[asterisk-dev] Asterisk 14.0.0 Now Available!

The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

Asterisk 14 is the next major release series of Asterisk. It is a Standard
Support release, similar to Asterisk 12. For more information about support
time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 14, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14

A short list of new features includes:
* A complete overhaul of the core DNS support in Asterisk, including
  implementing full NAPTR and SRV support in the PJSIP stack via the
  libunbound library.

* The ability to publish extension state to a SIP Subscription server,
  such as Kamailio. This includes the ability to automatically generate
  a hint in the dialplan based on device state changes using the new
  autohint setting.

* Playback of media from a remote HTTP server via a URI is now supported
  by all dialplan applications and AGI. Media retrieved using a URI is
  cached in a media cache and re-used when possible.

* When using ARI to manipulate media on a resource, a list of media
  resources can now be supplied. The media resources will be played back
  sequentially in the order that they are provided.

* Channels created via ARI can now be created and handed off to Stasis
  for external control prior to performing the outbound dial. This
  enables applications to set additional state on the channel prior to
  dialing, as well as enabling certain early media scenarios.
And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation

A full list of all new features can also be found in the CHANGES file:

https://github.com/asterisk/asterisk/blob/14/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.0.0

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 13.11.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.11.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.11.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
———————————–
* ASTERISK-26349 –  13.11.1 res_pjsip/pjsip_distributor.c: Request
‘REGISTER’ failed (Reported by Dmitry Melekhov)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 11.6-cert15, 11.23.1, 13.8-cert3, 13.11.1 Now Available (Security Release)

The Asterisk Development Team has announced security releases for
Certified Asterisk 11.6, Asterisk 11, Certified Asterisk 13.8 and
Asterisk 13.
The available security releases are released as versions 11.6-cert15,
11.23.1, 13.8-cert3 and 13.11.1.
These releases are available for immediate download at
The release of these versions resolves the following security
vulnerabilities:
* AST-2016-006: Crash on ACK from unknown endpoint
  Asterisk can be crashed remotely by sending an ACK to it from an
  endpoint username that Asterisk does not recognize. Most SIP request
  types result in an “artificial” endpoint being looked up, but ACKs
  bypass this lookup. The resulting NULL pointer results in a crash
  when attempting to determine if ACLs should be applied.
  This issue was introduced in the Asterisk 13.10 release and only
  affects that release and later releases.
  This issue only affects users using the PJSIP stack with Asterisk.
  Those users that use chan_sip are unaffected.
* AST-2016-007: RTP Resource Exhaustion
 The overlap dialing feature in chan_sip allows chan_sip to report to a
 device that the number that has been dialed is incomplete and more
 digits are required. If this functionality is used with a device that
 has performed username/password authentication RTP resources are
 leaked. This occurs because the code fails to release the old RTP
 resources before allocating new ones in this scenario. If all
 resources are used then RTP port exhaustion will occur and no RTP
 sessions are able to be set up.
For a full list of changes in the current releases, please see the
ChangeLogs:
The security advisories are available at:
Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 11 – Security Fix Mode

Hello Everyone,

As many of you are already aware, we are rapidly approaching the time
when the Asterisk 11 branch will go into what is known as security fix
only mode.  Up to this point, bug fixes have been included and merged
into the 11 branch.  For Asterisk 11, this new phase of life shall
begin October 25th of this year.

This means that from a development perspective, the Asterisk
development team will not be putting effort into bug fixes for the 11
branch after the 25th of October.  Security related patches will be
merged for another year before completely putting the branch to rest.
During the course of that year, releases will be made as needed and as
security related patches are merged.

For any questions, you can either reply to this message or look at the
Asterisk Versioning Policy wiki page [1] as it explains most of this
process in greater detail.

Thanks so much again for all of your support and effort.  Asterisk
would not be what it is if it were not for all the great contributors
that are and have been involved.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions


Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW – Huntsville, AL 35806 – USA

[asterisk-dev] Asterisk 14.0.0-beta1 Now Available

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate

download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.0.0-beta1 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 beta:

New Features made in this release:

———————————–

* ASTERISK-25904 – PJSIP: add contact.updated event (Reported by

     Alexei Gradinari)

* ASTERISK-26058 – [Patch] Add uptime and last reloaded to

     FullyBooted AMI event (Reported by Niklas Larsson)

* ASTERISK-25925 – Allow Early Bridges on ARI Dials (Reported by

     Mark Michelson)

* ASTERISK-26068 – Multicast RTP Options (Reported by Mark

     Michelson)

* ASTERISK-26042 – ARI: Allow downloading of the media associated

     with a stored recording (Reported by Matt Jordan)

* ASTERISK-25425 – logger: Add JSON structured logging (Reported

     by Matt Jordan)

* ASTERISK-25900 – PJSIP Endpoint IP Access Controls (Reported by

     Alexei Gradinari)

* ASTERISK-25972 – res_pjsip_exten_state: Use body generator to

     publish extension state (Reported by Richard Mudgett)

* ASTERISK-25889 – ARI: Add separate “create” and “dial”

     operations for channels (Reported by Mark Michelson)

* ASTERISK-25803 – chan_sip: Optionally supply

     fromuser/fromdomain in SIP dial string (Reported by Walter

     Doekes)

* ASTERISK-24919 – res_pjsip_config_wizard: Ability to write

     contents to file (Reported by Ray Crumrine)

* ASTERISK-25670 – Add regcontext to PJSIP (Reported by Daniel

     Journo)

* ASTERISK-25660 – Add sipp-sendfax.xml and spandspflow2pcap.py to

     contrib/scripts. (Reported by Walter Doekes)

* ASTERISK-25591 – Complete List of Header Files

     (#include): iwyu (Reported by Alexander Traud)

* ASTERISK-25551 – Ability to add channel to an existing

     bridge by specifying an existing channel prefix (Reported by

     Alec Davis)

* ASTERISK-25419 – Dialplan Application for Integration of StatsD

     (Reported by Ashley Sanders)

* ASTERISK-25549 – Confbridge: Add participant timeout option

     (Reported by Mark Michelson)

* ASTERISK-24922 – ARI: Add the ability to intercept hold and

     raise an event (Reported by Matt Jordan)

* ASTERISK-25479 – Allow CDR’s to be modified before being

     dispatched to engines (Reported by Jonh Wendell)

* ASTERISK-25480 – Add field PauseReason on

     QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25377 – res_pjsip: Change default “From user” from UUID

     to something more palatable (Reported by Mark Michelson)

* ASTERISK-25252 – ARI: Add the ability to manipulate log channels

     (Reported by Matt Jordan)

* ASTERISK-25259 – chan_pjsip: Add rtptimeout support (Reported by

     Joshua Colp)

* ASTERISK-25238 – ARI: Support push configuration (Reported by

     Matt Jordan)

* ASTERISK-25173 – ARI: Add the ability to load/reload/unload an

     Asterisk module (Reported by Matt Jordan)

* ASTERISK-25006 – Add support set character for quoted

     identifiers  (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-23186 – Add usegmtime option to cel_pgsql

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24931 – dns: Add support for SRV records. (Reported by

     Joshua Colp)

* ASTERISK-24834 – DNS Overhaul: Implement the proposed core API –

     sync/async functions, resolver registration (Reported by Matt

     Jordan)

* ASTERISK-24836 – DNS Overhaul: Write a Resolver Implementation

     (Reported by Matt Jordan)

* ASTERISK-22591 – Prevent Asterisk from writing received

     SMS content in log (Reported by Jan Juergens)

* ASTERISK-17899 – Handle crypto lifetime in SDES-SRTP negotiation

     (Reported by Dwayne Hubbard)

* ASTERISK-24703 – ARI: Add the ability to “transfer” (redirect) a

     channel (Reported by Matt Jordan)

* ASTERISK-24363 – Add ability for Channel Drivers to

     provide Presence State information (Reported by Gareth Palmer)

* ASTERISK-24554 – AMI/ARI: Generate events on connected line

     changes (Reported by Matt Jordan)

* ASTERISK-24276 – [Patch] Option to make app MOH override channel

     musicclass (Reported by Kristian Høgh)

* ASTERISK-23871 – RLS Tests: Implement RLS off-nominal tests

     (Reported by Mark Michelson)

* ASTERISK-23823 – Option to keep queuerules in realtime

     (Reported by Michael K.)

Bugs fixed in this release:

———————————–

* ASTERISK-26227 – sqlalchemy error due to long identifier name

     (Reported by Mark Michelson)

* ASTERISK-26221 – chan_sip: iLBC does not include correct mode

     (Reported by Aaron Meriwether)

* ASTERISK-26216 – res_fax: Deadlock when detect fax while channel

     executing Playback (Reported by Richard Mudgett)

* ASTERISK-26214 – Allow arbitrary time for fax detection to end

     on a channel (Reported by Richard Mudgett)

* ASTERISK-23013 – Deadlock between ‘sip show channels’

     command and attended transfer handling (Reported by Ben

     Smithurst)

* ASTERISK-26212 – Makefile: Retain XML Declaration and

     DTD in docs. (Reported by Alexander Traud)

* ASTERISK-26211 – Unit tests: AST_TEST_DEFINE should be used in

     conditional code. (Reported by Corey Farrell)

* ASTERISK-26207 – sRTP: Count a roll-over of the sequence

     number even on lost packets. (Reported by Alexander Traud)

* ASTERISK-26038 – ‘make install’ doesn’t seem to install OS/X

     init files (Reported by Tzafrir Cohen)

* ASTERISK-26133 – app_queue: Queue members receive multiple calls

     (Reported by Richard Miller)

* ASTERISK-26196 – pbx: Time based includes can leak timezone

     string (Reported by Corey Farrell)

* ASTERISK-26193 – chan_sip: reference leak in mwi_event_cb

     (Reported by Corey Farrell)

* ASTERISK-26191 – threadpool: Leak on duplicate taskprocessor for

     ast_threadpool_serializer_group (Reported by Corey Farrell)

* ASTERISK-25659 – res_rtp_asterisk: ECDH not negotiated causing

     DTLS failure occurred on RTP instance (Reported by Edwin

     Vandamme)

* ASTERISK-26046 – Avoid obsolete warnings on autoconf.

     (Reported by Alexander Traud)

* ASTERISK-26160 – pjsip: Updated->Reachable during qualify

     (Reported by Matt Jordan)

* ASTERISK-26177 – func_odbc: Database handle is kept when it

     should be released (Reported by Leandro Dardini)

* ASTERISK-25289 – Build System does not respect CFLAGS and

     CXXFLAGS when building menuselect (Reported by Jeffrey Walton)

* ASTERISK-26119 – fix: memory leaks, resource leaks, out

     of bounds and bugs (Reported by Alexei Gradinari)

* ASTERISK-26184 – chan_sip: Reference leaks in error paths.

     (Reported by Corey Farrell)

* ASTERISK-26181 – REF_DEBUG: Node object incorrectly logged

     during duplicate replacement (Reported by Corey Farrell)

* ASTERISK-26179 – chan_sip: Second T.38 request fails (Reported

     by Joshua Colp)

* ASTERISK-26180 – PJSIP: provide valid tcp nodelay option for

     reuse (Reported by Scott Griepentrog)

* ASTERISK-25772 – res_pjsip: Unexpected two BYE when answered

     (Reported by Dmitriy Serov)

* ASTERISK-26099 – res_pjsip_pubsub: Crash when sending request

     due to server timeout (Reported by Ross Beer)

* ASTERISK-26144 – Crash on loading codecs g729/g723 (Reported by

     Alexei Gradinari)

* ASTERISK-26157 – Build:   Fix errors highlighted by GCC 6.x

     (Reported by George Joseph)

* ASTERISK-26021 – Build codecs siren7 and siren14 for Asterisk 13

     (Reported by Daniel Denson)

* ASTERISK-26141 – res_fax: fax_v21_session_new leaks reference to

     v21_details (Reported by Corey Farrell)

* ASTERISK-26140 – res_rtp_asterisk: gcc 6 caught a

     self-comparison (Reported by George Joseph)

* ASTERISK-26138 – chan_unistim:  Under FreeBSD, chan_unistim

     generates a compile error (Reported by George Joseph)

* ASTERISK-26128 – Alembic scripts are failing (Reported by Mark

     Michelson)

* ASTERISK-26139 – test_res_pjsip_scheduler:  Compile failure if

     pjproject isn’t installed in a system location (Reported by

     George Joseph)

* ASTERISK-26130 – WebRTC: Should use latest DTLS version.

     (Reported by Alexander Traud)

* ASTERISK-26132 – PJSIP: provide transport type with received

     messages (Reported by Scott Griepentrog)

* ASTERISK-26127 – res_pjsip_session: Crash due to race condition

     between res_pjsip_session unload and timer (Reported by Joshua

     Colp)

* ASTERISK-26045 – app_voicemail: fix bugs, imap mm_status

     log change to debug (Reported by Alexei Gradinari)

* ASTERISK-26083 – ARI: Announcer channels staying around after

     playback to a bridge is finished (Reported by Per Jensen)

* ASTERISK-26126 – leverage ‘bindaddr’ for TLS in

     http.conf (Reported by Alexander Traud)

* ASTERISK-26097 – CLI: show maximum file descriptors

     (Reported by Alexander Traud)

* ASTERISK-25262 – Memory leak when a caller channel does multiple

     dials and CEL is enabled (Reported by Etienne Lessard)

* ASTERISK-26047 – ARI allows certain commands to run on down

     channels. (Reported by Mark Michelson)

* ASTERISK-25959 –

     http_media_cache/retrieve_cache_control_directives: Sporadic

     failure (Reported by Joshua Colp)

* ASTERISK-26103 – cdr:  Assert on ‘dial end’ event during a blond

     transfer (Reported by George Joseph)

* ASTERISK-26092 – [Segfault] in res_rtp_asterisk.c:4268 after

     Remotely bridged channels (Reported by Niklas Larsson)

* ASTERISK-26089 – Invalid security events during boot using PJSIP

     Realtime (Reported by Scott Griepentrog)

* ASTERISK-26096 – res_hep: Crash when configuration file is

     missing (Reported by Niklas Larsson)

* ASTERISK-26074 – res_odbc: Deadlock within UnixODBC (Reported by

     Ross Beer)

* ASTERISK-26054 – Asterisk crashes (core dump) (Reported by B.

