[asterisk-users] Certified Asterisk 16.3-cert1 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Dec 24 11:12:53 CST 2019

The Asterisk Development Team would like to announce the release of
Certified Asterisk 16.3-cert1.
This release is available for immediate download at

The release of Certified Asterisk 16.3-cert1 resolves several issues
reported by the
community and would have not been possible without your participation.

*Thank you!*

The following issues are resolved in this release:

*Security bugs fixed in this release:*

   - [ASTERISK-28589
   <https://issues.asterisk.org/jira/browse/ASTERISK-28589>] -

chan_sip: Depending on configuration an INVITE can alter Addr of a peer
(Reported by Andrey V. T.)

   - [ASTERISK-28580
   <https://issues.asterisk.org/jira/browse/ASTERISK-28580>] -

Bypass SYSTEM write permission in manager action allows system commands
(Reported by Eliel Sardañons)

   - [ASTERISK-28495
   <https://issues.asterisk.org/jira/browse/ASTERISK-28495>] -

res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
(Reported by Alexei Gradinari)

   - [ASTERISK-28447
   <https://issues.asterisk.org/jira/browse/ASTERISK-28447>] -

res_pjsip_messaging: In-dialog MESSAGE with no body causes crash
(Reported by Gil Richard)

   - [ASTERISK-28465
   <https://issues.asterisk.org/jira/browse/ASTERISK-28465>] -

Broken SDP can cause a segfault in a T.38 reINVITE
(Reported by Francesco Castellano)

   - [ASTERISK-28260
   <https://issues.asterisk.org/jira/browse/ASTERISK-28260>] -

Asterisk segfault when rtp negotiation is wrong or fails
(Reported by Sotiris Ganouris)

   - [ASTERISK-28127
   <https://issues.asterisk.org/jira/browse/ASTERISK-28127>] -

Buffer overflow for DNS SRV/NAPTR records
(Reported by Jan Hoffmann)

   - [ASTERISK-28013
   <https://issues.asterisk.org/jira/browse/ASTERISK-28013>] -

res_http_websocket: Crash when reading HTTP Upgrade requests
(Reported by Sean Bright)

   - [ASTERISK-27807
   <https://issues.asterisk.org/jira/browse/ASTERISK-27807>] -

iostreams: Potential DoS when client connection closed prematurely
(Reported by Sean Bright)

   - [ASTERISK-27818
   <https://issues.asterisk.org/jira/browse/ASTERISK-27818>] -

Username bruteforce is possible when using ACL with PJSIP
(Reported by John)

   - [ASTERISK-27658
   <https://issues.asterisk.org/jira/browse/ASTERISK-27658>] -

WebSocket frames with 0 sized payload causes DoS
(Reported by Sean Bright)

   - [ASTERISK-27583
   <https://issues.asterisk.org/jira/browse/ASTERISK-27583>] -

Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute
(Reported by Sandro Gauci)

   - [ASTERISK-27582
   <https://issues.asterisk.org/jira/browse/ASTERISK-27582>] -

Segmentation fault occurs in Asterisk with an invalid SDP media format
(Reported by Sandro Gauci)

   - [ASTERISK-27618
   <https://issues.asterisk.org/jira/browse/ASTERISK-27618>] -

Crash occurs when sending a repeated number of INVITE messages over TCP or
TLS transport
(Reported by Sandro Gauci)

   - [ASTERISK-27640
   <https://issues.asterisk.org/jira/browse/ASTERISK-27640>] -

SUBSCRIBE message with a large Accept value causes stack corruption
(Reported by Sandro Gauci)

*New Features made in this release:*

   - [ASTERISK-28267
   <https://issues.asterisk.org/jira/browse/ASTERISK-28267>] -

res_stasis: Add ability to switch applications
(Reported by Benjamin Keith Ford)

   - [ASTERISK-28087
   <https://issues.asterisk.org/jira/browse/ASTERISK-28087>] -

add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip
(Reported by Torrey Searle)

   - [ASTERISK-27286
   <https://issues.asterisk.org/jira/browse/ASTERISK-27286>] -

Add the ability to read the media file type from HTTP header for playback
(Reported by Gaurav Khurana)

   - [ASTERISK-27704
   <https://issues.asterisk.org/jira/browse/ASTERISK-27704>] -

Add cache_pools debug option to pjproject.conf
(Reported by Richard Mudgett)

   - [ASTERISK-27581
   <https://issues.asterisk.org/jira/browse/ASTERISK-27581>] -

Add new AMI Action for PJSIPShowContacts
(Reported by sungtae kim)

   - [ASTERISK-27547
   <https://issues.asterisk.org/jira/browse/ASTERISK-27547>] -

res_pjsip: Add new AMI Action for PJSIPShowAuths
(Reported by sungtae kim)

   - [ASTERISK-27117
   <https://issues.asterisk.org/jira/browse/ASTERISK-27117>] -

core: Add support for timelen parsing to ast_parse_arg and ACO.
(Reported by Corey Farrell)

   - [ASTERISK-27478
   <https://issues.asterisk.org/jira/browse/ASTERISK-27478>] -

PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI.
(Reported by Richard Mudgett)

   - [ASTERISK-27413
   <https://issues.asterisk.org/jira/browse/ASTERISK-27413>] -

Add cache_media_frames debugging option.
(Reported by Richard Mudgett)

   - [ASTERISK-27206
   <https://issues.asterisk.org/jira/browse/ASTERISK-27206>] -

res_pjsip: No mechanism exists to limit endpoint identification to IP only
(Reported by Ben Merrills)

   - [ASTERISK-27215
   <https://issues.asterisk.org/jira/browse/ASTERISK-27215>] -

[patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)

   - [ASTERISK-27322
   <https://issues.asterisk.org/jira/browse/ASTERISK-27322>] -

[New Feature] Add mute and DTMF passthrough to ARI add channel to bridge
(Reported by Darren Sessions)

   - [ASTERISK-27162
   <https://issues.asterisk.org/jira/browse/ASTERISK-27162>] -

[patch]chan_sip: Access incoming SIP REFER headers in the dialplan
(Reported by Kirill Katsnelson)

   - [ASTERISK-27163
   <https://issues.asterisk.org/jira/browse/ASTERISK-27163>] -

chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER().
(Reported by Kirill Katsnelson)

   - [ASTERISK-27129
   <https://issues.asterisk.org/jira/browse/ASTERISK-27129>] -

ast_waitfordigit_full: add support for filtering DTMF keys which can break
the wait.
(Reported by Corey Farrell)

   - [ASTERISK-27063
   <https://issues.asterisk.org/jira/browse/ASTERISK-27063>] -

Add support for systemd socket activation
(Reported by Corey Farrell)

   - [ASTERISK-26995
   <https://issues.asterisk.org/jira/browse/ASTERISK-26995>] -

Add QUEUE_FLOAT_PENALTY to app_queue
(Reported by Steve Davies)

   - [ASTERISK-26878
   <https://issues.asterisk.org/jira/browse/ASTERISK-26878>] -

func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)

   - [ASTERISK-26863
   <https://issues.asterisk.org/jira/browse/ASTERISK-26863>] -

res_pjsip: Add endpoint identification scheme based on a configured SIP
(Reported by Matt Jordan)

   - [ASTERISK-17428
   <https://issues.asterisk.org/jira/browse/ASTERISK-17428>] -

[patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

   - [ASTERISK-26584
   <https://issues.asterisk.org/jira/browse/ASTERISK-26584>] -

[patch] RTCP feedback for codec modules
(Reported by Lorenzo Miniero)

   - [ASTERISK-19862
   <https://issues.asterisk.org/jira/browse/ASTERISK-19862>] -

app_queue: Update Data of Queues (use queues as outbound calls container)
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26630
   <https://issues.asterisk.org/jira/browse/ASTERISK-26630>] -

Make logging PJPROJECT messages a bit easier
(Reported by Richard Mudgett)

   - [ASTERISK-26587
   <https://issues.asterisk.org/jira/browse/ASTERISK-26587>] -

app_originate: Add option to execute gosub prior to dial
(Reported by dkerr)

   - [ASTERISK-26595
   <https://issues.asterisk.org/jira/browse/ASTERISK-26595>] -

ARI: Add the ability to control the source of video in a multi-party mixing
(Reported by Matt Jordan)

   - [ASTERISK-26492
   <https://issues.asterisk.org/jira/browse/ASTERISK-26492>] -

ARI: Add ability to specify channel variables on websocket events
(Reported by Mark Michelson)

   - [ASTERISK-26470
   <https://issues.asterisk.org/jira/browse/ASTERISK-26470>] -

ARI: Add an 'asterisk_id' field to outgoing events
(Reported by Matt Jordan)

   - [ASTERISK-26277
   <https://issues.asterisk.org/jira/browse/ASTERISK-26277>] -

Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session
refresh to update formats on a channel after session establishment
(Reported by Matt Jordan)

   - [ASTERISK-25904
   <https://issues.asterisk.org/jira/browse/ASTERISK-25904>] -

PJSIP: add contact.updated event
(Reported by Alexei Gradinari)

   - [ASTERISK-18995
   <https://issues.asterisk.org/jira/browse/ASTERISK-18995>] -

Support for OGG/Speex file format
(Reported by Timo Teräs)

   - [ASTERISK-26087
   <https://issues.asterisk.org/jira/browse/ASTERISK-26087>] -

Icelandic grammar support for voicemail and numbers
(Reported by Örn Arnarson)

   - [ASTERISK-26058
   <https://issues.asterisk.org/jira/browse/ASTERISK-26058>] -

[Patch] Add uptime and last reloaded to FullyBooted AMI event
(Reported by Niklas Larsson)

   - [ASTERISK-25925
   <https://issues.asterisk.org/jira/browse/ASTERISK-25925>] -

Allow Early Bridges on ARI Dials
(Reported by Mark Michelson)

   - [ASTERISK-26068
   <https://issues.asterisk.org/jira/browse/ASTERISK-26068>] -

Multicast RTP Options
(Reported by Mark Michelson)

   - [ASTERISK-26042
   <https://issues.asterisk.org/jira/browse/ASTERISK-26042>] -

ARI: Allow downloading of the media associated with a stored recording
(Reported by Matt Jordan)

   - [ASTERISK-26022
   <https://issues.asterisk.org/jira/browse/ASTERISK-26022>] -

ARI: Add media playlists
(Reported by Matt Jordan)

   - [ASTERISK-25425
   <https://issues.asterisk.org/jira/browse/ASTERISK-25425>] -

logger: Add JSON structured logging
(Reported by Matt Jordan)

   - [ASTERISK-25900
   <https://issues.asterisk.org/jira/browse/ASTERISK-25900>] -

PJSIP Endpoint IP Access Controls
(Reported by Alexei Gradinari)

   - [ASTERISK-25989
   <https://issues.asterisk.org/jira/browse/ASTERISK-25989>] -

apps/confbridge: add regcontext feature
(Reported by Jaco Kroon)

   - [ASTERISK-25903
   <https://issues.asterisk.org/jira/browse/ASTERISK-25903>] -

PJSIP AMI Event ContactStatus: add Useragent and RegExpire
(Reported by Alexei Gradinari)

   - [ASTERISK-25866
   <https://issues.asterisk.org/jira/browse/ASTERISK-25866>] -

ChanSpy: allow usage of a long queue to store audio frames, to avoid audio
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-25972
   <https://issues.asterisk.org/jira/browse/ASTERISK-25972>] -

res_pjsip_exten_state: Use body generator to publish extension state
(Reported by Richard Mudgett)

   - [ASTERISK-25901
   <https://issues.asterisk.org/jira/browse/ASTERISK-25901>] -

Add transport for outbound PUBLISH
(Reported by Alexei Gradinari)

   - [ASTERISK-25889
   <https://issues.asterisk.org/jira/browse/ASTERISK-25889>] -

ARI: Add separate "create" and "dial" operations for channels
(Reported by Mark Michelson)

   - [ASTERISK-25654
   <https://issues.asterisk.org/jira/browse/ASTERISK-25654>] -

Playback: Add the ability to play remote URIs
(Reported by Matt Jordan)

   - [ASTERISK-25652
   <https://issues.asterisk.org/jira/browse/ASTERISK-25652>] -

func_curl: Add the ability to CURL files down to a specified location
(Reported by Matt Jordan)

   - [ASTERISK-25803
   <https://issues.asterisk.org/jira/browse/ASTERISK-25803>] -

[patch] chan_sip: Optionally supply fromuser/fromdomain in SIP dial string
(Reported by Walter Doekes)

   - [ASTERISK-24919
   <https://issues.asterisk.org/jira/browse/ASTERISK-24919>] -

res_pjsip_config_wizard: Ability to write contents to file
(Reported by Ray Crumrine)

   - [ASTERISK-16394
   <https://issues.asterisk.org/jira/browse/ASTERISK-16394>] -

[patch] Last pause information to queue members
(Reported by Evandro César Arruda)

   - [ASTERISK-25670
   <https://issues.asterisk.org/jira/browse/ASTERISK-25670>] -

Add regcontext to PJSIP
(Reported by Daniel Journo)

   - [ASTERISK-25660
   <https://issues.asterisk.org/jira/browse/ASTERISK-25660>] -

Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts.
(Reported by Walter Doekes)

   - [ASTERISK-25591
   <https://issues.asterisk.org/jira/browse/ASTERISK-25591>] -

[patch] Complete List of Header Files (#include): iwyu
(Reported by Alexander Traud)

   - [ASTERISK-25551
   <https://issues.asterisk.org/jira/browse/ASTERISK-25551>] -

[patch]Ability to add channel to an existing bridge by specifying an
existing channel prefix
(Reported by Alec Davis)

   - [ASTERISK-25419
   <https://issues.asterisk.org/jira/browse/ASTERISK-25419>] -

Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)

   - [ASTERISK-25549
   <https://issues.asterisk.org/jira/browse/ASTERISK-25549>] -

Confbridge: Add participant timeout option
(Reported by Mark Michelson)

   - [ASTERISK-24922
   <https://issues.asterisk.org/jira/browse/ASTERISK-24922>] -

ARI: Add the ability to intercept hold and raise an event
(Reported by Matt Jordan)

   - [ASTERISK-25479
   <https://issues.asterisk.org/jira/browse/ASTERISK-25479>] -

Allow CDR's to be modified before being dispatched to engines
(Reported by Jonh Wendell)

   - [ASTERISK-25480
   <https://issues.asterisk.org/jira/browse/ASTERISK-25480>] -

[patch]Add field PauseReason on QueueMemberStatus
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25377
   <https://issues.asterisk.org/jira/browse/ASTERISK-25377>] -

res_pjsip: Change default "From user" from UUID to something more palatable
(Reported by Mark Michelson)

   - [ASTERISK-25252
   <https://issues.asterisk.org/jira/browse/ASTERISK-25252>] -

ARI: Add the ability to manipulate log channels
(Reported by Matt Jordan)

   - [ASTERISK-25259
   <https://issues.asterisk.org/jira/browse/ASTERISK-25259>] -

chan_pjsip: Add rtptimeout support
(Reported by Joshua C. Colp)

   - [ASTERISK-25238
   <https://issues.asterisk.org/jira/browse/ASTERISK-25238>] -

ARI: Support push configuration
(Reported by Matt Jordan)

   - [ASTERISK-25173
   <https://issues.asterisk.org/jira/browse/ASTERISK-25173>] -

ARI: Add the ability to load/reload/unload an Asterisk module
(Reported by Matt Jordan)

   - [ASTERISK-25006
   <https://issues.asterisk.org/jira/browse/ASTERISK-25006>] -

[patch] Add support set character for quoted identifiers
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-23186
   <https://issues.asterisk.org/jira/browse/ASTERISK-23186>] -

[patch] Add usegmtime option to cel_pgsql
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-24931
   <https://issues.asterisk.org/jira/browse/ASTERISK-24931>] -

dns: Add support for SRV records.
(Reported by Joshua C. Colp)

   - [ASTERISK-24834
   <https://issues.asterisk.org/jira/browse/ASTERISK-24834>] -

DNS Overhaul: Implement the proposed core API - sync/async functions,
resolver registration
(Reported by Matt Jordan)

   - [ASTERISK-24836
   <https://issues.asterisk.org/jira/browse/ASTERISK-24836>] -

DNS Overhaul: Write a Resolver Implementation
(Reported by Matt Jordan)

   - [ASTERISK-22591
   <https://issues.asterisk.org/jira/browse/ASTERISK-22591>] -

[patch]Prevent Asterisk from writing received SMS content in log
(Reported by Jan Juergens)

   - [ASTERISK-17899
   <https://issues.asterisk.org/jira/browse/ASTERISK-17899>] -

Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)

   - [ASTERISK-24703
   <https://issues.asterisk.org/jira/browse/ASTERISK-24703>] -

ARI: Add the ability to "transfer" (redirect) a channel
(Reported by Matt Jordan)

   - [ASTERISK-24341
   <https://issues.asterisk.org/jira/browse/ASTERISK-24341>] -

PJSIP Ability to get info per contact
(Reported by xrobau)

   - [ASTERISK-24363
   <https://issues.asterisk.org/jira/browse/ASTERISK-24363>] -

[patch] Add ability for Channel Drivers to provide Presence State
(Reported by Gareth Palmer)

   - [ASTERISK-24554
   <https://issues.asterisk.org/jira/browse/ASTERISK-24554>] -

AMI/ARI: Generate events on connected line changes
(Reported by Matt Jordan)

   - [ASTERISK-24276
   <https://issues.asterisk.org/jira/browse/ASTERISK-24276>] -

[Patch] Option to make app MOH override channel musicclass
(Reported by Kristian Høgh)

   - [ASTERISK-23871
   <https://issues.asterisk.org/jira/browse/ASTERISK-23871>] -

RLS Tests: Implement RLS off-nominal tests
(Reported by Mark Michelson)

   - [ASTERISK-23823
   <https://issues.asterisk.org/jira/browse/ASTERISK-23823>] -

[patch] Option to keep queuerules in realtime
(Reported by Michael K.)

*Bugs fixed in this release:*

   - [ASTERISK-28609
   <https://issues.asterisk.org/jira/browse/ASTERISK-28609>] -

Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)

   - [ASTERISK-28631
   <https://issues.asterisk.org/jira/browse/ASTERISK-28631>] -

res_parking: Doesn't park when parkee and parker are the same
(Reported by Ross Beer)

   - [ASTERISK-28624
   <https://issues.asterisk.org/jira/browse/ASTERISK-28624>] -

res_pjsip_outbound_registration: add SRV failover
(Reported by Kevin Harwell)

   - [ASTERISK-28616
   <https://issues.asterisk.org/jira/browse/ASTERISK-28616>] -

parking: Deadlock when multi call parking
(Reported by Joshua C. Colp)

   - [ASTERISK-28523
   <https://issues.asterisk.org/jira/browse/ASTERISK-28523>] -

Asterisk 16.5.0 Memory leak
(Reported by Cyril Ramière)

   - [ASTERISK-28538
   <https://issues.asterisk.org/jira/browse/ASTERISK-28538>] -

chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)

   - [ASTERISK-28362
   <https://issues.asterisk.org/jira/browse/ASTERISK-28362>] -

strtok_r() makes gcc compile warning
(Reported by sungtae kim)

   - [ASTERISK-27541
   <https://issues.asterisk.org/jira/browse/ASTERISK-27541>] -

app_queue: Queue paused reason was (big number) secs ago when reason is set
(Reported by César Benjamín García Martínez)

   - [ASTERISK-20986
   <https://issues.asterisk.org/jira/browse/ASTERISK-20986>] -

QUEUE_MEMBER 's description is inaccurate
(Reported by Olivier Krief)

   - [ASTERISK-28350
   <https://issues.asterisk.org/jira/browse/ASTERISK-28350>] -

manager: Stasis backed up due to locking
(Reported by Joshua C. Colp)

   - [ASTERISK-25792
   <https://issues.asterisk.org/jira/browse/ASTERISK-25792>] -

chan_sip: qualifygap bounds checking
(Reported by Paul Sandys)

   - [ASTERISK-28341
   <https://issues.asterisk.org/jira/browse/ASTERISK-28341>] -

res_config_odbc eliminates empty custom (“@” prefix) variables
(Reported by Alexei Gradinari)

   - [ASTERISK-28333
   <https://issues.asterisk.org/jira/browse/ASTERISK-28333>] -

StasisEnd event makes wrong timestamp value
(Reported by sungtae kim)

   - [ASTERISK-28306
   <https://issues.asterisk.org/jira/browse/ASTERISK-28306>] -

res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent
(Reported by Jared Hull)

   - [ASTERISK-28332
   <https://issues.asterisk.org/jira/browse/ASTERISK-28332>] -

Variable ALTCONF ignored when service is used in Debian
(Reported by Cirillo Ferreira)

   - [ASTERISK-28314
   <https://issues.asterisk.org/jira/browse/ASTERISK-28314>] -

ARI: API changed but "apiVersion" in rest-api\resources.json did not
(Reported by Stefan Repke)

   - [ASTERISK-28335
   <https://issues.asterisk.org/jira/browse/ASTERISK-28335>] -

stasis: Make topic and maybe subscription names unique and more useful
(Reported by Joshua C. Colp)

   - [ASTERISK-28321
   <https://issues.asterisk.org/jira/browse/ASTERISK-28321>] -

res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation
(Reported by sungtae kim)

   - [ASTERISK-28322
   <https://issues.asterisk.org/jira/browse/ASTERISK-28322>] -

chan_pjsip: Add option to allow ignoring of 183 without SDP
(Reported by Torrey Searle)

   - [ASTERISK-28328
   <https://issues.asterisk.org/jira/browse/ASTERISK-28328>] -

MeetMe global non-admin mute is muting admins that subsequently join
(Reported by Philip Mott)

   - [ASTERISK-27964
   <https://issues.asterisk.org/jira/browse/ASTERISK-27964>] -

app_queue: ring_entry accesses nativeformats without channel lock or
(Reported by Francisco Seratti)

   - [ASTERISK-28168
   <https://issues.asterisk.org/jira/browse/ASTERISK-28168>] -

app_queue: Adding a blank entry into sql queue_members crashes asterisk.
(Reported by Michael)

   - [ASTERISK-28323
   <https://issues.asterisk.org/jira/browse/ASTERISK-28323>] -

pjsip: sip.conf to pjsip.conf conversion script fails
(Reported by Guido Weckwerth)

   - [ASTERISK-28272
   <https://issues.asterisk.org/jira/browse/ASTERISK-28272>] -

The basic-pbx config samples don't produce a running asterisk
(Reported by George Joseph)

   - [ASTERISK-28312
   <https://issues.asterisk.org/jira/browse/ASTERISK-28312>] -

res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302
(Reported by Alex Odrov)

   - [ASTERISK-24173
   <https://issues.asterisk.org/jira/browse/ASTERISK-24173>] -

File menuselect/menuselect_gtk.c has no license header
(Reported by Jeremy Lainé)

   - [ASTERISK-28309
   <https://issues.asterisk.org/jira/browse/ASTERISK-28309>] -

res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces
(Reported by Nikolay shakin)

   - [ASTERISK-27992
   <https://issues.asterisk.org/jira/browse/ASTERISK-27992>] -

PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash
(Reported by Jonathan Harris)

   - [ASTERISK-28166
   <https://issues.asterisk.org/jira/browse/ASTERISK-28166>] -

app_voicemail: Asterisk unresponsive after changing voicemail password with
(Reported by Michael)

   - [ASTERISK-28213
   <https://issues.asterisk.org/jira/browse/ASTERISK-28213>] -

res_pjsip: Threads pile up needlessly when AOR is blocked
(Reported by Ross Beer)

   - [ASTERISK-28301
   <https://issues.asterisk.org/jira/browse/ASTERISK-28301>] -

Allow voicemail boxes to be subscribed to with a presence event package
(Reported by George Joseph)

   - [ASTERISK-28303
   <https://issues.asterisk.org/jira/browse/ASTERISK-28303>] -

res_rtp_asterisk: Interaction between smoother and DTMF can cause out of
order timestamps
(Reported by Torrey Searle)

   - [ASTERISK-28302
   <https://issues.asterisk.org/jira/browse/ASTERISK-28302>] -

ARI: "Error destroying mutex" when listing all ARI applications
(Reported by Stefan Repke)

   - [ASTERISK-28300
   <https://issues.asterisk.org/jira/browse/ASTERISK-28300>] -

AST_PBX_MAX_STACK is too low for some applications
(Reported by George Joseph)

   - [ASTERISK-28106
   <https://issues.asterisk.org/jira/browse/ASTERISK-28106>] -

Astricon Feedback: Unable to filter ARI events when GETting causes overload
of events
(Reported by George Joseph)

   - [ASTERISK-28284
   <https://issues.asterisk.org/jira/browse/ASTERISK-28284>] -

switching between native_bridge and simple_bridge can cause one way audio
(Reported by Torrey Searle)

   - [ASTERISK-28251
   <https://issues.asterisk.org/jira/browse/ASTERISK-28251>] -

CI: Fix CI so it reverifies commit message changes
(Reported by George Joseph)

   - [ASTERISK-28277
   <https://issues.asterisk.org/jira/browse/ASTERISK-28277>] -

database: Add some basic logging
(Reported by Joshua C. Colp)

   - [ASTERISK-28181
   <https://issues.asterisk.org/jira/browse/ASTERISK-28181>] -

ari: Originating overwrites channel start time
(Reported by sungtae kim)

   - [ASTERISK-28173
   <https://issues.asterisk.org/jira/browse/ASTERISK-28173>] -

Deadlock in chan_sip handling subscribe request during res_parking reload
(Reported by Giuseppe Sucameli)

   - [ASTERISK-28104
   <https://issues.asterisk.org/jira/browse/ASTERISK-28104>] -

AstriCon Feedback: Automatically create a 1 line dialplan context for
stasis apps
(Reported by George Joseph)

   - [ASTERISK-28271
   <https://issues.asterisk.org/jira/browse/ASTERISK-28271>] -

Opensuse Leap 15 --with-jannson-bundled will not compile
(Reported by David Wilcox)

   - [ASTERISK-28238
   <https://issues.asterisk.org/jira/browse/ASTERISK-28238>] -

PJSIP realtime. getcontext not working with DUNDI
(Reported by Ray)

   - [ASTERISK-28263
   <https://issues.asterisk.org/jira/browse/ASTERISK-28263>] -

codec_opus: errors setting max_playback_rate and bitrate to "sdp"
(Reported by Gianluca Merlo)

   - [ASTERISK-28250
   <https://issues.asterisk.org/jira/browse/ASTERISK-28250>] -

build: Cross-compilation fails for target arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-28257
   <https://issues.asterisk.org/jira/browse/ASTERISK-28257>] -

res_http_websocket: PING / PONG opcodes break data reception
(Reported by Jeremy Lainé)

   - [ASTERISK-28252
   <https://issues.asterisk.org/jira/browse/ASTERISK-28252>] -

HangupHandler manager events are never thrown
(Reported by Gerald Schnabel)

   - [ASTERISK-28249
   <https://issues.asterisk.org/jira/browse/ASTERISK-28249>] -

res_monitor: Segfault with Monitor(wav,file,i)
(Reported by Valentin Vidić)

   - [ASTERISK-28244
   <https://issues.asterisk.org/jira/browse/ASTERISK-28244>] -

stasis: Filter messages at publishing to AMI/ARI
(Reported by Joshua C. Colp)

   - [ASTERISK-28231
   <https://issues.asterisk.org/jira/browse/ASTERISK-28231>] -

res_http_websocket: Not responding to Connection Close Frame (opcode 8)
(Reported by Jeremy Lainé)

   - [ASTERISK-28197
   <https://issues.asterisk.org/jira/browse/ASTERISK-28197>] -

stasis: ast_endpoint struct holds the channel_ids of channels past
destruction in certain cases
(Reported by Mohit Dhiman)

   - [ASTERISK-28232
   <https://issues.asterisk.org/jira/browse/ASTERISK-28232>] -

core: RAII using clang use-after-scope issue
(Reported by Diederik de Groot)

   - [ASTERISK-28230
   <https://issues.asterisk.org/jira/browse/ASTERISK-28230>] -

res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks
GXV3140 video telephony
(Reported by David Kuehling)

   - [ASTERISK-28162
   <https://issues.asterisk.org/jira/browse/ASTERISK-28162>] -

[patch] need to reset DTMF last sequence number and timestamp on RTP
(Reported by Alexei Gradinari)

   - [ASTERISK-28225
   <https://issues.asterisk.org/jira/browse/ASTERISK-28225>] -

app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if
message marked "urgent"
(Reported by boatright)

   - [ASTERISK-28218
   <https://issues.asterisk.org/jira/browse/ASTERISK-28218>] -

app_queue: Asterisk crashes when using Queue with a pre-dial handler
(option b)
(Reported by Mark)

   - [ASTERISK-28212
   <https://issues.asterisk.org/jira/browse/ASTERISK-28212>] -

stasis: Statistics broke ABI under developer mode
(Reported by Joshua C. Colp)

   - [ASTERISK-28222
   <https://issues.asterisk.org/jira/browse/ASTERISK-28222>] -

Regression: MWI polling no longer works
(Reported by abelbeck)

   - [ASTERISK-28221
   <https://issues.asterisk.org/jira/browse/ASTERISK-28221>] -

Bug in ast_coredumper
(Reported by Andrew Nagy)

   - [ASTERISK-28215
   <https://issues.asterisk.org/jira/browse/ASTERISK-28215>] -

app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
(Reported by George Joseph)

   - [ASTERISK-27959
   <https://issues.asterisk.org/jira/browse/ASTERISK-27959>] -

[patch] Asterisk 15.4.1 h264 fmtp negotiation problem
(Reported by David Kuehling)

   - [ASTERISK-28201
   <https://issues.asterisk.org/jira/browse/ASTERISK-28201>] -

[patch] confbridge: no announce to the marked users when they join an empty
(Reported by Alexei Gradinari)

   - [ASTERISK-28117
   <https://issues.asterisk.org/jira/browse/ASTERISK-28117>] -

stasis: Add statistics for usage when in developer mode
(Reported by Joshua C. Colp)

   - [ASTERISK-28186
   <https://issues.asterisk.org/jira/browse/ASTERISK-28186>] -

stasis: Filter messages at publishing based on to_* presence
(Reported by Joshua C. Colp)

   - [ASTERISK-28194
   <https://issues.asterisk.org/jira/browse/ASTERISK-28194>] -

chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)

   - [ASTERISK-27095
   <https://issues.asterisk.org/jira/browse/ASTERISK-27095>] -

chan_pjsip: When connected_line_method is set to invite, we're not trying
(Reported by George Joseph)

   - [ASTERISK-28182
   <https://issues.asterisk.org/jira/browse/ASTERISK-28182>] -

chan_pjsip: When connected_line_method is set to invite, asterisk is not
trying UPDATE
(Reported by nappsoft)

   - [ASTERISK-28151
   <https://issues.asterisk.org/jira/browse/ASTERISK-28151>] -

app_voicemail: MWI fails with mailboxes=##@device instead of
(Reported by Ronald Raikes)

   - [ASTERISK-28125
   <https://issues.asterisk.org/jira/browse/ASTERISK-28125>] -

app_queue: Revert broken queue channel reference patch
(Reported by lvl)

   - [ASTERISK-28157
   <https://issues.asterisk.org/jira/browse/ASTERISK-28157>] -

Asterisk crashes when the res_pjsip_* modules unload
(Reported by sungtae kim)

   - [ASTERISK-28159
   <https://issues.asterisk.org/jira/browse/ASTERISK-28159>] -

SIGABRT caused by stack corruption in hashkeys_read when no matching keys
(Reported by Michael Walton)

   - [ASTERISK-28140
   <https://issues.asterisk.org/jira/browse/ASTERISK-28140>] -

repeated segmentation faults
(Reported by Eyal Hasson)

   - [ASTERISK-28169
   <https://issues.asterisk.org/jira/browse/ASTERISK-28169>] -

ARI /channels/create handler causes core dump
(Reported by sungtae kim)

   - [ASTERISK-28103
   <https://issues.asterisk.org/jira/browse/ASTERISK-28103>] -

stasis: Filter messages at publishing to reduce work done
(Reported by Joshua C. Colp)

   - [ASTERISK-28129
   <https://issues.asterisk.org/jira/browse/ASTERISK-28129>] -

Incorrect Behavior for rewrite_contact when Re-Invite omits routset
(Reported by Torrey Searle)

   - [ASTERISK-28158
   <https://issues.asterisk.org/jira/browse/ASTERISK-28158>] -

Some conditions prevent running of el_end, break the terminal.
(Reported by Corey Farrell)

   - [ASTERISK-28110
   <https://issues.asterisk.org/jira/browse/ASTERISK-28110>] -

rtp: Incorrect Packetization
(Reported by Robert Cripps)

   - [ASTERISK-28146
   <https://issues.asterisk.org/jira/browse/ASTERISK-28146>] -

pbx_config: Only the first [globals] section is processed.
(Reported by Corey Farrell)

   - [ASTERISK-28150
   <https://issues.asterisk.org/jira/browse/ASTERISK-28150>] -

Formatting error in documentation
(Reported by Scott Griepentrog)

   - [ASTERISK-28081
   <https://issues.asterisk.org/jira/browse/ASTERISK-28081>] -

chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in
(Reported by Luit van Drongelen)

   - [ASTERISK-28137
   <https://issues.asterisk.org/jira/browse/ASTERISK-28137>] -

res_pjsip_notify: improve realtime performance on CLI completion on the
(Reported by Alexei Gradinari)

   - [ASTERISK-27980
   <https://issues.asterisk.org/jira/browse/ASTERISK-27980>] -

Caller ID cannot be changed on Attended Transfer before dialing out
(Reported by Alexei Gradinari)

   - [ASTERISK-28107
   <https://issues.asterisk.org/jira/browse/ASTERISK-28107>] -

app_confbridge: Participant info labels aren't being added to the SDPs
(Reported by George Joseph)

