[asterisk-dev] Asterisk 16.6.0 Now Available

[Asterisk Development Team]

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

The release of Asterisk 16.6.0 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:

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-28521 - pjsip: Memory Leak
      (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      (Reported
      by Cyril Rami��re)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      FreeBSD
      (Reported by Guido Falsi)
 * 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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20191008/8f2b8c6e/attachment.html>