     Davis)

* ASTERISK-26069 – Asterisk truncates To: header, dropping the

     closing ‘>’ (Reported by Vasil Kolev)

* ASTERISK-24436 – Missing header in res/res_srtp.c when compiling

     against libsrtp-1.5.0 (Reported by Patrick Laimbock)

* ASTERISK-26091 – ar cru creates warning, instead use ar

     cr (Reported by Alexander Traud)

* ASTERISK-26070 – ari/channels:  Creating a local channel without

     an originator adds all audio formats to it’s capabilities

     (Reported by George Joseph)

* ASTERISK-26078 – core: Memory leak in logging (Reported by

     Etienne Lessard)

* ASTERISK-26065 – chan_pjsip: MWI NOTIFY contents not ordered

     properly (Reported by Ross Beer)

* ASTERISK-26063 – ${PJSIP_HEADER(read,Call-ID)} does not work –

     documentation needs clarification for when read/write is

     possible (Reported by Private Name)

* ASTERISK-25777 – data race in threadpool (Reported by Badalian

     Vyacheslav)

* ASTERISK-26053 – res_pjsip_outbound_publish: Crash when shutting

     down (Reported by Joshua Colp)

* ASTERISK-26049 – res_pjsip: Crash when our own request timer

     fires (Reported by Joshua Colp)

* ASTERISK-25669 – CURL incorrect trim for non ASCII

     characters (Reported by Jesper)

* ASTERISK-26029 – parking: ast_parking_park_call should return

     parking_space instead of parking_exten (Reported by Diederik de

     Groot)

* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin

     Vandamme)

* ASTERISK-25941 – chan_pjsip: Crash on an immediate SIP final

     response (Reported by Javier Riveros )

* ASTERISK-26014 – res_sorcery_astdb: Make tolerant of unknown

     fields (Reported by Joshua Colp)

* ASTERISK-24986 – keepalive INFO packages ignored by asterisk

     (Reported by Ilya Trikoz)

* ASTERISK-26034 – T.38 passthrough problem behind firewall due to

     early nosignal packet (Reported by George Joseph)

* ASTERISK-26030 – call cut because of double Session-Expires

     header in re-invite after proxy authentication is required

     (Reported by George Joseph)

* ASTERISK-25964 – Outbound registrations created via ARI/push

     configuration do not clean up outbound registrations currently

     in flight (Reported by Matt Jordan)

* ASTERISK-26005 – res_pjsip: Multiple SIP messages are combined

     into 1 TCP packet (Reported by Ross Beer)

* ASTERISK-25352 – res_hep_rtcp correlation_id is different then

     res_hep (Reported by Kevin Scott Adams)

* ASTERISK-26007 – res_pjsip: Endpoints deleting early after

     upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

* ASTERISK-25990 – PJSIP TLS registration should respect

     client_uri scheme when generating Contact URI (Reported by

     Sebastian Damm)

* ASTERISK-26008 – app_followme does not delete recorded name

     prompt (Reported by Tzafrir Cohen)

* ASTERISK-25978 – res_pjsip_authenticator_digest: Should not use

     source port in nonce verification (Reported by Mark Michelson)

* ASTERISK-26004 – res_pjsip:  The transport/method parameter is

     ignored (Reported by George Joseph)

* ASTERISK-25999 – res_pjsip_dialog_info_body_generator: Remove

     subscription requirement (Reported by Joshua Colp)

* ASTERISK-25993 – pjproject: Allow bundling to not require

     everything it does (Reported by Joshua Colp)

* ASTERISK-25998 – file: Crash when using nativeformats (Reported

     by Joshua Colp)

* ASTERISK-25826 – PJSIP / Sorcery slow load from realtime

     (Reported by Ross Beer)

* ASTERISK-25956 – Compilation error in conditionally compiled

     code in config_options.c (Reported by Chris Trobridge)

* ASTERISK-25968 – pjproject_bundled:  Configure and make need to

     be re-tested (Reported by George Joseph)

* ASTERISK-24463 – Voicemail email address corrupt or not sent

     when message is in the process of being recorded during reload

     (Reported by John Campbell)

* ASTERISK-25922 – res_pjsip_exten_state: Add configuration

     support for publishing (Reported by Joshua Colp)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25961 – tests/channels/SIP/sip_tls_call: Sporadic crash

     when running test (Reported by Joshua Colp)

* ASTERISK-16115 – problem with ringinuse=no, queue

     members receive sometimes two calls (Reported by nik600)

* ASTERISK-25917 – app_voicemail: passwordlocation=spooldir

     only works if you manually add secret.conf yourself (Reported by

     Jonathan R. Rose)

* ASTERISK-25954 – Manager QueueSummary and QueueStatus Actions

     are case sensitive to QueueName (Reported by Javier Acosta)

* ASTERISK-25951 – res_agi:  run_agi eats frames it shouldn’t

     (Reported by George Joseph)

* ASTERISK-25950 – SIP channel does not send PeerStatus

     events for autocreated peers (Reported by Kirill Katsnelson)

* ASTERISK-25927 – Removed option “registertrying” is still

     documented in sip.conf.sample (Reported by Etienne Lessard)

* ASTERISK-25947 – Protocol transfers to stasis applications are

     missing the StasisStart with the replace_channel object.

     (Reported by Richard Mudgett)

* ASTERISK-24649 – Pushing of channel into bridge fails; Stasis

     fails to get app name (Reported by John Bigelow)

* ASTERISK-24782 – StasisEnd event not present for channel that

     was swapped out for another after completing attended transfer

     (Reported by John Bigelow)

* ASTERISK-25942 – res_pjsip_caller_id: Transfer results in mixed

     ConnectedLine information (Reported by George Joseph)

* ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

     thread (Reported by Joshua Colp)

* ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

     not raised (Reported by Joshua Colp)

* ASTERISK-25934 – chan_sip should not require sipregs or

     updateable sippeers table unless rt (Reported by Jaco Kroon)

* ASTERISK-25888 – Frequent segfaults in function can_ring_entry()

     of app_queue.c (Reported by Sébastien Couture)

* ASTERISK-25796 – res_pjsip: DOS/Crash when TCP/TLS sockets

     exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George

     Joseph)

* ASTERISK-25707 – Long contact URIs or hostnames can crash

     pjproject/Asterisk under certain conditions (Reported by George

     Joseph)

* ASTERISK-25123 – Bracketed IPv6 Contact header parameter

     unparsable with Asterisk/PJSIP (Reported by Anthony Messina)

* ASTERISK-25874 – app_voicemail: Stack buffer overflow in

     test_voicemail_notify_endl (Reported by Badalian Vyacheslav)

* ASTERISK-25912 – chan_local passes AST_CONTROL_PVT_CAUSE_CODE

     without adding them to the local hangupcauses via

     ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)

* ASTERISK-25885 – res_pjsip: Race condition between adding

     contact and automatic expiration (Reported by Joshua Colp)

* ASTERISK-25910 – pjproject:  Via headers are not parsed when

     “received” contains an IPv6 address (Reported by George Joseph)

* ASTERISK-25890 – Asterisk 13.8.0 alembic database update fails

     (Reported by Harley Peters)

* ASTERISK-25894 – webrtc video broken due to missing

     marker bits in RTP streams (Reported by Jacek Konieczny)

* ASTERISK-25881 – pbx: Add support for autohints (Reported by

     Joshua Colp)

* ASTERISK-25854 – No audio after HOLD/RESUME – incorrect

     a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)

* ASTERISK-25868 – Sorcery “append to category” should allow

     filters (Reported by Nick Repin)

* ASTERISK-25873 – res_pjsip: Bundled pjproject: compile error,

     cannot find -lasteriskpj (Reported by Hans van Eijsden)

* ASTERISK-25882 – ARI: Crash can occur due to race condition when

     attempting to operate on a hung up channel (Part 2) (Reported by

     Richard Mudgett)

* ASTERISK-25867 – Video delay on app_echo (Reported by

     Jacek Konieczny)

* ASTERISK-24605 – res_parking option parkeddynamic does not work

     with the core Features ‘parkcall’ (DTMF initiated parking)

     (Reported by Philip Correia)

* ASTERISK-24596 – Unclear how to use Park application with

     res_parking ‘parkeddynamic’ enabled. Documentation? (Reported by

     Philip Correia)

* ASTERISK-25825 – Crashes during shutdown when running CLI

     commands (Reported by Mark Michelson)

* ASTERISK-24543 – Asterisk 13 responds to SIP Invite with all

     possible codecs configured for peer as opposed to intersection

     of configured codecs and offered codecs (Reported by Taylor

     Hawkes)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25510 – Log to syslog failing (Reported by

     Michael Newton)

* ASTERISK-25857 – func_aes: incorrect use of strlen() leads to

     data corruption (Reported by Gianluca Merlo)

* ASTERISK-25849 – chan_pjsip: transfers with direct media

     sometimes drops audio (Reported by Kevin Harwell)

* ASTERISK-25814 – Segfault at f ip in res_pjsip_refer.so

     (Reported by Sergio Medina Toledo)

* ASTERISK-25023 – Deadlock in chan_sip in

     update_provisional_keepalive (Reported by Arnd Schmitter)

* ASTERISK-25321 – DeadLock ChanSpy with call over Local

     channel (Reported by Filip Frank)

* ASTERISK-25829 – res_pjsip: PJSIP does not accept spaces when

     separating multiple AORs (Reported by Mateusz Kowalski)

* ASTERISK-25771 – ARI:Crash – Attended transfers of channels into

     Stasis application. (Reported by Javier Riveros )

* ASTERISK-25830 – Revision 2451d4e breaks NAT (Reported by Sean

     Bright)

* ASTERISK-25582 – Testsuite: Reactor timeout error in

     tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt

     Jordan)

* ASTERISK-25811 – Unable to delete object from sorcery cache

     (Reported by Ross Beer)

* ASTERISK-25800 – Calculate talktime when is first call

     answered (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25727 – RPM build requires OPTIONAL_API cflag due to

     PJSIP requirement (Reported by Gergely Dömsödi)

* ASTERISK-25337 – Crash on PJSIP_HEADER Add P-Asserted-Identity

     when calling from Gosub (Reported by Jacques Peacock)

* ASTERISK-25738 – res_pjsip_pubsub: Crash while executing

     OutboundSubscriptionDetail ami action (Reported by Kevin

     Harwell)

* ASTERISK-25721 – res_phoneprov: memory leak and

     heap-use-after-free (Reported by Badalian Vyacheslav)

* ASTERISK-25272 – The ICONV dialplan function sometimes

     returns garbage (Reported by Etienne Lessard)

* ASTERISK-25751 – res_pjsip: Support

     pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)

* ASTERISK-25606 – Core dump when using transports in sorcery

     (Reported by Martin Moučka)

* ASTERISK-20987 – non-admin users, who join muted conference are

     not being muted (Reported by hristo)

* ASTERISK-25737 – res_pjsip_outbound_registration: line option

     not in Alembic (Reported by Joshua Colp)

* ASTERISK-24972 – Transport Layer Security (TLS) Protocol BEAST

     Vulnerability – Investigate vulnerability of HTTP server

     (Reported by Alex A. Welzl)

* ASTERISK-25603 – udptl: Uninitialized lengths and bufs in

     udptl_rx_packet cause ast_frdup crash (Reported by Walter

     Doekes)

* ASTERISK-25742 – Secondary IFP Packets can result in accessing

     uninitialized pointers and a crash (Reported by Torrey Searle)

* ASTERISK-25397 – chan_sip: File descriptor leak with

     non-default timert1 (Reported by Alexander Traud)

* ASTERISK-25702 – PjSip realtime DB and Cache Errors since

     upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by

     Nic Colledge)

* ASTERISK-25730 – build:  make uninstall after make distclean

     tries to remove root (Reported by George Joseph)

* ASTERISK-25725 – core: Incorrect XML documentation may result in

     weird behavior (Reported by Joshua Colp)

* ASTERISK-25722 – ASAN & testsute: stack-buffer-overflow in

     sip_sipredirect (Reported by Badalian Vyacheslav)

* ASTERISK-25709 – ARI: Crash can occur due to race condition when

     attempting to operate on a hung up channel (Reported by Mark

     Michelson)

* ASTERISK-25714 – ASAN:heap-buffer-overflow in logger.c (Reported

     by Badalian Vyacheslav)

* ASTERISK-25685 – infrastructure: Run alembic in Jenkins build

     script (Reported by Joshua Colp)

* ASTERISK-25712 – Second call to already-on-call phone and

     Asterisk sends “Ready” (Reported by Richard Mudgett)

* ASTERISK-24801 – ASAN: ast_el_read_char stack-buffer-overflow

     (Reported by Badalian Vyacheslav)

* ASTERISK-25179 – CDR(billsec,f) and CDR(duration,f) report

     incorrect values (Reported by Gianluca Merlo)

* ASTERISK-25611 – core: threadpool thread_timeout_thrash unit

     test sporadically failing (Reported by Joshua Colp)

* ASTERISK-25686 – PJSIP: qualify_timeout is a double, database

     schema is an integer (Reported by Marcelo Terres)

* ASTERISK-25700 – main/config: Clean config maps on shutdown.

     (Reported by Corey Farrell)

* ASTERISK-25696 – bridge_basic: don’t cache xferfailsound during

     a transfer (Reported by Kevin Harwell)

* ASTERISK-25697 – bridge_basic: don’t play an attended transfer

     fail sound after target hangs up (Reported by Kevin Harwell)

* ASTERISK-25683 – res_ari: Asterisk fails to start if compiled

     with MALLOC_DEBUG  (Reported by yaron nahum)

* ASTERISK-24097 – Documentation – CHANNEL function help text

     missing ‘linkedid’ argument (Reported by Steven T. Wheeler)

* ASTERISK-25690 – Hanging up when executing connected line sub

     does not cause hangup (Reported by Joshua Colp)

* ASTERISK-25687 – res_musiconhold: Concurrent invocations of ‘moh

     reload’ cause a crash (Reported by Sean Bright)

* ASTERISK-25632 – res_pjsip_sdp_rtp: RTP is sent from wrong IP

     address when multihomed (Reported by Olivier Krief)

* ASTERISK-25637 – Multi homed server using wrong IP (Reported by

     Daniel Journo)

* ASTERISK-25394 – pbx: Incorrect device and presence state when

     changing hint details (Reported by Joshua Colp)

* ASTERISK-25640 – pbx: Deadlock on features reload and state

     change hint. (Reported by Krzysztof Trempala)

* ASTERISK-25681 – devicestate: Engine thread is not shut down

     (Reported by Corey Farrell)

* ASTERISK-25680 – manager: manager_channelvars is not cleaned at

     shutdown (Reported by Corey Farrell)

* ASTERISK-25679 – res_calendar leaks scheduler. (Reported by

     Corey Farrell)

* ASTERISK-25675 – Endpoint not listed as Unreachable (Reported by

     Daniel Journo)

* ASTERISK-25677 – pbx_dundi: leaks during failed load. (Reported

     by Corey Farrell)

* ASTERISK-25673 – res_crypto leaks CLI entries (Reported by Corey

     Farrell)

* ASTERISK-25668 – res_pjsip: Deadlock in distributor (Reported by

     Mark Michelson)

* ASTERISK-25664 – ast_format_cap_append_by_type leaks a reference

     (Reported by Corey Farrell)

* ASTERISK-25647 – bug of cel_radius.c: wrong point of

     ADD_VENDOR_CODE (Reported by Aaron An)

* ASTERISK-25137 – endpoint stasis messages are delivered twice

     (Reported by Vitezslav Novy)

* ASTERISK-25116 – res_pjsip:  Two PeerStatus AMI messages are

     sent for every status change (Reported by George Joseph)

* ASTERISK-25641 – bridge: GOTO_ON_BLINDXFR doesn’t work on

     transfer initiated channel (Reported by Dmitry Melekhov)

* ASTERISK-25614 – DTLS negotiation delays (Reported by Dade

     Brandon)

* ASTERISK-25625 – res_sorcery_memory_cache: Add full backend

     caching (Reported by Joshua Colp)

* ASTERISK-25601 – json: Audit reference usage and thread safety

     (Reported by Joshua Colp)

* ASTERISK-25624 – AMI Event OriginateResponse bug (Reported by

     sungtae kim)

* ASTERISK-25615 – res_pjsip: Setting transport async_operations >

     1 causes segfault on tls transports (Reported by George Joseph)

* ASTERISK-25442 – using realtime (mysql) queue members are never

     updated in wait_our_turn function (app_queue.c)  (Reported by

     Carlos Oliva)

* ASTERISK-25364 – Issue a TCP connection(kernel) and

     thread of asterisk is not released (Reported by Hiroaki Komatsu)

* ASTERISK-25569 – app_meetme: Audio quality issues (Reported by

     Corey Farrell)

* ASTERISK-25619 – res_chan_stats not sending the correct

     information to StatsD (Reported by Tyler Cambron)

* ASTERISK-24146 – No audio on WebRtc caller side when

     answer waiting time is more than ~7sec (Reported by Aleksei

     Kulakov)

* ASTERISK-25609 – Asterisk may crash when calling

     ast_channel_get_t38_state(c) (Reported by Filip Jenicek)

* ASTERISK-25599 – SLIN Resampling Codec only 80 msec

     (Reported by Alexander Traud)

* ASTERISK-25616 – Warning with a Codec Module which supports PLC

     with FEC (Reported by Alexander Traud)

* ASTERISK-25610 – Asterisk crash during “sip reload” (Reported by

     Dudás József)