   - [ASTERISK-28089
   <https://issues.asterisk.org/jira/browse/ASTERISK-28089>] -

function ast_sendtext() create RTP realtime packets with a trailing null
byte in the payload
(Reported by Emmanuel BUU)

   - [ASTERISK-28076
   <https://issues.asterisk.org/jira/browse/ASTERISK-28076>] -

bridging: Asterisk crashes when receiving an empty realtime text frame
(Reported by Emmanuel BUU)

   - [ASTERISK-28084
   <https://issues.asterisk.org/jira/browse/ASTERISK-28084>] -

app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)

   - [ASTERISK-28077
   <https://issues.asterisk.org/jira/browse/ASTERISK-28077>] -

res_pjsip: improve realtime performance on CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)

   - [ASTERISK-27920
   <https://issues.asterisk.org/jira/browse/ASTERISK-27920>] -

app_queue: Queue member considered inuse after immediately hanging up
during dialing.
(Reported by Cao Minh Hiep)

   - [ASTERISK-26094
   <https://issues.asterisk.org/jira/browse/ASTERISK-26094>] -

stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)

   - [ASTERISK-28065
   <https://issues.asterisk.org/jira/browse/ASTERISK-28065>] -

res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)

   - [ASTERISK-28057
   <https://issues.asterisk.org/jira/browse/ASTERISK-28057>] -

chan_sip: SipNotify via AMI behaves differently to CLI
(Reported by Peter Katzmann)

   - [ASTERISK-28045
   <https://issues.asterisk.org/jira/browse/ASTERISK-28045>] -

configure script does not enforce libunbound2 version
(Reported by Samuel Galarneau)

   - [ASTERISK-28070
   <https://issues.asterisk.org/jira/browse/ASTERISK-28070>] -

testsuite: Sniffer assumes pjmedia will use ports below 10000
(Reported by Joshua C. Colp)

   - [ASTERISK-27854
   <https://issues.asterisk.org/jira/browse/ASTERISK-27854>] -

rtp: Crash in off-nominal case where RTP instance can't be set up
(Reported by Lei Fu)

   - [ASTERISK-28034
   <https://issues.asterisk.org/jira/browse/ASTERISK-28034>] -

chan_sip unstable with TLS after asterisk start or reloads
(Reported by David Hajek)

   - [ASTERISK-28059
   <https://issues.asterisk.org/jira/browse/ASTERISK-28059>] -

PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)

   - [ASTERISK-27121
   <https://issues.asterisk.org/jira/browse/ASTERISK-27121>] -

res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)

   - [ASTERISK-28047
   <https://issues.asterisk.org/jira/browse/ASTERISK-28047>] -

chan_pjsip: Declined video stream is added when no video codecs configured
and session refresh with removed video stream occurs
(Reported by Will)

   - [ASTERISK-28033
   <https://issues.asterisk.org/jira/browse/ASTERISK-28033>] -

AMI event "NewExten" is set to the wrong class
(Reported by lvl)

   - [ASTERISK-28049
   <https://issues.asterisk.org/jira/browse/ASTERISK-28049>] -

res_pjproject build failure
(Reported by Jaco Kroon)

   - [ASTERISK-28029
   <https://issues.asterisk.org/jira/browse/ASTERISK-28029>] -

[patch] res_musiconhold : music on hold will not start if previous hold
just reached end of file
(Reported by Frederic LE FOLL)

   - [ASTERISK-28005
   <https://issues.asterisk.org/jira/browse/ASTERISK-28005>] -

channel.c: ARI ring only once
(Reported by Hajek Michal)

   - [ASTERISK-28032
   <https://issues.asterisk.org/jira/browse/ASTERISK-28032>] -

Realtime queuemembers are not updated during retry phase
(Reported by lvl)

   - [ASTERISK-27988
   <https://issues.asterisk.org/jira/browse/ASTERISK-27988>] -

alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not
(Reported by Joshua C. Colp)

   - [ASTERISK-27981
   <https://issues.asterisk.org/jira/browse/ASTERISK-27981>] -

res_fax: Fax session leak with fax gatewaying
(Reported by pasandev)

   - [ASTERISK-28020
   <https://issues.asterisk.org/jira/browse/ASTERISK-28020>] -

res_pjsip_transport_websocket: Properly set 'received' for IPv6
(Reported by Sean Bright)

   - [ASTERISK-28002
   <https://issues.asterisk.org/jira/browse/ASTERISK-28002>] -

When T.140 realtime text is negociated, a lot of debug traces are generated
(Reported by Emmanuel BUU)

   - [ASTERISK-27881
   <https://issues.asterisk.org/jira/browse/ASTERISK-27881>] -

PBX calls via chan_sip TCP trunk now get authentification error
(Reported by Ian Gilmour)

   - [ASTERISK-28022
   <https://issues.asterisk.org/jira/browse/ASTERISK-28022>] -

res_pjsip realtime: uri column in ps_contacts table can be too short
(Reported by Florian Floimair)

   - [ASTERISK-27944
   <https://issues.asterisk.org/jira/browse/ASTERISK-27944>] -

res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for
(Reported by Joshua Elson)

   - [ASTERISK-28007
   <https://issues.asterisk.org/jira/browse/ASTERISK-28007>] -

rtcp-mux is put in SDP answer regardless of offer
(Reported by Torrey Searle)

   - [ASTERISK-27398
   <https://issues.asterisk.org/jira/browse/ASTERISK-27398>] -

No joint capabilities with video and audio-only streams
(Reported by Benjamin Keith Ford)

   - [ASTERISK-27973
   <https://issues.asterisk.org/jira/browse/ASTERISK-27973>] -

(Reported by Valentin Safonov)

   - [ASTERISK-27997
   <https://issues.asterisk.org/jira/browse/ASTERISK-27997>] -

pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
(Reported by Alexander Traud)

   - [ASTERISK-27999
   <https://issues.asterisk.org/jira/browse/ASTERISK-27999>] -

Wrong SRTP use status report
(Reported by Salah Ahmed)

   - [ASTERISK-28001
   <https://issues.asterisk.org/jira/browse/ASTERISK-28001>] -

res_pjsip_registrar: Improve performance of inbound handling
(Reported by Joshua C. Colp)

   - [ASTERISK-27966
   <https://issues.asterisk.org/jira/browse/ASTERISK-27966>] -

pjsip: Race condition in 183 re transmission can result in a deadlock
(Reported by Torrey Searle)

   - [ASTERISK-15331
   <https://issues.asterisk.org/jira/browse/ASTERISK-15331>] -

make menuselect fails due to undefined symbols (initscr32, w32addch) in
(Reported by Majdi Bsoul)

   - [ASTERISK-14935
   <https://issues.asterisk.org/jira/browse/ASTERISK-14935>] -

[regression] menuselect compilation failure on Solaris 10
(Reported by Samuel Owens)

   - [ASTERISK-12382
   <https://issues.asterisk.org/jira/browse/ASTERISK-12382>] -

menuselect compilation failure on Solaris 10 / gcc 3.4.3
(Reported by rleasure)

   - [ASTERISK-9107 <https://issues.asterisk.org/jira/browse/ASTERISK-9107>]

menuselect compilation failure on Solaris 10/gcc-4.1.1
(Reported by Bob Atkins)

   - [ASTERISK-27991
   <https://issues.asterisk.org/jira/browse/ASTERISK-27991>] -

BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)

   - [ASTERISK-27548
   <https://issues.asterisk.org/jira/browse/ASTERISK-27548>] -

res_pjsip_endpoint_identifier_ip only matches against "generic string"
(Reported by George Joseph)

   - [ASTERISK-27990
   <https://issues.asterisk.org/jira/browse/ASTERISK-27990>] -

res_rtp_asterisk: Requires OpenSSL in Developer Mode.
(Reported by Alexander Traud)

   - [ASTERISK-27591
   <https://issues.asterisk.org/jira/browse/ASTERISK-27591>] -

Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)

   - [ASTERISK-27978
   <https://issues.asterisk.org/jira/browse/ASTERISK-27978>] -

res_pjsip: Change default transport keepalive to preserve behavior
(Reported by Joshua C. Colp)

   - [ASTERISK-27968
   <https://issues.asterisk.org/jira/browse/ASTERISK-27968>] -

systemd: asterisk.service
(Reported by seanchann.zhou)

   - [ASTERISK-27880
   <https://issues.asterisk.org/jira/browse/ASTERISK-27880>] -

[patch] pjproject_bundled: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27810
   <https://issues.asterisk.org/jira/browse/ASTERISK-27810>] -

BASIC-RETRANS: Implement receive
(Reported by Benjamin Keith Ford)

   - [ASTERISK-27972
   <https://issues.asterisk.org/jira/browse/ASTERISK-27972>] -

res_sorcery_config: Allow object name based matching
(Reported by Joshua C. Colp)

   - [ASTERISK-27965
   <https://issues.asterisk.org/jira/browse/ASTERISK-27965>] -

module: Remove old modules, update support levels
(Reported by Joshua C. Colp)

   - [ASTERISK-25548
   <https://issues.asterisk.org/jira/browse/ASTERISK-25548>] -

stasis: Improve message type "Use of before init/after destruction" error
(Reported by Joshua C. Colp)

   - [ASTERISK-27967
   <https://issues.asterisk.org/jira/browse/ASTERISK-27967>] -

srtp: rejecting short sdes lifetimes incompatible with obihai ATAs
(Reported by Nick French)

   - [ASTERISK-27961
   <https://issues.asterisk.org/jira/browse/ASTERISK-27961>] -

res_pjsip: Spurious ERROR logging when printing headers in sip_msg
(Reported by Nick French)

   - [ASTERISK-27563
   <https://issues.asterisk.org/jira/browse/ASTERISK-27563>] -

pjsip modules always get -O2 even when DONT_OPTIMIZE is set
(Reported by George Joseph)

   - [ASTERISK-27347
   <https://issues.asterisk.org/jira/browse/ASTERISK-27347>] -

[patch] pjproject_bundled: Disable TCP/TLS keep-alives.
(Reported by Alexander Traud)

   - [ASTERISK-27957
   <https://issues.asterisk.org/jira/browse/ASTERISK-27957>] -

PJSIP proposes ICE candidates on answer even if not in offer
(Reported by Torrey Searle)

   - [ASTERISK-27938
   <https://issues.asterisk.org/jira/browse/ASTERISK-27938>] -

[patch] Compile fails with `IPTOS_MINCOST' undeclared.
(Reported by Alexander Traud)

   - [ASTERISK-27955
   <https://issues.asterisk.org/jira/browse/ASTERISK-27955>] -

res_pjsip_session: sdp group:BUNDLE attribute truncated
(Reported by Kevin Harwell)

   - [ASTERISK-27956
   <https://issues.asterisk.org/jira/browse/ASTERISK-27956>] -

res_pjsip_pubsub: segfault in function publish_expire
(Reported by Alexei Gradinari)

   - [ASTERISK-27949
   <https://issues.asterisk.org/jira/browse/ASTERISK-27949>] -

res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason
(Reported by Ross Beer)

   - [ASTERISK-27763
   <https://issues.asterisk.org/jira/browse/ASTERISK-27763>] -

res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of
declining stream
(Reported by Thiago Coutinho)

   - [ASTERISK-27657
   <https://issues.asterisk.org/jira/browse/ASTERISK-27657>] -

res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not
(Reported by Jared Hull)

   - [ASTERISK-27080
   <https://issues.asterisk.org/jira/browse/ASTERISK-27080>] -

res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled
(Reported by Torrey Searle)

   - [ASTERISK-26686
   <https://issues.asterisk.org/jira/browse/ASTERISK-26686>] -

res_pjsip: Lock inversion in transport management
(Reported by Ross Beer)

   - [ASTERISK-27939
   <https://issues.asterisk.org/jira/browse/ASTERISK-27939>] -

[patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
(Reported by Alexander Traud)

   - [ASTERISK-27783
   <https://issues.asterisk.org/jira/browse/ASTERISK-27783>] -

res_pjsip_pubsub: apparent crash on shutdown
(Reported by Kevin Harwell)

   - [ASTERISK-27870
   <https://issues.asterisk.org/jira/browse/ASTERISK-27870>] -

app_confbridge: Conference bridge and announcer channels are not removed if
conference is ended as soon as it starts
(Reported by Robert Mordec)

   - [ASTERISK-27909
   <https://issues.asterisk.org/jira/browse/ASTERISK-27909>] -

cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch
(Reported by Denis Lebedev)

   - [ASTERISK-26987
   <https://issues.asterisk.org/jira/browse/ASTERISK-26987>] -

pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi
(Reported by Kirsty Tyerman)

   - [ASTERISK-27943
   <https://issues.asterisk.org/jira/browse/ASTERISK-27943>] -

AMI: Action SendText needs to use the correct thread.
(Reported by Richard Mudgett)

   - [ASTERISK-27942
   <https://issues.asterisk.org/jira/browse/ASTERISK-27942>] -

res_pjsip_messaging doesn't accept application/* content-types.
(Reported by George Joseph)

   - [ASTERISK-27936
   <https://issues.asterisk.org/jira/browse/ASTERISK-27936>] -

res_pjsip_session doesn't update media when a 200 comes in with a different
port than a 183
(Reported by George Joseph)

   - [ASTERISK-27933
   <https://issues.asterisk.org/jira/browse/ASTERISK-27933>] -

[patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)

   - [ASTERISK-27625
   <https://issues.asterisk.org/jira/browse/ASTERISK-27625>] -

channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)

   - [ASTERISK-27931
   <https://issues.asterisk.org/jira/browse/ASTERISK-27931>] -

[patch] BuildSystem: Enable ./configure in Solaris 11.
(Reported by Alexander Traud)

   - [ASTERISK-27926
   <https://issues.asterisk.org/jira/browse/ASTERISK-27926>] -

[patch] bootstrap.sh: find -maxdepth is not POSIX compatible.
(Reported by Alexander Traud)

   - [ASTERISK-27903
   <https://issues.asterisk.org/jira/browse/ASTERISK-27903>] -

menuselect: GCC 8: restrict-qualified parameter passed and aliased.
(Reported by Alexander Traud)

   - [ASTERISK-27914
   <https://issues.asterisk.org/jira/browse/ASTERISK-27914>] -

[patch] tests/test_utils: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27705
   <https://issues.asterisk.org/jira/browse/ASTERISK-27705>] -

chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)

   - [ASTERISK-27848
   <https://issues.asterisk.org/jira/browse/ASTERISK-27848>] -

rtp: DTMF Breaks With telephony-event/16000
(Reported by Dominic)

   - [ASTERISK-27908
   <https://issues.asterisk.org/jira/browse/ASTERISK-27908>] -

[patch] crypto.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27905
   <https://issues.asterisk.org/jira/browse/ASTERISK-27905>] -

[patch] res_srtp: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27888
   <https://issues.asterisk.org/jira/browse/ASTERISK-27888>] -

SQL fetch error on query which return 0 columns
(Reported by Alexei Gradinari)

   - [ASTERISK-27902
   <https://issues.asterisk.org/jira/browse/ASTERISK-27902>] -

chan_pjsip isn't updating hangupcause on 4XX responses
(Reported by George Joseph)

   - [ASTERISK-27901
   <https://issues.asterisk.org/jira/browse/ASTERISK-27901>] -

[patch] ooh323c: GCC 8: output truncated before terminating nul.
(Reported by Alexander Traud)

   - [ASTERISK-27872
   <https://issues.asterisk.org/jira/browse/ASTERISK-27872>] -

res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload
(Reported by Alexei Gradinari)

   - [ASTERISK-27094
   <https://issues.asterisk.org/jira/browse/ASTERISK-27094>] -

res_fax: Deadlock when using Local channels and fax gateway
(Reported by David Brillert)

   - [ASTERISK-25261
   <https://issues.asterisk.org/jira/browse/ASTERISK-25261>] -

Manager events for MeetMe have incorrectly documented key name 'Usernum' -
should be 'User'
(Reported by Francois Blackburn)

   - [ASTERISK-27878
   <https://issues.asterisk.org/jira/browse/ASTERISK-27878>] -

[patch] tcptls.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27876
   <https://issues.asterisk.org/jira/browse/ASTERISK-27876>] -

[patch] tcptls: Allow OpenSSL configured with no-dh.
(Reported by Alexander Traud)

   - [ASTERISK-27874
   <https://issues.asterisk.org/jira/browse/ASTERISK-27874>] -

[patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method
(Reported by Alexander Traud)

   - [ASTERISK-27845
   <https://issues.asterisk.org/jira/browse/ASTERISK-27845>] -

Codec-Change Re-INVITE during DTMF can cause marker bit error
(Reported by Torrey Searle)

   - [ASTERISK-27831
   <https://issues.asterisk.org/jira/browse/ASTERISK-27831>] -

res_rtp_asterisk: Add support for abs-send-time RTP extension
(Reported by Joshua C. Colp)

   - [ASTERISK-27863
   <https://issues.asterisk.org/jira/browse/ASTERISK-27863>] -

config/ast_destroy_realtime_fields: successful DELETE is treated as failed
(Reported by Alexei Gradinari)

   - [ASTERISK-27865
   <https://issues.asterisk.org/jira/browse/ASTERISK-27865>] -

[patch]: tcptls: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27760
   <https://issues.asterisk.org/jira/browse/ASTERISK-27760>] -

Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version.
(Reported by Nic Colledge)

   - [ASTERISK-27853
   <https://issues.asterisk.org/jira/browse/ASTERISK-27853>] -

Incorrect error reported when leaving/retrieving a ODBC voicemail
(Reported by Nic Colledge)

   - [ASTERISK-27726
   <https://issues.asterisk.org/jira/browse/ASTERISK-27726>] -

chan_mobile: presents incorrect inbound Caller-ID names
(Reported by Brian)

   - [ASTERISK-27861
   <https://issues.asterisk.org/jira/browse/ASTERISK-27861>] -

[patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
(Reported by Alexander Traud)

   - [ASTERISK-27852
   <https://issues.asterisk.org/jira/browse/ASTERISK-27852>] -

cli: "manager show settings" mislabels HTTP timeout as being minutes.
(Reported by Corey Farrell)

   - [ASTERISK-27824
   <https://issues.asterisk.org/jira/browse/ASTERISK-27824>] -

Fix issues exposed by GCC 8
(Reported by George Joseph)

   - [ASTERISK-27850
   <https://issues.asterisk.org/jira/browse/ASTERISK-27850>] -

[patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on
egress again.
(Reported by Alexander Traud)

   - [ASTERISK-27811
   <https://issues.asterisk.org/jira/browse/ASTERISK-27811>] -

[patch] sip_to_pjsip: Enable python3 compatibility.
(Reported by Alexander Traud)

   - [ASTERISK-27841
   <https://issues.asterisk.org/jira/browse/ASTERISK-27841>] -

digest over for manager (ami) over http fails on too long uris
(Reported by Jaco Kroon)

   - [ASTERISK-26570
   <https://issues.asterisk.org/jira/browse/ASTERISK-26570>] -

Macro allows an infinite loop of dialplan inclusion resulting in a crash
(Reported by Tzafrir Cohen)

   - [ASTERISK-27572
   <https://issues.asterisk.org/jira/browse/ASTERISK-27572>] -

cdr_mysql creates empty records if reconnects when mysql was not up on
module load
(Reported by Tzafrir Cohen)

   - [ASTERISK-27801
   <https://issues.asterisk.org/jira/browse/ASTERISK-27801>] -

Asterisk got stuck while enabling "ari set debug all on"
(Reported by shaurya jain)

   - [ASTERISK-27795
   <https://issues.asterisk.org/jira/browse/ASTERISK-27795>] -

chan_sip: one way / no audio with srtp
(Reported by Florian Kaiser)

   - [ASTERISK-27800
   <https://issues.asterisk.org/jira/browse/ASTERISK-27800>] -

One way audio when calling from Asterisk(sip trunk) to another number where
both are connected to a SBC using TLS+SRTP
(Reported by Artur Pires)

   - [ASTERISK-26806
   <https://issues.asterisk.org/jira/browse/ASTERISK-26806>] -

pjsip_options: rework to make more efficient
(Reported by Kevin Harwell)

   - [ASTERISK-27814
   <https://issues.asterisk.org/jira/browse/ASTERISK-27814>] -

translate: interpolated frames are not passed through
(Reported by Kevin Harwell)

   - [ASTERISK-27812
   <https://issues.asterisk.org/jira/browse/ASTERISK-27812>] -

When the ooh323 debug is on there is no ringing signal to incoming calls
via H323 trunk.
(Reported by Dimos)

   - [ASTERISK-26893
   <https://issues.asterisk.org/jira/browse/ASTERISK-26893>] -

No "alert" or "progress" in chan_ooh323 if debug is enabled only on the
(Reported by Marco Giordani)

   - [ASTERISK-27804
   <https://issues.asterisk.org/jira/browse/ASTERISK-27804>] -

bridge_softmix / app_confbridge: Add support for combining REMB reports
(Reported by Joshua C. Colp)

   - [ASTERISK-27639
   <https://issues.asterisk.org/jira/browse/ASTERISK-27639>] -

[patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)

   - [ASTERISK-27418
   <https://issues.asterisk.org/jira/browse/ASTERISK-27418>] -

app_confbridge: "core show profile bridge" does not output "sfu" when
video_mode is sfu
(Reported by Carlos Chavez)

   - [ASTERISK-27809
   <https://issues.asterisk.org/jira/browse/ASTERISK-27809>] -

[patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally.
(Reported by Alexander Traud)

   - [ASTERISK-27808
   <https://issues.asterisk.org/jira/browse/ASTERISK-27808>] -

[patch] chan_vpb: Avoid GNU old-style field designator extension.
(Reported by Alexander Traud)

   - [ASTERISK-27806
   <https://issues.asterisk.org/jira/browse/ASTERISK-27806>] -

BASIC-RETRANS: Implement send
(Reported by Benjamin Keith Ford)

   - [ASTERISK-27774
   <https://issues.asterisk.org/jira/browse/ASTERISK-27774>] -

res_musiconhold: Music on hold restarts after every announcement
(Reported by lvl)

   - [ASTERISK-27782
   <https://issues.asterisk.org/jira/browse/ASTERISK-27782>] -

cdr_mysql: Missing MYSQL_PORT definition
(Reported by Evandro César Arruda)

   - [ASTERISK-27614
   <https://issues.asterisk.org/jira/browse/ASTERISK-27614>] -

res_pjsip_session: SDP origin does not use resolved address
(Reported by John M.)

   - [ASTERISK-27776
   <https://issues.asterisk.org/jira/browse/ASTERISK-27776>] -

res_rtp_asterisk: Add support for sending RTCP feedback messages
(Reported by Joshua C. Colp)

   - [ASTERISK-27740
   <https://issues.asterisk.org/jira/browse/ASTERISK-27740>] -

chan_sip: New Channel creation from new SIP dialog with Replaces failed to
be properly tracked and destroyed
(Reported by Shannon Price)

   - [ASTERISK-27786
   <https://issues.asterisk.org/jira/browse/ASTERISK-27786>] -

app_confbridge: Add ability to enable and configure REMB support
(Reported by Joshua C. Colp)

   - [ASTERISK-27706
   <https://issues.asterisk.org/jira/browse/ASTERISK-27706>] -

PJSIP: Deadlock shutting down subscription TCP connection and sending
subscription message.
(Reported by Ross Beer)

   - [ASTERISK-27688
   <https://issues.asterisk.org/jira/browse/ASTERISK-27688>] -

res_pjsip: Crash on TCP PJSIP Transport Disconnect
(Reported by Ross Beer)

   - [ASTERISK-27758
   <https://issues.asterisk.org/jira/browse/ASTERISK-27758>] -

res_rtp_asterisk: Add support for raising RTCP feedback messages
(Reported by Joshua C. Colp)

   - [ASTERISK-26366
   <https://issues.asterisk.org/jira/browse/ASTERISK-26366>] -

rtp: RTCP messages with REMB trigger fast picture update
(Reported by Joshua C. Colp)

   - [ASTERISK-27773
   <https://issues.asterisk.org/jira/browse/ASTERISK-27773>] -

Command line not being parsed correctly with getopt not from glibc
(Reported by Guido Falsi)

   - [ASTERISK-27435
   <https://issues.asterisk.org/jira/browse/ASTERISK-27435>] -

[patch] configure: pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)

   - [ASTERISK-27761
   <https://issues.asterisk.org/jira/browse/ASTERISK-27761>] -

[patch] BuildSystem: With external editline, do not require libs for
internal editline.
(Reported by Alexander Traud)

   - [ASTERISK-27755
   <https://issues.asterisk.org/jira/browse/ASTERISK-27755>] -

ConfBridge: raise ConfbridgeTalking when put on hold and clear talking
(Reported by Kevin Harwell)

   - [ASTERISK-27743
   <https://issues.asterisk.org/jira/browse/ASTERISK-27743>] -

Generic PLC doesn't work if the 2 codecs on a channel are equal
(Reported by George Joseph)

   - [ASTERISK-27745
   <https://issues.asterisk.org/jira/browse/ASTERISK-27745>] -

[patch] BuildSystem: Remove unused dependency on libltdl.
(Reported by Alexander Traud)

   - [ASTERISK-12841
   <https://issues.asterisk.org/jira/browse/ASTERISK-12841>] -

[patch] Make format_ogg_vorbis work on OpenBSD
(Reported by Michiel van Baak)

   - [ASTERISK-27720
   <https://issues.asterisk.org/jira/browse/ASTERISK-27720>] -

[patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in
(Reported by Alexander Traud)

   - [ASTERISK-27741
   <https://issues.asterisk.org/jira/browse/ASTERISK-27741>] -

res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't account for more than
one 'Reason' header
(Reported by Ross Beer)

   - [ASTERISK-27734
   <https://issues.asterisk.org/jira/browse/ASTERISK-27734>] -

[patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux.
(Reported by Alexander Traud)

   - [ASTERISK-27686
   <https://issues.asterisk.org/jira/browse/ASTERISK-27686>] -

[patch] install_prereq: Update FreeBSD libraries.
(Reported by Alexander Traud)

   - [ASTERISK-27733
   <https://issues.asterisk.org/jira/browse/ASTERISK-27733>] -

[patch] res_srtp: Add support for libsrtp2.x on openSUSE.
(Reported by Alexander Traud)

   - [ASTERISK-11015
   <https://issues.asterisk.org/jira/browse/ASTERISK-11015>] -

NetBSD Build Needs RPATH set in 1.2.25
(Reported by Curt Sampson)

   - [ASTERISK-27641
   <https://issues.asterisk.org/jira/browse/ASTERISK-27641>] -

BuildSystem: Enable Better Backtraces in FreeBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27671
   <https://issues.asterisk.org/jira/browse/ASTERISK-27671>] -

Deprecate legacy modules
(Reported by Corey Farrell)

   - [ASTERISK-25586
   <https://issues.asterisk.org/jira/browse/ASTERISK-25586>] -

uuid_generate_random detection failure
(Reported by John Nemeth)

   - [ASTERISK-27721
   <https://issues.asterisk.org/jira/browse/ASTERISK-27721>] -

[patch] BuildSystem: Enable PortAudio in NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27715
   <https://issues.asterisk.org/jira/browse/ASTERISK-27715>] -

[patch] BuildSystem: AC_PATH_PROG sets to colon character when not found.
(Reported by Alexander Traud)

   - [ASTERISK-27554
   <https://issues.asterisk.org/jira/browse/ASTERISK-27554>] -

res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints
(Reported by Ross Beer)

   - [ASTERISK-27703
   <https://issues.asterisk.org/jira/browse/ASTERISK-27703>] -

AMI Action VoicemailUsersList returns 0 MessageCount
(Reported by Sébastien Duthil)

   - [ASTERISK-27674
   <https://issues.asterisk.org/jira/browse/ASTERISK-27674>] -

chan_sip: RTP framing issues on outgoing calls
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-27441
   <https://issues.asterisk.org/jira/browse/ASTERISK-27441>] -

PJSIP: Forked INVITE SDP negotiation gets one way audio.
(Reported by lvl)

   - [ASTERISK-27718
   <https://issues.asterisk.org/jira/browse/ASTERISK-27718>] -

[patch] BuildSystem: Enable Lua in NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27722
   <https://issues.asterisk.org/jira/browse/ASTERISK-27722>] -

[patch] BuildSystem: Depend not implicitly but explicitly on external
(Reported by Alexander Traud)

   - [ASTERISK-27719
   <https://issues.asterisk.org/jira/browse/ASTERISK-27719>] -

[patch] res_http_post: Enable GMime in NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27716
   <https://issues.asterisk.org/jira/browse/ASTERISK-27716>] -

[patch] BuildSystem: Enable autotools in NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27714
   <https://issues.asterisk.org/jira/browse/ASTERISK-27714>] -

[patch] chan_unistim: NetBSD has an incompatible struct in_pktinfo.
(Reported by Alexander Traud)

   - [ASTERISK-27713
   <https://issues.asterisk.org/jira/browse/ASTERISK-27713>] -

[patch] BuildSystem: Cast any intptr_t explicitly to its proposed type.
(Reported by Alexander Traud)

   - [ASTERISK-27712
   <https://issues.asterisk.org/jira/browse/ASTERISK-27712>] -

[patch] BuildSystem: Detect whether uselocale(.) is available.
(Reported by Alexander Traud)

   - [ASTERISK-27711
   <https://issues.asterisk.org/jira/browse/ASTERISK-27711>] -

[patch] BuildSystem: Avoid re-defining of pthread_* on NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27710
   <https://issues.asterisk.org/jira/browse/ASTERISK-27710>] -

[patch] BuildSystem: Install init scripts on openSUSE Tumbleweed.
(Reported by Alexander Traud)

   - [ASTERISK-27709
   <https://issues.asterisk.org/jira/browse/ASTERISK-27709>] -

[patch] BuildSystem: Avoid == for comparison in ./configure.
(Reported by Alexander Traud)

   - [ASTERISK-27610
   <https://issues.asterisk.org/jira/browse/ASTERISK-27610>] -

app_amd.so returning TOOLONG before reaching the timeout
(Reported by Michael Cargile)

   - [ASTERISK-26688
   <https://issues.asterisk.org/jira/browse/ASTERISK-26688>] -

Documentation: voicemail.conf.sample shows 512 limit for emailbody field,
however this is only true if compiled with LOW_MEMORY option
(Reported by Fran Vicente)

   - [ASTERISK-27568
   <https://issues.asterisk.org/jira/browse/ASTERISK-27568>] -

PJSIP: Crash during SIP attended transfer.
(Reported by Bryan Walters)

   - [ASTERISK-27659
   <https://issues.asterisk.org/jira/browse/ASTERISK-27659>] -

Output from rawman truncated if output is long enough
(Reported by Bojan Nemčić)

   - [ASTERISK-27692
   <https://issues.asterisk.org/jira/browse/ASTERISK-27692>] -

bridging: Sometimes cloning the stream topology causes a crash
(Reported by Richard Mudgett)

   - [ASTERISK-27488
   <https://issues.asterisk.org/jira/browse/ASTERISK-27488>] -

core: If frame with unnegotiated format is read crash will occur
(Reported by Sébastien Duthil)

   - [ASTERISK-24488
   <https://issues.asterisk.org/jira/browse/ASTERISK-24488>] -

Wrong remote identity and target in dialog package XML in NOTIFY
(Reported by Alejandro Padilla)

   - [ASTERISK-24386
   <https://issues.asterisk.org/jira/browse/ASTERISK-24386>] -

Asterisk "doc/lang/language-criteria.txt" needs update or removal.
(Reported by Rusty Newton)

   - [ASTERISK-27646
   <https://issues.asterisk.org/jira/browse/ASTERISK-27646>] -

ICE fails with no candidate nominated
(Reported by Thomas Guebels)

   - [ASTERISK-27689
   <https://issues.asterisk.org/jira/browse/ASTERISK-27689>] -

[patch] rtp_engine: Load format name / mime type in uppercase again.
(Reported by Alexander Traud)

   - [ASTERISK-27679
   <https://issues.asterisk.org/jira/browse/ASTERISK-27679>] -

res_pjsip: Endpoint destruction does not free DTLS configuration
(Reported by Mak Dee)

   - [ASTERISK-27684
   <https://issues.asterisk.org/jira/browse/ASTERISK-27684>] -

[patch] install_prereq: Update OpenBSD libraries.
(Reported by Alexander Traud)

   - [ASTERISK-27680
   <https://issues.asterisk.org/jira/browse/ASTERISK-27680>] -

[patch] res_calendar: Specialized calendars depend on symbols of general
(Reported by Alexander Traud)

   - [ASTERISK-27681
   <https://issues.asterisk.org/jira/browse/ASTERISK-27681>] -

[patch] BuildSystem: Enable IMAP storage on OpenBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27677
   <https://issues.asterisk.org/jira/browse/ASTERISK-27677>] -

[patch] BuildSystem: Enable system provided libedit on OpenBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27670
   <https://issues.asterisk.org/jira/browse/ASTERISK-27670>] -

[patch] BuildSystem: Remove chan_h323 leftovers.
(Reported by Alexander Traud)

   - [ASTERISK-27595
   <https://issues.asterisk.org/jira/browse/ASTERISK-27595>] -

[patch] BuildSystem: Invoke ldconfig with previous paths.
(Reported by Alexander Traud)

   - [ASTERISK-27631
   <https://issues.asterisk.org/jira/browse/ASTERISK-27631>] -

[patch] BuildSystem: Do not warn when bash is not installed.
(Reported by Alexander Traud)

   - [ASTERISK-27666
   <https://issues.asterisk.org/jira/browse/ASTERISK-27666>] -

chan_sip: Crash processing CANCEL request
(Reported by Leandro Dardini)

   - [ASTERISK-27584
   <https://issues.asterisk.org/jira/browse/ASTERISK-27584>] -

Internal pjproject build doesn't disable bcg729
(Reported by Stuart Henderson)

