安全問(wèn)題：
-----------------------------------
* ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
（Reported by Jan Hoffmann）
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
（Reported by Sean Bright）添加了新功能：
-----------------------------------
* ASTERISK-28087 - add flag to allow CALLERID（num） to be placed
in Contact header in chan_pjsip
（Reported by Torrey
Searle）
修復了一些bug:-----------------------------------
* ASTERISK-28151 - app_voicemail: MWI fails with
mailboxes=##@device instead of mailboxes=##@default
（Reported by Ronald Raikes）
* ASTERISK-28125 - app_queue: Revert broken queue channel
reference patch
（Reported by lvl）
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on voice packet with marker bit
（Reported by Alexei Gradinari）
* ASTERISK-28159 - SIGABRT caused by stack corruption in
hashkeys_read when no matching keys present
（Reported by
Michael Walton）
* ASTERISK-28140 - repeated segmentation faults
（Reported by Eyal Hasson）
* ASTERISK-28169 - ARI /channels/create handler causes core
dump
（Reported by sungtae kim）
* ASTERISK-28103 - stasis: Filter messages at publishing to
reduce work done
（Reported by Joshua C. Colp）
* ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
Re-Invite omits routset
（Reported by Torrey Searle）
* ASTERISK-28158 - Some conditions prevent running of el_end,
break the terminal.
（Reported by Corey Farrell）
* ASTERISK-28110 - rtp: Incorrect Packetization
（Reported
by Robert Cripps）
* ASTERISK-28146 - pbx_config: Only the first [globals] section
is processed.
（Reported by Corey Farrell）
* ASTERISK-28150 - Formatting error in documentation
（Reported by Scott Griepentrog）
* ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
report AST_CEL_PICKUP in handle_invite_replaces
（Reported
by Luit van Drongelen）
* ASTERISK-28137 - res_pjsip_notify: improve realtime
performance on CLI completion on the endpoint
（Reported by
Alexei Gradinari）
* ASTERISK-27980 - Caller ID cannot be changed on Attended
Transfer before dialing out
（Reported by Alexei Gradinari）
* ASTERISK-28107 - app_confbridge: Participant info labels
aren't being added to the SDPs
（Reported by George Joseph）
* ASTERISK-28089 - function ast_sendtext（） create RTP realtime
packets with a trailing null byte in the payload
（Reported
by Emmanuel BUU）
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
（Reported by Emmanuel BUU）
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
AMI
（Reported by Andrej）
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
（Reported by Alexei Gradinari）
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
（Reported by
Cao Minh Hiep）
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
not work
（Reported by Cameron）
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
（Reported by Alexei Gradinari）
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
differently to CLI
（Reported by Peter Katzmann）
* ASTERISK-28045 - configure script does not enforce
libunbound2 version
（Reported by Samuel Galarneau）
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
（Reported by Joshua C. Colp）
* ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
instance can't be set up
（Reported by Lei Fu）
* ASTERISK-28034 - chan_sip unstable with TLS after asterisk
start or reloads
（Reported by David Hajek）
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
2.8
（Reported by Joshua C. Colp）
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
（Reported by Sergej Kasumovic）
* ASTERISK-28047 - chan_pjsip: Declined video stream is added
when no video codecs configured and session refresh with removed
video stream occurs
（Reported by Will）
* ASTERISK-28033 - AMI event "NewExten" is set to the wrong
class
（Reported by lvl）
* ASTERISK-28049 - res_pjproject build failure
（Reported
by Jaco Kroon）
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
（Reported by Frederic LE FOLL）
* ASTERISK-28005 - channel.c: ARI ring only once
（Reported by Hajek Michal）
* ASTERISK-28032 - Realtime queuemembers are not updated during
retry phase
（Reported by lvl）
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not
boolean
（Reported by Joshua C. Colp）
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
（Reported by Sean Bright）
* ASTERISK-28002 - When T.140 realtime text is negociated, a
lot of debug traces are generated
（Reported by Emmanuel
BUU）
* ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
authentification error
（Reported by Ian Gilmour）
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
（Reported by Florian
Floimair）
* ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
（Reported by
Joshua Elson）
* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
offer
（Reported by Torrey Searle）
* ASTERISK-27398 - No joint capabilities with video and
audio-only streams
（Reported by Benjamin Keith Ford）
* ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
LEAVEEMPTY
（Reported by Valentin Safonov）
* ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
Do not undef s_addr.
（Reported by Alexander Traud）
* ASTERISK-27999 - Wrong SRTP use status report
（Reported
by Salah Ahmed）
* ASTERISK-28001 - res_pjsip_registrar: Improve performance of
inbound handling
（Reported by Joshua C. Colp）
* ASTERISK-27966 - pjsip: Race condition in 183 re transmission
can result in a deadlock
（Reported by Torrey Searle）
* ASTERISK-15331 - make menuselect fails due to undefined
symbols （initscr32, w32addch） in menuselect_curses.o
（Reported by Majdi Bsoul）
* ASTERISK-14935 - [regression] menuselect compilation failure
on Solaris 10
（Reported by Samuel Owens）
* ASTERISK-12382 - menuselect compilation failure on Solaris 10
/ gcc 3.4.3
（Reported by rleasure）
* ASTERISK-9107 - menuselect compilation failure on Solaris
10/gcc-4.1.1
（Reported by Bob Atkins）
* ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
（Reported by Alexander Traud）
* ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
matches against "generic string" headers
（Reported by
George Joseph）
* ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
Developer Mode.
（Reported by Alexander Traud）
* ASTERISK-27591 - Frack errors in stasis.c and memory leakage
（Reported by Siruja Maharjan）
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
（Reported by Joshua C.
Colp）
* ASTERISK-27968 - systemd: asterisk.service
（Reported by
seanchann.zhou）
優(yōu)化升級：
-----------------------------------
* ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
parse an URI and return a specified part of the URI
（Reported by Alexei Gradinari）
* ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
a pipe
（Reported by Pascal Cadotte Michaud）
* ASTERISK-28046 - Remove stale nonoptreq references
（Reported by Walter Doekes）
* ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
（Reported by Adam Secombe）
* ASTERISK-28006 - PJSIP: Missing
"party=calling"/"party=called" in Remote-Party-ID
（Reported by Eric Dantie）
* ASTERISK-27995 - pjproject_bundled: Find shared libraries in
root --with-ssl=PATH.
（Reported by Alexander Traud）
* ASTERISK-27993 - pjsip_wizard example gives wrong info about
unsupported SRV records
（Reported by Jonathan Harris）
* ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
backspace or end of line are merged with regular text and it
causes some UA to break
（Reported by Emmanuel BUU）
源代碼下載：
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
參考資料：
https://www.rfc-editor.org/rfc/pdfrfc/rfc3262.txt.pdf
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