[asterisk-dev] Asterisk 18.1.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.1.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.1.0-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-29057 - pjsip: Crash on call rejection during high
      load
      (Reported by Sandro Gauci)

New Features made in this release:
-----------------------------------
 * ASTERISK-29027 - Implement support for History-Info
     
      (Reported by Torrey Searle)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28933 - res_pjsip.so fails to load when bundled
      pjproject is compiled without libssl
      (Reported by Walter
      Doekes)
 * ASTERISK-28825 - Any curl response checks out as valid even
      if 404 is returned.
      (Reported by dovid)
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
      invites (with auth) on 407 replies
      (Reported by Sebastian
      Damm)
 * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
      includes
      (Reported by Michael Newton)
 * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
     
      (Reported by Alexander Traud)
 * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
     
      (Reported by Alexander Traud)
 * ASTERISK-29146 - GCC Warnings: ���%s��� directive argument is
      null.
      (Reported by Alexander Traud)
 * ASTERISK-29124 - res_pjsip: flow transport broken for
      outbound requests
      (Reported by Nick French)
 * ASTERISK-29136 - config: Sample features.conf incorrectly
      includes " around sound files
      (Reported by Benjamin M.)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
      line 115
      (Reported by Andrew Siplas)
 * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
      progress calls due to codec negotiation after upgrading from
      Asterisk 16
      (Reported by Ross Beer)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
      errno != EBADF
      (Reported by under)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
      endpoint not found
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
      VMSayName
      (Reported by Eric Smith)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
      string when failing to add extension
      (Reported by Vieri)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
 
      (Reported by Jasper van der Neut)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
      values on RTP instance when "auto" DTMF is used
      (Reported
      by Sebastian Damm)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
      single entry
      (Reported by lvl)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
      judgment of frame format
      (Reported by ���������)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
      Frederiksen)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by P��ter
      Juh��sz)
 * ASTERISK-29089 - RTP Ports not cleared after hangup
     
      (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
      moh container
      (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
      slin
      (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
      aren't handled correctly
      (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29054 - Logger: Add debug logging categories
     
      (Reported by Kevin Harwell)
 * ASTERISK-29056 - Increase reg_server column size for
      ps_contacts table realtime
      (Reported by sungtae kim)
 * ASTERISK-29055 - Create a Bridge with video_single mode
     
      (Reported by sungtae kim)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0-rc1

Thank you for your continued support of Asterisk!
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