* ASTERISK-25608 – res_pjsip/contacts/statsd:  Lifecycle events

     aren’t consistent (Reported by George Joseph)

* ASTERISK-25584 – format-attribute module: VP8 missing

     (Reported by Alexander Traud)

* ASTERISK-25583 – format-attribute module: RFC 7587 (Opus

     Codec) (Reported by Alexander Traud)

* ASTERISK-25498 – Asterisk crashes when negotiating g729 without

     that module installed (Reported by Ben Langfeld)

* ASTERISK-25595 – Unescaped : in messge sent to statsd (Reported

     by Niklas Larsson)

* ASTERISK-25598 – res_pjsip:  Contact status messages are

     printing a hash instead of the uri (Reported by George Joseph)

* ASTERISK-25600 – bridging: Inconsistency in BRIDGEPEER (Reported

     by Jonathan Rose)

* ASTERISK-25476 – chan_sip loses registrations after a while

     (Reported by Michael Keuter)

* ASTERISK-25593 – fastagi: record file closed after sending

     result (Reported by Kevin Harwell)

* ASTERISK-25585 – rasterisk never hits most of main(), but

     it’s assumed to (Reported by Walter Doekes)

* ASTERISK-25590 – CLI Usage info for ‘pjsip send notify’

     references incorrect config (Reported by Corey Farrell)

* ASTERISK-25165 – Testsuite – Sorcery memory cache leaks

     (Reported by Corey Farrell)

* ASTERISK-25575 – res_pjsip: Dynamic outbound registrations

     created via ARI are not loaded into memory on Asterisk

     start/restart (Reported by Matt Jordan)

* ASTERISK-25545 – translation module gets cached not

     joint format (Reported by Alexander Traud)

* ASTERISK-25573 – H.264 format attribute module: resets

     whole SDP (Reported by Alexander Traud)

* ASTERISK-24958 – Forwarding loop detection inhibits certain

     desirable scenarios (Reported by Mark Michelson)

* ASTERISK-25561 – app_queue.c line 6503 (try_calling): mutex

     ‘qe->chan’ freed more times than we’ve locked! (Reported by Alec

     Davis)

* ASTERISK-25565 – DNS: System resolver only returns 1 record per

     result (Reported by George Joseph)

* ASTERISK-25552 – hashtab: Improve NULL tolerance (Reported by

     Joshua Colp)

* ASTERISK-25160 – Opus Codec: SIP/SDP line fmtp missing

     when called internally (Reported by Alexander Traud)

* ASTERISK-25535 – format creation on module load instead

     of cache (Reported by Alexander Traud)

* ASTERISK-25449 – main/sched: Regression introduced by

     5c713fdf18f causes erroneous duplicate RTCP messages; other

     potential scheduling issues in chan_sip/chan_skinny (Reported by

     Matt Jordan)

* ASTERISK-25546 – threadpool: Race condition between idle timeout

     and activation (Reported by Joshua Colp)

* ASTERISK-25537 – format-attribute module: RFC or

     internal defaults? (Reported by Alexander Traud)

* ASTERISK-25533 – buffer for ast_format_cap_get_names

     only 64 bytes (Reported by Alexander Traud)

* ASTERISK-25373 –  add documentation for CALLERID(pres) and also

     the CONNECTEDLINE and REDIRECTING variants (Reported by Walter

     Doekes)

* ASTERISK-25528 – DNS: System resolver issues with TTL parse

     (Reported by dtryba)

* ASTERISK-25527 – Quirky xmldoc description wrapping (Reported by

     Walter Doekes)

* ASTERISK-24779 – Passthrough OPUS codec not working with

     chan_pjsip (Reported by PowerPBX)

* ASTERISK-25522 – ARI: Crash when creating channel via ARI

     originate with requesting channel (Reported by Matt Jordan)

* ASTERISK-25434 – Compiler flags not reported in ‘core show

     settings’ despite usage during compilation (Reported by Rusty

     Newton)

* ASTERISK-24106 – WebSockets Automatically decides what driver it

     will use  (Reported by Andrew Nagy)

* ASTERISK-25513 – Crash: malloc failed with high load of

     subscriptions. (Reported by John Bigelow)

* ASTERISK-25505 – res_pjsip_pubsub: Crash on off-nominal when UAS

     dialog can’t be created (Reported by Joshua Colp)

* ASTERISK-25494 – build:  GCC 5.1.x catches some new const, array

     bounds and missing paren issues (Reported by George Joseph)

* ASTERISK-25485 – res_pjsip_outbound_registration: registration

     stops due to 400 response (Reported by Kevin Harwell)

* ASTERISK-25486 – res_pjsip: Fix deadlock when validating URIs

     (Reported by Joshua Colp)

* ASTERISK-7803 – Update the maximum packetization values

     in frame.c (Reported by dea)

* ASTERISK-25484 – autoframing=yes has no effect (Reported

     by Alexander Traud)

* ASTERISK-25308 – ari: Websocket leak (Reported by Joshua Colp)

* ASTERISK-25461 – Nested dialplan #includes don’t work as

     expected. (Reported by Richard Mudgett)

* ASTERISK-25455 – Deadlock of PJSIP realtime over

     res_config_pgsql  (Reported by mdu113)

* ASTERISK-25135 – RTP Timeout hangup cause code missing

     (Reported by Olle Johansson)

* ASTERISK-25108 – configure check for older unbound library

     (Reported by John Bigelow)

* ASTERISK-25435 – Asterisk periodically hangs. UDP Recv-Q greatly

     exceeds zero. (Reported by Dmitriy Serov)

* ASTERISK-25451 – Broken video – erased rtp marker bit (Reported

     by Stefan Engström)

* ASTERISK-25400 – Hints broken when “CustomPresence” doesn’t

     exist in AstDB (Reported by Andrew Nagy)

* ASTERISK-25443 – IPv6 – Potential issue in via header

     parsing (Reported by ffs)

* ASTERISK-25404 – segfault/crash in chan_pjsip_hangup … at

     chan_pjsip.c (Reported by Chet Stevens)

* ASTERISK-25391 – AMI GetConfigJSON returns invalid JSON

     (Reported by Bojan Nemčić)

* ASTERISK-25441 – Deadlock in res_sorcery_memory_cache. (Reported

     by Richard Mudgett)

* ASTERISK-25438 – res_rtp_asterisk: ICE role message even when

     ICE is not enabled (Reported by Joshua Colp)

* ASTERISK-25383 – Core dumps on startup and shutdown with

     MALLOC_DEBUG enabled (Reported by yaron nahum)

* ASTERISK-25423 – Caller gets no Connected line update during

     call pickup. (Reported by Richard Mudgett)

* ASTERISK-25305 – Dynamic logger channels can be added multiple

     times (Reported by Mark Michelson)

* ASTERISK-25418 – On-hold channels redirected out of a bridge

     appear to still be on hold (Reported by Mark Michelson)

* ASTERISK-25384 – Regular Asterisk crashes when using Page

     application. “user_data is NULL” (Reported by Chet Stevens)

* ASTERISK-25410 – app_record: RECORDED_FILE variable not being

     populated (Reported by Kevin Harwell)

* ASTERISK-25396 – chan_sip: Extremely long callerid name causes

     invalid SIP (Reported by Walter Doekes)

* ASTERISK-25399 – app_queue: AgentComplete event has wrong reason

     (Reported by Kevin Harwell)

* ASTERISK-25185 – Segfault in app_queue on transfer scenarios

     (Reported by Etienne Lessard)

* ASTERISK-25353 – Transcoding while different in Frame

     size = Frames lost (Reported by Alexander Traud)

* ASTERISK-25325 – ARI PUT reload chan_sip HTTP response 404

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25390 – default_from_user can crash with certain

     configuration backends (Reported by Mark Michelson)

* ASTERISK-25387 – res_pjsip_nat: Malformed REGISTER request

     causes NAT’d Contact header to not be rewritten (Reported by

     Matt Jordan)

* ASTERISK-25227 – No audio at in-band announcements in ooh323

     channel (Reported by Alexandr Dranchuk)

* ASTERISK-25295 – res_pjsip crash – pjsip_uri_get_uri at

     /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)

* ASTERISK-25381 – res_pjsip: AoRs deleted via ARI (or other

     mechanism) do not destroy their related contacts (Reported by

     Matt Jordan)

* ASTERISK-25369 – res_parking: ParkAndAnnounce – Inheritable

     variables aren’t applied to the announcer channel (Reported by

     Jonathan Rose)

* ASTERISK-25367 – pbx: Long pattern match hints may cause “core

     show hints” to crash (Reported by Joshua Colp)

* ASTERISK-25365 – Persistent subscriptions have extra

     Content-Length/corrupted messages (Reported by Mark Michelson)

* ASTERISK-25356 – res_pjsip_sdp_rtp: Multiple keepalive scheduled

     items may exist (Reported by Joshua Colp)

* ASTERISK-25355 – sched: ast_sched_del may return prematurely due

     to spurious wakeup (Reported by Joshua Colp)

* ASTERISK-25318 –

     tests/rest_api/applications/subscribe-endpoint/nominal/resource:

     Sporadically failing (Reported by Joshua Colp)

* ASTERISK-25346 – chan_sip: Overwriting answered elsewhere hangup

     cause on call pickup (Reported by Joshua Colp)

* ASTERISK-25342 – res_pjsip: Repeated usage of pj_gethostip may

     block (Reported by Joshua Colp)

* ASTERISK-25341 – bridge: Hangups may get lost when executing

     actions (Reported by Joshua Colp)

* ASTERISK-25339 – res_pjsip: Empty “auth” sections from

     non-config backgrounds are interpreted as valid (Reported by

     Matt Jordan)

* ASTERISK-25215 – Differences in queue.log between Set

     QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne

     Gaetz)

* ASTERISK-25322 – Crash occurs when using MixMonitor with t() or

     r() options. (Reported by Richard Mudgett)

* ASTERISK-25320 – chan_sip.c: sip_report_security_event searches

     for wrong or non existent peer on invite (Reported by Kevin

     Harwell)

* ASTERISK-25312 – res_http_websocket: Terminate connection on

     fatal cases (Reported by Joshua Colp)

* ASTERISK-25315 – DAHDI channels send shortened duration DTMF

     tones. (Reported by Richard Mudgett)

* ASTERISK-25306 – Persistent subscriptions can save multiple SIP

     messages at once, leading to potential crashes. (Reported by

     Mark Michelson)

* ASTERISK-25309 – iLBC 20 advertised (Reported by

     Alexander Traud)

* ASTERISK-25304 – res_pjsip: XML sanitization may write past

     buffer (Reported by Joshua Colp)

* ASTERISK-25265 – DTLS Failure when calling WebRTC-peer on

     Firefox 39 – add ECDH support and fallback to prime256v1

     (Reported by Stefan Engström)

* ASTERISK-24988 – func_talkdetect: Test is bouncing sporadically

     (Reported by Joshua Colp)

* ASTERISK-25181 – ARI: Channels added to Stasis application

     during WebSocket creation don’t receive a StasisStart event

     (Reported by Matt Jordan)

* ASTERISK-25296 – RTP performance issue with several channel

     drivers. (Reported by Richard Mudgett)

* ASTERISK-25297 – Crashes running

     channels/pjsip/resolver/srv/failover/in_dialog testsuite tests

     (Reported by Richard Mudgett)

* ASTERISK-25292 – Testuite:

     tests/apps/bridge/bridge_wait/bridge_wait_e_options fails

     (Reported by Kevin Harwell)

* ASTERISK-25271 – Parking & blind transfer: Transferer channel

     not hung up if no MOH (Reported by Kevin Harwell)

* ASTERISK-25250 – chan_sip – Despite the channel being answered,

     caller on a call established via Local channel continues to hear

     ringback (Reported by Etienne Lessard)

* ASTERISK-25253 – confbridge volume options and other volume

     controls such as func_volume don’t work (Reported by Dmitriy

     Serov)

* ASTERISK-25247 – choppy audio when spying on a g722 channel,

     chan_sip or chan_pjsip (Reported by hristo)

* ASTERISK-25263 – cdr_adaptive_odbc: CDR insert failure

     due to reversed if logic (Reported by Elazar Broad)

* ASTERISK-24867 – Docs for ‘e’ option in ResetCDR say to use

     CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty

     Newton)

* ASTERISK-24853 – Documentation claims chan_sip outbound

     registrations support WS or WSS as valid transports (not true)

     (Reported by PSDK)

* ASTERISK-25242 – PJSIP: No audio when Asterisk inside NAT and

     endpoints outside NAT – implement functionality similar to

     chan_sip ‘rtpkeepalive’? (Reported by Mark Michelson)

* ASTERISK-25258 – chan_pjsip: Incorrect format switch on received

     RTP packet (Reported by Joshua Colp)

* ASTERISK-25257 – channels/sig_pri.h -> sig_pri_span ->

     force_restart_unavailable_chans in wrong scope (Reported by

     Patric Marschall)

* ASTERISK-24934 – Asterisk manager output does not escape

     control characters (Reported by warren smith)

* ASTERISK-25255 – Missing AMI VarSet events when setting to an

     empty string. (Reported by Richard Mudgett)

* ASTERISK-25254 – Crash if dialplan sets ATTENDEDTRANSFER to an

     empty string before Park. (Reported by Richard Mudgett)

* ASTERISK-25183 – PJSIP: Crash on NULL channel in

     chan_pjsip_incoming_response despite previous checks for NULL

     channel (Reported by Matt Jordan)

* ASTERISK-25201 – Crash in PJSIP distributor on already free’d

     threadpool (Reported by Matt Jordan)

* ASTERISK-25240 – bridge_native_rtp: Direct media wrongfully

     started when completing attended transfer (Reported by Joshua

     Colp)

* ASTERISK-25103 – Roundup – investigate Asterisk DTLS crashes

     (Reported by Rusty Newton)

* ASTERISK-25146 – DNS: Create system level resolver (Reported by

     Joshua Colp)

* ASTERISK-22805 – res_rtp_asterisk: Crash when calling

     BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP

     (Reported by Dmitry Burilov)

* ASTERISK-24550 – res_rtp_asterisk: Crash in

     ast_rtp_on_ice_complete during DTLS handshake (Reported by

     Osaulenko Alexander)

* ASTERISK-24651 – Fix race condition in DTLS (Reported by

     Badalian Vyacheslav)

* ASTERISK-24832 – DTLS-crashes within openssl  (Reported

     by Stefan Engström)

* ASTERISK-25127 – DTLS crashes following “Unable to cancel

     schedule ID” in dtls_srtp_check_pending (Reported by Dade

     Brandon)

* ASTERISK-25168 – Random Core Dumps on Asterisk 13.4 PJSIP, in

     ast_channel_name at channel_internal_api.c (Reported by Carl

     Fortin)

* ASTERISK-25076 – res_pjsip: Failover does not occur on

     connection-less transport or 503 response (Reported by Joshua

     Colp)

* ASTERISK-25226 – chan_sip: Channel leak in branch 13 on early

     replaces call pickup (Reported by Walter Doekes)

* ASTERISK-25222 – Crash in recurring cancel callback called from

     ast_dns_resolve_cancel on junk pointer (Reported by Matt Jordan)

* ASTERISK-25220 – Closing of fd -1 in chan_mgcp.c

     (Reported by Walter Doekes)

* ASTERISK-25219 – Source and destination overlap in memcpy

     in rtp_engine.c (Reported by Walter Doekes)

* ASTERISK-25212 – Segfault when using DEBUG_FD_LEAKS

     (Reported by Walter Doekes)

* ASTERISK-19277 – endlessly repeating error: “poll failed:

     Bad file descriptor” (Reported by Barry Chern)

* ASTERISK-25202 – Hints extension state broken between 13.3.2 and

     13.4 (Reported by cervajs)

* ASTERISK-25196 – res_pjsip_nat: rewrite_contact should not be

     applied to Contact header when Record-Route headers are present

     (Reported by Mark Michelson)

