The release of Asterisk 15.2.0 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:
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:
-----------------------------------
| Compiler optimizations can break module load sequence. | (Reported by abelbeck) | |
| Security: Authenticated SUBSCRIBE without Contact crashes asterisk | (Reported by Ross Beer) | |
| 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) | |
| [patch] 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) | |
| [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. | (Reported by Alexander Traud) | |
| [patch] chan_sip/ICE: Square brackets around IPv6 addresses. | (Reported by Alexander Traud) | |
| [patch] 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) | |
| [patch] tcptls: Print notice when TLS is enabled but not configured. | (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) | |
| [patch] 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) | |
| [patch] 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) | |
| [patch] Crash on ast_ssl_teardown when stopping. | (Reported by Alexander Traud) |
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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
Thank you for your continued support of Asterisk!
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