   - [ASTERISK-27669
   <https://issues.asterisk.org/jira/browse/ASTERISK-27669>] -

[patch] codecs: Add support for WebRTC iLBC 2.0.
(Reported by Alexander Traud)

   - [ASTERISK-27634
   <https://issues.asterisk.org/jira/browse/ASTERISK-27634>] -

Determine if the internal editline and stdtime libraries are still relevant
(Reported by George Joseph)

   - [ASTERISK-27642
   <https://issues.asterisk.org/jira/browse/ASTERISK-27642>] -

[patch] backtrace: Avoid -Wlogical-not-parentheses.
(Reported by Alexander Traud)

   - [ASTERISK-27555
   <https://issues.asterisk.org/jira/browse/ASTERISK-27555>] -

[patch] install_prereq: Update Debian/Ubuntu libraries.
(Reported by Alexander Traud)

   - [ASTERISK-27656
   <https://issues.asterisk.org/jira/browse/ASTERISK-27656>] -

CDR: Leaking channel snapshots allocated by stasis_channel.c
(Reported by Kristijan Vrban)

   - [ASTERISK-27426
   <https://issues.asterisk.org/jira/browse/ASTERISK-27426>] -

chan_console: cannot read and write at the same time with alsa backend
(Reported by Tzafrir Cohen)

   - [ASTERISK-27621
   <https://issues.asterisk.org/jira/browse/ASTERISK-27621>] -

(null) string tailing after AsyncAGIEnd AMI event
(Reported by sungtae kim)

   - [ASTERISK-27652
   <https://issues.asterisk.org/jira/browse/ASTERISK-27652>] -

Null pointer Crash in PJSIP MWI
(Reported by Joshua Elson)

   - [ASTERISK-27571
   <https://issues.asterisk.org/jira/browse/ASTERISK-27571>] -

res_pjsip: If SIP response is received during shutdown a crash may occur
(Reported by Joshua C. Colp)

   - [ASTERISK-27619
   <https://issues.asterisk.org/jira/browse/ASTERISK-27619>] -

Build System: Require compiler to provide built-in support for atomic
(Reported by Corey Farrell)

   - [ASTERISK-27612
   <https://issues.asterisk.org/jira/browse/ASTERISK-27612>] -

Subscriptions Persist After Expiration and TCP/TLS Disconnect
(Reported by Ross Beer)

   - [ASTERISK-27637
   <https://issues.asterisk.org/jira/browse/ASTERISK-27637>] -

[patch] BuildSystem: Enable autotools in FreeBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27635
   <https://issues.asterisk.org/jira/browse/ASTERISK-27635>] -

[patch] app_voicemail: Avoid always true warnings with clang.
(Reported by Alexander Traud)

   - [ASTERISK-27599
   <https://issues.asterisk.org/jira/browse/ASTERISK-27599>] -

[patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
(Reported by Alexander Traud)

   - [ASTERISK-26563
   <https://issues.asterisk.org/jira/browse/ASTERISK-26563>] -

core: macOS devmode build fails: variable 'freeswap' set but not used
(Reported by David M. Lee)

   - [ASTERISK-27630
   <https://issues.asterisk.org/jira/browse/ASTERISK-27630>] -

[patch] editline: Avoid shifting a negative signed value.
(Reported by Alexander Traud)

   - [ASTERISK-16172
   <https://issues.asterisk.org/jira/browse/ASTERISK-16172>] -

Problems with siren14 codec; problems with siren7 sound files.
(Reported by Steve Murphy)

   - [ASTERISK-16951
   <https://issues.asterisk.org/jira/browse/ASTERISK-16951>] -

[patch] configure.ac in 1.4.37 broken with autoconf 2.60
(Reported by Stéphan Kochen)

   - [ASTERISK-27603
   <https://issues.asterisk.org/jira/browse/ASTERISK-27603>] -

[patch] install_prereq: Download latest Jansson.
(Reported by Alexander Traud)

   - [ASTERISK-27620
   <https://issues.asterisk.org/jira/browse/ASTERISK-27620>] -

New module loader aborts startup if a required module declines load.
(Reported by snuffy)

   - [ASTERISK-27607
   <https://issues.asterisk.org/jira/browse/ASTERISK-27607>] -

[patch] res_config_mysql: Avoid the header mysql_version.h.
(Reported by Alexander Traud)

   - [ASTERISK-24598
   <https://issues.asterisk.org/jira/browse/ASTERISK-24598>] -

When running ./contrib/scripts/install_prereq install-unpackaged pjproject
is installed in wrong place
(Reported by PowerPBX)

   - [ASTERISK-27602
   <https://issues.asterisk.org/jira/browse/ASTERISK-27602>] -

[patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
(Reported by Alexander Traud)

   - [ASTERISK-27600
   <https://issues.asterisk.org/jira/browse/ASTERISK-27600>] -

[patch] BuildSystem: Allow make clean all again.
(Reported by Alexander Traud)

   - [ASTERISK-27598
   <https://issues.asterisk.org/jira/browse/ASTERISK-27598>] -

[patch] install_prereq: Support package manager DNF.
(Reported by Alexander Traud)

   - [ASTERISK-26596
   <https://issues.asterisk.org/jira/browse/ASTERISK-26596>] -

Placing call on hold temporarily locks up set
(Reported by Igor Goncharovsky)

   - [ASTERISK-27596
   <https://issues.asterisk.org/jira/browse/ASTERISK-27596>] -

[patch] BuildSystem: Use the detected name for MD5 everywhere.
(Reported by Alexander Traud)

   - [ASTERISK-27594
   <https://issues.asterisk.org/jira/browse/ASTERISK-27594>] -

[patch] BuildSystem: Invoke install not in GNU but POSIX style.
(Reported by Alexander Traud)

   - [ASTERISK-27593
   <https://issues.asterisk.org/jira/browse/ASTERISK-27593>] -

[patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
(Reported by Alexander Traud)

   - [ASTERISK-27592
   <https://issues.asterisk.org/jira/browse/ASTERISK-27592>] -

[patch] BuildSystem: Detect external library Lua in version 5.3.
(Reported by Alexander Traud)

   - [ASTERISK-27491
   <https://issues.asterisk.org/jira/browse/ASTERISK-27491>] -

res_pjsip_endpoint_identifier_ip only matches against header if match by ip
(Reported by George Joseph)

   - [ASTERISK-26832
   <https://issues.asterisk.org/jira/browse/ASTERISK-26832>] -

res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581
(Reported by Ross Beer)

   - [ASTERISK-27589
   <https://issues.asterisk.org/jira/browse/ASTERISK-27589>] -

[patch] BuildSystem: Avoid $EUID and use id -u instead.
(Reported by Alexander Traud)

   - [ASTERISK-27585
   <https://issues.asterisk.org/jira/browse/ASTERISK-27585>] -

[patch] BuildSystem: Resolve resolv.h not via Generic but Particular
(Reported by Alexander Traud)

   - [ASTERISK-27575
   <https://issues.asterisk.org/jira/browse/ASTERISK-27575>] -

menuselect : remove obsolete TRACE_FRAMES compiler flag
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-27576
   <https://issues.asterisk.org/jira/browse/ASTERISK-27576>] -

[patch] res_config_pgsql: Avoid typecasting an int to unsigned char.
(Reported by Alexander Traud)

   - [ASTERISK-27560
   <https://issues.asterisk.org/jira/browse/ASTERISK-27560>] -

[patch] clang 5 does not know -Wno-format-truncation
(Reported by Alexander Traud)

   - [ASTERISK-27578
   <https://issues.asterisk.org/jira/browse/ASTERISK-27578>] -

[patch] app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)

   - [ASTERISK-27577
   <https://issues.asterisk.org/jira/browse/ASTERISK-27577>] -

[patch] chan_ooh323: Avoid typecasting an int to unsigned short.
(Reported by Alexander Traud)

   - [ASTERISK-27534
   <https://issues.asterisk.org/jira/browse/ASTERISK-27534>] -

chan_sip: Assumes iostream is non-NULL when it may not be
(Reported by Lubos Dolezel)

   - [ASTERISK-27549
   <https://issues.asterisk.org/jira/browse/ASTERISK-27549>] -

[patch] translate: Avoid absolute value on unsigned substraction.
(Reported by Alexander Traud)

   - [ASTERISK-27566
   <https://issues.asterisk.org/jira/browse/ASTERISK-27566>] -

res_pjsip_session: Improve WebRTC interop with bundling during renegotiation
(Reported by Joshua C. Colp)

   - [ASTERISK-27553
   <https://issues.asterisk.org/jira/browse/ASTERISK-27553>] -

[patch] res_curl: Avoid error message on unload.
(Reported by Alexander Traud)

   - [ASTERISK-27557
   <https://issues.asterisk.org/jira/browse/ASTERISK-27557>] -

[patch] clang 5.0: implicit conversion to char changes value to negative.
(Reported by Alexander Traud)

   - [ASTERISK-27550
   <https://issues.asterisk.org/jira/browse/ASTERISK-27550>] -

[patch] bridge_softmix: Avoid warning about an uninitialized variable.
(Reported by Alexander Traud)

   - [ASTERISK-27559
   <https://issues.asterisk.org/jira/browse/ASTERISK-27559>] -

[patch] editline: Avoid comparison between pointer and zero character
(Reported by Alexander Traud)

   - [ASTERISK-27558
   <https://issues.asterisk.org/jira/browse/ASTERISK-27558>] -

[patch] codec_gsm: Avoid shifting a negative signed value.
(Reported by Alexander Traud)

   - [ASTERISK-25329
   <https://issues.asterisk.org/jira/browse/ASTERISK-25329>] -

Asterisk configure fails on 'cannot find ptlib-config', despite
ptlib-config existing
(Reported by Rusty Newton)

   - [ASTERISK-27552
   <https://issues.asterisk.org/jira/browse/ASTERISK-27552>] -

[patch] chan_ooh323: Limit outgoinglimit to positive values as intended.
(Reported by Alexander Traud)

   - [ASTERISK-27551
   <https://issues.asterisk.org/jira/browse/ASTERISK-27551>] -

[patch] ooh323cDriver: Fix typo in header guard.
(Reported by Alexander Traud)

   - [ASTERISK-26046
   <https://issues.asterisk.org/jira/browse/ASTERISK-26046>] -

[patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)

   - [ASTERISK-20346
   <https://issues.asterisk.org/jira/browse/ASTERISK-20346>] -

Modules need to ensure that any functions, apps, AMI actions, etc. they
register are unregistered if the module declines loading
(Reported by Mark Michelson)

   - [ASTERISK-27539
   <https://issues.asterisk.org/jira/browse/ASTERISK-27539>] -

'cdr submit' fails: batch mode not enabled.
(Reported by Tzafrir Cohen)

   - [ASTERISK-27498
   <https://issues.asterisk.org/jira/browse/ASTERISK-27498>] -

ICE candidate parser - ICE foundation parsing too short
(Reported by Michele Prà)

   - [ASTERISK-25128
   <https://issues.asterisk.org/jira/browse/ASTERISK-25128>] -

Datastore: Implement automatic module references.
(Reported by Corey Farrell)

   - [ASTERISK-27366
   <https://issues.asterisk.org/jira/browse/ASTERISK-27366>] -

Asterisk Turkish Language Set Problem
(Reported by Halil İbrahim YILDIZ)

   - [ASTERISK-23133
   <https://issues.asterisk.org/jira/browse/ASTERISK-23133>] -

Documentation fix - MASTER_CHANNEL Unexpected Behaviour
(Reported by Shane Mitchell)

   - [ASTERISK-27531
   <https://issues.asterisk.org/jira/browse/ASTERISK-27531>] -

Compiler optimizations can break module load sequence.
(Reported by abelbeck)

   - [ASTERISK-27480
   <https://issues.asterisk.org/jira/browse/ASTERISK-27480>] -

Security: Authenticated SUBSCRIBE without Contact crashes asterisk
(Reported by Ross Beer)

   - [ASTERISK-24198
   <https://issues.asterisk.org/jira/browse/ASTERISK-24198>] -

(Reported by Walter Doekes)

   - [ASTERISK-27229
   <https://issues.asterisk.org/jira/browse/ASTERISK-27229>] -

bridge: Old channel video source not set to NULL after unref
(Reported by Richard Kenner)

   - [ASTERISK-27495
   <https://issues.asterisk.org/jira/browse/ASTERISK-27495>] -

DNS: Unexpected rr_type can cause crash
(Reported by Corey Farrell)

   - [ASTERISK-25079
   <https://issues.asterisk.org/jira/browse/ASTERISK-25079>] -

AMI bridge of channels results in MOH not destroyed and robotic audio on
one channel
(Reported by Zane Conkle)

   - [ASTERISK-27490
   <https://issues.asterisk.org/jira/browse/ASTERISK-27490>] -

chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)

   - [ASTERISK-27299
   <https://issues.asterisk.org/jira/browse/ASTERISK-27299>] -

Asterisk Hangs with Bad file descriptor on read()
(Reported by Abhay Gupta)

   - [ASTERISK-24756
   <https://issues.asterisk.org/jira/browse/ASTERISK-24756>] -

ConfBridge sound_muted does not work from CLI or AMI
(Reported by Thomas Frederiksen)

   - [ASTERISK-25649
   <https://issues.asterisk.org/jira/browse/ASTERISK-25649>] -

Transfer application does not work with Local channels - documentation
(Reported by Ivan Ullmann)

   - [ASTERISK-25869
   <https://issues.asterisk.org/jira/browse/ASTERISK-25869>] -

chan_sip: "rejected because extension not found" should be logged as a
security event
(Reported by Brian J. Murrell)

   - [ASTERISK-27440
   <https://issues.asterisk.org/jira/browse/ASTERISK-27440>] -

Strictrtp has issues to qualify video rtp streams
(Reported by Wim De Vlaminck)

   - [ASTERISK-19657
   <https://issues.asterisk.org/jira/browse/ASTERISK-19657>] -

Coverity Report: Fix issues for error type CHAR_IO
(Reported by Matt Jordan)

   - [ASTERISK-27175
   <https://issues.asterisk.org/jira/browse/ASTERISK-27175>] -

iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)

   - [ASTERISK-27430
   <https://issues.asterisk.org/jira/browse/ASTERISK-27430>] -

README refers to security documents that do not exist.
(Reported by Corey Farrell)

   - [ASTERISK-20281
   <https://issues.asterisk.org/jira/browse/ASTERISK-20281>] -

"core set verbose" behaves strangely, can't alias it, cli.conf example
(Reported by Tim Ringenbach at Asteria Solutions Group)

   - [ASTERISK-27382
   <https://issues.asterisk.org/jira/browse/ASTERISK-27382>] -

crash after an invalid rtcp packet from GT48 FXS gateway
(Reported by Tzafrir Cohen)

   - [ASTERISK-27429
   <https://issues.asterisk.org/jira/browse/ASTERISK-27429>] -

res_rtp_asterisk: Multiple reports in an RTCP packet will write past where
it should
(Reported by Vitezslav Novy)

   - [ASTERISK-27408
   <https://issues.asterisk.org/jira/browse/ASTERISK-27408>] -

Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)

   - [ASTERISK-18411
   <https://issues.asterisk.org/jira/browse/ASTERISK-18411>] -

Queue members with hints for state_interface get stuck in "In Use" state.
(Reported by Steven Wheeler)

   - [ASTERISK-26131
   <https://issues.asterisk.org/jira/browse/ASTERISK-26131>] -

chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a
call to a single character in a dot pattern match
(Reported by Dwayne Hubbard)

   - [ASTERISK-27467
   <https://issues.asterisk.org/jira/browse/ASTERISK-27467>] -

pjsip_options: qualify_frequency sometimes not applied on reload
(Reported by John Bigelow)

   - [ASTERISK-27460
   <https://issues.asterisk.org/jira/browse/ASTERISK-27460>] -

CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=...
(Reported by Richard Mudgett)

   - [ASTERISK-27453
   <https://issues.asterisk.org/jira/browse/ASTERISK-27453>] -

RTP: Blind transfer direct media scenario results in one way audio.
(Reported by Richard Mudgett)

   - [ASTERISK-20643
   <https://issues.asterisk.org/jira/browse/ASTERISK-20643>] -

SIP ICE support - remove hardcoded limitation on SDP size, make ICE support
disabled by default in SIP, maybe provide a better warning message
(Reported by Roy)

   - [ASTERISK-27457
   <https://issues.asterisk.org/jira/browse/ASTERISK-27457>] -

chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP.
(Reported by Alexander Traud)

   - [ASTERISK-26980
   <https://issues.asterisk.org/jira/browse/ASTERISK-26980>] -

pjsip: Clean up WebRTC disables
(Reported by abelbeck)

   - [ASTERISK-27452
   <https://issues.asterisk.org/jira/browse/ASTERISK-27452>] -

Security: chan_skinny: Memory exhaustion if flooded with unauthenticated
(Reported by George Joseph)

   - [ASTERISK-27454
   <https://issues.asterisk.org/jira/browse/ASTERISK-27454>] -

res_http_post: Don't require GMIME_MAJOR_VERSION
(Reported by Joshua C. Colp)

   - [ASTERISK-23735
   <https://issues.asterisk.org/jira/browse/ASTERISK-23735>] -

Transcoding makes bad choice in high-rate translations
(Reported by Richard Kenner)

   - [ASTERISK-27445
   <https://issues.asterisk.org/jira/browse/ASTERISK-27445>] -

ARI: Updating a bridge gives wrong error message.
(Reported by Frank Durden)

   - [ASTERISK-24662
   <https://issues.asterisk.org/jira/browse/ASTERISK-24662>] -

[patch] column and row headers for Signed Linear format variants in output
of 'core show translation' are ambiguous
(Reported by Rusty Newton)

   - [ASTERISK-27353
   <https://issues.asterisk.org/jira/browse/ASTERISK-27353>] -

H323 audio starts with a delay of 2 seconds.
(Reported by Marco Giordani)

   - [ASTERISK-27442
   <https://issues.asterisk.org/jira/browse/ASTERISK-27442>] -

pjsip: 183 without To tag does not negotiate media
(Reported by Kevin Harwell)

   - [ASTERISK-27437
   <https://issues.asterisk.org/jira/browse/ASTERISK-27437>] -

[patch] ICE: server-reflexive candidates (srflx) with Dual-Stack.
(Reported by Alexander Traud)

   - [ASTERISK-27434
   <https://issues.asterisk.org/jira/browse/ASTERISK-27434>] -

[patch] chan_sip/ICE: Square brackets around IPv6 addresses.
(Reported by Alexander Traud)

   - [ASTERISK-27332
   <https://issues.asterisk.org/jira/browse/ASTERISK-27332>] -

Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)

   - [ASTERISK-27431
   <https://issues.asterisk.org/jira/browse/ASTERISK-27431>] -

Asterisk fails to build when openssl headers are not installed.
(Reported by Corey Farrell)

   - [ASTERISK-27421
   <https://issues.asterisk.org/jira/browse/ASTERISK-27421>] -

RTP source learning not working with devices that have some clock issues
(Reported by nappsoft)

   - [ASTERISK-27361
   <https://issues.asterisk.org/jira/browse/ASTERISK-27361>] -

Attended transfer crashes in Asterisk 13.17.2
(Reported by Alessandro Pimenta)

   - [ASTERISK-27238
   <https://issues.asterisk.org/jira/browse/ASTERISK-27238>] -

Bridging: Crash freeing a frame that's already been freed
(Reported by Richard Kenner)

   - [ASTERISK-27412
   <https://issues.asterisk.org/jira/browse/ASTERISK-27412>] -

core: Audiohook freeing interpolated frame when it shouldn't.
(Reported by Mikhail)

   - [ASTERISK-27423
   <https://issues.asterisk.org/jira/browse/ASTERISK-27423>] -

app_record: We set the RECORD_STATUS channel variable before closing the
(Reported by George Joseph)

   - [ASTERISK-26758
   <https://issues.asterisk.org/jira/browse/ASTERISK-26758>] -

res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in
"source ip address" and "destination ip address" fields in HEP packets
(Reported by Max Norba)

   - [ASTERISK-27363
   <https://issues.asterisk.org/jira/browse/ASTERISK-27363>] -

res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress)
(Reported by Vasilii Rogin)

   - [ASTERISK-27415
   <https://issues.asterisk.org/jira/browse/ASTERISK-27415>] -

asterisk.conf: Setting astctl without setting astrundir is ineffective.
(Reported by Corey Farrell)

   - [ASTERISK-27411
   <https://issues.asterisk.org/jira/browse/ASTERISK-27411>] -

pjsip: TCP connections may not be destroyed
(Reported by Joshua C. Colp)

   - [ASTERISK-27404
   <https://issues.asterisk.org/jira/browse/ASTERISK-27404>] -

DEBUG_FD_LEAKS does not record socketpair, timerfd_create or eventfd.
(Reported by Corey Farrell)

   - [ASTERISK-27345
   <https://issues.asterisk.org/jira/browse/ASTERISK-27345>] -

res_pjsip_session: RTP instances leak on 488 responses.
(Reported by Corey Farrell)

   - [ASTERISK-27337
   <https://issues.asterisk.org/jira/browse/ASTERISK-27337>] -

chan_sip: Security vulnerability with client code header (revisited)
(Reported by Richard Mudgett)

   - [ASTERISK-27319
   <https://issues.asterisk.org/jira/browse/ASTERISK-27319>] -

(Security) Function in PJSIP 2.7 miscalculates the length of an unsigned
long variable in 64bit machines
(Reported by Kim youngsung)

   - [ASTERISK-27391
   <https://issues.asterisk.org/jira/browse/ASTERISK-27391>] -

Regression: Deadlock between AOR named lock and pjproject grp lock
(Reported by shaurya jain)

   - [ASTERISK-27393
   <https://issues.asterisk.org/jira/browse/ASTERISK-27393>] -

res_pjsip: Crash occurs when an empty contact read from astdb or database
(Reported by Aaron An)

   - [ASTERISK-27290
   <https://issues.asterisk.org/jira/browse/ASTERISK-27290>] -

res_pjsip: PIDF contact field has malformed/invalid XML
(Reported by basildane)

   - [ASTERISK-27032
   <https://issues.asterisk.org/jira/browse/ASTERISK-27032>] -

res_pjsip: TLS options do not handle empty values
(Reported by seanchann.zhou)

   - [ASTERISK-27395
   <https://issues.asterisk.org/jira/browse/ASTERISK-27395>] -

srtp: Add support for ephemeral DTLS certificates
(Reported by Sean Bright)

   - [ASTERISK-26426
   <https://issues.asterisk.org/jira/browse/ASTERISK-26426>] -

format_ogg_opus: remove from source
(Reported by Kevin Harwell)

   - [ASTERISK-27394
   <https://issues.asterisk.org/jira/browse/ASTERISK-27394>] -

[patch] tcptls: Print notice when TLS is enabled but not configured.
(Reported by Alexander Traud)

   - [ASTERISK-27356
   <https://issues.asterisk.org/jira/browse/ASTERISK-27356>] -

[patch] libsrtp-2.x.x + AES-GCM support
(Reported by Alexander Traud)

   - [ASTERISK-27378
   <https://issues.asterisk.org/jira/browse/ASTERISK-27378>] -

Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)

   - [ASTERISK-27387
   <https://issues.asterisk.org/jira/browse/ASTERISK-27387>] -

Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more
(Reported by Michael Maier)

   - [ASTERISK-27364
   <https://issues.asterisk.org/jira/browse/ASTERISK-27364>] -

channel: Crash when fax gateway is in use with PJSIP
(Reported by Jared Hull)

   - [ASTERISK-27390
   <https://issues.asterisk.org/jira/browse/ASTERISK-27390>] -

Audit menuselect module dependencies
(Reported by Corey Farrell)

   - [ASTERISK-27389
   <https://issues.asterisk.org/jira/browse/ASTERISK-27389>] -

Optional API modules should not allow unload.
(Reported by Corey Farrell)

   - [ASTERISK-27369
   <https://issues.asterisk.org/jira/browse/ASTERISK-27369>] -

Bridge() dialplan application fails without setting BRIDGERESULT channel
(Reported by James Terhune)

   - [ASTERISK-27067
   <https://issues.asterisk.org/jira/browse/ASTERISK-27067>] -

res_ari_channels: channel_state_invalid always leaks snapshot reference.
(Reported by Marin Odrljin)

   - [ASTERISK-27379
   <https://issues.asterisk.org/jira/browse/ASTERISK-27379>] -

stream: Allow streams on a topology to be put into groups
(Reported by Joshua C. Colp)

   - [ASTERISK-27374
   <https://issues.asterisk.org/jira/browse/ASTERISK-27374>] -

alembic: PJSIP scripts are missing column bundle in ps_endpoints table
(Reported by Florian Floimair)

   - [ASTERISK-27377
   <https://issues.asterisk.org/jira/browse/ASTERISK-27377>] -

Typo in CHANNEL(dtmf_features) usage documentation
(Reported by Igor Goncharovsky)

   - [ASTERISK-27181
   <https://issues.asterisk.org/jira/browse/ASTERISK-27181>] -

GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting'
(Reported by Anthony Messina)

   - [ASTERISK-27194
   <https://issues.asterisk.org/jira/browse/ASTERISK-27194>] -

jitterbuffer: Does not handle case where translator returns null frame.
(Reported by Joshua Elson)

   - [ASTERISK-27372
   <https://issues.asterisk.org/jira/browse/ASTERISK-27372>] -

ARI: Node ARI client broken in latest versions of 13 and 14
(Reported by Benjamin Keith Ford)

   - [ASTERISK-26639
   <https://issues.asterisk.org/jira/browse/ASTERISK-26639>] -

core: Disabling xmldoc support does not work. Also results in abort during
Asterisk startup.
(Reported by Mr Dini)

   - [ASTERISK-18140
   <https://issues.asterisk.org/jira/browse/ASTERISK-18140>] -

Expires handling in SUBSCRIBE confuses the absence of the Expires header
field with an unsubscribe action.
(Reported by Jonathan Cloots)

   - [ASTERISK-25960
   <https://issues.asterisk.org/jira/browse/ASTERISK-25960>] -

The config_hook unit test causes Asterisk to crash if run a second time
(Reported by George Joseph)

   - [ASTERISK-27198
   <https://issues.asterisk.org/jira/browse/ASTERISK-27198>] -

res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes
(Reported by Martin Cisárik)

   - [ASTERISK-27346
   <https://issues.asterisk.org/jira/browse/ASTERISK-27346>] -

res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
(Reported by Ronald Raikes)

   - [ASTERISK-27365
   <https://issues.asterisk.org/jira/browse/ASTERISK-27365>] -

[patch] chan_sip: Crypto attribute not last but first on SDP media level.
(Reported by Alexander Traud)

   - [ASTERISK-24483
   <https://issues.asterisk.org/jira/browse/ASTERISK-24483>] -

res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id ==
(Reported by Tzafrir Cohen)

   - [ASTERISK-23462
   <https://issues.asterisk.org/jira/browse/ASTERISK-23462>] -

Cannot disable SIP debugging via CLI after enabling with conf file option -
also 'sip set debug off' reports debugging disabled, when it really isn't
(Reported by Rusty Newton)

   - [ASTERISK-27350
   <https://issues.asterisk.org/jira/browse/ASTERISK-27350>] -

app_macro deprecation
(Reported by Corey Farrell)

   - [ASTERISK-27354
   <https://issues.asterisk.org/jira/browse/ASTERISK-27354>] -

bridge_softmix: When a channel leaves add in any missing participant streams
(Reported by Joshua C. Colp)

   - [ASTERISK-27333
   <https://issues.asterisk.org/jira/browse/ASTERISK-27333>] -

sip_to_pjsip not correctly handling disallow=all directive
(Reported by Torrey Searle)

   - [ASTERISK-27343
   <https://issues.asterisk.org/jira/browse/ASTERISK-27343>] -

Fails to build in FreeBSD due to sys/sysmacros.h not existing there
(Reported by Guido Falsi)

   - [ASTERISK-27341
   <https://issues.asterisk.org/jira/browse/ASTERISK-27341>] -

[patch] res_pjsip_session: SIP/SDP origin (o=) contains local address.
(Reported by Alexander Traud)

   - [ASTERISK-27259
   <https://issues.asterisk.org/jira/browse/ASTERISK-27259>] -

chan_pjsip: Outgoing leg does not use all configured codecs, but subset
based on caller
(Reported by lvl)

   - [ASTERISK-27340
   <https://issues.asterisk.org/jira/browse/ASTERISK-27340>] -

backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)

   - [ASTERISK-27339
   <https://issues.asterisk.org/jira/browse/ASTERISK-27339>] -

[patch] Crash on ast_ssl_teardown when stopping.
(Reported by Alexander Traud)

   - [ASTERISK-27047
   <https://issues.asterisk.org/jira/browse/ASTERISK-27047>] -

res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
(Reported by dtryba)

   - [ASTERISK-26988
   <https://issues.asterisk.org/jira/browse/ASTERISK-26988>] -

res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
(Reported by dtryba)

   - [ASTERISK-27301
   <https://issues.asterisk.org/jira/browse/ASTERISK-27301>] -

[patch] app_queue: Music On Hold for real-time queues is not reset to
(Reported by Nathan Bruning)

   - [ASTERISK-25266
   <https://issues.asterisk.org/jira/browse/ASTERISK-25266>] -

Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to
(Reported by Allen Ford)

   - [ASTERISK-27270
   <https://issues.asterisk.org/jira/browse/ASTERISK-27270>] -

cdr_mysql: various crashes at second module reload if cdr_mysql.conf is
(Reported by Tzafrir Cohen)

   - [ASTERISK-27328
   <https://issues.asterisk.org/jira/browse/ASTERISK-27328>] -

Missing openssl dependencies in res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)

   - [ASTERISK-27192
   <https://issues.asterisk.org/jira/browse/ASTERISK-27192>] -

res_pjsip: Loss of SIP registrations causing unavailable endpoints
(Reported by Richard Mudgett)

   - [ASTERISK-27305
   <https://issues.asterisk.org/jira/browse/ASTERISK-27305>] -

res_ari: Memory leaks in ARI when using Content-Type: application/json
(Reported by David Hajek)

   - [ASTERISK-26922
   <https://issues.asterisk.org/jira/browse/ASTERISK-26922>] -

chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)

   - [ASTERISK-27324
   <https://issues.asterisk.org/jira/browse/ASTERISK-27324>] -

[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
(Reported by Alexander Traud)

   - [ASTERISK-27317
   <https://issues.asterisk.org/jira/browse/ASTERISK-27317>] -

vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)

   - [ASTERISK-27318
   <https://issues.asterisk.org/jira/browse/ASTERISK-27318>] -

res_pjsip_mwi: uninitialized value from ast_strings_match
(Reported by Corey Farrell)

   - [ASTERISK-27284
   <https://issues.asterisk.org/jira/browse/ASTERISK-27284>] -

Status of RFC 3323 and PJSIP
(Reported by dtryba)

   - [ASTERISK-27296
   <https://issues.asterisk.org/jira/browse/ASTERISK-27296>] -

[patch] False positive busy checks when icalendar's recurrence-id mechanism
is involved
(Reported by Benoît Dereck-Tricot)

   - [ASTERISK-27216
   <https://issues.asterisk.org/jira/browse/ASTERISK-27216>] -

app_queue: does its check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engström)

   - [ASTERISK-27298
   <https://issues.asterisk.org/jira/browse/ASTERISK-27298>] -

Problem with expires on pjsip / outbound-publish
(Reported by Cyrille Demaret)

   - [ASTERISK-27295
   <https://issues.asterisk.org/jira/browse/ASTERISK-27295>] -

Contact is improperly translated after d178f497
(Reported by Sean Bright)

   - [ASTERISK-27292
   <https://issues.asterisk.org/jira/browse/ASTERISK-27292>] -

Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
(Reported by Ross Beer)

   - [ASTERISK-27289
   <https://issues.asterisk.org/jira/browse/ASTERISK-27289>] -

A codeblock that maintains a bug,but maybe the codeblock will never run
(Reported by Huangyx)

   - [ASTERISK-27277
   <https://issues.asterisk.org/jira/browse/ASTERISK-27277>] -

bridge: Renegotiate if source stream changes.
(Reported by Joshua C. Colp)

   - [ASTERISK-27264
   <https://issues.asterisk.org/jira/browse/ASTERISK-27264>] -

res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
(Reported by Daniel Heckl)

   - [ASTERISK-27283
   <https://issues.asterisk.org/jira/browse/ASTERISK-27283>] -

Realtime config fail with PostgreSQL version before 9.1
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-27260
   <https://issues.asterisk.org/jira/browse/ASTERISK-27260>] -

[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
(Reported by Daniel Heckl)

   - [ASTERISK-27257
   <https://issues.asterisk.org/jira/browse/ASTERISK-27257>] -

bridge_native_rtp: half-way direct media when using early bridging
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-16898
   <https://issues.asterisk.org/jira/browse/ASTERISK-16898>] -

SRTP unprotect: authentication failure when RTP sequence number switches
from 65535 -> 0
(Reported by Marcello Ceschia)

   - [ASTERISK-27279
   <https://issues.asterisk.org/jira/browse/ASTERISK-27279>] -

Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
(Reported by Ross Beer)

   - [ASTERISK-25524
   <https://issues.asterisk.org/jira/browse/ASTERISK-25524>] -

module reload res_calendar.so does not reload everything in calendar.conf
(Reported by Jesper)

   - [ASTERISK-27274
   <https://issues.asterisk.org/jira/browse/ASTERISK-27274>] -

RTCP needs better packet validation to resist port scans.
(Reported by Richard Mudgett)

   - [ASTERISK-27252
   <https://issues.asterisk.org/jira/browse/ASTERISK-27252>] -

RTP: One way audio with direct media and strictrtp=yes.
(Reported by Richard Mudgett)

   - [ASTERISK-24588
   <https://issues.asterisk.org/jira/browse/ASTERISK-24588>] -

res_calendar does not process CalDAV from Owncloud [fix included]
(Reported by Stefan Gofferje)

   - [ASTERISK-25523
   <https://issues.asterisk.org/jira/browse/ASTERISK-25523>] -

res_calendar: Warning about invalid channel value (for notification) occurs
even when event has no notification configured.
(Reported by Jesper)

   - [ASTERISK-21399
   <https://issues.asterisk.org/jira/browse/ASTERISK-21399>] -

RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
(Reported by Tzafrir Cohen)

   - [ASTERISK-27248
   <https://issues.asterisk.org/jira/browse/ASTERISK-27248>] -

[patch]external_media_address and external_signaling_address don't always
honor localnet
(Reported by Walter Doekes)

   - [ASTERISK-27165
   <https://issues.asterisk.org/jira/browse/ASTERISK-27165>] -

CDR: CDR(start,u) function won't work in cdr_custom config
(Reported by Jacek Konieczny)

   - [ASTERISK-24066
   <https://issues.asterisk.org/jira/browse/ASTERISK-24066>] -

res_smdi: convert to astobj2
(Reported by Corey Farrell)

   - [ASTERISK-27217
   <https://issues.asterisk.org/jira/browse/ASTERISK-27217>] -

chan_sip: Asterisk crashing when subscription doesn't get set
(Reported by Bryan Walters)

   - [ASTERISK-17540
   <https://issues.asterisk.org/jira/browse/ASTERISK-17540>] -

SDP origin attribute modified when issuing re-INVITE because of
(Reported by saghul)

   - [ASTERISK-27254
   <https://issues.asterisk.org/jira/browse/ASTERISK-27254>] -

alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)

   - [ASTERISK-27232
   <https://issues.asterisk.org/jira/browse/ASTERISK-27232>] -

When in queue on g722 with interruptions, music on hold can get stuck and
no longer play
(Reported by Jens T.)