* ASTERISK-24907 – res_pjsip_outbound_registration: crash during

     unload if registration attempts are still occuring (Reported by

     Kevin Harwell)

* ASTERISK-25204 – res_pjsip_refer: Duplicated Referred-By or

     Replaces headers on outbound INVITEs. (Reported by Mark

     Michelson)

* ASTERISK-25189 – AMI: Add Linkedid header to standard channel

     snapshot information. (Reported by Richard Mudgett)

* ASTERISK-25171 – Early completion of feature code attended

     transfer results in intermittent one-way audio, “ghost ringing”

     and robotic sound. (Reported by Rusty Newton)

* ASTERISK-25172 – Crash in channels/sip/sip blind

     transfer/caller_refer_only test in

     ast_format_cap_append_from_cap during ast_request (Reported by

     Matt Jordan)

* ASTERISK-25180 – res_pjsip_mwi: Unsolicited MWI requires reload

     (Reported by Joshua Colp)

* ASTERISK-25182 – on CLI sip reload, new codecs get

     appended only (Reported by Alexander Traud)

* ASTERISK-25163 – Deadlock in chan_sip between reload of sip peer

     container and MWI Stasis callback (Reported by Dmitriy Serov)

* ASTERISK-25091 – Asterisk REST API – bridge.addChannel crash

     asterisk when calling channel hangup while adding to bridge

     (Reported by Ilya Trikoz)

* ASTERISK-24900 – Manager event ParkedCallSwap is not documented

     (Reported by Rusty Newton)

* ASTERISK-25162 – func_pjsip_aor: Leak of contact in iterator

     (Reported by Corey Farrell)

* ASTERISK-25158 – res_pjsip: Add option to use AAL2 packing when

     negotiating g.726 (Reported by Kevin Harwell)

* ASTERISK-24344 – CDR_PROP(disable) disables CDR only for first

     dialed party (Reported by Janusz Karolak)

* ASTERISK-24443 – CDR fields (dst, dcontext) empty in transfer

     call started from Macro (Reported by Arveno Santoro)

* ASTERISK-25154 – fromtag may need to be updated after

     successful call dialog match (Reported by Damian Ivereigh)

* ASTERISK-25156 – chan_pjsip’s CHAN_START cel event lacks the

     correct context and exten (Reported by cloos)

* ASTERISK-25157 – bridging: Performing a blonde transfer does not

     result in connected line updates (Reported by Joshua Colp)

* ASTERISK-25087 – Asterisk segfault when using Directory

     application with alias option and specific mailbox configuration

     (Reported by Chet Stevens)

* ASTERISK-25115 – Crash related to func

     sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c

     (Reported by John Bigelow)

* ASTERISK-25096 – Segfault when registering over

     websockets with PJSIP (in ast_sockaddr_isnull at

     /include/asterisk/netsock2.h) (Reported by Josh Kitchens)

* ASTERISK-24963 – ASAN: heap-use-after-free with PJSIP and WSS

     (Reported by Badalian Vyacheslav)

* ASTERISK-22559 – gcc 4.6 and higher supports weakref attribute

     but asterisk doesn’t detect it. (Reported by ibercom)

* ASTERISK-25094 – PBX core: Investigate thread safety issues

     (Reported by Corey Farrell)

* ASTERISK-25113 – install_prereq in Debian 8 without “standard

     system utilities” (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25148 – res_pjsip NULL channel audit (Reported by Mark

     Michelson)

* ASTERISK-25131 – chan_pjsip: In-dialog authentication not

     handled. (Reported by Richard Mudgett)

* ASTERISK-24717 – ASAN: global-buffer-overflow codec_{ilbc | gsm

     | adpcm | ipc10} (Reported by Badalian Vyacheslav)

* ASTERISK-25100 – asterisk coredump if host has an IPv6 address

     that end with ::80 (Reported by Mark Petersen)

* ASTERISK-25122 – Large SIP packet received via pjsip over

     websocket crashes Asterisk  (Reported by Ivan Poddubny)

* ASTERISK-25121 – Stasis: Fix unsafe use of stasis_unsubscribe in

     modules. (Reported by Corey Farrell)

* ASTERISK-25120 – Astobj2: Weakproxy subscriptions should be run

     in reverse order. (Reported by Corey Farrell)

* ASTERISK-25105 – res_pjsip:  Possible incompatibility between

     qualify_timeout and pjproject-2.4 (Reported by George Joseph)

* ASTERISK-25117 – res_mwi_external_ami: Fix manager action

     registrations. (Reported by Corey Farrell)

* ASTERISK-25112 – Logger: Configuration settings are not reset to

     default during reload. (Reported by Corey Farrell)

* ASTERISK-24983 – IAX deadlock between hangup and scheduled

     actions (ex. largrq) (Reported by Y Ateya)

* ASTERISK-24944 – main/audiohook.c change prevents G722 call

     recording (Reported by Ronald Raikes)

* ASTERISK-25110 – res_resolver_unbound.c compilation failure:

     SIGURG is undeclared in func unbound_resolver_stop (Reported by

     John Bigelow)

* ASTERISK-24887 – tags in a=crypto lines do not accept 2

     or more digits (Reported by Makoto Dei)

* ASTERISK-25086 – PJSIP crashes if endpoint missing in

     Dial() (Reported by snuffy)

* ASTERISK-25089 – res_pjsip_config_wizard: Variable specified in

     templates aren’t being processed correctly (Reported by George

     Joseph)

* ASTERISK-25090 – CLI core show channel truncates cdr variables

     (Reported by snuffy)

* ASTERISK-25083 – Message.c: Message channel becomes saturated

     with frames leading to spammy log messages (Reported by Jonathan

     Rose)

* ASTERISK-25085 – Potential crash after unload of

     func_periodic_hook or test_message (Reported by Corey Farrell)

* ASTERISK-25082 – Asterisk deletes message after doing a playback

     of an INBOX message using ast_vm_play when the Old folder is

     full for that mailbox. (Reported by Jonathan Rose)

* ASTERISK-21893 – Segfault after call hangup, in

     ast_channel_hangupcause_set, at channel_internal_api.c (Reported

     by Aleksandr Gordeev)

* ASTERISK-25042 – asterisk.conf options override command-line

     options. (Reported by Corey Farrell)

* ASTERISK-25074 – Regression: Recent clang-related change broke

     cross compiling of Asterisk (Reported by Sebastian Kemper)

* ASTERISK-24442 – Outgoing call files don’t work properly when

     set in the future (Reported by tootai)

* ASTERISK-18252 – queue_log mysql time column data format

     (Reported by Gareth Blades)

* ASTERISK-25041 – Broken column type checking in

     res_config_mysql addon (Reported by Alexandre Fournier)

* ASTERISK-25057 – res_pjsip_pubsub: Crash in send_notify due to

     invalid root pointer in sub_tree (Reported by Matt Jordan)

* ASTERISK-24938 – ARI Snoop Channel results in excessive

     escalating CPU usage (Reported by George Ladoff)

* ASTERISK-25034 – chan_dahdi: Some telco switches occasionally

     ignore ISDN RESTART requests. (Reported by Richard Mudgett)

* ASTERISK-25003 – Asterisk crashes on attended transfer (using

     feature) (Reported by Artem Volodin)

* ASTERISK-25038 – Queue log “EXITWITHTIMEOUT” does not always

     contain waiting time (Reported by Etienne Lessard)

* ASTERISK-25027 – Build System: Many ARI modules are missing

     dependencies. (Reported by Corey Farrell)

* ASTERISK-25061 – pbx_config: Register manager actions with

     module version of macro. (Reported by Corey Farrell)

* ASTERISK-24967 – Problem support schema for pgsql on CEL

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25025 – Periodic crashes (in

     ast_channel_snapshot_create at stasis_channels.c) with Certified

     Asterisk 13. (Reported by Chet Stevens)

* ASTERISK-25053 – Unit test category /main/presence missing

     trailing slash. (Reported by Corey Farrell)

* ASTERISK-22708 – res_odbc.conf negative_connection_cache option

     not respected, failover between DSNs doesn’t work (Reported by

     JoshE)

* ASTERISK-25054 – Formats interface’s cannot be unregistered,

     needs to hold modules until shutdown. (Reported by Corey

     Farrell)

* ASTERISK-24976 – cdr_odbc not include new columns added on 1.8

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25033 – Asterisk 13 (branch head) won’t compile without

     PJSip (Reported by Peter Whisker)

* ASTERISK-24896 – Using force black background leads to

     colours not being reset (Reported by dant)

* ASTERISK-25048 – Astobj2: Initialization order wrong when both

     refdebug and AO2_DEBUG are both enabled. (Reported by Corey

     Farrell)

* ASTERISK-19608 – Asterisk-1.8.x  starts rejecting calls with

     cause code 44 after some time. (Reported by Denis Alberto

     Martinez)

* ASTERISK-25037 – res_pjsip_outbound_registration: Potential

     crash in off-nominal failure case when sending message (Reported

     by Joshua Colp)

* ASTERISK-25022 – Memory leak setting up DTLS/SRTP calls

     (Reported by Steve Davies)

* ASTERISK-22790 – check_modem_rate() may return incorrect rate

     for V.27 (Reported by not here)

* ASTERISK-23231 – Since 405693 If we have res_fax.conf file set

     to minrate=2400, then res_fax refuse to load (Reported by David

     Brillert)

* ASTERISK-24955 – res_fax: v.27ter support baud rate of 2400,

     which is disallowed in res_fax’s check_modem_rate (Reported by

     Matt Jordan)

* ASTERISK-25020 – Mismatched response to outgoing REGISTER

     request (Reported by Mark Michelson)

* ASTERISK-25028 – Build System: Unneeded defines in

     asterisk/buildopts.h (Reported by Corey Farrell)

* ASTERISK-25026 – Git conversion: Non-C files not switched to

     ASTERISK_REGISTER_FILE (Reported by Corey Farrell)

* ASTERISK-24996 – chan_pjsip: Creating Channel Causes Asterisk to

     Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported

     by Ashley Sanders)

* ASTERISK-25018 – pjsip show endpoints crashes asterisk when

     qualified aors present (Reported by Ivan Poddubny)

* ASTERISK-24749 – ConfBridge: Wrong language on playing

     conf-hasjoin and conf-hasleft when played to bridge (Reported by

     Philippe Bolduc)

* ASTERISK-24845 – pjsip send notify not working with Cisco phone

     (Reported by Carl Fortin)

* ASTERISK-25004 – Crash in authenticated reinvite after

     originated T.38 FAX (Reported by Mark Michelson)

* ASTERISK-24999 – PJSIP crashes with malformed contact line

     (Reported by snuffy)

* ASTERISK-24998 – res_corosync:  res_corosync tries to load even

     if res_corosync.conf is missing (Reported by George Joseph)

* ASTERISK-24997 – Astobj2: Some callers of __adjust_lock do not

     pre-check the object (Reported by Corey Farrell)

* ASTERISK-24994 – dns: Query set unit tests are failing due to

     race condition (Reported by Joshua Colp)

* ASTERISK-24982 – res_pjsip_mwi: Unsolicited MWI NOTIFY only sent

     on mailbox changes (Reported by Joshua Colp)

* ASTERISK-24991 – Check for ao2_alloc failure in

     __ast_channel_internal_alloc (Reported by Corey Farrell)

* ASTERISK-24895 – After hangup on the side of the ISDN network no

     HangupRequest event comes for the dahdi channel. (Reported by

     Andrew Zherdin)

* ASTERISK-24977 – Contacts that don’t use qualify are being

     marked as unavailable (Reported by George Joseph)

* ASTERISK-24774 – Segfault in ast_context_destroy with

     extensions.ael and extensions.conf (Reported by Corey Farrell)

* ASTERISK-24841 – ConfBridge: Strange sampling rates chosen when

     channels have multiple native formats (Reported by Matt Jordan)

* ASTERISK-24975 – Enabling ‘DEBUG_THREADLOCALS’ Causes the Build

     to Fail (Reported by Ashley Sanders)

* ASTERISK-24863 – res_pjsip: No endpoint events raised via AMI

     when contacts cannot be reached/qualified (Reported by Dmitriy

     Serov)

* ASTERISK-24869 – Asterisk segfaults on DAHDI attended transfer

     due to application (appl) being NULL on unbridged channel

     (Reported by viniciusfontes)

* ASTERISK-24970 – Crash in res_pjsip_pubsub handling of failed

     notify (Reported by Scott Griepentrog)

* ASTERISK-13271 – menuselect sets defaults too late (Reported by

     John Nemeth)

* ASTERISK-24959 – CLI command cdr show pgsql status

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-20524 – AMI improperly handles lines of exactly 1025

     characters (Reported by David M. Lee)

* ASTERISK-24936 – New Feature: AO2 weakproxy objects (Reported by

     Corey Farrell)

* ASTERISK-24954 – Git migration: Asterisk version numbers are

     incompatible with the Test Suite (Reported by Matt Jordan)

* ASTERISK-17608 – func_aes.so cannot be loaded if res_crypto /

     openssl not compiled (Reported by Warren Selby)

* ASTERISK-24928 – t38_udptl_maxdatagram in pjsip.conf not

     honored (Reported by Juergen Spies)

* ASTERISK-24835 – Early Media Not working with Chan SIP and

     Asterisk 13 (Reported by Andrew Nagy)

* ASTERISK-21777 – Asterisk tries to transcode video instead of

     audio (Reported by Nick Ruggles)

* ASTERISK-24380 – core: Native formats are set to h264 with

     certain audio/video codec configuration, resulting in path

     translation WARNINGs (Reported by Matt Jordan)

* ASTERISK-22352 – IAX2 custom qualify timer is not taken

     into account (Reported by Frederic Van Espen)

* ASTERISK-24894 – iax2_poke_noanswer expiration timer too

     short (Reported by Y Ateya)

* ASTERISK-24935 – res_pjsip_phoneprov_provider: Fix leaked

     OBJ_MULTIPLE iterator. (Reported by Corey Farrell)

* ASTERISK-23319 – Segmentation fault in queue_exec at app_queue.c

     (Reported by Vadim)

* ASTERISK-24933 – T38 fails negotiation (Reported by Jonathan

     Rose)

* ASTERISK-24847 – [security] tcptls: certificate CN NULL

     byte prefix bug (Reported by Matt Jordan)

* ASTERISK-21211 – chan_iax2 – unprotected access of

     iaxs[peer->callno] potentially results in segfault (Reported by

     Jaco Kroon)

* ASTERISK-18032 – – IPv6 and IPv4 NAT not working

     (Reported by Christoph Timm)

* ASTERISK-24910 – “timer=no” and “timer=required” settings in

     pjsip.conf fail (Reported by Ray Crumrine)

* ASTERISK-24932 – Asterisk 13.x does not build with GCC 5.0

     (Reported by Jeffrey C. Ollie)

* ASTERISK-24914 – Division by zero in file.c when playback of

     voicemail with video as h264 (Reported by Marcello Ceschia)

* ASTERISK-24899 – Parking fall-through behavior different in 13

     (Reported by Malcolm Davenport)

* ASTERISK-24937 – res_pjsip_messaging: Messages may be

     sent out of order (Reported by Mark Michelson)

* ASTERISK-24920 – Asterisk handles duplicate SIP requests as if

     they were each a new request (Reported by Mark Michelson)

* ASTERISK-24781 – PJSIP: Unnecessary 180 Ringing messages sent

     with undesireabe consequences. (Reported by Richard Mudgett)

* ASTERISK-24857 – “timing test”, pjsip incoming/outgoing

     calls, voicemail prompts and recordings all fail when using the

     kqueue timer source on FreeBSD 10.x (Reported by Justin T.

