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Asterisk 16.6.0-rc1 Now Available

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!