   - [ASTERISK-27024
   <https://issues.asterisk.org/jira/browse/ASTERISK-27024>] -

nat/external_media settings ignored in 14.4.1
(Reported by Christopher van de Sande)

   - [ASTERISK-26879
   <https://issues.asterisk.org/jira/browse/ASTERISK-26879>] -

PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)

   - [ASTERISK-27236
   <https://issues.asterisk.org/jira/browse/ASTERISK-27236>] -

Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during
T.38 Fax Receive
(Reported by Ross Beer)

   - [ASTERISK-27225
   <https://issues.asterisk.org/jira/browse/ASTERISK-27225>] -

Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)

   - [ASTERISK-27177
   <https://issues.asterisk.org/jira/browse/ASTERISK-27177>] -

ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)

   - [ASTERISK-27241
   <https://issues.asterisk.org/jira/browse/ASTERISK-27241>] -

libc segfault upon entry into app_directory
(Reported by David Moore)

   - [ASTERISK-27152
   <https://issues.asterisk.org/jira/browse/ASTERISK-27152>] -

Sending a "tel" uri in a From or To header in an unauthenticated message
causes asterisk to crash
(Reported by Ross Beer)

   - [ASTERISK-27103
   <https://issues.asterisk.org/jira/browse/ASTERISK-27103>] -

core: ast_safe_system command injection possible.
(Reported by Corey Farrell)

   - [ASTERISK-27013
   <https://issues.asterisk.org/jira/browse/ASTERISK-27013>] -

res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
(Reported by Joshua C. Colp)

   - [ASTERISK-27231
   <https://issues.asterisk.org/jira/browse/ASTERISK-27231>] -

res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
(Reported by Joshua C. Colp)

   - [ASTERISK-26994
   <https://issues.asterisk.org/jira/browse/ASTERISK-26994>] -

Confbridge: CBAnn channels intermittently become stuck when caller hangs up
before recording name
(Reported by James Terhune)

   - [ASTERISK-27222
   <https://issues.asterisk.org/jira/browse/ASTERISK-27222>] -

core: Don't queue up multiple video update frames.
(Reported by Joshua C. Colp)

   - [ASTERISK-20858
   <https://issues.asterisk.org/jira/browse/ASTERISK-20858>] -

app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)

   - [ASTERISK-16777
   <https://issues.asterisk.org/jira/browse/ASTERISK-16777>] -

several filename bugs in Record() application
(Reported by klaus3000)

   - [ASTERISK-27168
   <https://issues.asterisk.org/jira/browse/ASTERISK-27168>] -

alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints
(Reported by Florian Floimair)

   - [ASTERISK-27209
   <https://issues.asterisk.org/jira/browse/ASTERISK-27209>] -

Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
(Reported by Torrey Searle)

   - [ASTERISK-19103
   <https://issues.asterisk.org/jira/browse/ASTERISK-19103>] -

When using realtime queues, function QUEUE_MEMBER_LIST() will return an
error if no other app/function has loaded the queues first. This problem
does not exist if queues.conf is used.
(Reported by Jim Van Meggelen)

   - [ASTERISK-21241
   <https://issues.asterisk.org/jira/browse/ASTERISK-21241>] -

When using voicemail as announce only (maxmsg=0), the star dtmf to enter
the voicemail is not honored
(Reported by Eelco Brolman)

   - [ASTERISK-27212
   <https://issues.asterisk.org/jira/browse/ASTERISK-27212>] -

bridge_softmix: Quickly joining/leaving may cause video stream to remain in
(Reported by Richard Mudgett)

   - [ASTERISK-27204
   <https://issues.asterisk.org/jira/browse/ASTERISK-27204>] -

[patch] app_queue: Wrong queue stat calculation
(Reported by sungtae kim)

   - [ASTERISK-27207
   <https://issues.asterisk.org/jira/browse/ASTERISK-27207>] -

XMPP OAuth not working due to inverted logic
(Reported by Michael Kuron)

   - [ASTERISK-27174
   <https://issues.asterisk.org/jira/browse/ASTERISK-27174>] -

res_calendar_icalendar: Recurring events not being loaded from Google
calendar using ical
(Reported by Mark Thompson)

   - [ASTERISK-27202
   <https://issues.asterisk.org/jira/browse/ASTERISK-27202>] -

If wget is not installed and "or" is not available, external components
(excluding pjsip) are not installed
(Reported by Seán C. McCord)

   - [ASTERISK-27200
   <https://issues.asterisk.org/jira/browse/ASTERISK-27200>] -

manager: hook event is not being raised
(Reported by Kevin Harwell)

   - [ASTERISK-27147
   <https://issues.asterisk.org/jira/browse/ASTERISK-27147>] -

Either asterisk or pjproject isn't re-using tcp connections (again)
(Reported by George Joseph)

   - [ASTERISK-27193
   <https://issues.asterisk.org/jira/browse/ASTERISK-27193>] -

IPv6 receive address in message doesn't include brackets
(Reported by Scott Griepentrog)

   - [ASTERISK-27158
   <https://issues.asterisk.org/jira/browse/ASTERISK-27158>] -

[patch] res_rtp_asterisk: RTCP statistics are not available when native
bridge is used
(Reported by Torrey Searle)

   - [ASTERISK-26745
   <https://issues.asterisk.org/jira/browse/ASTERISK-26745>] -

Asymmetric codecs when asymmetric_rtp_codec=no
(Reported by Jesse Ross)

   - [ASTERISK-27189
   <https://issues.asterisk.org/jira/browse/ASTERISK-27189>] -

Make --with-pjproject-bundled the default for Asterisk 15
(Reported by George Joseph)

   - [ASTERISK-27110
   <https://issues.asterisk.org/jira/browse/ASTERISK-27110>] -

RTP session is not fully destroyed on channel hangup
(Reported by Matt Jordan)

   - [ASTERISK-27182
   <https://issues.asterisk.org/jira/browse/ASTERISK-27182>] -

bridge: Crash when mapping streams
(Reported by Joshua C. Colp)

   - [ASTERISK-27180
   <https://issues.asterisk.org/jira/browse/ASTERISK-27180>] -

channel: requester leaks joint_cap on success.
(Reported by Corey Farrell)

   - [ASTERISK-27179
   <https://issues.asterisk.org/jira/browse/ASTERISK-27179>] -

res_pjsip_session: Handling of 'msid' is incorrect
(Reported by Kevin Harwell)

   - [ASTERISK-27119
   <https://issues.asterisk.org/jira/browse/ASTERISK-27119>] -

res_pjsip: parse/add msid attribute when webrtc is enabled
(Reported by Kevin Harwell)

   - [ASTERISK-27171
   <https://issues.asterisk.org/jira/browse/ASTERISK-27171>] -

Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)

   - [ASTERISK-26659
   <https://issues.asterisk.org/jira/browse/ASTERISK-26659>] -

res_pjsip: PJSIP presence - missing braces around the status element in XML
(Reported by Abraham Liebsch)

   - [ASTERISK-27156
   <https://issues.asterisk.org/jira/browse/ASTERISK-27156>] -

Asterisk won't compile on Fedora 26 with devmode enabled.
(Reported by Corey Farrell)

   - [ASTERISK-27001
   <https://issues.asterisk.org/jira/browse/ASTERISK-27001>] -

res_pjsip: TLS connection not stable
(Reported by Ian Gilmour)

   - [ASTERISK-27130
   <https://issues.asterisk.org/jira/browse/ASTERISK-27130>] -

Applications ARI: Unsubscribe action for deviceStates does not remove old
subscriptions properly
(Reported by Sergej Kasumovic)

   - [ASTERISK-25810
   <https://issues.asterisk.org/jira/browse/ASTERISK-25810>] -

say.c calls for sounds in the subdir "digits" that don't exist (in Core).
SayUnixTime or other Say... apps will fail out when they call these sounds.
(Reported by Nicolas Riendeau)

   - [ASTERISK-27142
   <https://issues.asterisk.org/jira/browse/ASTERISK-27142>] -

sounds: Conflict between files in asterisk-sounds-core-1.6 and
(Reported by Corey Farrell)

   - [ASTERISK-27143
   <https://issues.asterisk.org/jira/browse/ASTERISK-27143>] -

bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
(Reported by Joshua C. Colp)

   - [ASTERISK-27136
   <https://issues.asterisk.org/jira/browse/ASTERISK-27136>] -

bridge_softmix: Don't reorder SFU streams
(Reported by Joshua C. Colp)

   - [ASTERISK-27134
   <https://issues.asterisk.org/jira/browse/ASTERISK-27134>] -

bridge_softmix: Reuse any removed streams for video
(Reported by Joshua C. Colp)

   - [ASTERISK-27133
   <https://issues.asterisk.org/jira/browse/ASTERISK-27133>] -

res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
(Reported by Joshua C. Colp)

   - [ASTERISK-27123
   <https://issues.asterisk.org/jira/browse/ASTERISK-27123>] -

confbridge: Name recordings are left on filesystem
(Reported by Sergej Kasumovic)

   - [ASTERISK-27122
   <https://issues.asterisk.org/jira/browse/ASTERISK-27122>] -

chan_iax2: On reload MWI taskprocessors keep adding up
(Reported by Sergej Kasumovic)

   - [ASTERISK-26807
   <https://issues.asterisk.org/jira/browse/ASTERISK-26807>] -

sounds: New 3-D Binaural audio features require new sound prompts
(Reported by Rusty Newton)

   - [ASTERISK-25816
   <https://issues.asterisk.org/jira/browse/ASTERISK-25816>] -

French conf-adminmenu, conf-usermenu prompts differ in content from the
English files
(Reported by Benoit Duverger)

   - [ASTERISK-26274
   <https://issues.asterisk.org/jira/browse/ASTERISK-26274>] -

Resolve open sounds issues and then create a new sounds release (1.5.1? or
(Reported by Rusty Newton)

   - [ASTERISK-27118
   <https://issues.asterisk.org/jira/browse/ASTERISK-27118>] -

res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
(Reported by Joshua C. Colp)

   - [ASTERISK-27128
   <https://issues.asterisk.org/jira/browse/ASTERISK-27128>] -

[patch]res_stasis_snoop: When recording a snoop channel (using ARI) where
no media is being received, no recording happens when theres no media
(Reported by Dan Jenkins)

   - [ASTERISK-27124
   <https://issues.asterisk.org/jira/browse/ASTERISK-27124>] -

app_playback.c:say_date_generic use timezonename parameter
(Reported by Holger Hans Peter Freyther)

   - [ASTERISK-27127
   <https://issues.asterisk.org/jira/browse/ASTERISK-27127>] -

configs: Erroneous load directive in sample configuration results in "Error
loading module 'res_pjsip_multihomed.so'"
(Reported by HZMI8gkCvPpom0tM)

   - [ASTERISK-27073
   <https://issues.asterisk.org/jira/browse/ASTERISK-27073>] -

manager: AMI "queues" action outputs freeform text that doesn't follow the
AMI spec
(Reported by Brian)

   - [ASTERISK-27105
   <https://issues.asterisk.org/jira/browse/ASTERISK-27105>] -

[patch]core: when setting 'maxfiles' in asterisk.conf, a message is
printed, even in rasterisk -x
(Reported by Tzafrir Cohen)

   - [ASTERISK-27036
   <https://issues.asterisk.org/jira/browse/ASTERISK-27036>] -

res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with
from_user containing '@'
(Reported by Maxim Vasilev)

   - [ASTERISK-27023
   <https://issues.asterisk.org/jira/browse/ASTERISK-27023>] -

res_rtp_asterisk: Deadlock when TURN session in use
(Reported by Jatin Jain)

   - [ASTERISK-27106
   <https://issues.asterisk.org/jira/browse/ASTERISK-27106>] -

[patch] autodomain (SIP Domain Support): Add only really different domain
with TLS.
(Reported by Alexander Traud)

   - [ASTERISK-27093
   <https://issues.asterisk.org/jira/browse/ASTERISK-27093>] -

ODBC deadlocks when app_directory tries to play back non-existent voicemail
(Reported by James Terhune)

   - [ASTERISK-27100
   <https://issues.asterisk.org/jira/browse/ASTERISK-27100>] -

channel: ast_waitfordigit_full fails to clear flag in an error branch.
(Reported by Corey Farrell)

   - [ASTERISK-27090
   <https://issues.asterisk.org/jira/browse/ASTERISK-27090>] -

PJSIP: Deadlock using TCP transport
(Reported by Richard Mudgett)

   - [ASTERISK-26997
   <https://issues.asterisk.org/jira/browse/ASTERISK-26997>] -

Create an StreamEcho dialplan application
(Reported by Kevin Harwell)

   - [ASTERISK-27096
   <https://issues.asterisk.org/jira/browse/ASTERISK-27096>] -

res_rtp_asterisk: add a control frame for when dtls is established
(Reported by Kevin Harwell)

   - [ASTERISK-27097
   <https://issues.asterisk.org/jira/browse/ASTERISK-27097>] -

pjproject_bundled: We don't pass options needed for cross-compile to
pjproject configure
(Reported by George Joseph)

   - [ASTERISK-27076
   <https://issues.asterisk.org/jira/browse/ASTERISK-27076>] -

chan_pjsip: Add support for multiple streams
(Reported by Joshua C. Colp)

   - [ASTERISK-27088
   <https://issues.asterisk.org/jira/browse/ASTERISK-27088>] -

res_rtp_asterisk: Better handle ICE renegotiation and unidirectional
(Reported by Joshua C. Colp)

   - [ASTERISK-26978
   <https://issues.asterisk.org/jira/browse/ASTERISK-26978>] -

rtp: Crash in ast_rtp_codecs_payload_code()
(Reported by Ross Beer)

   - [ASTERISK-25665
   <https://issues.asterisk.org/jira/browse/ASTERISK-25665>] -

Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)

   - [ASTERISK-27065
   <https://issues.asterisk.org/jira/browse/ASTERISK-27065>] -

call hangup after leaving app_queue
(Reported by Marek Cervenka)

   - [ASTERISK-24052
   <https://issues.asterisk.org/jira/browse/ASTERISK-24052>] -

app_voicemail reloads result in leaked IMAP sockets.
(Reported by Louis Jocelyn Paquet)

   - [ASTERISK-27074
   <https://issues.asterisk.org/jira/browse/ASTERISK-27074>] -

core_local: local channel data not being properly unref'ed and unlocked
(Reported by Kevin Harwell)

   - [ASTERISK-27075
   <https://issues.asterisk.org/jira/browse/ASTERISK-27075>] -

bridge: stuck channel(s) after failed attended transfer
(Reported by Kevin Harwell)

   - [ASTERISK-27051
   <https://issues.asterisk.org/jira/browse/ASTERISK-27051>] -

res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the
endpoint's last contact
(Reported by Alexei Gradinari)

   - [ASTERISK-27059
   <https://issues.asterisk.org/jira/browse/ASTERISK-27059>] -

res_stasis: Stolen channel references are leaking
(Reported by George Joseph)

   - [ASTERISK-27060
   <https://issues.asterisk.org/jira/browse/ASTERISK-27060>] -

Comment typo format_g729.c
(Reported by Matthew Fredrickson)

   - [ASTERISK-27041
   <https://issues.asterisk.org/jira/browse/ASTERISK-27041>] -

Core/PBX: [patch] Deadlock between dialplan execution and application
(Reported by Frederic LE FOLL)

   - [ASTERISK-26919
   <https://issues.asterisk.org/jira/browse/ASTERISK-26919>] -

res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference
between chan_sip and res_pjsip
(Reported by Zach R)

   - [ASTERISK-25370
   <https://issues.asterisk.org/jira/browse/ASTERISK-25370>] -

res_corosync segfaults at startup with corosync version > 2.x
(Reported by mdu113)

   - [ASTERISK-27026
   <https://issues.asterisk.org/jira/browse/ASTERISK-27026>] -

res_ari: Crash when no ari.conf configuration file exists
(Reported by Ronald Raikes)

   - [ASTERISK-27016
   <https://issues.asterisk.org/jira/browse/ASTERISK-27016>] -

Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted
multiple times.
(Reported by Chris Howard)

   - [ASTERISK-27057
   <https://issues.asterisk.org/jira/browse/ASTERISK-27057>] -

Seg Fault in ast_sorcery_object_get_id at sorcery.c
(Reported by Ryan Smith)

   - [ASTERISK-27022
   <https://issues.asterisk.org/jira/browse/ASTERISK-27022>] -

res_rtp_asterisk: Incorrect SSRC change for RTCP component
(Reported by Michael Walton)

   - [ASTERISK-26923
   <https://issues.asterisk.org/jira/browse/ASTERISK-26923>] -

bridging: T.38 request is lost when channels are added to bridge
(Reported by Torrey Searle)

   - [ASTERISK-27053
   <https://issues.asterisk.org/jira/browse/ASTERISK-27053>] -

res_pjsip_refer/session: Calls dropped during transfer
(Reported by Kevin Harwell)

   - [ASTERISK-27052
   <https://issues.asterisk.org/jira/browse/ASTERISK-27052>] -

Asterisk build process fails with flag --with-pjproject-bundled with curl
download command and slow network
(Reported by alex)

   - [ASTERISK-27046
   <https://issues.asterisk.org/jira/browse/ASTERISK-27046>] -

res_pjsip_transport_websocket: segfault in get_write_timeout
(Reported by Jørgen H)

   - [ASTERISK-27039
   <https://issues.asterisk.org/jira/browse/ASTERISK-27039>] -

chan_pjsip: Device state is idle when channel from endpoint is in early
(Reported by Joshua C. Colp)

   - [ASTERISK-26996
   <https://issues.asterisk.org/jira/browse/ASTERISK-26996>] -

chan_pjsip: Flipping between codecs
(Reported by Michael Maier)

   - [ASTERISK-26281
   <https://issues.asterisk.org/jira/browse/ASTERISK-26281>] -

chan_pjsip would send INVITE to 'Unreachable' endpoints
(Reported by Jacek Konieczny)

   - [ASTERISK-26973
   <https://issues.asterisk.org/jira/browse/ASTERISK-26973>] -

bridge: Crash when freeing frame and snooping
(Reported by Michel R. Vaillancourt)

   - [ASTERISK-19291
   <https://issues.asterisk.org/jira/browse/ASTERISK-19291>] -

Background in realtime
(Reported by Andrew Nowrot)

   - [ASTERISK-27025
   <https://issues.asterisk.org/jira/browse/ASTERISK-27025>] -

channel / meetme: Fix missing parentheses
(Reported by Joshua C. Colp)

   - [ASTERISK-27021
   <https://issues.asterisk.org/jira/browse/ASTERISK-27021>] -

GET /recordings/stored returns 500 Internal Server Error
(Reported by Tim Morgan)

   - [ASTERISK-24858
   <https://issues.asterisk.org/jira/browse/ASTERISK-24858>] -

[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel
platform when using slin codec
(Reported by Frankie Chin)

   - [ASTERISK-23951
   <https://issues.asterisk.org/jira/browse/ASTERISK-23951>] -

Asterisk attempts and fails to build format_mp3 even if mp3lib was not
(Reported by Tzafrir Cohen)

   - [ASTERISK-25294
   <https://issues.asterisk.org/jira/browse/ASTERISK-25294>] -

srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)

   - [ASTERISK-23839
   <https://issues.asterisk.org/jira/browse/ASTERISK-23839>] -

AGI - RECORD FILE - documentation doesn't describe BEEP argument
(Reported by Rusty Newton)

   - [ASTERISK-22432
   <https://issues.asterisk.org/jira/browse/ASTERISK-22432>] -

Async AGI crashes Asterisk when issuing "set variable" command without args
(Reported by Antoine Pitrou)

   - [ASTERISK-25662
   <https://issues.asterisk.org/jira/browse/ASTERISK-25662>] -

Malformed AGI 520 Usage response
(Reported by Tony Mountifield)

   - [ASTERISK-27008
   <https://issues.asterisk.org/jira/browse/ASTERISK-27008>] -

res_format_attr_h264: SDP parse fails if fmtp optional parameters have a
(Reported by John Harris)

   - [ASTERISK-26399
   <https://issues.asterisk.org/jira/browse/ASTERISK-26399>] -

app_queue: Agent not called when caller is parked
(Reported by wushumasters)

   - [ASTERISK-26400
   <https://issues.asterisk.org/jira/browse/ASTERISK-26400>] -

app_queue: Queue member stops being called after AMI "Redirect" action for
queues with wrapuptime
(Reported by Etienne Lessard)

   - [ASTERISK-26715
   <https://issues.asterisk.org/jira/browse/ASTERISK-26715>] -

app_queue: Member will not receive any new calls after doing a transfer if
wrapuptime = greater than 0 and using Local channel
(Reported by David Brillert)

   - [ASTERISK-26975
   <https://issues.asterisk.org/jira/browse/ASTERISK-26975>] -

app_queue: Non-zero wrapup time can cause agents not to receive queue calls
after transfer queue call
(Reported by Lorne Gaetz)

   - [ASTERISK-27012
   <https://issues.asterisk.org/jira/browse/ASTERISK-27012>] -

app_confbridge: ConfBridge sometimes does not play user name recording
while leaving
(Reported by Robert Mordec)

   - [ASTERISK-26979
   <https://issues.asterisk.org/jira/browse/ASTERISK-26979>] -

res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or
(Reported by Javier Riveros )

   - [ASTERISK-26982
   <https://issues.asterisk.org/jira/browse/ASTERISK-26982>] -

chan_sip: rtcp_mux setting may cause ice completion failure/delay if client
offers rtcp-mux as negotiable
(Reported by Stefan Engström)

   - [ASTERISK-26939
   <https://issues.asterisk.org/jira/browse/ASTERISK-26939>] -

Out of bound memory access in PJSIP multipart parser crashes Asterisk
(Reported by Sandro Gauci)

   - [ASTERISK-26940
   <https://issues.asterisk.org/jira/browse/ASTERISK-26940>] -

Asterisk Skinny memory exhaustion vulnerability leads to DoS
(Reported by Sandro Gauci)

   - [ASTERISK-26938
   <https://issues.asterisk.org/jira/browse/ASTERISK-26938>] -

Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
(Reported by Sandro Gauci)

   - [ASTERISK-26789
   <https://issues.asterisk.org/jira/browse/ASTERISK-26789>] -

Audit manipulation of channel flags without locks
(Reported by Joshua C. Colp)

   - [ASTERISK-26998
   <https://issues.asterisk.org/jira/browse/ASTERISK-26998>] -

res_pjsip_session: INVITE retransmissions could still setup the same call
(Reported by Richard Mudgett)

   - [ASTERISK-26143
   <https://issues.asterisk.org/jira/browse/ASTERISK-26143>] -

res_rtp_asterisk: One way audio when transcoding
(Reported by Henning Holtschneider)

   - [ASTERISK-26333
   <https://issues.asterisk.org/jira/browse/ASTERISK-26333>] -

Problems with Blind Transfer, PJSIP (Aastra 6869i)
(Reported by Matthias Binder)

   - [ASTERISK-26606
   <https://issues.asterisk.org/jira/browse/ASTERISK-26606>] -

tcptls: Incorrect OpenSSL function call leads to misleading error report
(Reported by Bob Ham)

   - [ASTERISK-26983
   <https://issues.asterisk.org/jira/browse/ASTERISK-26983>] -

Crash in Manager Reload when TLS Config Changes
(Reported by Joshua Elson)

   - [ASTERISK-25032
   <https://issues.asterisk.org/jira/browse/ASTERISK-25032>] -

[patch]cel_odbc sometimes inserts CEL with wrong eventtime
(Reported by Etienne Lessard)

   - [ASTERISK-26173
   <https://issues.asterisk.org/jira/browse/ASTERISK-26173>] -

func_cdr: CDR function does not permit empty values to be assigned
(Reported by gkloepfer)

   - [ASTERISK-25506
   <https://issues.asterisk.org/jira/browse/ASTERISK-25506>] -

[patch]CONFBRIDGE failure after an app_confbrige.so module reload results
in segfault or error/warning messages.
(Reported by Frederic LE FOLL)

   - [ASTERISK-24529
   <https://issues.asterisk.org/jira/browse/ASTERISK-24529>] -

Using AMI Action Bridge to on an already bridged channel causes the
incorrect return priority to be used
(Reported by Corey Farrell)

   - [ASTERISK-26966
   <https://issues.asterisk.org/jira/browse/ASTERISK-26966>] -

bridge_simple: Add support for streams
(Reported by Kevin Harwell)

   - [ASTERISK-26860
   <https://issues.asterisk.org/jira/browse/ASTERISK-26860>] -

Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port
missing in (null)
(Reported by Evers Lab)

   - [ASTERISK-26974
   <https://issues.asterisk.org/jira/browse/ASTERISK-26974>] -

res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)

   - [ASTERISK-26908
   <https://issues.asterisk.org/jira/browse/ASTERISK-26908>] -

res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
(Reported by Richard Mudgett)

   - [ASTERISK-26959
   <https://issues.asterisk.org/jira/browse/ASTERISK-26959>] -

dial: Allow topology of dialing channel to influence dialed channel
(Reported by Joshua C. Colp)

   - [ASTERISK-25823
   <https://issues.asterisk.org/jira/browse/ASTERISK-25823>] -

SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or
(Reported by Andreas Krüger)

   - [ASTERISK-26926
   <https://issues.asterisk.org/jira/browse/ASTERISK-26926>] -

func_speex: Crash caused by frame with no datalen
(Reported by Richard Kenner)

   - [ASTERISK-26964
   <https://issues.asterisk.org/jira/browse/ASTERISK-26964>] -

res_pjsip_session: Wrong From on reinvite when request and To URI differ
(Reported by Yasin CANER)

   - [ASTERISK-26930
   <https://issues.asterisk.org/jira/browse/ASTERISK-26930>] -

pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2
instrunction Linux
(Reported by abelbeck)

   - [ASTERISK-26929
   <https://issues.asterisk.org/jira/browse/ASTERISK-26929>] -

pjsip: Add database tables for RLS
(Reported by Joshua C. Colp)

   - [ASTERISK-26949
   <https://issues.asterisk.org/jira/browse/ASTERISK-26949>] -

sdp: Implement T.38
(Reported by Joshua C. Colp)

   - [ASTERISK-26953
   <https://issues.asterisk.org/jira/browse/ASTERISK-26953>] -

Asterisk crash if hep.conf have some missing parameters
(Reported by Joel Vandal)

   - [ASTERISK-26890
   <https://issues.asterisk.org/jira/browse/ASTERISK-26890>] -

STUN server with non-default-route transport causes INVITE delay
(Reported by George Joseph)

   - [ASTERISK-26951
   <https://issues.asterisk.org/jira/browse/ASTERISK-26951>] -

chan_sip: ACK with SDP does not update a direct media bridge
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-26692
   <https://issues.asterisk.org/jira/browse/ASTERISK-26692>] -

res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk
(using chan_sip)
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26835
   <https://issues.asterisk.org/jira/browse/ASTERISK-26835>] -

res_rtp_asterisk: Crash when freeing RTCP address string
(Reported by Niklas Larsson)

   - [ASTERISK-26853
   <https://issues.asterisk.org/jira/browse/ASTERISK-26853>] -

res_rtp_asterisk: Crash in pjnath when receiving packet
(Reported by Adagio)

   - [ASTERISK-26613
   <https://issues.asterisk.org/jira/browse/ASTERISK-26613>] -

format_wav: wav16 format read file only by 320 - half of frame
(Reported by Vitaly K)

   - [ASTERISK-26169
   <https://issues.asterisk.org/jira/browse/ASTERISK-26169>] -

format_ogg_vorbis: Memory leak using OGG in MixMonitor
(Reported by Ivan Myalkin)

   - [ASTERISK-21856
   <https://issues.asterisk.org/jira/browse/ASTERISK-21856>] -

STUN never works when asterisk started without internet access
(Reported by Jeremy Kister)

   - [ASTERISK-20984
   <https://issues.asterisk.org/jira/browse/ASTERISK-20984>] -

Audible clicks when playing sox encoded au file with STREAM FILE AGI command
(Reported by Roman S.)