     Gibbs)

* ASTERISK-24155 – Non-portable and non-reliable recursion

     detection in ast_malloc (Reported by Timo Teräs)

* ASTERISK-24142 – CCSS: crash during shutdown due to device

     lookup in destroyed container (Reported by David Brillert)

* ASTERISK-24683 – Crash in PBX ast_hashtab_lookup_internal during

     core restart now (Reported by Peter Katzmann)

* ASTERISK-24805 – – ASAN: Race condition

     (heap-use-after-free) on asterisk closing (Reported by Badalian

     Vyacheslav)

* ASTERISK-24881 – ast_register_atexit should only be used when

     absolutely needed (Reported by Corey Farrell)

* ASTERISK-24731 – res_pjsip_session cannot be unloaded (Reported

     by Corey Farrell)

* ASTERISK-24864 – app_confbridge: file playback blocks dtmf

     (Reported by Kevin Harwell)

* ASTERISK-14233 – Buddies are always auto-registered when

     processing the roster (Reported by Simon Arlott)

* ASTERISK-24780 – – Buddies are always auto-registered

     when processing the roster (Reported by Simon Arlott)

* ASTERISK-24879 – Compilation fails due to 64bit time

     under OpenBSD (Reported by snuffy)

* ASTERISK-24880 – Compilation under OpenBSD  (Reported by

     snuffy)

* ASTERISK-21765 – – FILE function’s length argument

     counts from beginning of file rather than the offset (Reported

     by John Zhong)

* ASTERISK-24817 – init_logger_chain: unreachable code block

     (Reported by Corey Farrell)

* ASTERISK-24882 – chan_sip: Improve usage of REF_DEBUG (Reported

     by Corey Farrell)

* ASTERISK-24876 – Investigate reference leaks from

     tests/channels/local/local_optimize_away (Reported by Corey

     Farrell)

* ASTERISK-24840 – res_pjsip: conflicting endpoint identifiers

     (Reported by Kevin Harwell)

* ASTERISK-16779 – Cannot disallow unknown format ” (Reported by

     Atis Lezdins)

* ASTERISK-18708 – func_curl hangs channel under load (Reported by

     Dave Cabot)

* ASTERISK-21038 – Bad command completion of “core set debug

     channel” (Reported by Richard Kenner)

* ASTERISK-19470 – Documentation on app_amd is incorrect (Reported

     by Frank DiGennaro)

* ASTERISK-24872 – AMI PJSIPShowEndpoint closes AMI

     connection on error (Reported by Dmitriy Serov)

* ASTERISK-23666 – CLONE – nested functions aren’t portable

     (Reported by Diederik de Groot)

* ASTERISK-20399 – Compilation on some systems requires the

     -fnested-functions flag (Reported by David M. Lee)

* ASTERISK-20850 – Nested functions aren’t portable.

     Adapting RAII_VAR to use clang/llvm blocks to get the

     same/similar functionality. (Reported by Diederik de Groot)

* ASTERISK-24807 – Missing mandatory field Max-Forwards (Reported

     by Anatoli)

* ASTERISK-24808 – res_config_odbc: Improper escaping of

     backslashes occurs with MySQL (Reported by Javier Acosta)

* ASTERISK-23390 – NewExten Event with application AGI shows up

     before and after AGI runs (Reported by Benjamin Keith Ford)

* ASTERISK-24786 – – Asterisk terminates when playing a

     voicemail stored in LDAP (Reported by Graham Barnett)

* ASTERISK-24739 – – Out of files — call fails —

     numerous files with inodes from under /usr/share/zoneinfo,

     mostly posixrules (Reported by Ed Hynan)

* ASTERISK-24755 – Asterisk sends unexpected early BYE to

     transferrer during attended transfer when using a Stasis bridge

     (Reported by John Bigelow)

* ASTERISK-24830 – res_rtp_asterisk.c checks USE_PJPROJECT not

     HAVE_PJPROJECT (Reported by Stefan Engström)

* ASTERISK-24825 – Caller ID not recognized using

     Centrex/Distinctive dialing (Reported by Richard Mudgett)

* ASTERISK-17588 – Caller ID on TDM410P *UK* PSTN (Reported by

     Daniel Flounders)

* ASTERISK-24838 – chan_sip: Locking inversion occurs when

     building a peer causes a peer poke during request handling

     (Reported by Richard Mudgett)

* ASTERISK-24751 – Integer values in json payload to ARI cause

     asterisk to crash (Reported by jeffrey putnam)

* ASTERISK-24828 – Fix Frame Leaks (Reported by Kevin Harwell)

* ASTERISK-18105 – most of asterisk modules are unbuildable in

     cygwin environment (Reported by feyfre)

* ASTERISK-21845 – maxcalls exceeded, Asterisk sends out 480 and

     also BYE (Reported by Tony Ching)

* ASTERISK-15434 – When ast_pbx_start failed, both an

     error response and BYE are sent to the caller (Reported by

     Makoto Dei)

* ASTERISK-23214 – chan_sip WARNING message ‘We are requesting

     SRTP for audio, but they responded without it’ is ambiguous and

     wrong in some cases (Reported by Rusty Newton)

* ASTERISK-17721 – Incoming SRTP calls that specify a key lifetime

     fail (Reported by Terry Wilson)

* ASTERISK-20233 – SRTP not working with some devices (Eg

     Grandstream gxv3175) – Message “Can’t provide secure audio

     requested in SDP offer” (Reported by tootai)

* ASTERISK-22748 – SRTP Crypto Offer With Lifetime Not Accepted

     (Reported by Alejandro Mejia)

* ASTERISK-24800 – Crash in __sip_reliable_xmit due to invalid

     thread ID being passed to pthread_kill (Reported by JoshE)

* ASTERISK-24812 – ARI: Creating channels through /channels

     resource always uses SLIN, which results in unneeded transcoding

     (Reported by Matt Jordan)

* ASTERISK-24797 – bridge_softmix: G.729 codec license held

     (Reported by Kevin Harwell)

* ASTERISK-24677 – ARI GET variable on channel provides unhelpful

     response on non-existent variable (Reported by Joshua Colp)

* ASTERISK-24785 – ‘Expires’ header missing from 200 OK on

     REGISTER (Reported by Ross Beer)

* ASTERISK-24499 – Need more explicit debug when PJSIP dialstring

     is invalid (Reported by Rusty Newton)

* ASTERISK-24724 – ‘httpstatus’ Web Page Produces Incomplete HTML

     (Reported by Ashley Sanders)

* ASTERISK-24796 – Codecs and bucket schema’s prevent module

     unload (Reported by Corey Farrell)

* ASTERISK-24814 – asterisk/lock.h: Fix syntax errors for non-gcc

     OSX with 64 bit integers (Reported by Corey Farrell)

* ASTERISK-24787 – – Microsoft exchange incompatibility

     for playing back messages stored in IMAP – play_message: No

     origtime (Reported by Graham Barnett)

* ASTERISK-22670 – Asterisk crashes when processing ISDN AoC

     Events (Reported by klaus3000)

* ASTERISK-24689 – Segfault on hangup after outgoing PRI-Euroisdn

     call (Reported by Marcel Manz)

* ASTERISK-24740 – Segmentation fault on aoc-e event

     (Reported by Panos Gkikakis)

* ASTERISK-24799 – make fails with undefined reference to

     SSLv3_client_method (Reported by Alexander Traud)

* ASTERISK-24451 – chan_iax2: reference leak in sched_delay_remove

     (Reported by Corey Farrell)

* ASTERISK-24700 – CRASH: NULL channel is being passed to

     ast_bridge_transfer_attended() (Reported by Zane Conkle)

* ASTERISK-24791 – Crash in ast_rtcp_write_report (Reported by

     JoshE)

* ASTERISK-24085 – Documentation – We should remove or further

     document the ‘contact’ section in pjsip.conf (Reported by Rusty

     Newton)

* ASTERISK-24632 – install_prereq script installs pjproject

     without IPv6 support (Reported by Rusty Newton)

* ASTERISK-24685 – “pjsip show version” CLI command (Reported by

     Joshua Colp)

* ASTERISK-24768 – res_timing_pthread: file descriptor leak

     (Reported by Matthias Urlichs)

* ASTERISK-24612 – res_pjsip: No information if a required sorcery

     wizard is not loaded (Reported by Joshua Colp)

* ASTERISK-24716 – Improve pjsip log messages for presence

     subscription failure (Reported by Rusty Newton)

* ASTERISK-24771 – ${CHANNEL(pjsip)} – segfault (Reported by

     Niklas Larsson)

* ASTERISK-24727 – PJSIP: Crash experienced during multi-Asterisk

     transfer scenario. (Reported by Mark Michelson)

* ASTERISK-24015 – app_transfer fails with PJSIP channels

     (Reported by Private Name)

* ASTERISK-24741 – dtls_handler causes Asterisk to crash (Reported

     by Zane Conkle)

* ASTERISK-24701 – Stasis: Write timeout on WebSocket fails to

     fully disconnect underlying socket, leading to events being

     dropped with no additional information (Reported by Matt Jordan)

* ASTERISK-24752 – Crash in bridge_manager_service_req when bridge

     is destroyed by ARI during shutdown (Reported by Richard

     Mudgett)

* ASTERISK-24772 – ODBC error in realtime sippeers when device

     unregisters under MariaDB (Reported by Richard Miller)

* ASTERISK-24479 – Enable REF_DEBUG for module references

     (Reported by Corey Farrell)

* ASTERISK-24742 – Fix ast_odbc_find_table function in

     res_odbc (Reported by ibercom)

* ASTERISK-24769 – res_pjsip_sdp_rtp: Local ICE candidates leaked

     (Reported by Matt Jordan)

* ASTERISK-24748 – res_pjsip: If wizards explicitly configured in

     sorcery.conf false ERROR messages may occur (Reported by Joshua

     Colp)

* ASTERISK-24616 – Crash in res_format_attr_h264 due to invalid

     string copy (Reported by Yura Kocyuba)

* ASTERISK-24737 – When agent not logged in, agent status shows

     unavailable, queue status shows agent invalid (Reported by

     Richard Mudgett)

* ASTERISK-24635 – PJSIP outbound PUBLISH crashes when no response

     is ever received (Reported by Marco Paland)

* ASTERISK-24736 – Memory Leak Fixes (Reported by Mark Michelson)

* ASTERISK-24646 – PJSIP changeset 4899 breaks TLS (Reported by

     Stephan Eisvogel)

* ASTERISK-24711 – DTLS handshake broken with latest OpenSSL

     versions (Reported by Jared Biel)

* ASTERISK-24666 – Security Vulnerability: RTP not closed after

     sip call using unsupported codec (Reported by Y Ateya)

* ASTERISK-24676 – Security Vulnerability: URL request injection

     in libCURL (CVE-2014-8150) (Reported by Matt Jordan)

* ASTERISK-24729 – Outbound registration not occuring on new

     registrations after reload. (Reported by Richard Mudgett)

* ASTERISK-24728 – tcptls: Bad file descriptor error when

     reloading chan_sip (Reported by Kevin Harwell)

* ASTERISK-24721 – manager: ModuleLoad action incorrectly reports

     ‘module not found’ during a Reload operation (Reported by Matt

     Jordan)

* ASTERISK-24715 – chan_sip: stale nonce causes failure (Reported

     by Kevin Harwell)

* ASTERISK-24485 – res_pjsip cannot be unloaded or shutdown

     (Reported by Corey Farrell)

* ASTERISK-24719 – ConfBridge recording channels get stuck when

     recording started/stopped more than once (Reported by Richard

     Mudgett)

* ASTERISK-24723 – confbridge: CLI command ‘confbridge list XXXX’

     no longer displays user menus (Reported by Matt Jordan)

* ASTERISK-24539 – Compile fails on OSX because of sem_timedwait

     in bridge_channel.c (Reported by George Joseph)

* ASTERISK-24544 – Compile fails on OSX Yosemite because of

     incorrect detection of htonll and ntohll (Reported by George

     Joseph)

* ASTERISK-24231 – crash: CLI execution of realtime destroy

     sippeers id 1 causes crash due to NULL name provided to

     ast_variable (Reported by Niklas Larsson)

* ASTERISK-24626 – Voicemail passwords not being stored in ARA

     (Reported by Paddy Grice)

* ASTERISK-24693 – Investigate and fix memory leaks in Asterisk

     (Reported by Kevin Harwell)

* ASTERISK-24355 – chan_sip realtime uses case sensitive

     column comparison for ‘defaultuser’ (Reported by

     HZMI8gkCvPpom0tM)

* ASTERISK-24709 – msg_create_from_file used by MixMonitor

     m() option does not queue an MWI event (Reported by Gareth

     Palmer)

* ASTERISK-24673 – outgoing sip registers cannot be removed or

     modified without doing restart (or doing module unload

     chan_sip.so) (Reported by Stefan Engström)

* ASTERISK-24640 – Registration pending stays forever after sip

     reload (Reported by Max Man)

* ASTERISK-24682 – app_dial: Multiple DialEnd events emitted when

     MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported

     by Matt Jordan)

* ASTERISK-24560 – Creating a named ARI bridge twice causes a

     crash (Reported by Kinsey Moore)

* ASTERISK-24600 – Stuck IAX channels, Asterisk stops responding

     to most traffic, potential deadlock (Reported by Jeff Collell)

* ASTERISK-24048 – contrib/scripts/install_prereq selects

     32-bit packages on 64-bit hosts (Reported by Ben Klang)

* ASTERISK-24288 – – ODBC usage with app_voicemail –

     voicemail is not deleted after review, hangup (Reported by LEI

     FU)

* ASTERISK-24615 – When Multiple Transports Exist in pjsip.conf,

     Incorrect External Addresses is Used in SIP Packets When

     Responding to INVITE (Reported by David Justl)

* ASTERISK-24624 – Transfer to invalid extension results in hung

     channel. (Reported by Zane Conkle)

* ASTERISK-24663 – Unnamed semaphore autoconf check fails

     on cross compilation (Reported by abelbeck)

* ASTERISK-24655 – res_pjsip_outbound_publish: Hang on shutdown

     while attempting to publish (Reported by Kevin Harwell)

* ASTERISK-23991 – asterisk.pc file contains a small error

     in the CFlags returned (Reported by Diederik de Groot)

* ASTERISK-23850 – Park Application does not respect Return

     Context Priority (Reported by Andrew Nagy)

* ASTERISK-24665 – Configure check required for

     pjsip_get_dest_info() (Reported by Mark Michelson)

* ASTERISK-24049 – Asterisk Manager Interface: A number of list

     type responses aren’t using astman_send_listack (Reported by

     Jonathan Rose)

* ASTERISK-20744 – Security event logging does not work

     over syslog (Reported by Michael Keuter)

* ASTERISK-24672 – [PATCH] Memory leak in func_curl CURLOPT

     (Reported by Kristian Høgh)

* ASTERISK-24474 – sip_to_pjsip.py lacks documentation and does

     not function (Reported by John Kiniston)

* ASTERISK-24637 – Channel re-enters Stasis() when it should not

     (Reported by John Bigelow)

* ASTERISK-24591 – Stasis() side of an ARI originated channel

     cannot be Redirected (Reported by Kinsey Moore)

* ASTERISK-24376 – res_pjsip_refer: REFER request for remote

     session attempts to direct channel to external_replaces

     extension instead of context, without providing for the

     Referred-To SIP URI (Reported by Matt Jordan)

* ASTERISK-24513 – Local channel apparently leaked in off-nominal

     DTMF attended transfer (Reported by Mark Michelson)

* ASTERISK-24367 – PJSIP: allow all results in failure to send

     INVITE (Reported by Scott Griepentrog)

* ASTERISK-24267 – Queue variables associated with

     setinterfacevar, setqueueentryvar, setqueuevar are not passed to

     local channel (Reported by Mitch Claborn)

* ASTERISK-24641 – Deadlock in Trunk (Reported by Malcolm

     Davenport)

* ASTERISK-23841 – DTMF atxfer doesn’t set CallerID for the recall

     calls to the transferrer. (Reported by Richard Mudgett)

