[asterisk-dev] Asterisk 18.16.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Dec 15 07:39:57 CST 2022


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.16.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.16.0-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:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-30338 - pjproject: Backport security fixes from
      2.13
      (Reported by Benjamin Keith Ford)
 * ASTERISK-30176 - manager: GetConfig can read files outside of
      Asterisk
      (Reported by shawty)
 * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
      party IE
      (Reported by Michael Bradeen)

Improvements made in this release:
-----------------------------------
 * ASTERISK-30328 - Typo in from_domain description on res_pjsip
      configuration documentation
      (Reported by Marcel Wagner)
 * ASTERISK-30316 - res_pjsip: Documentation should point out
      default if contact_user is not being set for outbound
      registrations
      (Reported by Marcel Wagner)
 * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded
     
      (Reported by N A)
 * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage

      (Reported by N A)
 * ASTERISK-30286 - app_mixmonitor: Add option to use real
      Caller ID for Caller ID
      (Reported by N A)
 * ASTERISK-30308 - pbx_builtins: Allow Answer to return
      immediately
      (Reported by N A)
 * ASTERISK-30295 - test_json: Remove duplicated static
      function
      (Reported by N A)
 * ASTERISK-30290 - file.c: Don't emit warnings on winks.
     
      (Reported by N A)
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
      scope trace debugs to DEBUG level
      (Reported by N A)
 * ASTERISK-30223 - features: add no-answer option to Bridge
      application
      (Reported by N A)
 * ASTERISK-30158 - PJSIP: Add new 100rel option
      "peer_supported"
      (Reported by Maximilian Fridrich)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30349 - app_if:  Format truncation error
     
      (Reported by George Joseph)
 * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support
      on INVITEs
      (Reported by N A)
 * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not
      sending email to user
      (Reported by N A)
 * ASTERISK-29793 - adsi: CAS is malformed
      (Reported by N
      A)
 * ASTERISK-30344 - ari: Memory leak in create when specifying
      JSON
      (Reported by Saken)
 * ASTERISK-30311 - func_presencestate: Fix invalid memory
      access.
      (Reported by N A)
 * ASTERISK-30336 - sig_analog: Fix no timeout duration
     
      (Reported by N A)
 * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
      TCP/TLS connection terminated and subscription persistence is
      removed
      (Reported by nappsoft)
 * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering
      results in wrong stream direction of first call leg
     
      (Reported by Maximilian Fridrich)
 * ASTERISK-29998 - sla: deadlock when calling SLAStation
      application
      (Reported by N A)
 * ASTERISK-30321 - Build:  Embedded blobs have executable
      stacks
      (Reported by George Joseph)
 * ASTERISK-30293 - Memory leak in JSON_DECODE
      (Reported
      by David Uczen)
 * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect
      "transmit_silence" asterisk.conf option
      (Reported by
      Joshua C. Colp)
 * ASTERISK-30285 - manager.c: Remove outdated documentation
   
      (Reported by N A)
 * ASTERISK-30076 - app_stack: Incorrect exit location in
      predial handlers logged
      (Reported by N A)
 * ASTERISK-30282 - CI: Coredump output isn't saved when running
      unittests
      (Reported by George Joseph)
 * ASTERISK-30281 - chan_rtp: Local address being used before
      being set
      (Reported by George Joseph)
 * ASTERISK-28689 - res_pjsip: Crash when locking group lock
      when sending stateful response
      (Reported by Jesse Ross)
 * ASTERISK-30217 - Registration do not allow multiple proxies
 
      (Reported by Igor Goncharovsky)
 * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged
      if MALLOC_DEBUG is enabled
      (Reported by N A)
 * ASTERISK-30273 - test_mwi: compilation fails on 32-bit
      Debian
      (Reported by N A)
 * ASTERISK-30193 - chan_pjsip should return all codecs on a
      re-INVITE without SDP
      (Reported by Henning Westerholt)
 * ASTERISK-30258 - Dialing API: Cancel a running async thread,
      does not always cancel all calls
      (Reported by Frederic LE
      FOLL)
 * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY
  
      (Reported by N A)
 * ASTERISK-30248 - ast_get_digit_str adds bogus initial
      delimiter if first character not to be spoken
      (Reported by
      David Woolley)
 * ASTERISK-30264 - res_pjsip: Subscription handlers do not get
      cleanly unregistered, causing crash
      (Reported by N A)
 * ASTERISK-30213 - Make crypto_load() reentrant and handle
      symlinks correctly
      (Reported by Philip Prindeville)
 * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings
 
      (Reported by N A)
 * ASTERISK-30239 - Prometheus plugin crashes Asterisk when
      using local channel
      (Reported by Joeran Vinzens)
 * ASTERISK-30237 - res_prometheus: Crash when scraping bridges

      (Reported by Igor Yeroshev)
 * ASTERISK-30245 - db: ListItems is incorrect
      (Reported
      by N A)
 * ASTERISK-30243 - func_logic: IF function complains if both
      branches are empty
      (Reported by N A)
 * ASTERISK-30232 - Initialize stack-based ast_test_capture
      structures correctly
      (Reported by Philip Prindeville)
 * ASTERISK-30220 - func_scramble: Fix segfault due to null
      pointer deref
      (Reported by N A)
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
      uninitialized variable error
      (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
      uninitialized error in geoloc_config.c
      (Reported by George
      Joseph)
 * ASTERISK-30226 - REGRESSION: res_crypto complains about the
      stir_shaken directory in /var/lib/asterisk/keys
      (Reported
      by George Joseph)

New Features made in this release:
-----------------------------------
 * ASTERISK-21502 - New SIP Channel Driver - add Advice of
      Charge support
      (Reported by Matt Jordan)
 * ASTERISK-30150 - res_pjsip_session: Add support for custom
      parameters
      (Reported by N A)
 * ASTERISK-30322 - res_hep: Add capture agent name support
    
      (Reported by N A)
 * ASTERISK-29497 - Add conditional branch applications
     
      (Reported by N A)
 * ASTERISK-30305 - chan_dahdi: Allow FXO channels to start
      immediately
      (Reported by N A)
 * ASTERISK-30284 - app_mixmonitor: Add option to delete
      recording file when done
      (Reported by N A)
 * ASTERISK-30146 - res_pjsip_logger: Add method-based log
      filtering
      (Reported by N A)
 * ASTERISK-30263 - res_pjsip_notify: Allow using
      pjsip_notify.conf from AMI
      (Reported by N A)
 * ASTERISK-30254 - res_tonedetect: Add audible ringback
      detection to TONE_DETECT
      (Reported by N A)
 * ASTERISK-30091 - cdr: Allow CDRs to ignore call state
      changes
      (Reported by N A)
 * ASTERISK-30032 - Support of mediasec SIP headers and SDP
      attributes
      (Reported by Maximilian Fridrich)
 * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to
      not answer
      (Reported by N A)
 * ASTERISK-30179 - app_amd: Allow audio to be played while AMD
      is running
      (Reported by N A)
 * ASTERISK-29432 - New function to allow access to any channel

      (Reported by N A)
 * ASTERISK-30222 - func_strings: Add trim functions
     
      (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0-rc1

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20221215/add4497a/attachment-0001.html>


More information about the asterisk-dev mailing list