   - [ASTERISK-26528
   <https://issues.asterisk.org/jira/browse/ASTERISK-26528>] -

[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
(Reported by Badalian Vyacheslav)

   - [ASTERISK-26851
   <https://issues.asterisk.org/jira/browse/ASTERISK-26851>] -

res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)

   - [ASTERISK-26903
   <https://issues.asterisk.org/jira/browse/ASTERISK-26903>] -

Listening TCP/TLS sockets stop when temporarily out of open files
(Reported by Walter Doekes)

   - [ASTERISK-26928
   <https://issues.asterisk.org/jira/browse/ASTERISK-26928>] -

pjsip: Add database tables for PUBLISH support
(Reported by Joshua C. Colp)

   - [ASTERISK-26927
   <https://issues.asterisk.org/jira/browse/ASTERISK-26927>] -

pjproject_bundled: Crash on pj_ssl_get_info() while
(Reported by Alexander Traud)

   - [ASTERISK-26905
   <https://issues.asterisk.org/jira/browse/ASTERISK-26905>] -

pjproject_bundled: Merge 3 upstream deadlock patches into bundled
(Reported by Ross Beer)

   - [ASTERISK-26920
   <https://issues.asterisk.org/jira/browse/ASTERISK-26920>] -

app_queue: PAUSEALL/UNPAUSEALL does not log reason
(Reported by Troy Bowman)

   - [ASTERISK-26897
   <https://issues.asterisk.org/jira/browse/ASTERISK-26897>] -

chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)

   - [ASTERISK-25974
   <https://issues.asterisk.org/jira/browse/ASTERISK-25974>] -

Unused realtime MOH classes not purged on 'moh reload'
(Reported by Sébastien Couture)

   - [ASTERISK-26916
   <https://issues.asterisk.org/jira/browse/ASTERISK-26916>] -

res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)

   - [ASTERISK-21721
   <https://issues.asterisk.org/jira/browse/ASTERISK-21721>] -

SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)

   - [ASTERISK-26915
   <https://issues.asterisk.org/jira/browse/ASTERISK-26915>] -

chan_sip: Session Timers required but refused wrongly.
(Reported by Alexander Traud)

   - [ASTERISK-26363
   <https://issues.asterisk.org/jira/browse/ASTERISK-26363>] -

res_pjsip: Bye sent to sip trunk is not authenticated even after receiving
a 407 error code
(Reported by Yaacov Akiba Slama)

   - [ASTERISK-26896
   <https://issues.asterisk.org/jira/browse/ASTERISK-26896>] -

Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
(Reported by twisted)

   - [ASTERISK-26705
   <https://issues.asterisk.org/jira/browse/ASTERISK-26705>] -

libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)

   - [ASTERISK-26900
   <https://issues.asterisk.org/jira/browse/ASTERISK-26900>] -

sdp: Add support for connection address management and topology updating
(Reported by Joshua C. Colp)

   - [ASTERISK-21009
   <https://issues.asterisk.org/jira/browse/ASTERISK-21009>] -

xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub
unsubscription on client
(Reported by Marcello Ceschia)

   - [ASTERISK-25490
   <https://issues.asterisk.org/jira/browse/ASTERISK-25490>] -

[patch]SDP crypto tag is validated incorrectly
(Reported by Joerg Sonnenberger)

   - [ASTERISK-26885
   <https://issues.asterisk.org/jira/browse/ASTERISK-26885>] -

channel: Support dynamic number of file descriptors
(Reported by Joshua C. Colp)

   - [ASTERISK-26086
   <https://issues.asterisk.org/jira/browse/ASTERISK-26086>] -

res_musiconhold: format option is not documented adequately
(Reported by Jens Bürger)

   - [ASTERISK-23996
   <https://issues.asterisk.org/jira/browse/ASTERISK-23996>] -

No core dumps because of res_musiconhold chdir.
(Reported by Walter Doekes)

   - [ASTERISK-24712
   <https://issues.asterisk.org/jira/browse/ASTERISK-24712>] -

xmpp: starttls problem causes connection spew
(Reported by Matthias Urlichs)

   - [ASTERISK-26814
   <https://issues.asterisk.org/jira/browse/ASTERISK-26814>] -

pjproject_bundled build fails to download pjproject source when using cURL
(Reported by Gergely Dömsödi)

   - [ASTERISK-23510
   <https://issues.asterisk.org/jira/browse/ASTERISK-23510>] -

JABBER_STATUS fails with improper code 7 for unavailable clients
(Reported by Anthony Critelli)

   - [ASTERISK-21855
   <https://issues.asterisk.org/jira/browse/ASTERISK-21855>] -

Asterisk crashes when XMPP message is sent (JabberSend) and no internet
connection is available
(Reported by Jeremy Kister)

   - [ASTERISK-25622
   <https://issues.asterisk.org/jira/browse/ASTERISK-25622>] -

WARNING for "JABBER: socket read error" should be more specific
(Reported by Sean Darcy)

   - [ASTERISK-26515
   <https://issues.asterisk.org/jira/browse/ASTERISK-26515>] -

rtp_engine: Allocate RTP payloads on a per-session basis
(Reported by Joshua C. Colp)

   - [ASTERISK-26818
   <https://issues.asterisk.org/jira/browse/ASTERISK-26818>] -

cdr: Problem setting variables in h exten
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26850
   <https://issues.asterisk.org/jira/browse/ASTERISK-26850>] -

res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)

   - [ASTERISK-26484
   <https://issues.asterisk.org/jira/browse/ASTERISK-26484>] -

res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
(Reported by Vinod Dharashive)

   - [ASTERISK-26776
   <https://issues.asterisk.org/jira/browse/ASTERISK-26776>] -

res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)

   - [ASTERISK-26880
   <https://issues.asterisk.org/jira/browse/ASTERISK-26880>] -

Asterisk crashes when multiple speex users join confbridge with pp_vad and
dtx enabled
(Reported by Kirsty Tyerman)

   - [ASTERISK-26875
   <https://issues.asterisk.org/jira/browse/ASTERISK-26875>] -

app_mixmonitor: Recording out of sync when 183 but no RTP
(Reported by Aaron An)

   - [ASTERISK-26862
   <https://issues.asterisk.org/jira/browse/ASTERISK-26862>] -

app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)

   - [ASTERISK-26732
   <https://issues.asterisk.org/jira/browse/ASTERISK-26732>] -

res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)

   - [ASTERISK-26867
   <https://issues.asterisk.org/jira/browse/ASTERISK-26867>] -

autochan: Locking in a function ast_autochan_destroy() on destroyed channel
(after masquerade).
(Reported by Krzysztof Trempala)

   - [ASTERISK-26869
   <https://issues.asterisk.org/jira/browse/ASTERISK-26869>] -

res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
(Reported by Torrey Searle)

   - [ASTERISK-26668
   <https://issues.asterisk.org/jira/browse/ASTERISK-26668>] -

core: Malformed pattern matching extension (various factors) results in
(Reported by xrobau)

   - [ASTERISK-26865
   <https://issues.asterisk.org/jira/browse/ASTERISK-26865>] -

chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)

   - [ASTERISK-26872
   <https://issues.asterisk.org/jira/browse/ASTERISK-26872>] -

Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)

   - [ASTERISK-26717
   <https://issues.asterisk.org/jira/browse/ASTERISK-26717>] -

Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)

   - [ASTERISK-26643
   <https://issues.asterisk.org/jira/browse/ASTERISK-26643>] -

Extra new line in Device field of DeviceStateChange AMI Event after restart
of Asterisk
(Reported by Roman Bedros)

   - [ASTERISK-25237
   <https://issues.asterisk.org/jira/browse/ASTERISK-25237>] -

stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)

   - [ASTERISK-26857
   <https://issues.asterisk.org/jira/browse/ASTERISK-26857>] -

chan_pjsip: Dialplan function race condition
(Reported by Joshua C. Colp)

   - [ASTERISK-26822
   <https://issues.asterisk.org/jira/browse/ASTERISK-26822>] -

pjsip/cli_commands: pjsip show channelstats shows wrong codec
(Reported by Kevin Harwell)

   - [ASTERISK-26353
   <https://issues.asterisk.org/jira/browse/ASTERISK-26353>] -

res_musiconhold: musiconhold seems to think that the general section is a
class and issues warning
(Reported by Jonathan Harris)

   - [ASTERISK-26685
   <https://issues.asterisk.org/jira/browse/ASTERISK-26685>] -

res_pjsip: Crash when using IPv6 and Transport ws,wss
(Reported by Michael Balen)

   - [ASTERISK-24562
   <https://issues.asterisk.org/jira/browse/ASTERISK-24562>] -

app_voicemail: Cannot set fromstring on a per-mailbox basis
(Reported by Mark Scholten)

   - [ASTERISK-26842
   <https://issues.asterisk.org/jira/browse/ASTERISK-26842>] -

Websocket becomes disconnected when trying to place call from browser
(Reported by Mark Michelson)

   - [ASTERISK-26841
   <https://issues.asterisk.org/jira/browse/ASTERISK-26841>] -

chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-26839
   <https://issues.asterisk.org/jira/browse/ASTERISK-26839>] -

core: Implement stream topology changing in channels
(Reported by Joshua C. Colp)

   - [ASTERISK-26598
   <https://issues.asterisk.org/jira/browse/ASTERISK-26598>] -

Saynumber is trying to get "and" from "digits/" subfolder
(Reported by Jonathan Harris)

   - [ASTERISK-17067
   <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] -

Long lines in call files cause spurious syntax error
(Reported by Dave Olszewski)

   - [ASTERISK-26796
   <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] -

res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
(Reported by Jørgen H)

   - [ASTERISK-26816
   <https://issues.asterisk.org/jira/browse/ASTERISK-26816>] -

Implement ast_read_stream in channels
(Reported by Joshua C. Colp)

   - [ASTERISK-25628
   <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] -

res_config_pgsql: should match the behavior of other drivers so that
queue_log can disable adaptive logging
(Reported by Dmitry Wagin)

   - [ASTERISK-26774
   <https://issues.asterisk.org/jira/browse/ASTERISK-26774>] -

core: Playback URL fails after some time
(Reported by Igor Gamayunov)

   - [ASTERISK-26825
   <https://issues.asterisk.org/jira/browse/ASTERISK-26825>] -

pjsip.conf.sample: user_agent: still refers to branch 12
(Reported by Tzafrir Cohen)

   - [ASTERISK-26823
   <https://issues.asterisk.org/jira/browse/ASTERISK-26823>] -

PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
(Reported by Mark Michelson)

   - [ASTERISK-26623
   <https://issues.asterisk.org/jira/browse/ASTERISK-26623>] -

res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by Jørgen H)

   - [ASTERISK-26808
   <https://issues.asterisk.org/jira/browse/ASTERISK-26808>] -

res_pjsip_outbound_registration doesn't know about network change events
(Reported by George Joseph)

   - [ASTERISK-26781
   <https://issues.asterisk.org/jira/browse/ASTERISK-26781>] -

bridge: Passing the 'p' (play tone) flag to Bridge() application results in
garbled audio
(Reported by Sean Bright)

   - [ASTERISK-26782
   <https://issues.asterisk.org/jira/browse/ASTERISK-26782>] -

res_pjsip: URI requirement for fields is not consistently documented and
error does not provide indication
(Reported by Peter Sokolov)

   - [ASTERISK-26793
   <https://issues.asterisk.org/jira/browse/ASTERISK-26793>] -

Implement ast_write_stream in channels
(Reported by George Joseph)

   - [ASTERISK-26812
   <https://issues.asterisk.org/jira/browse/ASTERISK-26812>] -

[patch] Fix download_externals To Allow The Use Of curl Or wget
(Reported by Michael L. Young)

   - [ASTERISK-18271
   <https://issues.asterisk.org/jira/browse/ASTERISK-18271>] -

Pattern matching with res_config_mysql extensions does not behave as
(Reported by Charlie Smurthwaite)

   - [ASTERISK-26811
   <https://issues.asterisk.org/jira/browse/ASTERISK-26811>] -

stream: Add streams to "core show channel"
(Reported by Joshua C. Colp)

   - [ASTERISK-18731
   <https://issues.asterisk.org/jira/browse/ASTERISK-18731>] -

[patch] DUNDi weight parameter not processed correctly
(Reported by Peter Racz)

   - [ASTERISK-26799
   <https://issues.asterisk.org/jira/browse/ASTERISK-26799>] -

res_pjsip: Using an auth object for inbound and outbound authentication
(Reported by Richard Mudgett)

   - [ASTERISK-26669
   <https://issues.asterisk.org/jira/browse/ASTERISK-26669>] -

PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)

   - [ASTERISK-26738
   <https://issues.asterisk.org/jira/browse/ASTERISK-26738>] -

Frequent segfaults since activation of DNS SRV, in
pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and
pj_atomic_inc_and_get at pj/os_core_unix.c
(Reported by Michael Maier)

   - [ASTERISK-25893
   <https://issues.asterisk.org/jira/browse/ASTERISK-25893>] -

Function vmauthenticate accesses uninitialized memory
(Reported by Filip Jenicek)

   - [ASTERISK-26580
   <https://issues.asterisk.org/jira/browse/ASTERISK-26580>] -

[patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)

   - [ASTERISK-26802
   <https://issues.asterisk.org/jira/browse/ASTERISK-26802>] -

[patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)

   - [ASTERISK-15858
   <https://issues.asterisk.org/jira/browse/ASTERISK-15858>] -

[patch] Fix query with double backslash in string literals and stop log
(Reported by Humberto Figuera)

   - [ASTERISK-26057
   <https://issues.asterisk.org/jira/browse/ASTERISK-26057>] -

res_config_sqlite3 uses incorrect query - unnecessary escape
(Reported by Stepan)

   - [ASTERISK-23457
   <https://issues.asterisk.org/jira/browse/ASTERISK-23457>] -

SQlite3: Realtime queue loading fails after PRAGMA query result
(Reported by Scott Griepentrog)

   - [ASTERISK-26794
   <https://issues.asterisk.org/jira/browse/ASTERISK-26794>] -

http: Crash on Reload Only in ast_tcptls_server_start
(Reported by Joshua Elson)

   - [ASTERISK-26714
   <https://issues.asterisk.org/jira/browse/ASTERISK-26714>] -

Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)

   - [ASTERISK-26696
   <https://issues.asterisk.org/jira/browse/ASTERISK-26696>] -

pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on
subscription refresh
(Reported by Zach R)

   - [ASTERISK-26756
   <https://issues.asterisk.org/jira/browse/ASTERISK-26756>] -

res_pjsip_mwi: Asterisk does not terminate MWI subscription
(Reported by Carl Fortin)

   - [ASTERISK-26790
   <https://issues.asterisk.org/jira/browse/ASTERISK-26790>] -

Implement stream topology (non-change request) API usage in channels
(Reported by George Joseph)

   - [ASTERISK-26723
   <https://issues.asterisk.org/jira/browse/ASTERISK-26723>] -

VoiceMailPlayMsg not playing messages via realtime
(Reported by Ryan Rittgarn)

   - [ASTERISK-18286
   <https://issues.asterisk.org/jira/browse/ASTERISK-18286>] -

[patch] 'Silence' is truncated in Record()
(Reported by var)

   - [ASTERISK-26775
   <https://issues.asterisk.org/jira/browse/ASTERISK-26775>] -

app_queue: reset abandoned in service level
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26786
   <https://issues.asterisk.org/jira/browse/ASTERISK-26786>] -

Implement ast_stream_topology API
(Reported by George Joseph)

   - [ASTERISK-26248
   <https://issues.asterisk.org/jira/browse/ASTERISK-26248>] -

chan_pjsip: Error when calling PJSIP client with domain specified
(Reported by Norbert Varga)

   - [ASTERISK-26788
   <https://issues.asterisk.org/jira/browse/ASTERISK-26788>] -

core: Protect flags during ast_waitfor
(Reported by Joshua C. Colp)

   - [ASTERISK-26115
   <https://issues.asterisk.org/jira/browse/ASTERISK-26115>] -

pbx: AMI Originate ignore "failed" extension on call failure
(Reported by Nasir Iqbal)

   - [ASTERISK-26773
   <https://issues.asterisk.org/jira/browse/ASTERISK-26773>] -

stream: Add basic API
(Reported by Joshua C. Colp)

   - [ASTERISK-26785
   <https://issues.asterisk.org/jira/browse/ASTERISK-26785>] -

configs/samples: The 'identify' entry is in the wrong section in
(Reported by Torrey Searle)

   - [ASTERISK-26772
   <https://issues.asterisk.org/jira/browse/ASTERISK-26772>] -

Crash in srv.c on startup with pjsip
(Reported by nappsoft)

   - [ASTERISK-26770
   <https://issues.asterisk.org/jira/browse/ASTERISK-26770>] -

res_stasis_device_state: Duplicate subscriptions when multiple received at
same time
(Reported by Joshua C. Colp)

   - [ASTERISK-26767
   <https://issues.asterisk.org/jira/browse/ASTERISK-26767>] -

ARI channelvars cause memory leak
(Reported by Sébastien Duthil)

   - [ASTERISK-26716
   <https://issues.asterisk.org/jira/browse/ASTERISK-26716>] -

ari: Channels with pre-dial handlers cannot be hung up via ARI
(Reported by Tom Pawelek)

   - [ASTERISK-26632
   <https://issues.asterisk.org/jira/browse/ASTERISK-26632>] -

core: Possibility of a frame "imbalance" leading to stuck channels.
(Reported by Mark Michelson)

   - [ASTERISK-25951
   <https://issues.asterisk.org/jira/browse/ASTERISK-25951>] -

res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)

   - [ASTERISK-26343
   <https://issues.asterisk.org/jira/browse/ASTERISK-26343>] -

ASTERISK-25951 causes issues for callerid manipulation through agi
(Reported by Morten Tryfoss)

   - [ASTERISK-26704
   <https://issues.asterisk.org/jira/browse/ASTERISK-26704>] -

res_odbc.conf contains deprecated configuration: 'pooling',
'shared_connections', 'limit', and 'idlecheck' options were replaced by
(Reported by Anthony Messina)

   - [ASTERISK-26765
   <https://issues.asterisk.org/jira/browse/ASTERISK-26765>] -

res_resolver_unbound: FRACK! Excessive ref count trap tripped.
(Reported by Richard Mudgett)

   - [ASTERISK-21094
   <https://issues.asterisk.org/jira/browse/ASTERISK-21094>] -

MixMonitorMute mutes through stream if already slinear (e.g. Originate)
(Reported by David Woolley)

   - [ASTERISK-26679
   <https://issues.asterisk.org/jira/browse/ASTERISK-26679>] -

Crash on invalid contact domain (pjsip aor)
(Reported by Dmitriy)

   - [ASTERISK-26699
   <https://issues.asterisk.org/jira/browse/ASTERISK-26699>] -

res_pjsip: Assertion when sending OPTIONS request to endpoint
(Reported by Ross Beer)

   - [ASTERISK-26754
   <https://issues.asterisk.org/jira/browse/ASTERISK-26754>] -

build_tools: make_build_h does not handle \ in user name
(Reported by Kirill Katsnelson)

   - [ASTERISK-26755
   <https://issues.asterisk.org/jira/browse/ASTERISK-26755>] -

app_queue: Random queues disappear on "core reload queue all"
(Reported by Kirill Katsnelson)

   - [ASTERISK-26735
   <https://issues.asterisk.org/jira/browse/ASTERISK-26735>] -

res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no
(Reported by Michael Maier)

   - [ASTERISK-26693
   <https://issues.asterisk.org/jira/browse/ASTERISK-26693>] -

res_pjsip_endpoint_identifier_ip: Add support for SRV
(Reported by Joshua C. Colp)

   - [ASTERISK-26743
   <https://issues.asterisk.org/jira/browse/ASTERISK-26743>] -

PJPROJECT: Detecting compiled max log level does not work.
(Reported by Richard Mudgett)

   - [ASTERISK-26731
   <https://issues.asterisk.org/jira/browse/ASTERISK-26731>] -

res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
(Reported by Ustinov Artem)

   - [ASTERISK-26739
   <https://issues.asterisk.org/jira/browse/ASTERISK-26739>] -

voicemail API test: confuses expected and actual values
(Reported by Tzafrir Cohen)

   - [ASTERISK-26740
   <https://issues.asterisk.org/jira/browse/ASTERISK-26740>] -

voicemail API test: uses varlibdir instead of datadir for a sound file
(Reported by Tzafrir Cohen)

   - [ASTERISK-26665
   <https://issues.asterisk.org/jira/browse/ASTERISK-26665>] -

app_queue: Agent ringing, Caller hangup before timeout, no agent name
logged - missing RINGNOANSWER?
(Reported by Marek Cervenka)

   - [ASTERISK-26710
   <https://issues.asterisk.org/jira/browse/ASTERISK-26710>] -

[patch] res_rtp_asterisk: CHANNEL arguments,
(rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
(Reported by Aaron An)

   - [ASTERISK-26672
   <https://issues.asterisk.org/jira/browse/ASTERISK-26672>] -

Crash when setting remote address on RTP instance
(Reported by Richard Mudgett)

   - [ASTERISK-26670
   <https://issues.asterisk.org/jira/browse/ASTERISK-26670>] -

[patch] Outgoing SIP-URI Dialing via PJSIP
(Reported by Alexander Traud)

   - [ASTERISK-26691
   <https://issues.asterisk.org/jira/browse/ASTERISK-26691>] -

Remember SDP negotiation on SIP_CODEC_INBOUND.
(Reported by Alexander Traud)

   - [ASTERISK-26673
   <https://issues.asterisk.org/jira/browse/ASTERISK-26673>] -

chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
(Reported by Joshua C. Colp)

   - [ASTERISK-26684
   <https://issues.asterisk.org/jira/browse/ASTERISK-26684>] -

res_pjsip: Various issues with compact SIP headers
(Reported by Joshua Elson)

   - [ASTERISK-26683
   <https://issues.asterisk.org/jira/browse/ASTERISK-26683>] -

res_calendar: Calendars duplicated after module reload
(Reported by Martin Tomec)

   - [ASTERISK-26655
   <https://issues.asterisk.org/jira/browse/ASTERISK-26655>] -

[patch]pjsip: Transfers Broken with Compact Headers Enabled
(Reported by JoshE)

   - [ASTERISK-26621
   <https://issues.asterisk.org/jira/browse/ASTERISK-26621>] -

app_queue: Queue application does not ring members with Local interface
(Reported by Jonas Kellens)

   - [ASTERISK-26586
   <https://issues.asterisk.org/jira/browse/ASTERISK-26586>] -

chan_sip: Segfaults upon reload if client with MWI wasn't registered
(Reported by Michael Kuron)

   - [ASTERISK-25494
   <https://issues.asterisk.org/jira/browse/ASTERISK-25494>] -

build: GCC 5.1.x catches some new const, array bounds and missing paren
(Reported by George Joseph)

   - [ASTERISK-24499
   <https://issues.asterisk.org/jira/browse/ASTERISK-24499>] -

Need more explicit debug when PJSIP dialstring is invalid
(Reported by Rusty Newton)

   - [ASTERISK-25083
   <https://issues.asterisk.org/jira/browse/ASTERISK-25083>] -

Message.c: Message channel becomes saturated with frames leading to spammy
log messages
(Reported by Jonathan Rose)

   - [ASTERISK-26653
   <https://issues.asterisk.org/jira/browse/ASTERISK-26653>] -

pjproject_bundled doesn't verify already downloaded tarballs
(Reported by George Joseph)

   - [ASTERISK-26433
   <https://issues.asterisk.org/jira/browse/ASTERISK-26433>] -

chan_sip: Allows To-tag checks to be bypassed, setting up new calls
(Reported by Walter Doekes)

   - [ASTERISK-26579
   <https://issues.asterisk.org/jira/browse/ASTERISK-26579>] -

codec_opus: Recursiveness when parsing fmtp line
(Reported by Jørgen H)

   - [ASTERISK-26644
   <https://issues.asterisk.org/jira/browse/ASTERISK-26644>] -

PJSIPShowRegistrationsInbound just dumps all aors
(Reported by George Joseph)

   - [ASTERISK-26647
   <https://issues.asterisk.org/jira/browse/ASTERISK-26647>] -

Support older DNS style for OpenBSD
(Reported by snuffy)

   - [ASTERISK-26490
   <https://issues.asterisk.org/jira/browse/ASTERISK-26490>] -

res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction
branch parameter contains "_"
(Reported by Juris Breicis)

   - [ASTERISK-26629
   <https://issues.asterisk.org/jira/browse/ASTERISK-26629>] -

tests/manager: 4 test failures as a result of iostream change
(Reported by Joshua C. Colp)

   - [ASTERISK-26109
   <https://issues.asterisk.org/jira/browse/ASTERISK-26109>] -

Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)

   - [ASTERISK-26617
   <https://issues.asterisk.org/jira/browse/ASTERISK-26617>] -

res_rtp_asterisk: Can't bind on systems without IPv6
(Reported by Guido Falsi)

   - [ASTERISK-26603
   <https://issues.asterisk.org/jira/browse/ASTERISK-26603>] -

[patch] chan_pjsip: not switching sending codec to receiving codec when
(Reported by Alexei Gradinari)

   - [ASTERISK-24330
   <https://issues.asterisk.org/jira/browse/ASTERISK-24330>] -

Requirement for 'wss' value in Contact header transport parameter on
inbound traffic violates RFC7118
(Reported by Marek Cervenka)

   - [ASTERISK-26566
   <https://issues.asterisk.org/jira/browse/ASTERISK-26566>] -

res_rtp_asterisk: RTT miscalculation in RTCP
(Reported by Hector Royo Concepcion)

   - [ASTERISK-26604
   <https://issues.asterisk.org/jira/browse/ASTERISK-26604>] -

chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
(Reported by Michael Kuron)

   - [ASTERISK-26608
   <https://issues.asterisk.org/jira/browse/ASTERISK-26608>] -

Compile and link failures on OpenBSD
(Reported by snuffy)

   - [ASTERISK-26520
   <https://issues.asterisk.org/jira/browse/ASTERISK-26520>] -

codec_opus: Generated fmtp line has no content
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26605
   <https://issues.asterisk.org/jira/browse/ASTERISK-26605>] -

codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
(Reported by Richard Mudgett)

   - [ASTERISK-26516
   <https://issues.asterisk.org/jira/browse/ASTERISK-26516>] -

pjsip: Memory corruption with possible memory leak.
(Reported by Richard Mudgett)

   - [ASTERISK-24515
   <https://issues.asterisk.org/jira/browse/ASTERISK-24515>] -

Unconditional use of fopencookie() / funopen() is non-portable
(Reported by Timo Teräs)

   - [ASTERISK-26556
   <https://issues.asterisk.org/jira/browse/ASTERISK-26556>] -

manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax
(Reported by Michelle Dupuis)

   - [ASTERISK-26592
   <https://issues.asterisk.org/jira/browse/ASTERISK-26592>] -

Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
(Reported by George Joseph)

   - [ASTERISK-26575
   <https://issues.asterisk.org/jira/browse/ASTERISK-26575>] -

testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
(Reported by Joshua C. Colp)

   - [ASTERISK-26565
   <https://issues.asterisk.org/jira/browse/ASTERISK-26565>] -

chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
(Reported by Ruse)

   - [ASTERISK-26573
   <https://issues.asterisk.org/jira/browse/ASTERISK-26573>] -

Some typos in documentation of chan_sip.c
(Reported by C.J. Collier)

   - [ASTERISK-26571
   <https://issues.asterisk.org/jira/browse/ASTERISK-26571>] -

res_pjsip: Resolution incorrect when explicit IPv6 transport configured
(Reported by Joshua C. Colp)

   - [ASTERISK-26468
   <https://issues.asterisk.org/jira/browse/ASTERISK-26468>] -

ari: Bridge events stop working after this sequence of ARI calls
(Reported by Daniele Pallastrelli)

   - [ASTERISK-24400
   <https://issues.asterisk.org/jira/browse/ASTERISK-24400>] -

ooh323 sends wrong hangup code
(Reported by Dmitry Melekhov)

   - [ASTERISK-26555
   <https://issues.asterisk.org/jira/browse/ASTERISK-26555>] -

Multi-party Video: Fix some post Asterisk-11 regressions
(Reported by Matt Jordan)

   - [ASTERISK-26412
   <https://issues.asterisk.org/jira/browse/ASTERISK-26412>] -

build: Prepare for gcc 6.2
(Reported by George Joseph)

   - [ASTERISK-26509
   <https://issues.asterisk.org/jira/browse/ASTERISK-26509>] -

A few non-critical deprecation warnings when building on Ubuntu 16.10
(Reported by Jonathan Harris)

   - [ASTERISK-26523
   <https://issues.asterisk.org/jira/browse/ASTERISK-26523>] -

chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes -
rtptimeout behaving badly - regression
(Reported by Michael Keuter)

   - [ASTERISK-26549
   <https://issues.asterisk.org/jira/browse/ASTERISK-26549>] -

app_dial: When PickupChan() is used some channels may have incorrect device
(Reported by Joshua C. Colp)

   - [ASTERISK-24274
   <https://issues.asterisk.org/jira/browse/ASTERISK-24274>] -

[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48
Codec Is Used
(Reported by Frankie Chin)

   - [ASTERISK-26311
   <https://issues.asterisk.org/jira/browse/ASTERISK-26311>] -

[patch] rtp_engine: Allow more than 32 dynamic payload types.
(Reported by Alexander Traud)

   - [ASTERISK-26546
   <https://issues.asterisk.org/jira/browse/ASTERISK-26546>] -

mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
(Reported by Tzafrir Cohen)

   - [ASTERISK-26541
   <https://issues.asterisk.org/jira/browse/ASTERISK-26541>] -

res_pjsip_sdp_rtp: Restrict number of formats to maximum
(Reported by Joshua C. Colp)

   - [ASTERISK-26476
   <https://issues.asterisk.org/jira/browse/ASTERISK-26476>] -

chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show
(Reported by Sergey Grachev)

   - [ASTERISK-25070
   <https://issues.asterisk.org/jira/browse/ASTERISK-25070>] -

Fix FTBFS on Hurd
(Reported by Gabriele Giacone)

   - [ASTERISK-26537
   <https://issues.asterisk.org/jira/browse/ASTERISK-26537>] -

AMI: NewConnectedLine event is not documented
(Reported by Etienne Lessard)

   - [ASTERISK-26526
   <https://issues.asterisk.org/jira/browse/ASTERISK-26526>] -

[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
(Reported by Badalian Vyacheslav)

   - [ASTERISK-26524
   <https://issues.asterisk.org/jira/browse/ASTERISK-26524>] -

astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
(Reported by Corey Farrell)

   - [ASTERISK-26344
   <https://issues.asterisk.org/jira/browse/ASTERISK-26344>] -

Asterisk 13.11.0 + PJSIP crash
(Reported by Ian Gilmour)

   - [ASTERISK-26387
   <https://issues.asterisk.org/jira/browse/ASTERISK-26387>] -

Asterisk segfaults shortly after starting even with no active calls.
(Reported by Harley Peters)

   - [ASTERISK-26506
   <https://issues.asterisk.org/jira/browse/ASTERISK-26506>] -

[patch]res_pjsip_outbound_publish: Crash when publishing, in
publisher_client_send at res_pjsip_outbound_publish.c
(Reported by Matt Krokosz)

   - [ASTERISK-26513
   <https://issues.asterisk.org/jira/browse/ASTERISK-26513>] -

tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
(Reported by Joshua C. Colp)

   - [ASTERISK-26514
   <https://issues.asterisk.org/jira/browse/ASTERISK-26514>] -

Super Awesome Company: Don't specify transport in pjsip.conf
(Reported by Rusty Newton)

   - [ASTERISK-26510
   <https://issues.asterisk.org/jira/browse/ASTERISK-26510>] -

pjproject_bundled uses the --strip-components option of tar which isn't
supported in older versions
(Reported by George Joseph)

   - [ASTERISK-22480
   <https://issues.asterisk.org/jira/browse/ASTERISK-22480>] -

Embedded pjproject: build.mak contains hardcoded full path to version.mak
(Reported by Matt Jordan)

   - [ASTERISK-26480
   <https://issues.asterisk.org/jira/browse/ASTERISK-26480>] -

[patch] CLI: core set debug: Auto-completes File not Module
(Reported by Alexander Traud)

   - [ASTERISK-26307
   <https://issues.asterisk.org/jira/browse/ASTERISK-26307>] -

res_pjsip_caller_id: Crash on outgoing change
(Reported by Bill Brigden)

   - [ASTERISK-26503
   <https://issues.asterisk.org/jira/browse/ASTERISK-26503>] -

app_voicemail: Asterisk crashes when MailboxExists is used
(Reported by Doug Lytle)

   - [ASTERISK-26423
   <https://issues.asterisk.org/jira/browse/ASTERISK-26423>] -

res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
(Reported by Andreas Wetzel)

   - [ASTERISK-26309
   <https://issues.asterisk.org/jira/browse/ASTERISK-26309>] -

[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
(Reported by Alexander Traud)

   - [ASTERISK-26482
   <https://issues.asterisk.org/jira/browse/ASTERISK-26482>] -

[patch] chan_pjsip: segfault on already disconnected session
(Reported by Alexei Gradinari)

   - [ASTERISK-26455
   <https://issues.asterisk.org/jira/browse/ASTERISK-26455>] -

cdr_radius / cel_radius: try fix memory leak
(Reported by Badalian Vyacheslav)

   - [ASTERISK-26421
   <https://issues.asterisk.org/jira/browse/ASTERISK-26421>] -

Segmentation Fault with ARI originate into mixing bridge with 43 clients
(Reported by Andrew Nagy)

   - [ASTERISK-26444
   <https://issues.asterisk.org/jira/browse/ASTERISK-26444>] -

'features show' command in CLI does not return prompt.
(Reported by John Kiniston)

   - [ASTERISK-26356
   <https://issues.asterisk.org/jira/browse/ASTERISK-26356>] -

menuselect: invalid test for GTK2
(Reported by Tzafrir Cohen)

   - [ASTERISK-26477
   <https://issues.asterisk.org/jira/browse/ASTERISK-26477>] -

pjproject: SEGV during SSL operations
(Reported by George Joseph)

   - [ASTERISK-26462
   <https://issues.asterisk.org/jira/browse/ASTERISK-26462>] -

[patch] app_queue: While using queues with realtime, setting back to an
empty context doesn't stop the exit key usage
(Reported by Leandro Dardini)

   - [ASTERISK-26439
   <https://issues.asterisk.org/jira/browse/ASTERISK-26439>] -

chan_rtp: Crash when originating
(Reported by Kayode)

   - [ASTERISK-17470
   <https://issues.asterisk.org/jira/browse/ASTERISK-17470>] -

[patch] - When videosupport=yes, asterisk allows one end peer to send
video, even though the other end supports only audio.
(Reported by effie mouzeli)

   - [ASTERISK-26416
   <https://issues.asterisk.org/jira/browse/ASTERISK-26416>] -

pjproject-bundled: configure fails to check for all required utilities
(Reported by Corey Farrell)

   - [ASTERISK-26466
   <https://issues.asterisk.org/jira/browse/ASTERISK-26466>] -

core: Be forgiving on external callerid that may be flawed so we don't drop
(Reported by Richard Mudgett)

   - [ASTERISK-26362
   <https://issues.asterisk.org/jira/browse/ASTERISK-26362>] -

res_config_mysql: Broken after 13.10
(Reported by Carlos Chavez)

   - [ASTERISK-26446
   <https://issues.asterisk.org/jira/browse/ASTERISK-26446>] -

app_dial: There's no way to override the hangupcause on unanswered channels
(Reported by George Joseph)

   - [ASTERISK-26457
   <https://issues.asterisk.org/jira/browse/ASTERISK-26457>] -

[patch] force_rport,auto_comedia: No NAT detection triggered.
(Reported by Alexander Traud)

   - [ASTERISK-26453
   <https://issues.asterisk.org/jira/browse/ASTERISK-26453>] -

res_pjsip_config_wizard: Memory leak in module_unload
(Reported by Badalian Vyacheslav)

   - [ASTERISK-26410
   <https://issues.asterisk.org/jira/browse/ASTERISK-26410>] -

core: Asterisk 14 doesn't show the header in the console or verbose when
(Reported by Dan Jenkins)

   - [ASTERISK-24311
   <https://issues.asterisk.org/jira/browse/ASTERISK-24311>] -

Populating database via Alembic fails when using same database for multiple
schema sets
(Reported by Dafi Ni)

   - [ASTERISK-26438
   <https://issues.asterisk.org/jira/browse/ASTERISK-26438>] -

[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
(Reported by Alexander Traud)

   - [ASTERISK-26330
   <https://issues.asterisk.org/jira/browse/ASTERISK-26330>] -

app_queue: Changing the "ringinuse" parameter of a queue doesn't affect
dynamic members
(Reported by Etienne Lessard)

   - [ASTERISK-18232
   <https://issues.asterisk.org/jira/browse/ASTERISK-18232>] -

Broken REGISTER sent to IPv4 server when bindaddr=[::]
(Reported by Jacek)

   - [ASTERISK-25468
   <https://issues.asterisk.org/jira/browse/ASTERISK-25468>] -

Deadlock in chan_sip - core show locks shows do_monitor lock
(Reported by Barry Flanagan)

   - [ASTERISK-26396
   <https://issues.asterisk.org/jira/browse/ASTERISK-26396>] -

chan_pjsip: HANGUPCAUSE return the wrong code when dialed channel answer.
(Reported by Aaron An)

   - [ASTERISK-26397
   <https://issues.asterisk.org/jira/browse/ASTERISK-26397>] -

manager: PresenceState action crashes Asterisk 14
(Reported by Andrew Nagy)

   - [ASTERISK-26389
   <https://issues.asterisk.org/jira/browse/ASTERISK-26389>] -

res_odbc: Clean up pooling options
(Reported by Joshua C. Colp)

   - [ASTERISK-26273
   <https://issues.asterisk.org/jira/browse/ASTERISK-26273>] -

core: Won't compile when LOW_MEMORY is enabled
(Reported by Anthony Messina)

