[asterisk-dev] Asterisk 17.5.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.5.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.5.0-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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28794 - res_pjsip: Crash when escaping during URI
      printing
      (Reported by nappsoft)
 * ASTERISK-28884 - x-ast-orig-host not filtered out from
      request URI and To header
      (Reported by nappsoft)
 * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on
      call answer
      (Reported by Alexei Gradinari)
 * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be
      wrong in SDP/SDES.
      (Reported by Alexander Traud)
 * ASTERISK-28898 - bridge_softmix: Conference bridge not
      passing silent rtp packets
      (Reported by Jonathan Hunter)
 * ASTERISK-28892 - res_musiconhold: Module res_musiconhold
      throws false warning
      (Reported by Nicholas John Koch)
 * ASTERISK-28904 - RTP ICE leaks the memory
      (Reported by
      sungtae kim)
 * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when
      transport=transport-udp6
      (Reported by Peter Sokolov)
 * ASTERISK-28854 - SIGSEGV when pjsip show history encounters
      IPV6 address
      (Reported by Roger James)
 * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
      truncation.
      (Reported by Alexander Traud)
 * ASTERISK-28776 - Non async-signal-safe syscalls used after
      fork before exec
      (Reported by nappsoft)
 * ASTERISK-28870 - streams: One memory leak and one issue
      cloning streams
      (Reported by George Joseph)
 * ASTERISK-28829 - app_queue: leaking stasis subscription when
      Redirecting call 
      (Reported by lvl)
 * ASTERISK-25844 - app_queue: Ghost channels in "core show
      channels" output
      (Reported by Etienne Lessard)
 * ASTERISK-28859 - pjsip: Increase maximum candidate count
    
      (Reported by Joshua C. Colp)
 * ASTERISK-22920 - Crash while Forwarding from TLS extension
      with CHANNEL args secure_bridge_media and
      secure_bridge_signaling
      (Reported by Shlomi Gutman)
 * ASTERISK-28852 - Unprotected access to nochecksums variable,
      causes build failures
      (Reported by Guido Falsi)
 * ASTERISK-28848 - app_fax: Compile.
      (Reported by
      Alexander Traud)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
 
      (Reported by Joshua C. Colp)
 * ASTERISK-28896 - ari: Add support for specifying variables on
      channel create
      (Reported by Joshua C. Colp)
 * ASTERISK-28879 - pjproject has race conditions in it's build
      system
      (Reported by Guido Falsi)
 * ASTERISK-28866 - third-party/pjproject/configure.m4 contains
      bashisms
      (Reported by Guido Falsi)
 * ASTERISK-28853 - Missing include on FreeBSD
      (Reported
      by Guido Falsi)
 * ASTERISK-28832 - chan_mobile creates PCMA streams that make
      some VoIP clients crash or not render received audio
     
      (Reported by Peter Turczak)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0-rc1

Thank you for your continued support of Asterisk!
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