Asterisk Development Team
2017-Mar-23 22:38 UTC
[asterisk-users] Asterisk 14.4.0-rc1 Now Available
The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.4.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26878 - func_channel: Add ability to get the callid so
dialplan has access to it. (Reported by Richard Mudgett)
* ASTERISK-26863 - res_pjsip: Add endpoint identification scheme
based on a configured SIP header/value (Reported by Matt Jordan)
* ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
removed (Reported by John Covert)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong protocol
name in "Protocol ID" field in HEP packets (Reported by Max
Norba)
* ASTERISK-26484 - res_pjsip_messaging: Crash when using invalid
URI in MessageSend 'from' argument. (Reported by Vinod
Dharashive)
* ASTERISK-26776 - res_pjsip_pubsub: Crash when generating xpidf
content (Reported by Andrew Green)
* ASTERISK-26880 - Asterisk crashes when multiple speex users join
confbridge with pp_vad and dtx enabled (Reported by Kirsty
Tyerman)
* ASTERISK-26862 - app_queue: Queue stops calling members with
local interface after forwarding in previous call (Reported by
Robert Mordec)
* ASTERISK-26732 - res_rtp_asterisk: Implement RTCP Multiplexing -
breaking WebRTC in Chrome (Reported by Dan Jenkins)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided (Reported by Matt Jordan)
* ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
same IP as explicit transport (Reported by Richard Begg)
* ASTERISK-26867 - autochan: Locking in a function
ast_autochan_destroy() on destroyed channel (after masquerade).
(Reported by Krzysztof Trempala)
* ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a user
name doesn't go to the s extension (Reported by Torrey Searle)
* ASTERISK-26668 - core: Malformed pattern matching extension
(various factors) results in crash (Reported by xrobau)
* ASTERISK-26865 - chan_iax2: Reload of iax peer results in loss
of host address/port (Reported by Richard Begg)
* ASTERISK-26872 - Bundled pjproject fails to build when tarball
downloaded with curl due to md5 verification failure in Docker
containers (or when there is no terminal) (Reported by Matt
Jordan)
* ASTERISK-26717 - Document the fact that Asterisk HEP support
only works with the PJSIP channel driver (Reported by Olivier
Krief)
* ASTERISK-26643 - Extra new line in Device field of
DeviceStateChange AMI Event after restart of Asterisk (Reported
by Roman Bedros)
* ASTERISK-25237 - stasis_cache.c:845 caching_topic_exec: -
misleading ERROR message (Reported by Smirnov Aleksey)
* ASTERISK-26857 - chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
* ASTERISK-26841 - chan_sip: Call not cancelled after receiving a
422 response (Reported by Jean Aunis - Prescom)
* ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
shows wrong codec (Reported by Kevin Harwell)
* ASTERISK-26353 - res_musiconhold: musiconhold seems to think
that the general section is a class and issues warning (Reported
by Jonathan Harris)
* ASTERISK-26685 - res_pjsip: Crash when using IPv6 and Transport
ws,wss (Reported by Michael Balen)
* ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
per-mailbox basis (Reported by Mark Scholten)
* ASTERISK-26598 - Saynumber is trying to get "and" from
"digits/"
subfolder (Reported by Jonathan Harris)
* ASTERISK-17067 - Long lines in call files cause spurious syntax
error (Reported by Dave Olszewski)
* ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
'WS' when it should be 'WSS' (Reported by J??rgen H)
* ASTERISK-25628 - res_config_pgsql: should match the behavior of
other drivers so that queue_log can disable adaptive logging
(Reported by Dmitry Wagin)
* ASTERISK-26774 - core: Playback URL fails after some time
(Reported by Igor Gamayunov)
* ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers to
branch 12 (Reported by Tzafrir Cohen)
* ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
FRACKs if endpoint does not exist (Reported by Mark Michelson)
* ASTERISK-26623 - res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by J??rgen H)
* ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
about network change events (Reported by George Joseph)
* ASTERISK-26705 - libasteriskssl.so not found when asterisk is
installed for the 1st time (Reported by George Joseph)
* ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to
Bridge() application results in garbled audio (Reported by Sean
Bright)
* ASTERISK-26782 - res_pjsip: URI requirement for fields is not
consistently documented and error does not provide indication
(Reported by Peter Sokolov)
* ASTERISK-26812 - [patch] Fix download_externals To Allow The Use
Of curl Or wget (Reported by Michael L. Young)
* ASTERISK-18271 - Pattern matching with res_config_mysql
extensions does not behave as expected (Reported by Charlie
Smurthwaite)
* ASTERISK-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
* ASTERISK-18731 - [patch] DUNDi weight parameter not processed
correctly (Reported by Peter Racz)
* ASTERISK-26799 - res_pjsip: Using an auth object for inbound and
outbound authentication fails. (Reported by Richard Mudgett)
* ASTERISK-26738 - Frequent segfaults since activation of DNS SRV,
in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and
pj_atomic_inc_and_get at pj/os_core_unix.c (Reported by Michael
Maier)
* ASTERISK-25893 - Function vmauthenticate accesses uninitialized
memory (Reported by Filip Jenicek)
* ASTERISK-26580 - [patch] Error during LDAP modify action when
user unregisters (Reported by Nicholas John Koch)
* ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)
* ASTERISK-15858 - [patch] Fix query with double backslash in
string literals and stop log warnings (Reported by Humberto
Figuera)
* ASTERISK-26057 - res_config_sqlite3 uses incorrect query -
unnecessary escape (Reported by Stepan)
* ASTERISK-23457 - SQlite3: Realtime queue loading fails after
PRAGMA query result (Reported by Scott Griepentrog)
* ASTERISK-26794 - http: Crash on Reload Only in
ast_tcptls_server_start (Reported by Joshua Elson)
* ASTERISK-26714 - Phone default have not ringing on ARM (Reported
by Igor Goncharovsky)
* ASTERISK-26696 - pjsip_pubsub: PJSIP Subscription Persistence in
AstDB Does not update on subscription refresh (Reported by Zach
R)
* ASTERISK-26756 - res_pjsip_mwi: Asterisk does not terminate MWI
subscription (Reported by Carl Fortin)
* ASTERISK-26109 - Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
* ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
realtime (Reported by Ryan Rittgarn)
* ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
(Reported by var)
* ASTERISK-26248 - chan_pjsip: Error when calling PJSIP client
with domain specified (Reported by Norbert Varga)
* ASTERISK-26788 - core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
* ASTERISK-26115 - pbx: AMI Originate ignore "failed" extension on
call failure (Reported by Nasir Iqbal)
* ASTERISK-26785 - configs/samples: The 'identify' entry is in
the wrong section in sorcery.conf.sample (Reported by Torrey
Searle)
* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
by nappsoft)
* ASTERISK-26770 - res_stasis_device_state: Duplicate
subscriptions when multiple received at same time (Reported by
Joshua Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-26864 - res_pjsip_session: Add support for overlap
dialling (Reported by Richard Begg)
* ASTERISK-26846 - chan_sip: Add rtcp-mux support (Reported by
Sean Bright)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.4.0-rc1
Thank you for your continued support of Asterisk!