The Asterisk Development Team would like to announce the release of Asterisk 13.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.15.0 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:
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-26851 - res_pjsip_sdp_rtp: RTP instance does not use
same IP as explicit transport
(Reported by Richard Begg)
* ASTERISK-26897 - chan_sip: Security vulnerability with client
code header
(Reported by Alex Villacís Lasso)
* ASTERISK-26916 - res_pjsip: Excessive refcount reached on
transport ao2 object
(Reported by Ross Beer)
* ASTERISK-26705 - libasteriskssl.so not found when asterisk is
installed for the 1st time
(Reported by George Joseph)
* 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-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-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 Jørgen 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-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 Jørgen H)
* ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
about network change events
(Reported by George Joseph)
* ASTERISK-26313 - chan_sip : Asterisk restart seems to be
required for changing encryption option
(Reported by
benasse)
* 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-26580 - [patch] Error during LDAP modify action when
user unregisters
(Reported by Nicholas John Koch)
* 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-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-13.15.0
Thank you for your continued support of Asterisk!