Asterisk 15.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
This release is available for immediate download at

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:
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
      (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
      endpoint identification to IP only
      (Reported by Ben

Bugs fixed in this release:
 * ASTERISK-27531 - Compiler optimizations can break module load
      (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
      Contact crashes asterisk
      (Reported by Ross Beer)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
      (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
      destroyed and robotic audio on one channel
      (Reported by
      Zane Conkle)
 * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
      (Reported by Corey Farrell)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
      (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
      CLI or AMI
      (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
      Local channels - documentation misleading
      (Reported by
      Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
      found" should be logged as a security event
      (Reported by
      Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
      (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
      (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
      (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
      alias it, cli.conf example broken
      (Reported by Tim
      Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
      FXS gateway
      (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
      RTCP packet will write past where it should
      (Reported by
      Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
      (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
      get stuck in "In Use" state.
      (Reported by Steven T.
 * ASTERISK-26131 - 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
 * ASTERISK-27475 - codec_opus requires libcurl
      by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
      by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-20643 - 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
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
      (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George
 * ASTERISK-27454 - res_http_post: Don't require
      (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
      (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
      (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
      Linear format variants in output of 'core show translation' are
      (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
      (Reported by Marco Giordani)
 * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
      (Reported by Kevin Harwell)
 * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
      (srflx) with Dual-Stack.
      (Reported by Alexander Traud)
 * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
      IPv6 addresses.
      (Reported by Alexander Traud)
 * ASTERISK-27435 - [patch] configure:
      pjsip_evsub_set_uas_timeout not found.
      (Reported by
      Alexander Traud)
 * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra

      (Reported by Ivan Larionov)
 * ASTERISK-27431 - Asterisk fails to build when openssl headers
      are not installed.
      (Reported by Corey Farrell)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27361 - Attended transfer crashes in Asterisk
      (Reported by Alessandro Pimenta)
 * ASTERISK-27238 - Bridging: Crash freeing a frame that's
      already been freed
      (Reported by Richard Kenner)
 * ASTERISK-27412 - core: Audiohook freeing interpolated frame
      when it shouldn't.
      (Reported by Mikhail)
 * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
      channel variable before closing the file
      (Reported by
      George Joseph)
 * ASTERISK-26758 - 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)
 * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
      is equal to RemoteAddress)
      (Reported by Vasilii Rogin)
 * ASTERISK-27415 - asterisk.conf: Setting astctl without
      setting astrundir is ineffective.
      (Reported by Corey
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed

      (Reported by Joshua Colp)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      (Reported by Corey Farrell)
 * ASTERISK-27337 - chan_sip: Security vulnerability with client
      code header (revisited)
      (Reported by Richard Mudgett)
 * ASTERISK-27319 - (Security) Function in PJSIP 2.7
      miscalculates the length of an unsigned long variable in 64bit
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
      contact read from astdb or database
      (Reported by Aaron An)
 * ASTERISK-27290 - res_pjsip: PIDF contact field has
      malformed/invalid XML
      (Reported by basildane)
 * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
      (Reported by seanchann.zhou)
 * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
      (Reported by Sean Bright)
 * ASTERISK-26426 - format_ogg_opus: remove from source
      (Reported by Kevin Harwell)
 * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
      (Reported by Alexander Traud)
 * ASTERISK-27378 - Modules: Fix issues with CLI completion.
      (Reported by Corey Farrell)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael
 * ASTERISK-27364 - channel: Crash when fax gateway is in use
      with PJSIP
      (Reported by Jared Hull)
 * ASTERISK-27390 - Audit menuselect module dependencies
      (Reported by Corey Farrell)
 * ASTERISK-27389 - Optional API modules should not allow
      (Reported by Corey Farrell)
 * ASTERISK-27369 - Bridge() dialplan application fails without
      setting BRIDGERESULT channel variable
      (Reported by James
 * ASTERISK-27067 - res_ari_channels: channel_state_invalid
      always leaks snapshot reference.
      (Reported by Marin
 * ASTERISK-27379 - stream: Allow streams on a topology to be
      put into groups
      (Reported by Joshua Colp)
 * ASTERISK-27374 - alembic: PJSIP scripts are missing column
      bundle in ps_endpoints table
      (Reported by Florian
 * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
      (Reported by Igor Goncharovsky)
 * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
      (Reported by Anthony Messina)
 * ASTERISK-27194 - jitterbuffer: Does not handle case where
      translator returns null frame.
      (Reported by Joshua Elson)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-26639 - core: Disabling xmldoc support does not
      work. Also results in abort during Asterisk startup.
      (Reported by Mr Dini)
 * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
      absence of the Expires header field with an unsubscribe action.

      (Reported by Jonathan Cloots)
 * ASTERISK-25960 - The config_hook unit test causes Asterisk to
      crash if run a second time
      (Reported by George Joseph)
 * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
      when rtp_ipv6 set to yes
      (Reported by Martin Cisárik)
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
      but first on SDP media level.
      (Reported by Alexander
 * ASTERISK-24483 -,
      Assertion on un/re-load: == -1
      (Reported by Tzafrir
 * ASTERISK-23462 - 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
      by Rusty Newton)
 * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
      any missing participant streams
      (Reported by Joshua Colp)
 * ASTERISK-27333 - sip_to_pjsip not correctly handling
      disallow=all directive
      (Reported by Torrey Searle)
 * ASTERISK-27328 - Missing openssl dependencies in
      res_rtp_asterisk and tcptls
      (Reported by Tzafrir Cohen)
 * ASTERISK-27343 - Fails to build in FreeBSD due to
      sys/sysmacros.h not existing there
      (Reported by Guido
 * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
      (o=) contains local address.
      (Reported by Alexander Traud)
 * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
      configured codecs, but subset based on caller
      (Reported by
 * ASTERISK-27340 - backtrace.c: Crash due to double-free.
      (Reported by Corey Farrell)
 * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
      (Reported by Alexander Traud)

Improvements made in this release:
 * ASTERISK-24297 - cdr.c: Minor code optimizations.
      (Reported by Richard Mudgett)
 * ASTERISK-27449 - [PATCH] When failing to acquire target
      during attended transfer, display wanted extension
      (Reported by Niklas Larsson)
 * ASTERISK-27456 - app_voicemail: Add new object for
      (Reported by sungtae kim)
 * ASTERISK-27380 - ast_coredumper: allow pointing out the
      asterisk binary explicitly
      (Reported by Tzafrir Cohen)
 * ASTERISK-23556 - Compilation warning for invert.c (array
      subscript is above array bounds)
      (Reported by Marcello
 * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
      (Reported by Richard Mudgett)
 * ASTERISK-27335 - CDR performance needs improvement.
      (Reported by Richard Mudgett)

For a full list of changes in this release, please see the ChangeLog:

Thank you for your continued support of Asterisk!