   - [ASTERISK-26391
   <https://issues.asterisk.org/jira/browse/ASTERISK-26391>] -

Consoles do not display verbose logger messages even when requested.
(Reported by Marcelo Terres)

   - [ASTERISK-26352
   <https://issues.asterisk.org/jira/browse/ASTERISK-26352>] -

Astcanary dies when doing "core restart"
(Reported by Walter Doekes)

   - [ASTERISK-19867
   <https://issues.asterisk.org/jira/browse/ASTERISK-19867>] -

asterisk fails to lower its priority when astcanary dies
(Reported by Xavier Hienne)

   - [ASTERISK-26263
   <https://issues.asterisk.org/jira/browse/ASTERISK-26263>] -

SQL error when using realtime and registering extension / inserting into
(Reported by Jeppe Ryskov Larsen)

   - [ASTERISK-26365
   <https://issues.asterisk.org/jira/browse/ASTERISK-26365>] -

rtp: Offer with multiple payloads for same codec is incorrectly handled
(Reported by Joshua C. Colp)

   - [ASTERISK-26374
   <https://issues.asterisk.org/jira/browse/ASTERISK-26374>] -

res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
(Reported by Joshua C. Colp)

   - [ASTERISK-26359
   <https://issues.asterisk.org/jira/browse/ASTERISK-26359>] -

[patch] cdr_mysql: fails to use UTC if so instructed
(Reported by Tzafrir Cohen)

   - [ASTERISK-26367
   <https://issues.asterisk.org/jira/browse/ASTERISK-26367>] -

rtp: Timestamps broken when video frame is across multiple RTP packets
(Reported by Joshua C. Colp)

   - [ASTERISK-26375
   <https://issues.asterisk.org/jira/browse/ASTERISK-26375>] -

res_pjsip_transport_management: Log message states seconds, but time value
is milliseconds
(Reported by Joshua C. Colp)

   - [ASTERISK-19968
   <https://issues.asterisk.org/jira/browse/ASTERISK-19968>] -

TCP Session-Timers not dropping call
(Reported by Aaron Hamstra)

   - [ASTERISK-26364
   <https://issues.asterisk.org/jira/browse/ASTERISK-26364>] -

res_pjsip: Don't assume a request will have target addresses
(Reported by Joshua C. Colp)

   - [ASTERISK-26360
   <https://issues.asterisk.org/jira/browse/ASTERISK-26360>] -

app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
(Reported by Richard Mudgett)

   - [ASTERISK-26358
   <https://issues.asterisk.org/jira/browse/ASTERISK-26358>] -

chan_sip: Contact is updated on re-200, but not on re-INVITE
(Reported by Walter Doekes)

   - [ASTERISK-26316
   <https://issues.asterisk.org/jira/browse/ASTERISK-26316>] -

res_pjsip_callerid: Irregular URI causes unexpected callerid
(Reported by Kevin Harwell)

   - [ASTERISK-26349
   <https://issues.asterisk.org/jira/browse/ASTERISK-26349>] -

13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
(Reported by Dmitry Melekhov)

   - [ASTERISK-26317
   <https://issues.asterisk.org/jira/browse/ASTERISK-26317>] -

res_pjsip_session: Add ability to use preferred codec only
(Reported by Aaron An)

   - [ASTERISK-26264
   <https://issues.asterisk.org/jira/browse/ASTERISK-26264>] -

res_pjsip: Crash when applying ACL from non-existent endpoint
(Reported by nappsoft)

   - [ASTERISK-26272
   <https://issues.asterisk.org/jira/browse/ASTERISK-26272>] -

chan_sip: File descriptors leak (UDP sockets)
(Reported by Etienne Lessard)

   - [ASTERISK-20234
   <https://issues.asterisk.org/jira/browse/ASTERISK-20234>] -

SRTP not working with some devices (Eg snom320) - Message "We are
requesting SRTP for audio, but they responded without it!"
(Reported by tootai)

   - [ASTERISK-26341
   <https://issues.asterisk.org/jira/browse/ASTERISK-26341>] -

ARI: Stopping a media playlist only stops the current media URI being
played back, and not the whole list
(Reported by Matt Jordan)

   - [ASTERISK-26291
   <https://issues.asterisk.org/jira/browse/ASTERISK-26291>] -

res_pjsip_session: segfault on already disconnected session
(Reported by Alexei Gradinari)

   - [ASTERISK-23989
   <https://issues.asterisk.org/jira/browse/ASTERISK-23989>] -

[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
(Reported by Olle Johansson)

   - [ASTERISK-25691
   <https://issues.asterisk.org/jira/browse/ASTERISK-25691>] -

Crash occurs when screening mode (Dial's 'p' argument) is enabled and
callee rejects a call or hangs up.
(Reported by Etienne Lessard)

   - [ASTERISK-26331
   <https://issues.asterisk.org/jira/browse/ASTERISK-26331>] -

Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
(Reported by CGI.NET)

   - [ASTERISK-26085
   <https://issues.asterisk.org/jira/browse/ASTERISK-26085>] -

app_mp3: results in timeout for streams
(Reported by Jens Bürger)

   - [ASTERISK-26319
   <https://issues.asterisk.org/jira/browse/ASTERISK-26319>] -

[patch] res_pjsip: qualify/unqualify added/deleted realtime endpoints
(Reported by Alexei Gradinari)

   - [ASTERISK-26269
   <https://issues.asterisk.org/jira/browse/ASTERISK-26269>] -

res_pjsip: Wrong state for aors without registered contacts after startup
(Reported by nappsoft)

   - [ASTERISK-26226
   <https://issues.asterisk.org/jira/browse/ASTERISK-26226>] -

pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular
dependency between contexts
(Reported by Etienne Lessard)

   - [ASTERISK-26299
   <https://issues.asterisk.org/jira/browse/ASTERISK-26299>] -

app_queue: Queue application sometimes stops calling members with Local
(Reported by Etienne Lessard)

   - [ASTERISK-26279
   <https://issues.asterisk.org/jira/browse/ASTERISK-26279>] -

pjproject-bundled: Fails to compile on Debian 6
(Reported by George Joseph)

   - [ASTERISK-26306
   <https://issues.asterisk.org/jira/browse/ASTERISK-26306>] -

channel: Hang-up crashes, chan_pjsip not cleaning up properly
(Reported by Alexander Traud)

   - [ASTERISK-26203
   <https://issues.asterisk.org/jira/browse/ASTERISK-26203>] -

res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
(Reported by Etienne Lessard)

   - [ASTERISK-24822
   <https://issues.asterisk.org/jira/browse/ASTERISK-24822>] -

Deadlock: Fax Gateway framehook creates locking inversion in T.38 query
option with features bridging code
(Reported by David Brillert)

   - [ASTERISK-22732
   <https://issues.asterisk.org/jira/browse/ASTERISK-22732>] -

Deadlock potential in res_fax and CCSS with local channels.
(Reported by Richard Mudgett)

   - [ASTERISK-26282
   <https://issues.asterisk.org/jira/browse/ASTERISK-26282>] -

AEL: macro-call in Dial application, macro "lacks 's' extension"
(Reported by chris de rock)

   - [ASTERISK-22820
   <https://issues.asterisk.org/jira/browse/ASTERISK-22820>] -

[patch] Plaintext auth is still supported in IAX2
(Reported by Eugene)

   - [ASTERISK-22374
   <https://issues.asterisk.org/jira/browse/ASTERISK-22374>] -

Finish mapping the sip.conf parameters to res_sip.conf parameters
(Reported by Matt Jordan)

   - [ASTERISK-24425
   <https://issues.asterisk.org/jira/browse/ASTERISK-24425>] -

[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE
(Reported by abelbeck)

   - [ASTERISK-26228
   <https://issues.asterisk.org/jira/browse/ASTERISK-26228>] -

res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
(Reported by Ali Ghavidel)

   - [ASTERISK-25472
   <https://issues.asterisk.org/jira/browse/ASTERISK-25472>] -

Swagger scripts are not replacing format variable in file brief
(Reported by Corey Farrell)

   - [ASTERISK-25984
   <https://issues.asterisk.org/jira/browse/ASTERISK-25984>] -

res_odbc relies on res_odbc_transaction, but it's not mandatory to compile
(Reported by József Dudás)

   - [ASTERISK-26305
   <https://issues.asterisk.org/jira/browse/ASTERISK-26305>] -

Asterisk 14: Two resolver unbound testsuite tests fail
(Reported by Richard Mudgett)

   - [ASTERISK-26288
   <https://issues.asterisk.org/jira/browse/ASTERISK-26288>] -

followme: fails to reset config items to default values on reload
(Reported by Tzafrir Cohen)

   - [ASTERISK-26303
   <https://issues.asterisk.org/jira/browse/ASTERISK-26303>] -

[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
(Reported by Alexander Traud)

   - [ASTERISK-25492
   <https://issues.asterisk.org/jira/browse/ASTERISK-25492>] -

ARI: Path parameters are case sensitive
(Reported by Joshua C. Colp)

   - [ASTERISK-26164
   <https://issues.asterisk.org/jira/browse/ASTERISK-26164>] -

XMPP no longer triggers NOTIFY to device via chan_pjsip
(Reported by Ross Beer)

   - [ASTERISK-26233
   <https://issues.asterisk.org/jira/browse/ASTERISK-26233>] -

pbx: Failure to remove inconsistent extension names
(Reported by Corey Farrell)

   - [ASTERISK-26246
   <https://issues.asterisk.org/jira/browse/ASTERISK-26246>] -

Security: Privilege escalation by AMI adding dialplan extensions.
(Reported by Richard Mudgett)

   - [ASTERISK-26267
   <https://issues.asterisk.org/jira/browse/ASTERISK-26267>] -

ast_register_atexit callbacks should be run on failed startup.
(Reported by Corey Farrell)

   - [ASTERISK-26241
   <https://issues.asterisk.org/jira/browse/ASTERISK-26241>] -

res_pjsip: When using compact headers, rpid and pai are incorrectly
(Reported by George Joseph)

   - [ASTERISK-25797
   <https://issues.asterisk.org/jira/browse/ASTERISK-25797>] -

app_queue: Crash when calling a queue with a member with a forward to an
nonexistent extension
(Reported by Etienne Lessard)

   - [ASTERISK-26239
   <https://issues.asterisk.org/jira/browse/ASTERISK-26239>] -

res_pjsip_logger: An empty global/debug option is treated as a "match all"
(Reported by George Joseph)

   - [ASTERISK-26238
   <https://issues.asterisk.org/jira/browse/ASTERISK-26238>] -

res_pjsip: Empty global default_from_user causes crash
(Reported by Joshua C. Colp)

   - [ASTERISK-26268
   <https://issues.asterisk.org/jira/browse/ASTERISK-26268>] -

alembic: 'auth_username' not in PJSIP 'identify_by' enum
(Reported by Joshua C. Colp)

   - [ASTERISK-26253
   <https://issues.asterisk.org/jira/browse/ASTERISK-26253>] -

sdp_srtp: libsrtp now a required dependency, shouldn't be
(Reported by Ben Merrills)

   - [ASTERISK-26145
   <https://issues.asterisk.org/jira/browse/ASTERISK-26145>] -

pjsip: Deadlock with suspend + masquerade + indicate
(Reported by Ross Beer)

   - [ASTERISK-26183
   <https://issues.asterisk.org/jira/browse/ASTERISK-26183>] -

alembic: error when using sqlalchemy version 1.1.0b2
(Reported by Kevin Harwell)

   - [ASTERISK-26283
   <https://issues.asterisk.org/jira/browse/ASTERISK-26283>] -

res_resolver_unbound: fails configure on older Ubuntu and CentOS
(Reported by George Joseph)

   - [ASTERISK-26280
   <https://issues.asterisk.org/jira/browse/ASTERISK-26280>] -

DNS lookups can block channel media paths
(Reported by Mark Michelson)

   - [ASTERISK-26278
   <https://issues.asterisk.org/jira/browse/ASTERISK-26278>] -

asterisk.h should produce a reasonable error for external modules that fail
(Reported by Corey Farrell)

   - [ASTERISK-25217
   <https://issues.asterisk.org/jira/browse/ASTERISK-25217>] -

[patch]res_pjsip_outbound_publish.c needs a similar treatment for module
unloading as res_pjsip_outbound_registration.c
(Reported by Richard Mudgett)

   - [ASTERISK-26265
   <https://issues.asterisk.org/jira/browse/ASTERISK-26265>] -

Errors ignored from some parts of system initialization.
(Reported by Corey Farrell)

   - [ASTERISK-26206
   <https://issues.asterisk.org/jira/browse/ASTERISK-26206>] -

[patch] res_pjsip: Use more compatible regex for get all
(Reported by Dmitry Wagin)

   - [ASTERISK-26256
   <https://issues.asterisk.org/jira/browse/ASTERISK-26256>] -

[patch] SIP/SDP origin (o=) contains brackets with IP6
(Reported by Alexander Traud)

   - [ASTERISK-25996
   <https://issues.asterisk.org/jira/browse/ASTERISK-25996>] -

Remove "live_dangerously" requirement on DB(read)
(Reported by Andrew Nagy)

   - [ASTERISK-26148
   <https://issues.asterisk.org/jira/browse/ASTERISK-26148>] -

pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference
(Reported by Hans van Eijsden)

   - [ASTERISK-26237
   <https://issues.asterisk.org/jira/browse/ASTERISK-26237>] -

Fax is detected on regular calls.
(Reported by Richard Mudgett)

   - [ASTERISK-26227
   <https://issues.asterisk.org/jira/browse/ASTERISK-26227>] -

sqlalchemy error due to long identifier name
(Reported by Mark Michelson)

   - [ASTERISK-14 <https://issues.asterisk.org/jira/browse/ASTERISK-14>] -

asterisk leaves zombie mpg123
(Reported by dcarr)

   - [ASTERISK-23013
   <https://issues.asterisk.org/jira/browse/ASTERISK-23013>] -

[patch] Deadlock between 'sip show channels' command and attended transfer
(Reported by Ben Smithurst)

   - [ASTERISK-26199
   <https://issues.asterisk.org/jira/browse/ASTERISK-26199>] -

PJSIP: tx_data_destroy called twice
(Reported by Scott Griepentrog)

   - [ASTERISK-26166
   <https://issues.asterisk.org/jira/browse/ASTERISK-26166>] -

res_pjsip_pubsub: Crash when decrementing reference count of message
(Reported by Ross Beer)

   - [ASTERISK-26174
   <https://issues.asterisk.org/jira/browse/ASTERISK-26174>] -

res_pjsip: Crash when freeing cloned message in distributor
(Reported by Ross Beer)

   - [ASTERISK-26216
   <https://issues.asterisk.org/jira/browse/ASTERISK-26216>] -

res_fax: Deadlock when detect fax while channel executing Playback
(Reported by Richard Mudgett)

   - [ASTERISK-26214
   <https://issues.asterisk.org/jira/browse/ASTERISK-26214>] -

Allow arbitrary time for fax detection to end on a channel
(Reported by Richard Mudgett)

   - [ASTERISK-26212
   <https://issues.asterisk.org/jira/browse/ASTERISK-26212>] -

[patch] Makefile: Retain XML Declaration and DTD in docs.
(Reported by Alexander Traud)

   - [ASTERISK-26211
   <https://issues.asterisk.org/jira/browse/ASTERISK-26211>] -

Unit tests: AST_TEST_DEFINE should be used in conditional code.
(Reported by Corey Farrell)

   - [ASTERISK-26200
   <https://issues.asterisk.org/jira/browse/ASTERISK-26200>] -

[patch] res_pjsip_mwi: improve realtime performance - remove unneeded check
on endpoint's contacts.
(Reported by Alexei Gradinari)

   - [ASTERISK-26207
   <https://issues.asterisk.org/jira/browse/ASTERISK-26207>] -

[patch] sRTP: Count a roll-over of the sequence number even on lost packets.
(Reported by Alexander Traud)

   - [ASTERISK-26038
   <https://issues.asterisk.org/jira/browse/ASTERISK-26038>] -

'make install' doesn't seem to install OS/X init files
(Reported by Tzafrir Cohen)

   - [ASTERISK-26133
   <https://issues.asterisk.org/jira/browse/ASTERISK-26133>] -

app_queue: Queue members receive multiple calls
(Reported by Richard Miller)

   - [ASTERISK-26196
   <https://issues.asterisk.org/jira/browse/ASTERISK-26196>] -

pbx: Time based includes can leak timezone string
(Reported by Corey Farrell)

   - [ASTERISK-26193
   <https://issues.asterisk.org/jira/browse/ASTERISK-26193>] -

chan_sip: reference leak in mwi_event_cb
(Reported by Corey Farrell)

   - [ASTERISK-26191
   <https://issues.asterisk.org/jira/browse/ASTERISK-26191>] -

threadpool: Leak on duplicate taskprocessor for
(Reported by Corey Farrell)

   - [ASTERISK-25659
   <https://issues.asterisk.org/jira/browse/ASTERISK-25659>] -

res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP
(Reported by Edwin Vandamme)

   - [ASTERISK-26160
   <https://issues.asterisk.org/jira/browse/ASTERISK-26160>] -

pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)

   - [ASTERISK-26177
   <https://issues.asterisk.org/jira/browse/ASTERISK-26177>] -

func_odbc: Database handle is kept when it should be released
(Reported by Leandro Dardini)

   - [ASTERISK-25289
   <https://issues.asterisk.org/jira/browse/ASTERISK-25289>] -

Build System does not respect CFLAGS and CXXFLAGS when building menuselect
(Reported by Jeffrey Walton)

   - [ASTERISK-26119
   <https://issues.asterisk.org/jira/browse/ASTERISK-26119>] -

[patch] fix: memory leaks, resource leaks, out of bounds and bugs
(Reported by Alexei Gradinari)

   - [ASTERISK-26184
   <https://issues.asterisk.org/jira/browse/ASTERISK-26184>] -

chan_sip: Reference leaks in error paths.
(Reported by Corey Farrell)

   - [ASTERISK-26181
   <https://issues.asterisk.org/jira/browse/ASTERISK-26181>] -

REF_DEBUG: Node object incorrectly logged during duplicate replacement
(Reported by Corey Farrell)

   - [ASTERISK-26172
   <https://issues.asterisk.org/jira/browse/ASTERISK-26172>] -

res_sorcery_realtime: fix bug when successful sql UPDATE is treated as
failed if there is no affected rows.
(Reported by Alexei Gradinari)

   - [ASTERISK-26179
   <https://issues.asterisk.org/jira/browse/ASTERISK-26179>] -

chan_sip: Second T.38 request fails
(Reported by Joshua C. Colp)

   - [ASTERISK-26180
   <https://issues.asterisk.org/jira/browse/ASTERISK-26180>] -

PJSIP: provide valid tcp nodelay option for reuse
(Reported by Scott Griepentrog)

   - [ASTERISK-25772
   <https://issues.asterisk.org/jira/browse/ASTERISK-25772>] -

res_pjsip: Unexpected two BYE when answered
(Reported by Dmitriy Serov)

   - [ASTERISK-26099
   <https://issues.asterisk.org/jira/browse/ASTERISK-26099>] -

res_pjsip_pubsub: Crash when sending request due to server timeout
(Reported by Ross Beer)

   - [ASTERISK-26144
   <https://issues.asterisk.org/jira/browse/ASTERISK-26144>] -

Crash on loading codecs g729/g723
(Reported by Alexei Gradinari)

   - [ASTERISK-26157
   <https://issues.asterisk.org/jira/browse/ASTERISK-26157>] -

Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)

   - [ASTERISK-26021
   <https://issues.asterisk.org/jira/browse/ASTERISK-26021>] -

Build codecs siren7 and siren14 for Asterisk 13
(Reported by Daniel Denson)

   - [ASTERISK-26141
   <https://issues.asterisk.org/jira/browse/ASTERISK-26141>] -

res_fax: fax_v21_session_new leaks reference to v21_details
(Reported by Corey Farrell)

   - [ASTERISK-26061
   <https://issues.asterisk.org/jira/browse/ASTERISK-26061>] -

[patch] res_pjsip: improve realtime performance - remove updating all
endpoints status on startup
(Reported by Alexei Gradinari)

   - [ASTERISK-26140
   <https://issues.asterisk.org/jira/browse/ASTERISK-26140>] -

res_rtp_asterisk: gcc 6 caught a self-comparison
(Reported by George Joseph)

   - [ASTERISK-26138
   <https://issues.asterisk.org/jira/browse/ASTERISK-26138>] -

chan_unistim: Under FreeBSD, chan_unistim generates a compile error
(Reported by George Joseph)

   - [ASTERISK-26128
   <https://issues.asterisk.org/jira/browse/ASTERISK-26128>] -

Alembic scripts are failing
(Reported by Mark Michelson)

   - [ASTERISK-26139
   <https://issues.asterisk.org/jira/browse/ASTERISK-26139>] -

test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a
system location
(Reported by George Joseph)

   - [ASTERISK-26129
   <https://issues.asterisk.org/jira/browse/ASTERISK-26129>] -

res_rtp_asterisk: Memory leak of CERT bio in DTLS implementation
(Reported by Torrey Searle)

   - [ASTERISK-26130
   <https://issues.asterisk.org/jira/browse/ASTERISK-26130>] -

[patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)

   - [ASTERISK-26132
   <https://issues.asterisk.org/jira/browse/ASTERISK-26132>] -

PJSIP: provide transport type with received messages
(Reported by Scott Griepentrog)

   - [ASTERISK-26127
   <https://issues.asterisk.org/jira/browse/ASTERISK-26127>] -

res_pjsip_session: Crash due to race condition between res_pjsip_session
unload and timer
(Reported by Joshua C. Colp)

   - [ASTERISK-26045
   <https://issues.asterisk.org/jira/browse/ASTERISK-26045>] -

[patch]app_voicemail: fix bugs, imap mm_status log change to debug
(Reported by Alexei Gradinari)

   - [ASTERISK-26083
   <https://issues.asterisk.org/jira/browse/ASTERISK-26083>] -

ARI: Announcer channels staying around after playback to a bridge is
(Reported by Per Jensen)

   - [ASTERISK-26126
   <https://issues.asterisk.org/jira/browse/ASTERISK-26126>] -

[patch] leverage 'bindaddr' for TLS in http.conf
(Reported by Alexander Traud)

   - [ASTERISK-26097
   <https://issues.asterisk.org/jira/browse/ASTERISK-26097>] -

[patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)

   - [ASTERISK-25262
   <https://issues.asterisk.org/jira/browse/ASTERISK-25262>] -

Memory leak when a caller channel does multiple dials and CEL is enabled
(Reported by Etienne Lessard)

   - [ASTERISK-26047
   <https://issues.asterisk.org/jira/browse/ASTERISK-26047>] -

ARI allows certain commands to run on down channels.
(Reported by Mark Michelson)

   - [ASTERISK-25959
   <https://issues.asterisk.org/jira/browse/ASTERISK-25959>] -

http_media_cache/retrieve_cache_control_directives: Sporadic failure
(Reported by Joshua C. Colp)

   - [ASTERISK-26103
   <https://issues.asterisk.org/jira/browse/ASTERISK-26103>] -

cdr: Assert on 'dial end' event during a blond transfer
(Reported by George Joseph)

   - [ASTERISK-26092
   <https://issues.asterisk.org/jira/browse/ASTERISK-26092>] -

[Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
(Reported by Niklas Larsson)

   - [ASTERISK-26089
   <https://issues.asterisk.org/jira/browse/ASTERISK-26089>] -

Invalid security events during boot using PJSIP Realtime
(Reported by Scott Griepentrog)

   - [ASTERISK-26096
   <https://issues.asterisk.org/jira/browse/ASTERISK-26096>] -

res_hep: Crash when configuration file is missing
(Reported by Niklas Larsson)

   - [ASTERISK-26074
   <https://issues.asterisk.org/jira/browse/ASTERISK-26074>] -

res_odbc: Deadlock within UnixODBC
(Reported by Ross Beer)

   - [ASTERISK-26054
   <https://issues.asterisk.org/jira/browse/ASTERISK-26054>] -

Asterisk crashes (core dump)
(Reported by B. Davis)

   - [ASTERISK-26069
   <https://issues.asterisk.org/jira/browse/ASTERISK-26069>] -

Asterisk truncates To: header, dropping the closing '>'
(Reported by Vasil Kolev)

   - [ASTERISK-24436
   <https://issues.asterisk.org/jira/browse/ASTERISK-24436>] -

Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
(Reported by Patrick Laimbock)

   - [ASTERISK-26091
   <https://issues.asterisk.org/jira/browse/ASTERISK-26091>] -

[patch] ar cru creates warning, instead use ar cr
(Reported by Alexander Traud)

   - [ASTERISK-26070
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-26078>] -

core: Memory leak in logging
(Reported by Etienne Lessard)

   - [ASTERISK-26065
   <https://issues.asterisk.org/jira/browse/ASTERISK-26065>] -

chan_pjsip: MWI NOTIFY contents not ordered properly
(Reported by Ross Beer)

   - [ASTERISK-26063
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25777>] -

data race in threadpool
(Reported by Badalian Vyacheslav)

   - [ASTERISK-26053
   <https://issues.asterisk.org/jira/browse/ASTERISK-26053>] -

res_pjsip_outbound_publish: Crash when shutting down
(Reported by Joshua C. Colp)

   - [ASTERISK-26049
   <https://issues.asterisk.org/jira/browse/ASTERISK-26049>] -

res_pjsip: Crash when our own request timer fires
(Reported by Joshua C. Colp)

   - [ASTERISK-25669
   <https://issues.asterisk.org/jira/browse/ASTERISK-25669>] -

[patch]CURL incorrect trim for non ASCII characters
(Reported by Jesper)

   - [ASTERISK-26029
   <https://issues.asterisk.org/jira/browse/ASTERISK-26029>] -

parking: ast_parking_park_call should return parking_space instead of
(Reported by Diederik de Groot)

   - [ASTERISK-25938
   <https://issues.asterisk.org/jira/browse/ASTERISK-25938>] -

res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
(Reported by Edwin Vandamme)

   - [ASTERISK-25941
   <https://issues.asterisk.org/jira/browse/ASTERISK-25941>] -

chan_pjsip: Crash on an immediate SIP final response
(Reported by Javier Riveros )

   - [ASTERISK-26014
   <https://issues.asterisk.org/jira/browse/ASTERISK-26014>] -

res_sorcery_astdb: Make tolerant of unknown fields
(Reported by Joshua C. Colp)

   - [ASTERISK-24986
   <https://issues.asterisk.org/jira/browse/ASTERISK-24986>] -

keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)

   - [ASTERISK-26034
   <https://issues.asterisk.org/jira/browse/ASTERISK-26034>] -

T.38 passthrough problem behind firewall due to early nosignal packet
(Reported by George Joseph)

   - [ASTERISK-26030
   <https://issues.asterisk.org/jira/browse/ASTERISK-26030>] -

call cut because of double Session-Expires header in re-invite after proxy
authentication is required
(Reported by George Joseph)

   - [ASTERISK-25964
   <https://issues.asterisk.org/jira/browse/ASTERISK-25964>] -

Outbound registrations created via ARI/push configuration do not clean up
outbound registrations currently in flight
(Reported by Matt Jordan)

   - [ASTERISK-26005
   <https://issues.asterisk.org/jira/browse/ASTERISK-26005>] -

res_pjsip: Multiple SIP messages are combined into 1 TCP packet
(Reported by Ross Beer)

   - [ASTERISK-25352
   <https://issues.asterisk.org/jira/browse/ASTERISK-25352>] -

res_hep_rtcp correlation_id is different then res_hep
(Reported by Kevin Scott Adams)

   - [ASTERISK-26007
   <https://issues.asterisk.org/jira/browse/ASTERISK-26007>] -

res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
(Reported by Greg Siemon)

   - [ASTERISK-25990
   <https://issues.asterisk.org/jira/browse/ASTERISK-25990>] -

PJSIP TLS registration should respect client_uri scheme when generating
Contact URI
(Reported by Sebastian Damm)

   - [ASTERISK-25538
   <https://issues.asterisk.org/jira/browse/ASTERISK-25538>] -

[patch]Missing PID in syslog logger messages
(Reported by Javier Acosta)

   - [ASTERISK-26008
   <https://issues.asterisk.org/jira/browse/ASTERISK-26008>] -

app_followme does not delete recorded name prompt
(Reported by Tzafrir Cohen)

   - [ASTERISK-25978
   <https://issues.asterisk.org/jira/browse/ASTERISK-25978>] -

res_pjsip_authenticator_digest: Should not use source port in nonce
(Reported by Mark Michelson)

   - [ASTERISK-26004
   <https://issues.asterisk.org/jira/browse/ASTERISK-26004>] -

res_pjsip: The transport/method parameter is ignored
(Reported by George Joseph)

   - [ASTERISK-25999
   <https://issues.asterisk.org/jira/browse/ASTERISK-25999>] -

res_pjsip_dialog_info_body_generator: Remove subscription requirement
(Reported by Joshua C. Colp)

   - [ASTERISK-25993
   <https://issues.asterisk.org/jira/browse/ASTERISK-25993>] -

pjproject: Allow bundling to not require everything it does
(Reported by Joshua C. Colp)

   - [ASTERISK-25998
   <https://issues.asterisk.org/jira/browse/ASTERISK-25998>] -

file: Crash when using nativeformats
(Reported by Joshua C. Colp)

   - [ASTERISK-25826
   <https://issues.asterisk.org/jira/browse/ASTERISK-25826>] -

PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)

   - [ASTERISK-25982
   <https://issues.asterisk.org/jira/browse/ASTERISK-25982>] -

[patch]res_fax/t38_gateway: Peer V.21 session is created on wrong channel
(Reported by Alexei Gradinari)

   - [ASTERISK-25956
   <https://issues.asterisk.org/jira/browse/ASTERISK-25956>] -

Compilation error in conditionally compiled code in config_options.c
(Reported by Chris Trobridge)

   - [ASTERISK-25968
   <https://issues.asterisk.org/jira/browse/ASTERISK-25968>] -

pjproject_bundled: Configure and make need to be re-tested
(Reported by George Joseph)

   - [ASTERISK-24463
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25922>] -

res_pjsip_exten_state: Add configuration support for publishing
(Reported by Joshua C. Colp)

   - [ASTERISK-25970
   <https://issues.asterisk.org/jira/browse/ASTERISK-25970>] -

Segfault in pjsip_url_compare
(Reported by Dmitriy Serov)

   - [ASTERISK-25963
   <https://issues.asterisk.org/jira/browse/ASTERISK-25963>] -

func_odbc requires reconnect checks for stale connections
(Reported by Ross Beer)

   - [ASTERISK-25961
   <https://issues.asterisk.org/jira/browse/ASTERISK-25961>] -

tests/channels/SIP/sip_tls_call: Sporadic crash when running test
(Reported by Joshua C. Colp)

   - [ASTERISK-16115
   <https://issues.asterisk.org/jira/browse/ASTERISK-16115>] -

[patch] problem with ringinuse=no, queue members receive sometimes two calls
(Reported by nik600)

   - [ASTERISK-25917
   <https://issues.asterisk.org/jira/browse/ASTERISK-25917>] -

[patch]app_voicemail: passwordlocation=spooldir only works if you manually
add secret.conf yourself
(Reported by Jonathan R. Rose)

   - [ASTERISK-25954
   <https://issues.asterisk.org/jira/browse/ASTERISK-25954>] -

Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
(Reported by Javier Acosta)

   - [ASTERISK-25950
   <https://issues.asterisk.org/jira/browse/ASTERISK-25950>] -

[patch]SIP channel does not send PeerStatus events for autocreated peers
(Reported by Kirill Katsnelson)

   - [ASTERISK-25927
   <https://issues.asterisk.org/jira/browse/ASTERISK-25927>] -

Removed option "registertrying" is still documented in sip.conf.sample
(Reported by Etienne Lessard)

   - [ASTERISK-25948
   <https://issues.asterisk.org/jira/browse/ASTERISK-25948>] -

ast_pthread_mutex_lock calling ast_reentrancy_lock with lt=0x0
(Reported by Diederik de Groot)

   - [ASTERISK-25947
   <https://issues.asterisk.org/jira/browse/ASTERISK-25947>] -

Protocol transfers to stasis applications are missing the StasisStart with
the replace_channel object.
(Reported by Richard Mudgett)

   - [ASTERISK-24649
   <https://issues.asterisk.org/jira/browse/ASTERISK-24649>] -

Pushing of channel into bridge fails; Stasis fails to get app name
(Reported by John Bigelow)

   - [ASTERISK-24782
   <https://issues.asterisk.org/jira/browse/ASTERISK-24782>] -

StasisEnd event not present for channel that was swapped out for another
after completing attended transfer
(Reported by John Bigelow)

   - [ASTERISK-25942
   <https://issues.asterisk.org/jira/browse/ASTERISK-25942>] -

res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
(Reported by George Joseph)

   - [ASTERISK-25928
   <https://issues.asterisk.org/jira/browse/ASTERISK-25928>] -

res_pjsip: URI validation done outside of PJSIP thread
(Reported by Joshua C. Colp)

   - [ASTERISK-25929
   <https://issues.asterisk.org/jira/browse/ASTERISK-25929>] -

res_pjsip_registrar: AOR_CONTACT_ADDED events not raised
(Reported by Joshua C. Colp)

   - [ASTERISK-25934
   <https://issues.asterisk.org/jira/browse/ASTERISK-25934>] -

chan_sip should not require sipregs or updateable sippeers table unless rt
(Reported by Jaco Kroon)

   - [ASTERISK-25888
   <https://issues.asterisk.org/jira/browse/ASTERISK-25888>] -

Frequent segfaults in function can_ring_entry() of app_queue.c
(Reported by Sébastien Couture)