* ASTERISK-24628 – chan_sip – CANCEL is sent to wrong

     destination when ‘sendrpid=yes’ (in proxy environment) (Reported

     by Karsten Wemheuer)

* ASTERISK-23733 – ‘reload acl’ fails if acl.conf is not present

     on startup (Reported by Richard Kenner)

* ASTERISK-24566 – Uninit buf in WS write (Reported by Badalian

     Vyacheslav)

* ASTERISK-24337 – Spammy DEBUG message needs to be at a higher

     level – ‘Remote address is null, most likely RTP has been

     stopped’ (Reported by Rusty Newton)

* ASTERISK-24459 – bridge_native_rtp: Native RTP bridging is

     chosen for RTP compatible channels when the DTMF mode is not

     compatible (Reported by Yaniv Simhi)

* ASTERISK-24536 – AMI redirect with PJSIP fails to move extra

     channel (Reported by Niklas Larsson)

* ASTERISK-24619 – Gcc 4.10 fixes in r413589 (1.8) wrongly

     casts char to unsigned int (Reported by Walter Doekes)

* ASTERISK-24449 – Reinvite for T.38 UDPTL fails if SRTP is

     enabled (Reported by Andreas Steinmetz)

* ASTERISK-22455 – Asterisk 12 on Ubuntu Lucid deadlocks with

     DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)

* ASTERISK-24614 – Deadlock when DEBUG_THREADS compiler flag

     enabled (Reported by Richard Mudgett)

* ASTERISK-24604 – res_rtp_asterisk: Crash during restart due to

     race condition in accessing codec in stored ast_frame and codec

     core (Reported by Matt Jordan)

* ASTERISK-24563 – Direct Media calls within private network

     sometimes get one way audio (Reported by Kevin Harwell)

* ASTERISK-24607 – res_pjsip_session: re-INVITE with declined

     media streams results in 488 (Reported by Matt Jordan)

* ASTERISK-24472 – Asterisk Crash in OpenSSL when calling over WSS

     from JSSIP (Reported by Badalian Vyacheslav)

* ASTERISK-24514 – res_pjsip_outbound_registration: stack overflow

     when using non-default sorcery wizard (Reported by Kevin

     Harwell)

* ASTERISK-24342 – PJSIP: Qualifying endpoints attempts to do them

     all at the same time. (Reported by Richard Mudgett)

* ASTERISK-24556 – Asterisk 13 core dumps when calling from pjsip

     extension to another pjsip extension  (Reported by Abhay Gupta)

* ASTERISK-24537 – Stasis: StasisStart/StasisEnd events are not

     reliably transmitted during transfers (Reported by Matt Jordan)

* ASTERISK-24573 – Out of sync conversation recording when

     divided in multiple recordings (Reported by Nuno Borges)

* ASTERISK-24572 – App_meetme is loaded without its

     defaults when the configuration file is missing (Reported by

     Nuno Borges)

* ASTERISK-24516 – Asterisk segfaults when playing back

     voicemail under high concurrency with an IMAP backend (Reported

     by David Duncan Ross Palmer)

* ASTERISK-24274 – Codec Format Is Not Included in the SDP

     Media Attributes When SLIN48 Codec Is Used (Reported by Frankie

     Chin)

* ASTERISK-24533 – 2 threads created per chan_sip entry (Reported

     by xrobau)

* ASTERISK-24542 – Failure showing codecs via ‘core show

     channeltype <tech>’ (Reported by snuffy)

* ASTERISK-24469 – Security Vulnerability: Mixed IPv4/IPv6 ACLs

     allow blocked addresses through (Reported by Matt Jordan)

* ASTERISK-24534 – Register DB() as escalating to prevent

     users from writing to astdb (Reported by Gareth Palmer)

* ASTERISK-24531 – res_pjsip_acl: ACLs not applied on initial

     module load (Reported by Matt Jordan)

* ASTERISK-24490 – Security Vulnerability: CONFBRIDGE function’s

     record_command option allows arbitrary parameters to be passed

     to MixMonitor, allowing remote execution of commands (Reported

     by Matt Jordan)

* ASTERISK-24528 – res_pjsip_refer: Sending INVITE with Replaces

     in-dialog with invalid target causes crash (Reported by Joshua

     Colp)

* ASTERISK-24471 – Crash – assert_fail in libc in

     pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2

     (Reported by yaron nahum)

* ASTERISK-24535 – stringfields: Fix regression from fix for

     unintentional memory retention and another issue exposed by the

     fix (Reported by Corey Farrell)

* ASTERISK-24508 – pjsip – REFER request from SNOM is rejected

     with “400 bad request” – DEBUG shows “Received a REFER without a

     parseable Refer-To” (Reported by Beppo Mazzucato)

* ASTERISK-15242 – transmit_refer leaks sip_refer structures

     (Reported by David Woolley)

* ASTERISK-24522 – ConfBridge: delay occurs between kicking all

     endmarked users when last marked user leaves (Reported by Matt

     Jordan)

* ASTERISK-23651 – Reloading some modules that are loaded already,

     results in ‘No such module’ before a successful reload (Reported

     by Rusty Newton)

* ASTERISK-24336 – PJSIP timer_min_se value under 90 causes crash

     (Reported by Leon Rowland)

* ASTERISK-24501 – ARI: Moving a channel between bridges followed

     by a hangup can cause an ARI client to not receive an expected

     ChannelLeftBridge event before StasisEnd (Reported by Matt

     Jordan)

* ASTERISK-24489 – Crash: Asterisk crashes when converting RTCP

     packet to JSON for res_hep_rtcp and report blocks are greater

     than 1 (Reported by Gregory Malsack)

* ASTERISK-24498 – Segmentation fault in res_hep_rtcp on attended

     transfer (Reported by Beppo Mazzucato)

* ASTERISK-24281 – When bridging 2 chan_sip channels, MOH not

     removed from on-hold channels and bridge is never destroyed

     after hangup. (Reported by Stefan Engström)

* ASTERISK-24444 – PBX: Crash when generating extension for

     pattern matching hint (Reported by Leandro Dardini)

* ASTERISK-24502 – Build fails when dev-mode, dont optimize and

     coverage are enabled (Reported by Corey Farrell)

* ASTERISK-24505 – manager: http connections leak references

     (Reported by Corey Farrell)

* ASTERISK-24500 – Regression introduced in chan_mgcp by SVN

     revision r227276 (Reported by Xavier Hienne)

* ASTERISK-24468 – Incoming UCS2 encoded SMS truncated if SMS

     length exceeds 50 (roughly) national symbols (Reported by

     Dmitriy Bubnov)

* ASTERISK-24250 – Voicemail with multi-recipients To:

     header fix (Reported by abelbeck)

* ASTERISK-24504 – chan_console: Fix reference leaks to pvt

     (Reported by Corey Farrell)

* ASTERISK-24447 – Bridge DTMF hooks: Audio doesn’t pass when

     waiting for more matching digits. (Reported by Richard Mudgett)

* ASTERISK-24257 – agent must dial acceptdtmf twice to bridge to

     queue caller (Reported by Steve Pitts)

* ASTERISK-24492 – main/file.c: ast_filestream sometimes causes

     extra calls to ast_module_unref (Reported by Corey Farrell)

* ASTERISK-24491 – Memory leak in res_hep (Reported by Zane

     Conkle)

* ASTERISK-24307 – Unintentional memory retention in stringfields

     (Reported by Etienne Lessard)

* ASTERISK-24438 – res_pjsip_multihomed.so blocks Asterisk reload

     when DNS settings invalid (Reported by Melissa Shepherd)

* ASTERISK-20127 – [Regression] Config.c config_text_file_load()

     unescapes semicolons (“\;” -> “;”) turning them into comments

     (corruption) on rewrite of a config file (Reported by George

     Joseph)

* ASTERISK-24487 – configuration: sections should be loadable as

     template even when not marked (Reported by Scott Griepentrog)

* ASTERISK-24482 – func_talkdetect: Fix stasis message leak in

     audiohook callback (Reported by Corey Farrell)

* ASTERISK-24480 – res_http_websockets: Module reference decrease

     below zero (Reported by Corey Farrell)

* ASTERISK-24476 – main/app.c / app_voicemail: ast_writestream

     leaks (Reported by Corey Farrell)

* ASTERISK-24411 – Status of outbound registration is not

     changed upon unregistering. (Reported by John Bigelow)

* ASTERISK-24432 – Install refcounter.py when REF_DEBUG is enabled

     (Reported by Corey Farrell)

* ASTERISK-24466 – app_queue: fix a couple leaks to struct

     call_queue (Reported by Corey Farrell)

* ASTERISK-24465 – audiohooks list leaks reference to formats

     (Reported by Corey Farrell)

* ASTERISK-24462 – res_pjsip: Stale qualify statistics after

     disablementation (Reported by Kevin Harwell)

* ASTERISK-24190 – IMAP voicemail causes segfault (Reported by

     Nick Adams)

* ASTERISK-24304 – asterisk crashing randomly because of unistim

     channel (Reported by dhanapathy sathya)

* ASTERISK-21721 – SIP Failed to parse multiple Supported: headers

     (Reported by Olle Johansson)

* ASTERISK-24458 – chan_phone fails to build on big endian systems

     (Reported by Tzafrir Cohen)

* ASTERISK-24457 – res_fax: fax gateway frames leak (Reported by

     Corey Farrell)

* ASTERISK-24453 – manager: acl_change_sub leaks (Reported by

     Corey Farrell)

* ASTERISK-24437 – Review implementation of ast_bridge_impart for

     leaks and document proper usage (Reported by Scott Griepentrog)

* ASTERISK-24430 – missing letter “p” in word response in

     OriginateResponse event documentation (Reported by Dafi Ni)

* ASTERISK-24323 – Bug in documentation AGI STREAM FILE CONTROL

     (Reported by Martin Cisárik)

* ASTERISK-24419 – Incorrect syntax for setting language in

     configs/extensions.conf.sample (Reported by Ben Klang)

* ASTERISK-24454 – app_queue: ao2_iterator not destroyed, causing

     leak (Reported by Corey Farrell)

* ASTERISK-24455 – func_cdr: CDR_PROP leaks payload (Reported by

     Corey Farrell)

* ASTERISK-24435 – Asterisk 13 with TC400P segfault (Reported by

     Marian Koniuszko)

* ASTERISK-24425 – jabber/xmpp to use TLS instead of

     SSLv3, security fix POODLE (CVE-2014-3566) (Reported by

     abelbeck)

* ASTERISK-24122 – Documentaton for res_pjsip option use_avpf

     needs to be fixed (Reported by James Van Vleet)

* ASTERISK-24381 – res_pjsip_sdp_rtp: Declined media streams are

     interpreted, leading to erroneous 488 rejections (Reported by

     Matt Jordan)

* ASTERISK-24063 – Asterisk does not respect outbound proxy

     when sending qualify requests (Reported by Damian Ivereigh)

* ASTERISK-24415 – Missing AMI VarSet events when channels inherit

     variables. (Reported by Richard Mudgett)

* ASTERISK-24327 – bridge_native_rtp: Smart bridge operation to

     softmix sometimes fails to properly re-INVITE remotely bridged

     participants (Reported by Matt Jordan)

* ASTERISK-24426 – CDR Batch mode: size used as time value after

     first expire (Reported by Shane Blaser)

* ASTERISK-24312 – SIGABRT when improperly configured realtime

     pjsip  (Reported by Dafi Ni)

* ASTERISK-23846 – Unistim multilines. Loss of voice after second

     call drops (on a second line). (Reported by Rustam Khankishyiev)

* ASTERISK-24413 – parking/parking_tests: Crash due to assertion

     in unit tests when MoH is started on channel in holding bridge

     (Reported by Matt Jordan)

* ASTERISK-24393 – rtptimeout=0 doesn’t disable rtptimeout

     (Reported by Dmitry Melekhov)

* ASTERISK-24321 – SIP deadlock when running automated queues

     tests (Reported by Steve Pitts)

* ASTERISK-24392 – res_fax: fax gateway sessions leak (Reported by

     Corey Farrell)

* ASTERISK-24237 – CDR: FRACK With PJSIP blonde transfer.

     (Reported by Richard Mudgett)

* ASTERISK-24394 – CDR: FRACK with PJSIP directed pickup.

     (Reported by Richard Mudgett)

* ASTERISK-18923 – res_fax_spandsp usage counter is wrong

     (Reported by Grigoriy Puzankin)

* ASTERISK-22791 – asterisk sends Re-INVITE after receiving a BYE

     (Reported by not here)

* ASTERISK-13797 – relax badshell tilde test (Reported by

     Tzafrir Cohen)

* ASTERISK-24325 – res_calendar_ews: cannot be used with neon 0.30

     (Reported by Tzafrir Cohen)

* ASTERISK-24406 – Some caller ID strings are parsed differently

     since 11.13.0 (Reported by Etienne Lessard)

* ASTERISK-24387 – res_pjsip: rport sent from UAS MUST include the

     port that the UAC sent the request on (Reported by Matt Jordan)

* ASTERISK-20784 – Failure to receive an ACK to a SIP Re-INVITE

     results in a SIP channel leak (Reported by NITESH BANSAL)

* ASTERISK-15879 – Failure to receive an ACK to a SIP

     Re-INVITE results in a SIP channel leak (Reported by Torrey

     Searle)

* ASTERISK-24383 – res_rtp_asterisk: Crash if no candidates

     received for component (Reported by Kevin Harwell)

* ASTERISK-24011 – safe_asterisk tries to set ulimit -n too

     high on linux systems with lots of RAM (Reported by Michael

     Myles)

* ASTERISK-24326 – res_rtp_asterisk: ICE-TCP candidates are

     incorrectly attempted (Reported by Joshua Colp)

* ASTERISK-24389 – chan_iax2: Unit test on Bamboo failing

     (Reported by Kevin Harwell)

* ASTERISK-24398 – Initialize auth_rejection_permanent on client

     state to the configuration parameter value (Reported by Matt

     Jordan)

* ASTERISK-24354 – AMI sendMessage closes AMI connection on error

     (Reported by Peter Katzmann)

* ASTERISK-24224 – When using Bridge() dialplan application,

     surrogate channel appears in list and call count is inflated.

