floimair) |
|
|
Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) |
|
|
GCC 7 warning: app_voicemail.c: In function ‘imap_delete_old_greeting’ (Reported by Anthony Messina) |
|
|
jitterbuffer: Does not handle case where translator returns null frame. (Reported by Joshua Elson) |
|
|
ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) |
|
|
core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. (Reported by Mr Dini) |
|
|
Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. (Reported by Jonathan Cloots) |
|
|
The config_hook unit test causes Asterisk to crash if run a second time (Reported by George Joseph) |
|
|
res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes (Reported by Martin Cisárik) |
|
|
res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) |
|
|
chan_sip: Crypto attribute not last but first on SDP media level. (Reported by Alexander Traud) |
|
|
res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 (Reported by Tzafrir Cohen) |
|
|
Cannot disable SIP debugging via CLI after enabling with conf file option – also ‘sip set debug off’ reports debugging disabled, when it really isn’t (Reported by Rusty Newton) |
|
|
bridge_softmix: When a channel leaves add in any missing participant streams (Reported by Joshua Colp) |
|
|
sip_to_pjsip not correctly handling disallow=all directive (Reported by Torrey Searle) |
|
|
Missing openssl dependencies in res_rtp_asterisk and tcptls (Reported by Tzafrir Cohen) |
|
|
Fails to build in FreeBSD due to sys/sysmacros.h not existing there (Reported by Guido Falsi) |
|
|
res_pjsip_session: SIP/SDP origin (o=) contains local address. (Reported by Alexander Traud) |
|
|
chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller (Reported by lvl) |
|
|
backtrace.c: Crash due to double-free. (Reported by Corey Farrell) |
|
|
Crash on ast_ssl_teardown when stopping. (Reported by Alexander Traud) |
|
|
Can’t load res_corosync.so module on Asterisk 13.18.2 (Reported by Anton Mosin) |
Improvements made in this release:
———————————–
|
cdr.c: Minor code optimizations. (Reported by Richard Mudgett) |
|
|
[PATCH] When failing to acquire target during attended transfer, display wanted extension (Reported by Niklas Larsson) |
|
|
app_voicemail: Add new object for VoicemailUserEntry (Reported by sungtae kim) |
|
|
ast_coredumper: allow pointing out the asterisk binary explicitly (Reported by Tzafrir Cohen) |
|
|
Compilation warning for invert.c (array subscript is above array bounds) (Reported by Marcello Ceschia) |
|
|
Upgrade bundled PJPROJECT to 2.7 (Reported by Richard Mudgett) |
|
|
CDR performance needs improvement. (Reported by Richard Mudgett) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0-rc1
Thank you for your continued support of Asterisk!