   - [ASTERISK-25914
   <https://issues.asterisk.org/jira/browse/ASTERISK-25914>] -

PJSIP: failed registration with wrong codec name on allow/disallow
(Reported by Alexei Gradinari)

   - [ASTERISK-25796
   <https://issues.asterisk.org/jira/browse/ASTERISK-25796>] -

res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject
(Reported by George Joseph)

   - [ASTERISK-25707
   <https://issues.asterisk.org/jira/browse/ASTERISK-25707>] -

Long contact URIs or hostnames can crash pjproject/Asterisk under certain
(Reported by George Joseph)

   - [ASTERISK-25123
   <https://issues.asterisk.org/jira/browse/ASTERISK-25123>] -

Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP
(Reported by Anthony Messina)

   - [ASTERISK-25874
   <https://issues.asterisk.org/jira/browse/ASTERISK-25874>] -

app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24927
   <https://issues.asterisk.org/jira/browse/ASTERISK-24927>] -

app_voicemail (IMAP support) function save_to_folder: creates wrong folder
(Reported by Alexei Gradinari)

   - [ASTERISK-25912
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25885>] -

res_pjsip: Race condition between adding contact and automatic expiration
(Reported by Joshua C. Colp)

   - [ASTERISK-25910
   <https://issues.asterisk.org/jira/browse/ASTERISK-25910>] -

pjproject: Via headers are not parsed when "received" contains an IPv6
(Reported by George Joseph)

   - [ASTERISK-25899
   <https://issues.asterisk.org/jira/browse/ASTERISK-25899>] -

IMAP access FATAL error: Out of memory
(Reported by Alexei Gradinari)

   - [ASTERISK-25890
   <https://issues.asterisk.org/jira/browse/ASTERISK-25890>] -

Asterisk 13.8.0 alembic database update fails
(Reported by Harley Peters)

   - [ASTERISK-25894
   <https://issues.asterisk.org/jira/browse/ASTERISK-25894>] -

[patch] webrtc video broken due to missing marker bits in RTP streams
(Reported by Jacek Konieczny)

   - [ASTERISK-25881
   <https://issues.asterisk.org/jira/browse/ASTERISK-25881>] -

pbx: Add support for autohints
(Reported by Joshua C. Colp)

   - [ASTERISK-25854
   <https://issues.asterisk.org/jira/browse/ASTERISK-25854>] -

No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk
(Reported by Robert McGilvray)

   - [ASTERISK-25868
   <https://issues.asterisk.org/jira/browse/ASTERISK-25868>] -

Sorcery "append to category" should allow filters
(Reported by Nick Repin)

   - [ASTERISK-25873
   <https://issues.asterisk.org/jira/browse/ASTERISK-25873>] -

res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj
(Reported by Hans van Eijsden)

   - [ASTERISK-25882
   <https://issues.asterisk.org/jira/browse/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-25642
   <https://issues.asterisk.org/jira/browse/ASTERISK-25642>] -

res_rtp_asterisk: SRTCP broken with DTLS - bad video is one of the
(Reported by Stefan Engström)

   - [ASTERISK-25867
   <https://issues.asterisk.org/jira/browse/ASTERISK-25867>] -

[patch] Video delay on app_echo
(Reported by Jacek Konieczny)

   - [ASTERISK-24605
   <https://issues.asterisk.org/jira/browse/ASTERISK-24605>] -

res_parking option parkeddynamic does not work with the core Features
'parkcall' (DTMF initiated parking)
(Reported by Philip Correia)

   - [ASTERISK-24596
   <https://issues.asterisk.org/jira/browse/ASTERISK-24596>] -

Unclear how to use Park application with res_parking 'parkeddynamic'
enabled. Documentation?
(Reported by Philip Correia)

   - [ASTERISK-25825
   <https://issues.asterisk.org/jira/browse/ASTERISK-25825>] -

Crashes during shutdown when running CLI commands
(Reported by Mark Michelson)

   - [ASTERISK-24543
   <https://issues.asterisk.org/jira/browse/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-25612
   <https://issues.asterisk.org/jira/browse/ASTERISK-25612>] -

Configuration parser handles unsigned integers as signed integers
(Reported by Gianluca Merlo)

   - [ASTERISK-25407
   <https://issues.asterisk.org/jira/browse/ASTERISK-25407>] -

Asterisk fails to log to multiple syslog destinations
(Reported by Elazar Broad)

   - [ASTERISK-25510
   <https://issues.asterisk.org/jira/browse/ASTERISK-25510>] -

[patch]Log to syslog failing
(Reported by Michael Newton)

   - [ASTERISK-21301
   <https://issues.asterisk.org/jira/browse/ASTERISK-21301>] -

ERROR and failure to resolve socket address due to whitespace after port
number in SIP Via header
(Reported by Martin Vit)

   - [ASTERISK-25857
   <https://issues.asterisk.org/jira/browse/ASTERISK-25857>] -

func_aes: incorrect use of strlen() leads to data corruption
(Reported by Gianluca Merlo)

   - [ASTERISK-25849
   <https://issues.asterisk.org/jira/browse/ASTERISK-25849>] -

chan_pjsip: transfers with direct media sometimes drops audio
(Reported by Kevin Harwell)

   - [ASTERISK-25814
   <https://issues.asterisk.org/jira/browse/ASTERISK-25814>] -

Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)

   - [ASTERISK-25023
   <https://issues.asterisk.org/jira/browse/ASTERISK-25023>] -

Deadlock in chan_sip in update_provisional_keepalive
(Reported by Arnd Schmitter)

   - [ASTERISK-25321
   <https://issues.asterisk.org/jira/browse/ASTERISK-25321>] -

[patch]DeadLock ChanSpy with call over Local channel
(Reported by Filip Frank)

   - [ASTERISK-25829
   <https://issues.asterisk.org/jira/browse/ASTERISK-25829>] -

res_pjsip: PJSIP does not accept spaces when separating multiple AORs
(Reported by Mateusz Kowalski)

   - [ASTERISK-25771
   <https://issues.asterisk.org/jira/browse/ASTERISK-25771>] -

ARI:Crash - Attended transfers of channels into Stasis application.
(Reported by Javier Riveros )

   - [ASTERISK-25830
   <https://issues.asterisk.org/jira/browse/ASTERISK-25830>] -

Revision 2451d4e breaks NAT
(Reported by Sean Bright)

   - [ASTERISK-25582
   <https://issues.asterisk.org/jira/browse/ASTERISK-25582>] -

Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
(Reported by Matt Jordan)

   - [ASTERISK-25811
   <https://issues.asterisk.org/jira/browse/ASTERISK-25811>] -

Unable to delete object from sorcery cache
(Reported by Ross Beer)

   - [ASTERISK-25800
   <https://issues.asterisk.org/jira/browse/ASTERISK-25800>] -

[patch] Calculate talktime when is first call answered
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25727
   <https://issues.asterisk.org/jira/browse/ASTERISK-25727>] -

RPM build requires OPTIONAL_API cflag due to PJSIP requirement
(Reported by Gergely Dömsödi)

   - [ASTERISK-25337
   <https://issues.asterisk.org/jira/browse/ASTERISK-25337>] -

Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
(Reported by Jacques Peacock)

   - [ASTERISK-25738
   <https://issues.asterisk.org/jira/browse/ASTERISK-25738>] -

res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami
(Reported by Kevin Harwell)

   - [ASTERISK-25721
   <https://issues.asterisk.org/jira/browse/ASTERISK-25721>] -

[patch] res_phoneprov: memory leak and heap-use-after-free
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25272
   <https://issues.asterisk.org/jira/browse/ASTERISK-25272>] -

[patch]The ICONV dialplan function sometimes returns garbage
(Reported by Etienne Lessard)

   - [ASTERISK-25751
   <https://issues.asterisk.org/jira/browse/ASTERISK-25751>] -

res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock
(Reported by Joshua C. Colp)

   - [ASTERISK-25606
   <https://issues.asterisk.org/jira/browse/ASTERISK-25606>] -

Core dump when using transports in sorcery
(Reported by Martin Moučka)

   - [ASTERISK-20987
   <https://issues.asterisk.org/jira/browse/ASTERISK-20987>] -

non-admin users, who join muted conference are not being muted
(Reported by hristo)

   - [ASTERISK-25737
   <https://issues.asterisk.org/jira/browse/ASTERISK-25737>] -

res_pjsip_outbound_registration: line option not in Alembic
(Reported by Joshua C. Colp)

   - [ASTERISK-24972
   <https://issues.asterisk.org/jira/browse/ASTERISK-24972>] -

Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate
vulnerability of HTTP server
(Reported by Alex A. Welzl)

   - [ASTERISK-25603
   <https://issues.asterisk.org/jira/browse/ASTERISK-25603>] -

[patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause
ast_frdup crash
(Reported by Walter Doekes)

   - [ASTERISK-25742
   <https://issues.asterisk.org/jira/browse/ASTERISK-25742>] -

Secondary IFP Packets can result in accessing uninitialized pointers and a
(Reported by Torrey Searle)

   - [ASTERISK-25397
   <https://issues.asterisk.org/jira/browse/ASTERISK-25397>] -

[patch]chan_sip: File descriptor leak with non-default timert1
(Reported by Alexander Traud)

   - [ASTERISK-25702
   <https://issues.asterisk.org/jira/browse/ASTERISK-25702>] -

PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from
(Reported by Nic Colledge)

   - [ASTERISK-25735
   <https://issues.asterisk.org/jira/browse/ASTERISK-25735>] -

[patch] res_xmpp: Does not connect in component mode
(Reported by Karsten Wemheuer)

   - [ASTERISK-25730
   <https://issues.asterisk.org/jira/browse/ASTERISK-25730>] -

build: make uninstall after make distclean tries to remove root
(Reported by George Joseph)

   - [ASTERISK-25725
   <https://issues.asterisk.org/jira/browse/ASTERISK-25725>] -

core: Incorrect XML documentation may result in weird behavior
(Reported by Joshua C. Colp)

   - [ASTERISK-25722
   <https://issues.asterisk.org/jira/browse/ASTERISK-25722>] -

ASAN & testsute: stack-buffer-overflow in sip_sipredirect
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25709
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25714>] -

ASAN:heap-buffer-overflow in logger.c
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25685
   <https://issues.asterisk.org/jira/browse/ASTERISK-25685>] -

infrastructure: Run alembic in Jenkins build script
(Reported by Joshua C. Colp)

   - [ASTERISK-25712
   <https://issues.asterisk.org/jira/browse/ASTERISK-25712>] -

Second call to already-on-call phone and Asterisk sends "Ready"
(Reported by Richard Mudgett)

   - [ASTERISK-24801
   <https://issues.asterisk.org/jira/browse/ASTERISK-24801>] -

ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25179
   <https://issues.asterisk.org/jira/browse/ASTERISK-25179>] -

CDR(billsec,f) and CDR(duration,f) report incorrect values
(Reported by Gianluca Merlo)

   - [ASTERISK-25611
   <https://issues.asterisk.org/jira/browse/ASTERISK-25611>] -

core: threadpool thread_timeout_thrash unit test sporadically failing
(Reported by Joshua C. Colp)

   - [ASTERISK-24833
   <https://issues.asterisk.org/jira/browse/ASTERISK-24833>] -

[patch] audit of startup order reveals logger concerns
(Reported by Corey Farrell)

   - [ASTERISK-25732
   <https://issues.asterisk.org/jira/browse/ASTERISK-25732>] -

[patch] persist queue member pause reason through restart
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25686
   <https://issues.asterisk.org/jira/browse/ASTERISK-25686>] -

PJSIP: qualify_timeout is a double, database schema is an integer
(Reported by Marcelo Terres)

   - [ASTERISK-25700
   <https://issues.asterisk.org/jira/browse/ASTERISK-25700>] -

main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)

   - [ASTERISK-25696
   <https://issues.asterisk.org/jira/browse/ASTERISK-25696>] -

bridge_basic: don't cache xferfailsound during a transfer
(Reported by Kevin Harwell)

   - [ASTERISK-25697
   <https://issues.asterisk.org/jira/browse/ASTERISK-25697>] -

bridge_basic: don't play an attended transfer fail sound after target hangs
(Reported by Kevin Harwell)

   - [ASTERISK-25683
   <https://issues.asterisk.org/jira/browse/ASTERISK-25683>] -

res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG
(Reported by yaron nahum)

   - [ASTERISK-24097
   <https://issues.asterisk.org/jira/browse/ASTERISK-24097>] -

Documentation - CHANNEL function help text missing 'linkedid' argument
(Reported by Steven Wheeler)

   - [ASTERISK-25690
   <https://issues.asterisk.org/jira/browse/ASTERISK-25690>] -

Hanging up when executing connected line sub does not cause hangup
(Reported by Joshua C. Colp)

   - [ASTERISK-25687
   <https://issues.asterisk.org/jira/browse/ASTERISK-25687>] -

res_musiconhold: Concurrent invocations of 'moh reload' cause a crash
(Reported by Sean Bright)

   - [ASTERISK-25632
   <https://issues.asterisk.org/jira/browse/ASTERISK-25632>] -

res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed
(Reported by Olivier Krief)

   - [ASTERISK-25637
   <https://issues.asterisk.org/jira/browse/ASTERISK-25637>] -

Multi homed server using wrong IP
(Reported by Daniel Journo)

   - [ASTERISK-25394
   <https://issues.asterisk.org/jira/browse/ASTERISK-25394>] -

pbx: Incorrect device and presence state when changing hint details
(Reported by Joshua C. Colp)

   - [ASTERISK-25640
   <https://issues.asterisk.org/jira/browse/ASTERISK-25640>] -

pbx: Deadlock on features reload and state change hint.
(Reported by Krzysztof Trempala)

   - [ASTERISK-25681
   <https://issues.asterisk.org/jira/browse/ASTERISK-25681>] -

devicestate: Engine thread is not shut down
(Reported by Corey Farrell)

   - [ASTERISK-25680
   <https://issues.asterisk.org/jira/browse/ASTERISK-25680>] -

manager: manager_channelvars is not cleaned at shutdown
(Reported by Corey Farrell)

   - [ASTERISK-25679
   <https://issues.asterisk.org/jira/browse/ASTERISK-25679>] -

res_calendar leaks scheduler.
(Reported by Corey Farrell)

   - [ASTERISK-25645
   <https://issues.asterisk.org/jira/browse/ASTERISK-25645>] -

res_rtp_asterisk: Lock inversion
(Reported by Steve Davies)

   - [ASTERISK-25675
   <https://issues.asterisk.org/jira/browse/ASTERISK-25675>] -

Endpoint not listed as Unreachable
(Reported by Daniel Journo)

   - [ASTERISK-25677
   <https://issues.asterisk.org/jira/browse/ASTERISK-25677>] -

pbx_dundi: leaks during failed load.
(Reported by Corey Farrell)

   - [ASTERISK-25673
   <https://issues.asterisk.org/jira/browse/ASTERISK-25673>] -

res_crypto leaks CLI entries
(Reported by Corey Farrell)

   - [ASTERISK-25668
   <https://issues.asterisk.org/jira/browse/ASTERISK-25668>] -

res_pjsip: Deadlock in distributor
(Reported by Mark Michelson)

   - [ASTERISK-25664
   <https://issues.asterisk.org/jira/browse/ASTERISK-25664>] -

ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)

   - [ASTERISK-25647
   <https://issues.asterisk.org/jira/browse/ASTERISK-25647>] -

bug of cel_radius.c: wrong point of ADD_VENDOR_CODE
(Reported by Aaron An)

   - [ASTERISK-19820
   <https://issues.asterisk.org/jira/browse/ASTERISK-19820>] -

wrapuptime is intermittently disregarded for queue calls
(Reported by WRP)

   - [ASTERISK-25307
   <https://issues.asterisk.org/jira/browse/ASTERISK-25307>] -

Hangup on channel using FastAGI does not hang up child channels
(Reported by David Cunningham)

   - [ASTERISK-25458
   <https://issues.asterisk.org/jira/browse/ASTERISK-25458>] -

Unable to set CDR variable in h extension or hangup_handler
(Reported by Ross Beer)

   - [ASTERISK-25137
   <https://issues.asterisk.org/jira/browse/ASTERISK-25137>] -

endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)

   - [ASTERISK-25116
   <https://issues.asterisk.org/jira/browse/ASTERISK-25116>] -

res_pjsip: Two PeerStatus AMI messages are sent for every status change
(Reported by George Joseph)

   - [ASTERISK-25641
   <https://issues.asterisk.org/jira/browse/ASTERISK-25641>] -

bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel
(Reported by Dmitry Melekhov)

   - [ASTERISK-25639
   <https://issues.asterisk.org/jira/browse/ASTERISK-25639>] -

app_amd: system maxwords discrepency
(Reported by Dade Brandon)

   - [ASTERISK-25614
   <https://issues.asterisk.org/jira/browse/ASTERISK-25614>] -

DTLS negotiation delays
(Reported by Dade Brandon)

   - [ASTERISK-25625
   <https://issues.asterisk.org/jira/browse/ASTERISK-25625>] -

res_sorcery_memory_cache: Add full backend caching
(Reported by Joshua C. Colp)

   - [ASTERISK-25601
   <https://issues.asterisk.org/jira/browse/ASTERISK-25601>] -

json: Audit reference usage and thread safety
(Reported by Joshua C. Colp)

   - [ASTERISK-25624
   <https://issues.asterisk.org/jira/browse/ASTERISK-25624>] -

AMI Event OriginateResponse bug
(Reported by sungtae kim)

   - [ASTERISK-25615
   <https://issues.asterisk.org/jira/browse/ASTERISK-25615>] -

res_pjsip: Setting transport async_operations > 1 causes segfault on tls
(Reported by George Joseph)

   - [ASTERISK-25442
   <https://issues.asterisk.org/jira/browse/ASTERISK-25442>] -

using realtime (mysql) queue members are never updated in wait_our_turn
function (app_queue.c)
(Reported by Carlos Oliva)

   - [ASTERISK-25364
   <https://issues.asterisk.org/jira/browse/ASTERISK-25364>] -

[patch]Issue a TCP connection(kernel) and thread of asterisk is not released
(Reported by Hiroaki Komatsu)

   - [ASTERISK-25569
   <https://issues.asterisk.org/jira/browse/ASTERISK-25569>] -

app_meetme: Audio quality issues
(Reported by Corey Farrell)

   - [ASTERISK-25619
   <https://issues.asterisk.org/jira/browse/ASTERISK-25619>] -

res_chan_stats not sending the correct information to StatsD
(Reported by Tyler Cambron)

   - [ASTERISK-24146
   <https://issues.asterisk.org/jira/browse/ASTERISK-24146>] -

[patch]No audio on WebRtc caller side when answer waiting time is more than
(Reported by Aleksei Kulakov)

   - [ASTERISK-25609
   <https://issues.asterisk.org/jira/browse/ASTERISK-25609>] -

[patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
(Reported by Filip Jenicek)

   - [ASTERISK-25599
   <https://issues.asterisk.org/jira/browse/ASTERISK-25599>] -

[patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)

   - [ASTERISK-25616
   <https://issues.asterisk.org/jira/browse/ASTERISK-25616>] -

Warning with a Codec Module which supports PLC with FEC
(Reported by Alexander Traud)

   - [ASTERISK-25610
   <https://issues.asterisk.org/jira/browse/ASTERISK-25610>] -

Asterisk crash during "sip reload"
(Reported by Dudás József)

   - [ASTERISK-25608
   <https://issues.asterisk.org/jira/browse/ASTERISK-25608>] -

res_pjsip/contacts/statsd: Lifecycle events aren't consistent
(Reported by George Joseph)

   - [ASTERISK-25584
   <https://issues.asterisk.org/jira/browse/ASTERISK-25584>] -

[patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)

   - [ASTERISK-25583
   <https://issues.asterisk.org/jira/browse/ASTERISK-25583>] -

[patch] format-attribute module: RFC 7587 (Opus Codec)
(Reported by Alexander Traud)

   - [ASTERISK-25498
   <https://issues.asterisk.org/jira/browse/ASTERISK-25498>] -

Asterisk crashes when negotiating g729 without that module installed
(Reported by Ben Langfeld)

   - [ASTERISK-25595
   <https://issues.asterisk.org/jira/browse/ASTERISK-25595>] -

Unescaped : in messge sent to statsd
(Reported by Niklas Larsson)

   - [ASTERISK-25598
   <https://issues.asterisk.org/jira/browse/ASTERISK-25598>] -

res_pjsip: Contact status messages are printing a hash instead of the uri
(Reported by George Joseph)

   - [ASTERISK-25600
   <https://issues.asterisk.org/jira/browse/ASTERISK-25600>] -

bridging: Inconsistency in BRIDGEPEER
(Reported by Jonathan Rose)

   - [ASTERISK-25476
   <https://issues.asterisk.org/jira/browse/ASTERISK-25476>] -

chan_sip loses registrations after a while
(Reported by Michael Keuter)

   - [ASTERISK-25593
   <https://issues.asterisk.org/jira/browse/ASTERISK-25593>] -

fastagi: record file closed after sending result
(Reported by Kevin Harwell)

   - [ASTERISK-25585
   <https://issues.asterisk.org/jira/browse/ASTERISK-25585>] -

[patch]rasterisk never hits most of main(), but it's assumed to
(Reported by Walter Doekes)

   - [ASTERISK-25590
   <https://issues.asterisk.org/jira/browse/ASTERISK-25590>] -

CLI Usage info for 'pjsip send notify' references incorrect config
(Reported by Corey Farrell)

   - [ASTERISK-25165
   <https://issues.asterisk.org/jira/browse/ASTERISK-25165>] -

Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)

   - [ASTERISK-25575
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25545>] -

[patch] translation module gets cached not joint format
(Reported by Alexander Traud)

   - [ASTERISK-25573
   <https://issues.asterisk.org/jira/browse/ASTERISK-25573>] -

[patch] H.264 format attribute module: resets whole SDP
(Reported by Alexander Traud)

   - [ASTERISK-24958
   <https://issues.asterisk.org/jira/browse/ASTERISK-24958>] -

Forwarding loop detection inhibits certain desirable scenarios
(Reported by Mark Michelson)

   - [ASTERISK-25561
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25565>] -

DNS: System resolver only returns 1 record per result
(Reported by George Joseph)

   - [ASTERISK-25552
   <https://issues.asterisk.org/jira/browse/ASTERISK-25552>] -

hashtab: Improve NULL tolerance
(Reported by Joshua C. Colp)

   - [ASTERISK-25160
   <https://issues.asterisk.org/jira/browse/ASTERISK-25160>] -

[patch] Opus Codec: SIP/SDP line fmtp missing when called internally
(Reported by Alexander Traud)

   - [ASTERISK-25535
   <https://issues.asterisk.org/jira/browse/ASTERISK-25535>] -

[patch] format creation on module load instead of cache
(Reported by Alexander Traud)

   - [ASTERISK-25449
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25546>] -

threadpool: Race condition between idle timeout and activation
(Reported by Joshua C. Colp)

   - [ASTERISK-25537
   <https://issues.asterisk.org/jira/browse/ASTERISK-25537>] -

[patch] format-attribute module: RFC or internal defaults?
(Reported by Alexander Traud)

   - [ASTERISK-25533
   <https://issues.asterisk.org/jira/browse/ASTERISK-25533>] -

[patch] buffer for ast_format_cap_get_names only 64 bytes
(Reported by Alexander Traud)

   - [ASTERISK-25373
   <https://issues.asterisk.org/jira/browse/ASTERISK-25373>] -

add documentation for CALLERID(pres) and also the CONNECTEDLINE and
(Reported by Walter Doekes)

   - [ASTERISK-25528
   <https://issues.asterisk.org/jira/browse/ASTERISK-25528>] -

DNS: System resolver issues with TTL parse
(Reported by dtryba)

   - [ASTERISK-25527
   <https://issues.asterisk.org/jira/browse/ASTERISK-25527>] -

Quirky xmldoc description wrapping
(Reported by Walter Doekes)

   - [ASTERISK-24779
   <https://issues.asterisk.org/jira/browse/ASTERISK-24779>] -

Passthrough OPUS codec not working with chan_pjsip
(Reported by PowerPBX)

   - [ASTERISK-23904
   <https://issues.asterisk.org/jira/browse/ASTERISK-23904>] -

#define AST_MAX_ACCOUNT_CODE 20 causes truncation
(Reported by Ben Merrills)

   - [ASTERISK-25522
   <https://issues.asterisk.org/jira/browse/ASTERISK-25522>] -

ARI: Crash when creating channel via ARI originate with requesting channel
(Reported by Matt Jordan)

   - [ASTERISK-25434
   <https://issues.asterisk.org/jira/browse/ASTERISK-25434>] -

Compiler flags not reported in 'core show settings' despite usage during
(Reported by Rusty Newton)

   - [ASTERISK-24106
   <https://issues.asterisk.org/jira/browse/ASTERISK-24106>] -

WebSockets Automatically decides what driver it will use
(Reported by Andrew Nagy)

   - [ASTERISK-25513
   <https://issues.asterisk.org/jira/browse/ASTERISK-25513>] -

Crash: malloc failed with high load of subscriptions.
(Reported by John Bigelow)

   - [ASTERISK-25505
   <https://issues.asterisk.org/jira/browse/ASTERISK-25505>] -

res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
(Reported by Joshua C. Colp)

   - [ASTERISK-25485
   <https://issues.asterisk.org/jira/browse/ASTERISK-25485>] -

res_pjsip_outbound_registration: registration stops due to 400 response
(Reported by Kevin Harwell)

   - [ASTERISK-25486
   <https://issues.asterisk.org/jira/browse/ASTERISK-25486>] -

res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua C. Colp)

   - [ASTERISK-7803 <https://issues.asterisk.org/jira/browse/ASTERISK-7803>]

[patch] Update the maximum packetization values in frame.c
(Reported by dea)

   - [ASTERISK-25484
   <https://issues.asterisk.org/jira/browse/ASTERISK-25484>] -

[patch] autoframing=yes has no effect
(Reported by Alexander Traud)

   - [ASTERISK-25308
   <https://issues.asterisk.org/jira/browse/ASTERISK-25308>] -

ari: Websocket leak
(Reported by Joshua C. Colp)

   - [ASTERISK-25461
   <https://issues.asterisk.org/jira/browse/ASTERISK-25461>] -

Nested dialplan #includes don't work as expected.
(Reported by Richard Mudgett)

   - [ASTERISK-25455
   <https://issues.asterisk.org/jira/browse/ASTERISK-25455>] -

Deadlock of PJSIP realtime over res_config_pgsql
(Reported by mdu113)

   - [ASTERISK-25135
   <https://issues.asterisk.org/jira/browse/ASTERISK-25135>] -

[patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)

   - [ASTERISK-25108
   <https://issues.asterisk.org/jira/browse/ASTERISK-25108>] -

configure check for older unbound library
(Reported by John Bigelow)

   - [ASTERISK-25435
   <https://issues.asterisk.org/jira/browse/ASTERISK-25435>] -

Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
(Reported by Dmitriy Serov)

   - [ASTERISK-25451
   <https://issues.asterisk.org/jira/browse/ASTERISK-25451>] -

Broken video - erased rtp marker bit
(Reported by Stefan Engström)

   - [ASTERISK-25400
   <https://issues.asterisk.org/jira/browse/ASTERISK-25400>] -

Hints broken when "CustomPresence" doesn't exist in AstDB
(Reported by Andrew Nagy)

   - [ASTERISK-25443
   <https://issues.asterisk.org/jira/browse/ASTERISK-25443>] -

[patch]IPv6 - Potential issue in via header parsing
(Reported by ffs)

   - [ASTERISK-25404
   <https://issues.asterisk.org/jira/browse/ASTERISK-25404>] -

segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
(Reported by Chet Stevens)

   - [ASTERISK-25391
   <https://issues.asterisk.org/jira/browse/ASTERISK-25391>] -

AMI GetConfigJSON returns invalid JSON
(Reported by Bojan Nemčić)

   - [ASTERISK-25441
   <https://issues.asterisk.org/jira/browse/ASTERISK-25441>] -

Deadlock in res_sorcery_memory_cache.
(Reported by Richard Mudgett)

   - [ASTERISK-25438
   <https://issues.asterisk.org/jira/browse/ASTERISK-25438>] -

res_rtp_asterisk: ICE role message even when ICE is not enabled
(Reported by Joshua C. Colp)

   - [ASTERISK-25383
   <https://issues.asterisk.org/jira/browse/ASTERISK-25383>] -

Core dumps on startup and shutdown with MALLOC_DEBUG enabled
(Reported by yaron nahum)

   - [ASTERISK-25423
   <https://issues.asterisk.org/jira/browse/ASTERISK-25423>] -

Caller gets no Connected line update during call pickup.
(Reported by Richard Mudgett)

   - [ASTERISK-25305
   <https://issues.asterisk.org/jira/browse/ASTERISK-25305>] -

Dynamic logger channels can be added multiple times
(Reported by Mark Michelson)

   - [ASTERISK-25418
   <https://issues.asterisk.org/jira/browse/ASTERISK-25418>] -

On-hold channels redirected out of a bridge appear to still be on hold
(Reported by Mark Michelson)

   - [ASTERISK-25384
   <https://issues.asterisk.org/jira/browse/ASTERISK-25384>] -

Regular Asterisk crashes when using Page application. "user_data is NULL"
(Reported by Chet Stevens)

   - [ASTERISK-25410
   <https://issues.asterisk.org/jira/browse/ASTERISK-25410>] -

app_record: RECORDED_FILE variable not being populated
(Reported by Kevin Harwell)

   - [ASTERISK-25396
   <https://issues.asterisk.org/jira/browse/ASTERISK-25396>] -

chan_sip: Extremely long callerid name causes invalid SIP
(Reported by Walter Doekes)

   - [ASTERISK-25399
   <https://issues.asterisk.org/jira/browse/ASTERISK-25399>] -

app_queue: AgentComplete event has wrong reason
(Reported by Kevin Harwell)

   - [ASTERISK-25185
   <https://issues.asterisk.org/jira/browse/ASTERISK-25185>] -

Segfault in app_queue on transfer scenarios
(Reported by Etienne Lessard)

   - [ASTERISK-25353
   <https://issues.asterisk.org/jira/browse/ASTERISK-25353>] -

[patch] Transcoding while different in Frame size = Frames lost
(Reported by Alexander Traud)

   - [ASTERISK-25325
   <https://issues.asterisk.org/jira/browse/ASTERISK-25325>] -

ARI PUT reload chan_sip HTTP response 404
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25390
   <https://issues.asterisk.org/jira/browse/ASTERISK-25390>] -

default_from_user can crash with certain configuration backends
(Reported by Mark Michelson)

   - [ASTERISK-25387
   <https://issues.asterisk.org/jira/browse/ASTERISK-25387>] -

res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to
not be rewritten
(Reported by Matt Jordan)

   - [ASTERISK-25227
   <https://issues.asterisk.org/jira/browse/ASTERISK-25227>] -

No audio at in-band announcements in ooh323 channel
(Reported by Alexandr Dranchuk)

   - [ASTERISK-25295
   <https://issues.asterisk.org/jira/browse/ASTERISK-25295>] -

res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
(Reported by Dmitriy Serov)

   - [ASTERISK-25299
   <https://issues.asterisk.org/jira/browse/ASTERISK-25299>] -

RTP port leaks with incoming OOH323 calls
(Reported by Alexandr Dranchuk)

   - [ASTERISK-25381
   <https://issues.asterisk.org/jira/browse/ASTERISK-25381>] -

res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their
related contacts
(Reported by Matt Jordan)

   - [ASTERISK-25369
   <https://issues.asterisk.org/jira/browse/ASTERISK-25369>] -

res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the
announcer channel
(Reported by Jonathan Rose)

   - [ASTERISK-25375
   <https://issues.asterisk.org/jira/browse/ASTERISK-25375>] -

Bad ao2 pointer on snapshot cleanup after creation
(Reported by Scott Griepentrog)

   - [ASTERISK-24602
   <https://issues.asterisk.org/jira/browse/ASTERISK-24602>] -

Unable to call WebRTC client via wss on chan_pjsip
(Reported by Oleg Kozlov)

   - [ASTERISK-25367
   <https://issues.asterisk.org/jira/browse/ASTERISK-25367>] -

pbx: Long pattern match hints may cause "core show hints" to crash
(Reported by Joshua C. Colp)

   - [ASTERISK-25365
   <https://issues.asterisk.org/jira/browse/ASTERISK-25365>] -

Persistent subscriptions have extra Content-Length/corrupted messages
(Reported by Mark Michelson)

   - [ASTERISK-25356
   <https://issues.asterisk.org/jira/browse/ASTERISK-25356>] -

res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist
(Reported by Joshua C. Colp)

   - [ASTERISK-25355
   <https://issues.asterisk.org/jira/browse/ASTERISK-25355>] -

sched: ast_sched_del may return prematurely due to spurious wakeup
(Reported by Joshua C. Colp)

   - [ASTERISK-25318
   <https://issues.asterisk.org/jira/browse/ASTERISK-25318>] -

Sporadically failing
(Reported by Joshua C. Colp)

   - [ASTERISK-25346
   <https://issues.asterisk.org/jira/browse/ASTERISK-25346>] -

chan_sip: Overwriting answered elsewhere hangup cause on call pickup
(Reported by Joshua C. Colp)

   - [ASTERISK-25342
   <https://issues.asterisk.org/jira/browse/ASTERISK-25342>] -

res_pjsip: Repeated usage of pj_gethostip may block
(Reported by Joshua C. Colp)

   - [ASTERISK-25341
   <https://issues.asterisk.org/jira/browse/ASTERISK-25341>] -

bridge: Hangups may get lost when executing actions
(Reported by Joshua C. Colp)

   - [ASTERISK-25339
   <https://issues.asterisk.org/jira/browse/ASTERISK-25339>] -

res_pjsip: Empty "auth" sections from non-config backgrounds are
interpreted as valid
(Reported by Matt Jordan)