     (Reported by Mark Michelson)

* ASTERISK-24370 – res_pjsip/pjsip_options: OPTIONS request sent

     to Asterisk with no user in request is always 404’d (Reported by

     Matt Jordan)

* ASTERISK-24382 – chan_pjsip: Calling PJSIP_MEDIA_OFFER on a

     non-PJSIP channel results in an invalid reference of a channel

     pvt and a FRACK (Reported by Matt Jordan)

* ASTERISK-24369 – res_pjsip: Large message on reliable transport

     can cause empty messages to be passed from the PJSIP stack up,

     causing crashes in multiple locations (Reported by Matt Jordan)

* ASTERISK-24368 – res_pjsip_pubsub: Subscription persistence

     causes crash when re-constructing stored subscription (Reported

     by Matt Jordan)

* ASTERISK-24378 – Release AMI connections on shutdown (Reported

     by Corey Farrell)

* ASTERISK-24384 – chan_motif: format capabilities leak on module

     load error (Reported by Corey Farrell)

* ASTERISK-24199 – ‘ALL’ is specified in pjsip.conf.sample for TLS

     cipher but it is not valid (Reported by Joshua Colp)

* ASTERISK-24195 – bridge_native_rtp: Removing mixmonitor from a

     native RTP capable smart bridge doesn’t cause the bridge to

     resume being a native rtp bridge (Reported by Jonathan Rose)

* ASTERISK-24356 – PJSIP: Directed pickup causes deadlock

     (Reported by Richard Mudgett)

* ASTERISK-24262 – AMI CoreShowChannel missing several output

     fields and event documentation (Reported by Mitch Claborn)

* ASTERISK-23781 – outgoing missing as enum from

     contrib/ast-db-manage/config (Reported by Stephen More)

* ASTERISK-24222 – PJSIP: Failed assertions when placing a call

     with no allow= specified (Reported by Mark Michelson)

* ASTERISK-24362 – res_hep leaks reference to configuration

     (Reported by Corey Farrell)

* ASTERISK-22945 – Memory leaks in chan_sip.c with

     realtime peers (Reported by ibercom)

* ASTERISK-24350 – PJSIP shows commands prints unneeded headers

     (Reported by snuffy)

* ASTERISK-20567 – bashism in autosupport (Reported by Tzafrir

     Cohen)

* ASTERISK-24357 – [fax] Out of bounds error in update_modem_bits

     (Reported by Jeremy Lainé)

* ASTERISK-24348 – Built-in editline tab complete segfault with

     MALLOC_DEBUG (Reported by Walter Doekes)

* ASTERISK-23768 – Asterisk man page contains a (new)

     unquoted minus sign (Reported by Jeremy Lainé)

* ASTERISK-24295 – crash: creating out of dialog OPTIONS request

     crashes (Reported by Rogger Padilla)

* ASTERISK-24335 – [PATCH] Asterisk incorrectly responds 503 to

     INVITE retransmissions of rejected calls (Reported by Torrey

     Searle)

* ASTERISK-24339 – Swagger API Docs have incorrect basePath

     (Reported by Bradley Watkins)

* ASTERISK-24265 – segfault in asterisk when try to make call to

     IAX  (Reported by Dafi Ni)

* ASTERISK-24290 – Endpoint identifier match value fails to parse

     when CIDR network format is specified (Reported by Ray Crumrine)

* ASTERISK-24301 – Security: Out of call MESSAGE requests

     processed via Message channel driver can crash Asterisk

     (Reported by Matt Jordan)

* ASTERISK-24136 – Security: Crash in Asterisk’s PJSIP code when

     subscribing to an event with an unexpected body type (Reported

     by Mark Michelson)

* ASTERISK-24161 – PJSIPShowEndpoint gives inaccurate count of

     list items (Reported by Mark Michelson)

* ASTERISK-24331 – Unexpected Errors in Asterisk Manager Interface

     Output (Reported by xrobau)

* ASTERISK-24328 – Use of MixMonitor ‘m’ option results in 0

     duration vm description file  (Reported by Scott Griepentrog)

* ASTERISK-23577 – res_rtp_asterisk: Crash in

     ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by

     Jay Jideliov)

* ASTERISK-23634 – With TURN Asterisk crashes on multiple (7-10)

     concurrent WebRTC (avpg/encryption/icesupport) calls (Reported

     by Roman Skvirsky)

* ASTERISK-24249 – SIP debugs do not stop (Reported by Avinash

     Mohod)

* ASTERISK-24181 – RLS: Large lists don’t get sent because they

     exceed the PJSIP message length limit (Reported by Jonathan

     Rose)

* ASTERISK-24254 – CDRs: Application/args/dialplan CEP updated

     during dial operation (Reported by Matt Jordan)

* ASTERISK-24241 – crash: CDRs recursively attempt to update Party

     B information in a multi-party bridge, overrunning the stack

     (Reported by Deepak Singh Rawat)

* ASTERISK-24208 – Channels with CDR Information Remain Active

     Even After ConfBrige Is Ended (Reported by Frankie Chin)

* ASTERISK-24223 – Gibberish Call-ID on Local channel on

     origination (Reported by Mark Michelson)

* ASTERISK-24271 – Unable to make WebRTC call through chan_PJSIP

     nor chan_SIP (Reported by Dafi Ni)

* ASTERISK-24212 – testsuite: Sporadic crash due to assert on

     stopping RTP engine (Reported by Matt Jordan)

* ASTERISK-24264 – ARI: Adding a channel to a holding bridge

     automatically starts MOH (Reported by Samuel Galarneau)

* ASTERISK-23767 – Dynamic IAX2 registration stops trying

     if ever not able to resolve (Reported by David Herselman)

* ASTERISK-24280 – Add ‘rtpbindaddr’ setting for chan_sip

     (Reported by Paul Belanger)

* ASTERISK-24019 – When a Music On Hold stream starts it restarts

     at beginning of file. (Reported by Jason Richards)

* ASTERISK-24143 – pjsip: Outbound call to WebRTC UA fails to

     transmit ACK on received 200 OK (Reported by Aleksei Kulakov)

* ASTERISK-23997 – chan_sip: port incorrectly incremented for RTCP

     ICE candidates in SDP answer (Reported by Badalian Vyacheslav)

* ASTERISK-24147 – ARI: channel hangup crashes asterisk process

     (Reported by Edvin Vidmar)

* ASTERISK-23994 – res_pjsip_sdp_rtp: owner address in SDP may not

     be fully qualified domainname (Reported by Private Name)

* ASTERISK-22252 – res_musiconhold cleanup – REF_DEBUG reload

     warnings and ref leaks (Reported by Walter Doekes)

* ASTERISK-24178 – fromdomainport used even if not set

     (Reported by Elazar Broad)

* ASTERISK-24229 – ARI: playback of sounds implicitly answers

     channel, preventing early media playback (Reported by Matt

     Jordan)

* ASTERISK-24245 – gcc 4.1.2 complains of files that do not end

     with newlines (Reported by Shaun Ruffell)

* ASTERISK-24246 – Quiet warning about type qualifiers ignored on

     function return type (Reported by Shaun Ruffell)

* ASTERISK-24043 – ARI /continue fails to actually continue into

     the dialplan (Reported by Krandon Bruse)

* ASTERISK-24215 – testsuite: ARI Live Dangerously test fails due

     to wrong response code from Asterisk (Reported by Matt Jordan)

* ASTERISK-24134 – ARI: GET /channels/{channel_id}/variable for

     channel in dialplan returns 409 conflict (Reported by Matt

     Jordan)

* ASTERISK-24138 – dial: Call forwarding information presented

     through AMI/ARI is wrong (Reported by Matt Jordan)

* ASTERISK-24234 – app_meetme: Crash on conference shutdown due to

     NULL channel passed to meetme_stasis_generate_msg() (Reported by

     Shaun Ruffell)

* ASTERISK-24225 – Dial option z is broken (Reported by

     dimitripietro)

* ASTERISK-24032 – Gentoo compilation emits warning:

     “_FORTIFY_SOURCE” redefined (Reported by Kilburn)

* ASTERISK-24027 – MixMonitor AMI action called during AGI

     execution from bridge feature causes channel to leave AGI has

     hung up (Reported by Matt Jordan)

* ASTERISK-24236 – res_hep_rtcp: Module incorrectly depends on

     pjsip (Reported by Matt Jordan)

* ASTERISK-23508 – Memory Corruption in

     __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)

Improvements made in this release:

———————————–

* ASTERISK-26218 – iLBC 20 (Reported by Alexander Traud)

* ASTERISK-26190 – SRTP: Enable AES-256 and AES-GCM.

     (Reported by Alexander Traud)

* ASTERISK-26220 – Add support for noreturn function attributes.

     (Reported by Corey Farrell)

* ASTERISK-22131 – Update the make dependencies script to pull,

     build, and install the correct pjproject (Reported by Matt

     Jordan)

* ASTERISK-25471 – Add subscribe_context to res_pjsip

     (Reported by JoshE)

* ASTERISK-26159 – res_hep: enabled by default and information

     sent to default address (Reported by Ross Beer)

* ASTERISK-26088 – Investigate heavy memory utilization by

     res_pjsip_pubsub (Reported by Richard Mudgett)

* ASTERISK-25578 – SIP/SDP: No rtpmap for static RTP

     payload IDs (Reported by Alexander Traud)

* ASTERISK-26011 – PJSIP: add “via_addr”, “via_port”,

     “call_id” to contacts (Reported by Alexei Gradinari)

* ASTERISK-25965 – res_pjsip_outbound_publish: Allow multiple

     clients per configuration (Reported by Kevin Harwell)

* ASTERISK-25994 – res_pjsip: module load priority

     (Reported by Alexei Gradinari)

* ASTERISK-25931 – PJSIP: add “reg_server” to contacts. (Reported

     by Alexei Gradinari)

* ASTERISK-25835 – Authentication using ‘Username’ field from

     Digest (Reported by Ross Beer)

* ASTERISK-25930 – PJSIP: disable multi domain to improve realtime

     performace (Reported by Alexei Gradinari)

* ASTERISK-25865 – Message-Account Missing From PJSIP MWI

     (Reported by Ross Beer)

* ASTERISK-25444 – Music On Hold Warning misleading

     (Reported by Conrad de Wet)

* ASTERISK-25846 – Gracefully deal with Absent Stasis Apps

     (Reported by Andrew Nagy)

* ASTERISK-25791 – res_pjsip_caller_id: Lack of support for

     Anonymous <anonymous@anonymous.invalid> (Reported by Anthony

     Messina)

* ASTERISK-25767 – Add check to configure for sanitizes

     (Reported by Badalian Vyacheslav)

* ASTERISK-25068 – Move commonly used FreePBX extra sounds to the

     core set (Reported by Rusty Newton)

* ASTERISK-25627 – Easily Preventable Compile Warning (Reported by

     Diederik de Groot)

* ASTERISK-25558 – chan_sip option ‘notifyringing’ doc fix

     and addition of ‘notifyringingprio’ (Reported by Ward van

     Wanrooij)

* ASTERISK-25618 – res_pjsip:  Check for readability of TLS files

     at startup (Reported by George Joseph)

* ASTERISK-25581 – Add value reason a pause on CLI

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25572 – Endpoints: Add StatsD stats for Asterisk

     endpoints (Reported by Matt Jordan)

* ASTERISK-25571 – PJSIP: Add StatsD stats for some common PJSIP

     objects (Reported by Matt Jordan)

* ASTERISK-25518 – taskprocessor: Add high water mark (Reported by

     Jonathan Rose)

* ASTERISK-25495 – Prevent old-update packages on

     repository Debian systems (Reported by Rodrigo Ramirez

     Norambuena)

* ASTERISK-25477 – pjsip show “command” like [criteria] (Reported

     by Bryant Zimmerman)

* ASTERISK-24718 – Add inital support of “sanitize” to

     configure (Reported by Badalian Vyacheslav)

* ASTERISK-24870 – ARI: Subscriptions to bridges generally not

     super useful (Reported by Matt Jordan)

* ASTERISK-25405 – CLI: core show fd: add timestamp

     (Reported by Alexander Traud)

* ASTERISK-25310 – on FreeBSD also pthread_attr_init()

     defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

* ASTERISK-25256 – Post AMI VarSet to empty string events

     when Asterisk deletes a dialplan variable. (Reported by Richard

     Mudgett)

* ASTERISK-25040 – pbx: Improve performance of reloads by making

     hint destruction more performant (Reported by Matt Jordan)

* ASTERISK-25067 – Sorcery Caching: Implement a new caching module

     (Reported by Matt Jordan)

* ASTERISK-25114 – res_pjsip:  Add AMI events for chan_pjsip

     contact lifecycle changes (Reported by George Joseph)

* ASTERISK-25072 – res_pjsip_outbound_registration: line

     functionality. Additional check for using the request URI

     (Reported by Dmitriy Serov)

* ASTERISK-24815 – Enable TLS Dual-Certificates (ECC+RSA)

     (Reported by Alexander Traud)

* ASTERISK-25063 – add X.509 subject alternative name

     support to Asterisk TLS support (Reported by Maciej Szmigiero)

* ASTERISK-25044 – sorcery:  Add ability to insert a new wizard

     into an object type’s list (Reported by George Joseph)

* ASTERISK-24892 – Super Awesome Company sound prompts (Reported

     by Rusty Newton)

* ASTERISK-24744 – Swedish Core Voice prompts (Reported by Tove

     Hjelm)

* ASTERISK-25049 – CLI: Enable automatic references to modules

     (Reported by Corey Farrell)

* ASTERISK-25056 – Modules: Make ast_module_info->self available

     to auxiliary sources.  (Reported by Corey Farrell)

* ASTERISK-25045 – vector:  Add new capabilities and unit tests

     (Reported by George Joseph)

* ASTERISK-25043 – Avoiding ERR_remove_state in OpenSSL

     (Reported by Alexander Traud)

* ASTERISK-24706 – add auto-dtmf mode for pjsip (Reported

     by yaron nahum)

* ASTERISK-24917 – clang compilation warnings (Reported by

     Diederik de Groot)

* ASTERISK-25051 – Remove unneeded uses of optional_api providers.

     (Reported by Corey Farrell)

* ASTERISK-24974 – Astobj2: Allow reference debugging to be

     enabled/disabled by config. (Reported by Corey Farrell)

* ASTERISK-24980 – cdr_adaptive_odbc: refactor lines to

     concatenate  of columns name (Reported by Rodrigo Ramirez

     Norambuena)

* ASTERISK-24947 – res_pjsip: Add a PJSIP resolver using core DNS

     (Reported by Joshua Colp)

* ASTERISK-24965 – cel_pgsql – log_error string references CDR

     instead of CEL (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24960 – Build System: Create MOD_ADD_SOURCE macro for

     module Makefiles (Reported by Corey Farrell)

* ASTERISK-24939 – IAX make calltoken expiration time

     configurable (Reported by Y Ateya)

* ASTERISK-24918 – pjsip: add CLI options to display global and

     system configuration (Reported by Scott Griepentrog)

* ASTERISK-24862 – Support in-dialog OPTIONS (Reported by

     yaron nahum)

* ASTERISK-24802 – stasis: set a channel variable on websocket

     disconnect error (Reported by Kevin Harwell)

* ASTERISK-24133 – Please support Clang; Allow no-exec

     stacks (Reported by Jeffrey Walton)

* ASTERISK-24790 – Reduce spurious noise in logs from voicemail –

     Couldn’t find mailbox %s in context (Reported by Graham Barnett)

* ASTERISK-24813 – asterisk.c: #if statement in listener()

     confuses code folding editors (Reported by Corey Farrell)

* ASTERISK-24811 – asterisk-publication sorcery object does not

     use realtime (Reported by Matt Hoskins)

* ASTERISK-24745 – Add no_answer to ARI hangup causes

     (Reported by Ben Merrills)

* ASTERISK-24316 – For httpd server, need option to define server

     name for security purposes (Reported by Andrew Nagy)

* ASTERISK-24671 – Missing docs for the CDR AMI Event (Reported by

     Dan Jenkins)

* ASTERISK-24575 – Make capath work for res_pjsip (Reported

     by cloos)

* ASTERISK-24678 – [PATCH] Added atxfer* settings to

     features.conf.sample (Reported by Niklas Larsson)

* ASTERISK-24412 – Incomplete channel originate/continue

     handling with ARI (Reported by Nir Simionovich (GreenfieldTech –

     Israel))

* ASTERISK-24351 – Allow passing options and command to

     MixMonitor when recording in ConfBridge (Reported by Gareth

     Palmer)

* ASTERISK-24553 – ARI/AMI: Include language in standard channel

     snapshot output (Reported by Matt Jordan)

* ASTERISK-24552 – ARI: Allow associating a channel as an

     initiator of an Origination for record keeping purposes

     (Reported by Matt Jordan)

* ASTERISK-24577 – Speed up loopback switches by avoiding unneeded

     lookups (Reported by Birger “WIMPy” Harzenetter)

* ASTERISK-24530 – app_record stripping 1/4 second from

     recordings (Reported by Ben Smithurst)

* ASTERISK-24283 – Microseconds precision in the eventtime

     column in the cel_odbc module (Reported by Etienne Lessard)

* ASTERISK-24128 – [Patch] Adding default dtls settings (Reported

     by Michael K.)

