Asterisk Development Team
2020-Jul-16 16:53 UTC
[asterisk-users] Asterisk 16.12.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
16.12.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.12.0 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-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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
Thank you for your continued support of Asterisk!
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