[asterisk-dev] Asterisk 16.6.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.6.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.6.0-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-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei
      Gradinari)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32
      (Reported by Ruddy G)
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
      PRI channel hangs up
      (Reported by Frederic LE FOLL)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
      unanswered=yes is set in cdr.conf
      (Reported by Frederic LE
      FOLL)
 * ASTERISK-28499 - translate: Crash when frame does not have a
      "src" field set
      (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
      type not at end of a struct
      (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
      re-register
      (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
     
      (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
      leave_voicemail because not checking mailstream
      (Reported
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      (Reported
      by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
     
      (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect
    
      (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

      (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
      entries
      (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
     
      (Reported by Torrey Searle)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
      res_config_sqlite3.conf
      (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
      res_config_sqlite3.conf
      (Reported by Dennis)
 * ASTERISK-26968 - chan_pjsip: Transfer() does not result in
      TRANSFERSTATUS reflecting SIP response to transfer
     
      (Reported by Dan Cropp)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
      deadlocks (in chan_sip)
      (Reported by Walter Doekes)

New Features made in this release:
-----------------------------------
 * ASTERISK-17808 - [patch] Unregister a realtime moh class
    
      (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
      chan_pjsip to setup From header URI domain
      (Reported by
      Stas Kobzar)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0-rc1

Thank you for your continued support of Asterisk!
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