[asterisk-announce] Asterisk 17.5.0 Now Available

[Asterisk Development Team]

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

The release of Asterisk 17.5.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28940 - /channels/create doesn't get any parameters
      from the body
      (Reported by sungtae kim)
 * ASTERISK-28932 - res_pjsip_logger writing too big packets
   
      (Reported by nappsoft)
 * ASTERISK-28921 - Wrong return value check for fwrite when
      writing to pcap file
      (Reported by nappsoft)
 * 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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0

Thank you for your continued support of Asterisk!
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