[asterisk-dev] Asterisk 16.12.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.12.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.12.0-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-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
      asterisk 16
      (Reported by Joseph Ades)
 * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
      contacts on AOR
      (Reported by Joshua C. Colp)
 * ASTERISK-28930 - ./configure --without-ssl build failure
    
      (Reported by Jaco Kroon)
 * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
      pjproject 2.7.2
      (Reported by Jared Smith)
 * ASTERISK-28957 - chan_sip: chan_sip does not process 400
      response to an INVITE.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28888 - res_corosync: causes asterisk crash in huge
      distributed environment.
      (Reported by Universit�� di
      Bologna - CESIA VoIP)
 * ASTERISK-28955 - "setvar" doesn't work properly in
      dahdi-channels.conf
      (Reported by Marin Odrljin)
 * ASTERISK-28954 - StreamEcho() only returns 1 active stream
  
      (Reported by Bill Kervaski)
 * ASTERISK-28942 - res_sorcery_memory_cache: Individual object
      expiration behaves unexpectedly with full backend caching
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28953 - res_pjsip_session: Preserve stream label
   
      (Reported by Joshua C. Colp)
 * ASTERISK-28952 - Queue wrapuptime sometimes not respected
      (based on stale lastcall time)
      (Reported by Walter Doekes)
 * ASTERISK-28950 - Stale code in app_queue to check untouched
      channel
      (Reported by Walter Doekes)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      exception
      (Reported by Walter Doekes)
 * ASTERISK-28948 - ARI channel create doesn't referencing the
      channel_id parameter
      (Reported by sungtae kim)
 * ASTERISK-28938 - core_unreal / core_local: Add support for
      multistream and re-negotiation
      (Reported by Joshua C.
      Colp)
 * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
      buffers on non-WebRTC
      (Reported by Joshua C. Colp)
 * ASTERISK-28944 - bridge_softmix: Transitioning a stream from
      inactive -> sendrecv/sendonly doesn't re-negotiation
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
   
      (Reported by Yury Kirsanov)
 * ASTERISK-28940 - /channels/create doesn't get any parameters
      from the body
      (Reported by sungtae kim)
 * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
  
      (Reported by Walter Doekes)
 * ASTERISK-28900 - res_fax: Double frame free when gateway in
      use with off-nominal format usage
      (Reported by Gregory
      Massel)
 * ASTERISK-28929 - pjproject_bundled: Honor
      --without-pjproject.
      (Reported by Alexander Traud)
 * 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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28959 - res_pjsip: Added option for disable rport
      parameter set
      (Reported by sungtae kim)
 * ASTERISK-28958 - Continue reading string when ping received
      by websocket
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-28945 - AMI SendText - add Content-Type parameter
  
      (Reported by Kevin Harwell)
 * ASTERISK-28949 - res_http_websocket: Add masking to websocket
      client
      (Reported by Moises Silva)
 * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
 
      (Reported by Kevin Harwell)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0-rc1

Thank you for your continued support of Asterisk!
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