* ASTERISK-24279 – Documentation: Clarify the behaviour of the CDR

     property ‘unanswered’ (Reported by Matt Jordan)

* ASTERISK-23512 – Inaccurate comment in manager.conf.sample

     (Reported by Richard Miller)

* ASTERISK-24365 – [Patch] Dialplan function to get first/head

     caller channel on queue (Reported by Kristian Høgh)

* ASTERISK-23324 – – QLOOG commiting Japanese translated

     prompts (Reported by Kevin McCoy)

* ASTERISK-24038 – device state: Report ONHOLD device state if

     channel driver defers device state calculation to core (Reported

     by Matt Jordan)

* ASTERISK-24171 – Provide a manpage for the aelparse

     utility (Reported by Jeremy Lainé)

* ASTERISK-23953 – Testsuite: Off-nominal Authenticate test

     (Reported by Matt Jordan)

* ASTERISK-24045 – Voicemail to email at multiple email

     addresses (Reported by Jacob Barber)

For a full list of changes in this beta, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1

Thank you for your continued support of Asterisk!

_____________________________________________________________________

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[asterisk-dev] Asterisk 13.10.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.10.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0 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:

Improvements made in this release:

———————————–

* ASTERISK-26088 – Investigate heavy memory utilization by

     res_pjsip_pubsub (Reported by Richard Mudgett)

* ASTERISK-26011 – PJSIP: add “via_addr”, “via_port”,

     “call_id” to contacts (Reported by Alexei Gradinari)

* ASTERISK-25994 – res_pjsip: module load priority

     (Reported by Alexei Gradinari)

* ASTERISK-25931 – PJSIP: add “reg_server” to contacts. (Reported

     by Alexei Gradinari)

* ASTERISK-25835 – Authentication using ‘Username’ field from

     Digest (Reported by Ross Beer)

* ASTERISK-25930 – PJSIP: disable multi domain to improve realtime

     performace (Reported by Alexei Gradinari)

Bugs fixed in this release:

———————————–

* ASTERISK-26160 – pjsip: Updated->Reachable during qualify

     (Reported by Matt Jordan)

* ASTERISK-26177 – func_odbc: Database handle is kept when it

     should be released (Reported by Leandro Dardini)

* ASTERISK-26099 – res_pjsip_pubsub: Crash when sending request

     due to server timeout (Reported by Ross Beer)

* ASTERISK-26141 – res_fax: fax_v21_session_new leaks reference to

     v21_details (Reported by Corey Farrell)

* ASTERISK-26140 – res_rtp_asterisk: gcc 6 caught a

     self-comparison (Reported by George Joseph)

* ASTERISK-26138 – chan_unistim:  Under FreeBSD, chan_unistim

     generates a compile error (Reported by George Joseph)

* ASTERISK-26128 – Alembic scripts are failing (Reported by Mark

     Michelson)

* ASTERISK-26139 – test_res_pjsip_scheduler:  Compile failure if

     pjproject isn’t installed in a system location (Reported by

     George Joseph)

* ASTERISK-26130 – WebRTC: Should use latest DTLS version.

     (Reported by Alexander Traud)

* ASTERISK-26127 – res_pjsip_session: Crash due to race condition

     between res_pjsip_session unload and timer (Reported by Joshua

     Colp)

* ASTERISK-26083 – ARI: Announcer channels staying around after

     playback to a bridge is finished (Reported by Per Jensen)

* ASTERISK-26126 – leverage ‘bindaddr’ for TLS in

     http.conf (Reported by Alexander Traud)

* ASTERISK-26069 – Asterisk truncates To: header, dropping the

     closing ‘>’ (Reported by Vasil Kolev)

* ASTERISK-26097 – CLI: show maximum file descriptors

     (Reported by Alexander Traud)

* ASTERISK-25262 – Memory leak when a caller channel does multiple

     dials and CEL is enabled (Reported by Etienne Lessard)

* ASTERISK-26092 – [Segfault] in res_rtp_asterisk.c:4268 after

     Remotely bridged channels (Reported by Niklas Larsson)

* ASTERISK-26096 – res_hep: Crash when configuration file is

     missing (Reported by Niklas Larsson)

* ASTERISK-26089 – Invalid security events during boot using PJSIP

     Realtime (Reported by Scott Griepentrog)

* ASTERISK-26074 – res_odbc: Deadlock within UnixODBC (Reported by

     Ross Beer)

* ASTERISK-26054 – Asterisk crashes (core dump) (Reported by B.

     Davis)

* ASTERISK-24436 – Missing header in res/res_srtp.c when compiling

     against libsrtp-1.5.0 (Reported by Patrick Laimbock)

* ASTERISK-26091 – ar cru creates warning, instead use ar

     cr (Reported by Alexander Traud)

* ASTERISK-26070 – ari/channels:  Creating a local channel without

     an originator adds all audio formats to it’s capabilities

     (Reported by George Joseph)

* ASTERISK-26078 – core: Memory leak in logging (Reported by

     Etienne Lessard)

* ASTERISK-26065 – chan_pjsip: MWI NOTIFY contents not ordered

     properly (Reported by Ross Beer)

* ASTERISK-26063 – ${PJSIP_HEADER(read,Call-ID)} does not work –

     documentation needs clarification for when read/write is

     possible (Reported by Private Name)

* ASTERISK-25777 – data race in threadpool (Reported by Badalian

     Vyacheslav)

* ASTERISK-26038 – ‘make install’ doesn’t seem to install OS/X

     init files (Reported by Tzafrir Cohen)

* ASTERISK-26029 – parking: ast_parking_park_call should return

     parking_space instead of parking_exten (Reported by Diederik de

     Groot)

* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin

     Vandamme)

* ASTERISK-25941 – chan_pjsip: Crash on an immediate SIP final

     response (Reported by Javier Riveros )

* ASTERISK-26014 – res_sorcery_astdb: Make tolerant of unknown

     fields (Reported by Joshua Colp)

* ASTERISK-24986 – keepalive INFO packages ignored by asterisk

     (Reported by Ilya Trikoz)

* ASTERISK-26034 – T.38 passthrough problem behind firewall due to

     early nosignal packet (Reported by George Joseph)

* ASTERISK-26030 – call cut because of double Session-Expires

     header in re-invite after proxy authentication is required

     (Reported by George Joseph)

* ASTERISK-25964 – Outbound registrations created via ARI/push

     configuration do not clean up outbound registrations currently

     in flight (Reported by Matt Jordan)

* ASTERISK-26005 – res_pjsip: Multiple SIP messages are combined

     into 1 TCP packet (Reported by Ross Beer)

* ASTERISK-25352 – res_hep_rtcp correlation_id is different then

     res_hep (Reported by Kevin Scott Adams)

* ASTERISK-26008 – app_followme does not delete recorded name

     prompt (Reported by Tzafrir Cohen)

* ASTERISK-26007 – res_pjsip: Endpoints deleting early after

     upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

* ASTERISK-25990 – PJSIP TLS registration should respect

     client_uri scheme when generating Contact URI (Reported by

     Sebastian Damm)

* ASTERISK-25978 – res_pjsip_authenticator_digest: Should not use

     source port in nonce verification (Reported by Mark Michelson)

* ASTERISK-25993 – pjproject: Allow bundling to not require

     everything it does (Reported by Joshua Colp)

* ASTERISK-25956 – Compilation error in conditionally compiled

     code in config_options.c (Reported by Chris Trobridge)

* ASTERISK-25998 – file: Crash when using nativeformats (Reported

     by Joshua Colp)

* ASTERISK-25826 – PJSIP / Sorcery slow load from realtime

     (Reported by Ross Beer)

* ASTERISK-25968 – pjproject_bundled:  Configure and make need to

     be re-tested (Reported by George Joseph)

* ASTERISK-24463 – Voicemail email address corrupt or not sent

     when message is in the process of being recorded during reload

     (Reported by John Campbell)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25961 – tests/channels/SIP/sip_tls_call: Sporadic crash

     when running test (Reported by Joshua Colp)

* ASTERISK-16115 – problem with ringinuse=no, queue

     members receive sometimes two calls (Reported by nik600)

* ASTERISK-25917 – app_voicemail: passwordlocation=spooldir

     only works if you manually add secret.conf yourself (Reported by

     Jonathan R. Rose)

* ASTERISK-25950 – SIP channel does not send PeerStatus

     events for autocreated peers (Reported by Kirill Katsnelson)

* ASTERISK-25954 – Manager QueueSummary and QueueStatus Actions

     are case sensitive to QueueName (Reported by Javier Acosta)

New Features made in this release:

———————————–

* ASTERISK-25904 – PJSIP: add contact.updated event (Reported by

     Alexei Gradinari)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 11.23.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.23.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.23.0 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:

Bugs fixed in this release:

———————————–

* ASTERISK-26141 – res_fax: fax_v21_session_new leaks reference to

     v21_details (Reported by Corey Farrell)

* ASTERISK-26140 – res_rtp_asterisk: gcc 6 caught a

     self-comparison (Reported by George Joseph)

* ASTERISK-26138 – chan_unistim:  Under FreeBSD, chan_unistim

     generates a compile error (Reported by George Joseph)

* ASTERISK-26130 – WebRTC: Should use latest DTLS version.

     (Reported by Alexander Traud)

* ASTERISK-26126 – leverage ‘bindaddr’ for TLS in

     http.conf (Reported by Alexander Traud)

* ASTERISK-26069 – Asterisk truncates To: header, dropping the

     closing ‘>’ (Reported by Vasil Kolev)

* ASTERISK-26097 – CLI: show maximum file descriptors

     (Reported by Alexander Traud)

* ASTERISK-24436 – Missing header in res/res_srtp.c when compiling

     against libsrtp-1.5.0 (Reported by Patrick Laimbock)

* ASTERISK-26091 – ar cru creates warning, instead use ar

     cr (Reported by Alexander Traud)

* ASTERISK-26038 – ‘make install’ doesn’t seem to install OS/X

     init files (Reported by Tzafrir Cohen)

* ASTERISK-26034 – T.38 passthrough problem behind firewall due to

     early nosignal packet (Reported by George Joseph)

* ASTERISK-26030 – call cut because of double Session-Expires

     header in re-invite after proxy authentication is required

     (Reported by George Joseph)

* ASTERISK-26008 – app_followme does not delete recorded name

     prompt (Reported by Tzafrir Cohen)

* ASTERISK-24463 – Voicemail email address corrupt or not sent

     when message is in the process of being recorded during reload

     (Reported by John Campbell)

* ASTERISK-25917 – app_voicemail: passwordlocation=spooldir

     only works if you manually add secret.conf yourself (Reported by

     Jonathan R. Rose)

* ASTERISK-25954 – Manager QueueSummary and QueueStatus Actions

     are case sensitive to QueueName (Reported by Javier Acosta)

* ASTERISK-16115 – problem with ringinuse=no, queue

     members receive sometimes two calls (Reported by nik600)

* ASTERISK-25934 – chan_sip should not require sipregs or

     updateable sippeers table unless rt (Reported by Jaco Kroon)

* ASTERISK-25888 – Frequent segfaults in function can_ring_entry()

     of app_queue.c (Reported by Sébastien Couture)

* ASTERISK-25874 – app_voicemail: Stack buffer overflow in

     test_voicemail_notify_endl (Reported by Badalian Vyacheslav)

* ASTERISK-25912 – chan_local passes AST_CONTROL_PVT_CAUSE_CODE

     without adding them to the local hangupcauses via

     ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25510 – Log to syslog failing (Reported by

     Michael Newton)

Improvements made in this release:

———————————–

* ASTERISK-25444 – Music On Hold Warning misleading

     (Reported by Conrad de Wet)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 13.9.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.9.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.9.0 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:

Bugs fixed in this release:

———————————–

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin

     Vandamme)

* ASTERISK-25927 – Removed option “registertrying” is still

     documented in sip.conf.sample (Reported by Etienne Lessard)

* ASTERISK-25947 – Protocol transfers to stasis applications are

     missing the StasisStart with the replace_channel object.

     (Reported by Richard Mudgett)

* ASTERISK-24649 – Pushing of channel into bridge fails; Stasis

     fails to get app name (Reported by John Bigelow)

* ASTERISK-24782 – StasisEnd event not present for channel that

     was swapped out for another after completing attended transfer

     (Reported by John Bigelow)

* ASTERISK-25942 – res_pjsip_caller_id: Transfer results in mixed

     ConnectedLine information (Reported by George Joseph)

* ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

     thread (Reported by Joshua Colp)

* ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

     not raised (Reported by Joshua Colp)

* ASTERISK-25934 – chan_sip should not require sipregs or

     updateable sippeers table unless rt (Reported by Jaco Kroon)

* ASTERISK-25888 – Frequent segfaults in function can_ring_entry()

     of app_queue.c (Reported by Sébastien Couture)

* ASTERISK-25796 – res_pjsip: DOS/Crash when TCP/TLS sockets

     exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George

     Joseph)

* ASTERISK-25707 – Long contact URIs or hostnames can crash

     pjproject/Asterisk under certain conditions (Reported by George

     Joseph)

* ASTERISK-25123 – Bracketed IPv6 Contact header parameter

     unparsable with Asterisk/PJSIP (Reported by Anthony Messina)

* ASTERISK-25874 – app_voicemail: Stack buffer overflow in

     test_voicemail_notify_endl (Reported by Badalian Vyacheslav)

* ASTERISK-25912 – chan_local passes AST_CONTROL_PVT_CAUSE_CODE

     without adding them to the local hangupcauses via

     ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)

* ASTERISK-25885 – res_pjsip: Race condition between adding

     contact and automatic expiration (Reported by Joshua Colp)

* ASTERISK-25910 – pjproject:  Via headers are not parsed when

     “received” contains an IPv6 address (Reported by George Joseph)

* ASTERISK-25890 – Asterisk 13.8.0 alembic database update fails

     (Reported by Harley Peters)

* ASTERISK-25894 – webrtc video broken due to missing

     marker bits in RTP streams (Reported by Jacek Konieczny)

* ASTERISK-25854 – No audio after HOLD/RESUME – incorrect

     a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)

* ASTERISK-25873 – res_pjsip: Bundled pjproject: compile error,

     cannot find -lasteriskpj (Reported by Hans van Eijsden)

* ASTERISK-25882 – ARI: Crash can occur due to race condition when

     attempting to operate on a hung up channel (Part 2) (Reported by

     Richard Mudgett)

* ASTERISK-25867 – Video delay on app_echo (Reported by

     Jacek Konieczny)

* ASTERISK-24605 – res_parking option parkeddynamic does not work

     with the core Features ‘parkcall’ (DTMF initiated parking)

     (Reported by Philip Correia)

* ASTERISK-25826 – PJSIP / Sorcery slow load from realtime

     (Reported by Ross Beer)

* ASTERISK-24596 – Unclear how to use Park application with

     res_parking ‘parkeddynamic’ enabled. Documentation? (Reported by

     Philip Correia)

* ASTERISK-24543 – Asterisk 13 responds to SIP Invite with all

     possible codecs configured for peer as opposed to intersection

     of configured codecs and offered codecs (Reported by Taylor

     Hawkes)

* ASTERISK-25825 – Crashes during shutdown when running CLI

     commands (Reported by Mark Michelson)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25510 – Log to syslog failing (Reported by

     Michael Newton)

* ASTERISK-25857 – func_aes: incorrect use of strlen() leads to

     data corruption (Reported by Gianluca Merlo)

Improvements made in this release:

———————————–

* ASTERISK-25865 – Message-Account Missing From PJSIP MWI

     (Reported by Ross Beer)

* ASTERISK-25444 – Music On Hold Warning misleading

     (Reported by Conrad de Wet)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 13.1-cert6 Now Available

The Asterisk Development Team has announced the release of Certified Asterisk 13.1-cert6.

This release is available for immediate download at

http://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 13.1-cert6 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:

Bugs fixed in this release:

———————————–

 * ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

      not raised (Reported by Joshua Colp)

 * ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

      thread (Reported by Joshua Colp)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.1-cert6

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 13.8.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.8.2.

This release is available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.8.2 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:

Bugs fixed in this release:

———————————–

* ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

     not raised (Reported by Joshua Colp)

* ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

     thread (Reported by Joshua Colp)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2

Thank you for your continued support of Asterisk!