   - [ASTERISK-17410
   <https://issues.asterisk.org/jira/browse/ASTERISK-17410>] -

Video dynamic RTP payload type negotiation incorrect when directmedia
(Reported by Boris Fox)

   - [ASTERISK-25331
   <https://issues.asterisk.org/jira/browse/ASTERISK-25331>] -

install_prereq is not installing sqlite 3 library on CentOS
(Reported by Scott Griepentrog)

   - [ASTERISK-25215
   <https://issues.asterisk.org/jira/browse/ASTERISK-25215>] -

Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
(Reported by Lorne Gaetz)

   - [ASTERISK-25322
   <https://issues.asterisk.org/jira/browse/ASTERISK-25322>] -

Crash occurs when using MixMonitor with t() or r() options.
(Reported by Richard Mudgett)

   - [ASTERISK-25320
   <https://issues.asterisk.org/jira/browse/ASTERISK-25320>] -

chan_sip.c: sip_report_security_event searches for wrong or non existent
peer on invite
(Reported by Kevin Harwell)

   - [ASTERISK-25312
   <https://issues.asterisk.org/jira/browse/ASTERISK-25312>] -

res_http_websocket: Terminate connection on fatal cases
(Reported by Joshua C. Colp)

   - [ASTERISK-25315
   <https://issues.asterisk.org/jira/browse/ASTERISK-25315>] -

DAHDI channels send shortened duration DTMF tones.
(Reported by Richard Mudgett)

   - [ASTERISK-25306
   <https://issues.asterisk.org/jira/browse/ASTERISK-25306>] -

Persistent subscriptions can save multiple SIP messages at once, leading to
potential crashes.
(Reported by Mark Michelson)

   - [ASTERISK-25309
   <https://issues.asterisk.org/jira/browse/ASTERISK-25309>] -

[patch] iLBC 20 advertised
(Reported by Alexander Traud)

   - [ASTERISK-25304
   <https://issues.asterisk.org/jira/browse/ASTERISK-25304>] -

res_pjsip: XML sanitization may write past buffer
(Reported by Joshua C. Colp)

   - [ASTERISK-25265
   <https://issues.asterisk.org/jira/browse/ASTERISK-25265>] -

[patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH
support and fallback to prime256v1
(Reported by Stefan Engström)

   - [ASTERISK-24988
   <https://issues.asterisk.org/jira/browse/ASTERISK-24988>] -

func_talkdetect: Test is bouncing sporadically
(Reported by Joshua C. Colp)

   - [ASTERISK-25181
   <https://issues.asterisk.org/jira/browse/ASTERISK-25181>] -

ARI: Channels added to Stasis application during WebSocket creation don't
receive a StasisStart event
(Reported by Matt Jordan)

   - [ASTERISK-25296
   <https://issues.asterisk.org/jira/browse/ASTERISK-25296>] -

RTP performance issue with several channel drivers.
(Reported by Richard Mudgett)

   - [ASTERISK-25297
   <https://issues.asterisk.org/jira/browse/ASTERISK-25297>] -

Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite
(Reported by Richard Mudgett)

   - [ASTERISK-25292
   <https://issues.asterisk.org/jira/browse/ASTERISK-25292>] -

Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
(Reported by Kevin Harwell)

   - [ASTERISK-25271
   <https://issues.asterisk.org/jira/browse/ASTERISK-25271>] -

Parking & blind transfer: Transferer channel not hung up if no MOH
(Reported by Kevin Harwell)

   - [ASTERISK-25250
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25253>] -

confbridge volume options and other volume controls such as func_volume
don't work
(Reported by Dmitriy Serov)

   - [ASTERISK-25247
   <https://issues.asterisk.org/jira/browse/ASTERISK-25247>] -

choppy audio when spying on a g722 channel, chan_sip or chan_pjsip
(Reported by hristo)

   - [ASTERISK-25263
   <https://issues.asterisk.org/jira/browse/ASTERISK-25263>] -

[patch]cdr_adaptive_odbc: CDR insert failure due to reversed if logic
(Reported by Elazar Broad)

   - [ASTERISK-24867
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24853>] -

Documentation claims chan_sip outbound registrations support WS or WSS as
valid transports (not true)
(Reported by PSDK)

   - [ASTERISK-25242
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25258>] -

chan_pjsip: Incorrect format switch on received RTP packet
(Reported by Joshua C. Colp)

   - [ASTERISK-25257
   <https://issues.asterisk.org/jira/browse/ASTERISK-25257>] -

[patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope
(Reported by Patric Marschall)

   - [ASTERISK-24934
   <https://issues.asterisk.org/jira/browse/ASTERISK-24934>] -

[patch]Asterisk manager output does not escape control characters
(Reported by warren smith)

   - [ASTERISK-25255
   <https://issues.asterisk.org/jira/browse/ASTERISK-25255>] -

Missing AMI VarSet events when setting to an empty string.
(Reported by Richard Mudgett)

   - [ASTERISK-25254
   <https://issues.asterisk.org/jira/browse/ASTERISK-25254>] -

Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park.
(Reported by Richard Mudgett)

   - [ASTERISK-25183
   <https://issues.asterisk.org/jira/browse/ASTERISK-25183>] -

PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite
previous checks for NULL channel
(Reported by Matt Jordan)

   - [ASTERISK-25201
   <https://issues.asterisk.org/jira/browse/ASTERISK-25201>] -

Crash in PJSIP distributor on already free'd threadpool
(Reported by Matt Jordan)

   - [ASTERISK-25240
   <https://issues.asterisk.org/jira/browse/ASTERISK-25240>] -

bridge_native_rtp: Direct media wrongfully started when completing attended
(Reported by Joshua C. Colp)

   - [ASTERISK-25103
   <https://issues.asterisk.org/jira/browse/ASTERISK-25103>] -

Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)

   - [ASTERISK-25146
   <https://issues.asterisk.org/jira/browse/ASTERISK-25146>] -

DNS: Create system level resolver
(Reported by Joshua C. Colp)

   - [ASTERISK-22805
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24550>] -

res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
(Reported by Osaulenko Alexander)

   - [ASTERISK-24651
   <https://issues.asterisk.org/jira/browse/ASTERISK-24651>] -

[patch] Fix race condition in DTLS
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24832
   <https://issues.asterisk.org/jira/browse/ASTERISK-24832>] -

[patch]DTLS-crashes within openssl
(Reported by Stefan Engström)

   - [ASTERISK-25127
   <https://issues.asterisk.org/jira/browse/ASTERISK-25127>] -

DTLS crashes following "Unable to cancel schedule ID" in
(Reported by Dade Brandon)

   - [ASTERISK-25168
   <https://issues.asterisk.org/jira/browse/ASTERISK-25168>] -

Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at
(Reported by Carl Fortin)

   - [ASTERISK-25076
   <https://issues.asterisk.org/jira/browse/ASTERISK-25076>] -

res_pjsip: Failover does not occur on connection-less transport or 503
(Reported by Joshua C. Colp)

   - [ASTERISK-25226
   <https://issues.asterisk.org/jira/browse/ASTERISK-25226>] -

chan_sip: Channel leak in branch 13 on early replaces call pickup
(Reported by Walter Doekes)

   - [ASTERISK-25222
   <https://issues.asterisk.org/jira/browse/ASTERISK-25222>] -

Crash in recurring cancel callback called from ast_dns_resolve_cancel on
junk pointer
(Reported by Matt Jordan)

   - [ASTERISK-25220
   <https://issues.asterisk.org/jira/browse/ASTERISK-25220>] -

[patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)

   - [ASTERISK-25219
   <https://issues.asterisk.org/jira/browse/ASTERISK-25219>] -

[patch]Source and destination overlap in memcpy in rtp_engine.c
(Reported by Walter Doekes)

   - [ASTERISK-25212
   <https://issues.asterisk.org/jira/browse/ASTERISK-25212>] -

[patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)

   - [ASTERISK-19277
   <https://issues.asterisk.org/jira/browse/ASTERISK-19277>] -

[patch]endlessly repeating error: "poll failed: Bad file descriptor"
(Reported by Barry Chern)

   - [ASTERISK-25202
   <https://issues.asterisk.org/jira/browse/ASTERISK-25202>] -

Hints extension state broken between 13.3.2 and 13.4
(Reported by Marek Cervenka)

   - [ASTERISK-25196
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24907>] -

res_pjsip_outbound_registration: crash during unload if registration
attempts are still occuring
(Reported by Kevin Harwell)

   - [ASTERISK-25204
   <https://issues.asterisk.org/jira/browse/ASTERISK-25204>] -

res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound
(Reported by Mark Michelson)

   - [ASTERISK-25189
   <https://issues.asterisk.org/jira/browse/ASTERISK-25189>] -

AMI: Add Linkedid header to standard channel snapshot information.
(Reported by Richard Mudgett)

   - [ASTERISK-25171
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25180>] -

res_pjsip_mwi: Unsolicited MWI requires reload
(Reported by Joshua C. Colp)

   - [ASTERISK-25182
   <https://issues.asterisk.org/jira/browse/ASTERISK-25182>] -

[patch] on CLI sip reload, new codecs get appended only
(Reported by Alexander Traud)

   - [ASTERISK-25163
   <https://issues.asterisk.org/jira/browse/ASTERISK-25163>] -

Deadlock in chan_sip between reload of sip peer container and MWI Stasis
(Reported by Dmitriy Serov)

   - [ASTERISK-25091
   <https://issues.asterisk.org/jira/browse/ASTERISK-25091>] -

Asterisk REST API - bridge.addChannel crash asterisk when calling channel
hangup while adding to bridge
(Reported by Ilya Trikoz)

   - [ASTERISK-24900
   <https://issues.asterisk.org/jira/browse/ASTERISK-24900>] -

Manager event ParkedCallSwap is not documented
(Reported by Rusty Newton)

   - [ASTERISK-25162
   <https://issues.asterisk.org/jira/browse/ASTERISK-25162>] -

func_pjsip_aor: Leak of contact in iterator
(Reported by Corey Farrell)

   - [ASTERISK-25158
   <https://issues.asterisk.org/jira/browse/ASTERISK-25158>] -

res_pjsip: Add option to use AAL2 packing when negotiating g.726
(Reported by Kevin Harwell)

   - [ASTERISK-24344
   <https://issues.asterisk.org/jira/browse/ASTERISK-24344>] -

CDR_PROP(disable) disables CDR only for first dialed party
(Reported by Janusz Karolak)

   - [ASTERISK-24443
   <https://issues.asterisk.org/jira/browse/ASTERISK-24443>] -

CDR fields (dst, dcontext) empty in transfer call started from Macro
(Reported by Arveno Santoro)

   - [ASTERISK-25154
   <https://issues.asterisk.org/jira/browse/ASTERISK-25154>] -

[patch]fromtag may need to be updated after successful call dialog match
(Reported by Damian Ivereigh)

   - [ASTERISK-25156
   <https://issues.asterisk.org/jira/browse/ASTERISK-25156>] -

chan_pjsip’s CHAN_START cel event lacks the correct context and exten
(Reported by cloos)

   - [ASTERISK-25157
   <https://issues.asterisk.org/jira/browse/ASTERISK-25157>] -

bridging: Performing a blonde transfer does not result in connected line
(Reported by Joshua C. Colp)

   - [ASTERISK-25087
   <https://issues.asterisk.org/jira/browse/ASTERISK-25087>] -

Asterisk segfault when using Directory application with alias option and
specific mailbox configuration
(Reported by Chet Stevens)

   - [ASTERISK-25115
   <https://issues.asterisk.org/jira/browse/ASTERISK-25115>] -

Crash related to func sip_resolve_invoke_user_callback of
(Reported by John Bigelow)

   - [ASTERISK-25096
   <https://issues.asterisk.org/jira/browse/ASTERISK-25096>] -

[patch]Segfault when registering over websockets with PJSIP (in
ast_sockaddr_isnull at /include/asterisk/netsock2.h)
(Reported by Josh Kitchens)

   - [ASTERISK-24963
   <https://issues.asterisk.org/jira/browse/ASTERISK-24963>] -

ASAN: heap-use-after-free with PJSIP and WSS
(Reported by Badalian Vyacheslav)

   - [ASTERISK-22559
   <https://issues.asterisk.org/jira/browse/ASTERISK-22559>] -

gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect
(Reported by ibercom)

   - [ASTERISK-25094
   <https://issues.asterisk.org/jira/browse/ASTERISK-25094>] -

PBX core: Investigate thread safety issues
(Reported by Corey Farrell)

   - [ASTERISK-25113
   <https://issues.asterisk.org/jira/browse/ASTERISK-25113>] -

install_prereq in Debian 8 without "standard system utilities"
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25148
   <https://issues.asterisk.org/jira/browse/ASTERISK-25148>] -

res_pjsip NULL channel audit
(Reported by Mark Michelson)

   - [ASTERISK-25141
   <https://issues.asterisk.org/jira/browse/ASTERISK-25141>] -

pjsip_options: Contact reference leak
(Reported by Corey Farrell)

   - [ASTERISK-25131
   <https://issues.asterisk.org/jira/browse/ASTERISK-25131>] -

chan_pjsip: In-dialog authentication not handled.
(Reported by Richard Mudgett)

   - [ASTERISK-24717
   <https://issues.asterisk.org/jira/browse/ASTERISK-24717>] -

ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25100
   <https://issues.asterisk.org/jira/browse/ASTERISK-25100>] -

asterisk coredump if host has an IPv6 address that end with ::80
(Reported by Mark Petersen)

   - [ASTERISK-25122
   <https://issues.asterisk.org/jira/browse/ASTERISK-25122>] -

Large SIP packet received via pjsip over websocket crashes Asterisk
(Reported by Ivan Poddubny)

   - [ASTERISK-25121
   <https://issues.asterisk.org/jira/browse/ASTERISK-25121>] -

Stasis: Fix unsafe use of stasis_unsubscribe in modules.
(Reported by Corey Farrell)

   - [ASTERISK-25120
   <https://issues.asterisk.org/jira/browse/ASTERISK-25120>] -

Astobj2: Weakproxy subscriptions should be run in reverse order.
(Reported by Corey Farrell)

   - [ASTERISK-25105
   <https://issues.asterisk.org/jira/browse/ASTERISK-25105>] -

res_pjsip: Possible incompatibility between qualify_timeout and
(Reported by George Joseph)

   - [ASTERISK-25117
   <https://issues.asterisk.org/jira/browse/ASTERISK-25117>] -

res_mwi_external_ami: Fix manager action registrations.
(Reported by Corey Farrell)

   - [ASTERISK-25112
   <https://issues.asterisk.org/jira/browse/ASTERISK-25112>] -

Logger: Configuration settings are not reset to default during reload.
(Reported by Corey Farrell)

   - [ASTERISK-24983
   <https://issues.asterisk.org/jira/browse/ASTERISK-24983>] -

IAX deadlock between hangup and scheduled actions (ex. largrq)
(Reported by Y Ateya)

   - [ASTERISK-24944
   <https://issues.asterisk.org/jira/browse/ASTERISK-24944>] -

main/audiohook.c change prevents G722 call recording
(Reported by Ronald Raikes)

   - [ASTERISK-25110
   <https://issues.asterisk.org/jira/browse/ASTERISK-25110>] -

res_resolver_unbound.c compilation failure: SIGURG is undeclared in func
(Reported by John Bigelow)

   - [ASTERISK-24887
   <https://issues.asterisk.org/jira/browse/ASTERISK-24887>] -

[patch]tags in a=crypto lines do not accept 2 or more digits
(Reported by Makoto Dei)

   - [ASTERISK-25086
   <https://issues.asterisk.org/jira/browse/ASTERISK-25086>] -

[patch]PJSIP crashes if endpoint missing in Dial()
(Reported by snuffy)

   - [ASTERISK-25089
   <https://issues.asterisk.org/jira/browse/ASTERISK-25089>] -

res_pjsip_config_wizard: Variable specified in templates aren't being
processed correctly
(Reported by George Joseph)

   - [ASTERISK-25090
   <https://issues.asterisk.org/jira/browse/ASTERISK-25090>] -

CLI core show channel truncates cdr variables
(Reported by snuffy)

   - [ASTERISK-25085
   <https://issues.asterisk.org/jira/browse/ASTERISK-25085>] -

[patch]Potential crash after unload of func_periodic_hook or test_message
(Reported by Corey Farrell)

   - [ASTERISK-25082
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-21893>] -

Segfault after call hangup, in ast_channel_hangupcause_set, at
(Reported by Aleksandr Gordeev)

   - [ASTERISK-25042
   <https://issues.asterisk.org/jira/browse/ASTERISK-25042>] -

asterisk.conf options override command-line options.
(Reported by Corey Farrell)

   - [ASTERISK-25074
   <https://issues.asterisk.org/jira/browse/ASTERISK-25074>] -

Regression: Recent clang-related change broke cross compiling of Asterisk
(Reported by Sebastian Kemper)

   - [ASTERISK-17069
   <https://issues.asterisk.org/jira/browse/ASTERISK-17069>] -

Callfile retries behave erratically as file size grows
(Reported by Jeremy Kister)

   - [ASTERISK-24442
   <https://issues.asterisk.org/jira/browse/ASTERISK-24442>] -

Outgoing call files don't work properly when set in the future
(Reported by tootai)

   - [ASTERISK-18252
   <https://issues.asterisk.org/jira/browse/ASTERISK-18252>] -

queue_log mysql time column data format
(Reported by Gareth Blades)

   - [ASTERISK-25041
   <https://issues.asterisk.org/jira/browse/ASTERISK-25041>] -

[patch]Broken column type checking in res_config_mysql addon
(Reported by Alexandre Fournier)

   - [ASTERISK-25057
   <https://issues.asterisk.org/jira/browse/ASTERISK-25057>] -

res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in
(Reported by Matt Jordan)

   - [ASTERISK-24938
   <https://issues.asterisk.org/jira/browse/ASTERISK-24938>] -

ARI Snoop Channel results in excessive escalating CPU usage
(Reported by George Ladoff)

   - [ASTERISK-25034
   <https://issues.asterisk.org/jira/browse/ASTERISK-25034>] -

chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
(Reported by Richard Mudgett)

   - [ASTERISK-25003
   <https://issues.asterisk.org/jira/browse/ASTERISK-25003>] -

Asterisk crashes on attended transfer (using feature)
(Reported by Artem Volodin)

   - [ASTERISK-25038
   <https://issues.asterisk.org/jira/browse/ASTERISK-25038>] -

Queue log "EXITWITHTIMEOUT" does not always contain waiting time
(Reported by Etienne Lessard)

   - [ASTERISK-25027
   <https://issues.asterisk.org/jira/browse/ASTERISK-25027>] -

Build System: Many ARI modules are missing dependencies.
(Reported by Corey Farrell)

   - [ASTERISK-25061
   <https://issues.asterisk.org/jira/browse/ASTERISK-25061>] -

pbx_config: Register manager actions with module version of macro.
(Reported by Corey Farrell)

   - [ASTERISK-24967
   <https://issues.asterisk.org/jira/browse/ASTERISK-24967>] -

Problem support schema for pgsql on CEL
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25025
   <https://issues.asterisk.org/jira/browse/ASTERISK-25025>] -

Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with
Certified Asterisk 13.
(Reported by Chet Stevens)

   - [ASTERISK-25053
   <https://issues.asterisk.org/jira/browse/ASTERISK-25053>] -

Unit test category /main/presence missing trailing slash.
(Reported by Corey Farrell)

   - [ASTERISK-22708
   <https://issues.asterisk.org/jira/browse/ASTERISK-22708>] -

res_odbc.conf negative_connection_cache option not respected, failover
between DSNs doesn't work
(Reported by JoshE)

   - [ASTERISK-25054
   <https://issues.asterisk.org/jira/browse/ASTERISK-25054>] -

Formats interface's cannot be unregistered, needs to hold modules until
(Reported by Corey Farrell)

   - [ASTERISK-24976
   <https://issues.asterisk.org/jira/browse/ASTERISK-24976>] -

cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25033
   <https://issues.asterisk.org/jira/browse/ASTERISK-25033>] -

Asterisk 13 (branch head) won't compile without PJSip
(Reported by Peter Whisker)

   - [ASTERISK-24896
   <https://issues.asterisk.org/jira/browse/ASTERISK-24896>] -

[patch] Using force black background leads to colours not being reset
(Reported by dant)

   - [ASTERISK-25048
   <https://issues.asterisk.org/jira/browse/ASTERISK-25048>] -

Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are
both enabled.
(Reported by Corey Farrell)

   - [ASTERISK-24969
   <https://issues.asterisk.org/jira/browse/ASTERISK-24969>] -

Named ACL's do not handle config errors.
(Reported by Corey Farrell)

   - [ASTERISK-19608
   <https://issues.asterisk.org/jira/browse/ASTERISK-19608>] -

Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
(Reported by Denis Alberto Martinez)

   - [ASTERISK-25037
   <https://issues.asterisk.org/jira/browse/ASTERISK-25037>] -

res_pjsip_outbound_registration: Potential crash in off-nominal failure
case when sending message
(Reported by Joshua C. Colp)

   - [ASTERISK-25022
   <https://issues.asterisk.org/jira/browse/ASTERISK-25022>] -

Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)

   - [ASTERISK-22790
   <https://issues.asterisk.org/jira/browse/ASTERISK-22790>] -

check_modem_rate() may return incorrect rate for V.27
(Reported by not here)

   - [ASTERISK-23231
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-25020>] -

Mismatched response to outgoing REGISTER request
(Reported by Mark Michelson)

   - [ASTERISK-25028
   <https://issues.asterisk.org/jira/browse/ASTERISK-25028>] -

Build System: Unneeded defines in asterisk/buildopts.h
(Reported by Corey Farrell)

   - [ASTERISK-25026
   <https://issues.asterisk.org/jira/browse/ASTERISK-25026>] -

Git conversion: Non-C files not switched to ASTERISK_REGISTER_FILE
(Reported by Corey Farrell)

   - [ASTERISK-24996
   <https://issues.asterisk.org/jira/browse/ASTERISK-24996>] -

chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR
Sections Exist in pjsip.conf
(Reported by Ashley Sanders)

   - [ASTERISK-25018
   <https://issues.asterisk.org/jira/browse/ASTERISK-25018>] -

pjsip show endpoints crashes asterisk when qualified aors present
(Reported by Ivan Poddubny)

   - [ASTERISK-24749
   <https://issues.asterisk.org/jira/browse/ASTERISK-24749>] -

ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when
played to bridge
(Reported by Philippe Bolduc)

   - [ASTERISK-24845
   <https://issues.asterisk.org/jira/browse/ASTERISK-24845>] -

pjsip send notify not working with Cisco phone
(Reported by Carl Fortin)

   - [ASTERISK-25004
   <https://issues.asterisk.org/jira/browse/ASTERISK-25004>] -

Crash in authenticated reinvite after originated T.38 FAX
(Reported by Mark Michelson)

   - [ASTERISK-24999
   <https://issues.asterisk.org/jira/browse/ASTERISK-24999>] -

PJSIP crashes with malformed contact line
(Reported by snuffy)

   - [ASTERISK-24998
   <https://issues.asterisk.org/jira/browse/ASTERISK-24998>] -

res_corosync: res_corosync tries to load even if res_corosync.conf is
(Reported by George Joseph)

   - [ASTERISK-24997
   <https://issues.asterisk.org/jira/browse/ASTERISK-24997>] -

Astobj2: Some callers of __adjust_lock do not pre-check the object
(Reported by Corey Farrell)

   - [ASTERISK-24994
   <https://issues.asterisk.org/jira/browse/ASTERISK-24994>] -

dns: Query set unit tests are failing due to race condition
(Reported by Joshua C. Colp)

   - [ASTERISK-24982
   <https://issues.asterisk.org/jira/browse/ASTERISK-24982>] -

res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes
(Reported by Joshua C. Colp)

   - [ASTERISK-24991
   <https://issues.asterisk.org/jira/browse/ASTERISK-24991>] -

Check for ao2_alloc failure in __ast_channel_internal_alloc
(Reported by Corey Farrell)

   - [ASTERISK-24895
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24977>] -

Contacts that don't use qualify are being marked as unavailable
(Reported by George Joseph)

   - [ASTERISK-24774
   <https://issues.asterisk.org/jira/browse/ASTERISK-24774>] -

Segfault in ast_context_destroy with extensions.ael and extensions.conf
(Reported by Corey Farrell)

   - [ASTERISK-24841
   <https://issues.asterisk.org/jira/browse/ASTERISK-24841>] -

ConfBridge: Strange sampling rates chosen when channels have multiple
native formats
(Reported by Matt Jordan)

   - [ASTERISK-24975
   <https://issues.asterisk.org/jira/browse/ASTERISK-24975>] -

Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
(Reported by Ashley Sanders)

   - [ASTERISK-24863
   <https://issues.asterisk.org/jira/browse/ASTERISK-24863>] -

res_pjsip: No endpoint events raised via AMI when contacts cannot be
(Reported by Dmitriy Serov)

   - [ASTERISK-24869
   <https://issues.asterisk.org/jira/browse/ASTERISK-24869>] -

Asterisk segfaults on DAHDI attended transfer due to application (appl)
being NULL on unbridged channel
(Reported by viniciusfontes)

   - [ASTERISK-24970
   <https://issues.asterisk.org/jira/browse/ASTERISK-24970>] -

Crash in res_pjsip_pubsub handling of failed notify
(Reported by Scott Griepentrog)

   - [ASTERISK-13271
   <https://issues.asterisk.org/jira/browse/ASTERISK-13271>] -

menuselect sets defaults too late
(Reported by John Nemeth)

   - [ASTERISK-24959
   <https://issues.asterisk.org/jira/browse/ASTERISK-24959>] -

[patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-20524
   <https://issues.asterisk.org/jira/browse/ASTERISK-20524>] -

AMI improperly handles lines of exactly 1025 characters
(Reported by David M. Lee)

   - [ASTERISK-24936
   <https://issues.asterisk.org/jira/browse/ASTERISK-24936>] -

New Feature: AO2 weakproxy objects
(Reported by Corey Farrell)

   - [ASTERISK-24954
   <https://issues.asterisk.org/jira/browse/ASTERISK-24954>] -

Git migration: Asterisk version numbers are incompatible with the Test Suite
(Reported by Matt Jordan)

   - [ASTERISK-17608
   <https://issues.asterisk.org/jira/browse/ASTERISK-17608>] -

func_aes.so cannot be loaded if res_crypto / openssl not compiled
(Reported by Warren Selby)

   - [ASTERISK-24928
   <https://issues.asterisk.org/jira/browse/ASTERISK-24928>] -

[patch]t38_udptl_maxdatagram in pjsip.conf not honored
(Reported by Juergen Spies)

   - [ASTERISK-24835
   <https://issues.asterisk.org/jira/browse/ASTERISK-24835>] -

Early Media Not working with Chan SIP and Asterisk 13
(Reported by Andrew Nagy)

   - [ASTERISK-21777
   <https://issues.asterisk.org/jira/browse/ASTERISK-21777>] -

Asterisk tries to transcode video instead of audio
(Reported by Nick Ruggles)

   - [ASTERISK-24380
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-22352>] -

[patch] IAX2 custom qualify timer is not taken into account
(Reported by Frederic Van Espen)

   - [ASTERISK-24894
   <https://issues.asterisk.org/jira/browse/ASTERISK-24894>] -

[patch] iax2_poke_noanswer expiration timer too short
(Reported by Y Ateya)

   - [ASTERISK-24935
   <https://issues.asterisk.org/jira/browse/ASTERISK-24935>] -

res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
(Reported by Corey Farrell)

   - [ASTERISK-23319
   <https://issues.asterisk.org/jira/browse/ASTERISK-23319>] -

Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)

   - [ASTERISK-24933
   <https://issues.asterisk.org/jira/browse/ASTERISK-24933>] -

T38 fails negotiation
(Reported by Jonathan Rose)

   - [ASTERISK-24847
   <https://issues.asterisk.org/jira/browse/ASTERISK-24847>] -

[security] [patch] tcptls: certificate CN NULL byte prefix bug
(Reported by Matt Jordan)

   - [ASTERISK-21211
   <https://issues.asterisk.org/jira/browse/ASTERISK-21211>] -

chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in
(Reported by Jaco Kroon)

   - [ASTERISK-18032
   <https://issues.asterisk.org/jira/browse/ASTERISK-18032>] -

[patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)

   - [ASTERISK-24910
   <https://issues.asterisk.org/jira/browse/ASTERISK-24910>] -

"timer=no" and "timer=required" settings in pjsip.conf fail
(Reported by Ray Crumrine)

   - [ASTERISK-24932
   <https://issues.asterisk.org/jira/browse/ASTERISK-24932>] -

Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)

   - [ASTERISK-24914
   <https://issues.asterisk.org/jira/browse/ASTERISK-24914>] -

Division by zero in file.c when playback of voicemail with video as h264
(Reported by Marcello Ceschia)

   - [ASTERISK-24899
   <https://issues.asterisk.org/jira/browse/ASTERISK-24899>] -

Parking fall-through behavior different in 13
(Reported by Malcolm Davenport)

   - [ASTERISK-24937
   <https://issues.asterisk.org/jira/browse/ASTERISK-24937>] -

[patch]res_pjsip_messaging: Messages may be sent out of order
(Reported by Mark Michelson)

   - [ASTERISK-24920
   <https://issues.asterisk.org/jira/browse/ASTERISK-24920>] -

Asterisk handles duplicate SIP requests as if they were each a new request
(Reported by Mark Michelson)

   - [ASTERISK-24781
   <https://issues.asterisk.org/jira/browse/ASTERISK-24781>] -

PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.
(Reported by Richard Mudgett)

   - [ASTERISK-24857
   <https://issues.asterisk.org/jira/browse/ASTERISK-24857>] -

[patch] "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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24155>] -

[patch]Non-portable and non-reliable recursion detection in ast_malloc
(Reported by Timo Teräs)

   - [ASTERISK-24142
   <https://issues.asterisk.org/jira/browse/ASTERISK-24142>] -

CCSS: crash during shutdown due to device lookup in destroyed container
(Reported by David Brillert)

   - [ASTERISK-24683
   <https://issues.asterisk.org/jira/browse/ASTERISK-24683>] -

Crash in PBX ast_hashtab_lookup_internal during core restart now
(Reported by Peter Katzmann)

   - [ASTERISK-24805
   <https://issues.asterisk.org/jira/browse/ASTERISK-24805>] -

[patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24881
   <https://issues.asterisk.org/jira/browse/ASTERISK-24881>] -

ast_register_atexit should only be used when absolutely needed
(Reported by Corey Farrell)

   - [ASTERISK-24731
   <https://issues.asterisk.org/jira/browse/ASTERISK-24731>] -

res_pjsip_session cannot be unloaded
(Reported by Corey Farrell)

   - [ASTERISK-24864
   <https://issues.asterisk.org/jira/browse/ASTERISK-24864>] -

app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)

   - [ASTERISK-14233
   <https://issues.asterisk.org/jira/browse/ASTERISK-14233>] -

[patch] Buddies are always auto-registered when processing the roster
(Reported by Simon Arlott)

   - [ASTERISK-24780
   <https://issues.asterisk.org/jira/browse/ASTERISK-24780>] -

[patch] - Buddies are always auto-registered when processing the roster
(Reported by Simon Arlott)

   - [ASTERISK-24879
   <https://issues.asterisk.org/jira/browse/ASTERISK-24879>] -

[patch]Compilation fails due to 64bit time under OpenBSD
(Reported by snuffy)

   - [ASTERISK-24880
   <https://issues.asterisk.org/jira/browse/ASTERISK-24880>] -

[patch]Compilation under OpenBSD
(Reported by snuffy)

   - [ASTERISK-21765
   <https://issues.asterisk.org/jira/browse/ASTERISK-21765>] -

[patch] - FILE function's length argument counts from beginning of file
rather than the offset
(Reported by John Zhong)

   - [ASTERISK-24817
   <https://issues.asterisk.org/jira/browse/ASTERISK-24817>] -

init_logger_chain: unreachable code block
(Reported by Corey Farrell)

   - [ASTERISK-24882
   <https://issues.asterisk.org/jira/browse/ASTERISK-24882>] -

chan_sip: Improve usage of REF_DEBUG
(Reported by Corey Farrell)

   - [ASTERISK-24876
   <https://issues.asterisk.org/jira/browse/ASTERISK-24876>] -

Investigate reference leaks from tests/channels/local/local_optimize_away
(Reported by Corey Farrell)

   - [ASTERISK-24840
   <https://issues.asterisk.org/jira/browse/ASTERISK-24840>] -

res_pjsip: conflicting endpoint identifiers
(Reported by Kevin Harwell)

   - [ASTERISK-16779
   <https://issues.asterisk.org/jira/browse/ASTERISK-16779>] -

Cannot disallow unknown format ''
(Reported by Atis Lezdins)

   - [ASTERISK-18708
   <https://issues.asterisk.org/jira/browse/ASTERISK-18708>] -

func_curl hangs channel under load
(Reported by Dave Cabot)

   - [ASTERISK-21038
   <https://issues.asterisk.org/jira/browse/ASTERISK-21038>] -

CLI: "core set debug channel" auto-complete returns "all", but not the
names of available channels
(Reported by Richard Kenner)

   - [ASTERISK-19470
   <https://issues.asterisk.org/jira/browse/ASTERISK-19470>] -

Documentation on app_amd is incorrect
(Reported by Frank DiGennaro)

   - [ASTERISK-24872
   <https://issues.asterisk.org/jira/browse/ASTERISK-24872>] -

[patch] AMI PJSIPShowEndpoint closes AMI connection on error
(Reported by Dmitriy Serov)

   - [ASTERISK-23666
   <https://issues.asterisk.org/jira/browse/ASTERISK-23666>] -

CLONE - nested functions aren't portable
(Reported by Diederik de Groot)

   - [ASTERISK-20399
   <https://issues.asterisk.org/jira/browse/ASTERISK-20399>] -

Compilation on some systems requires the -fnested-functions flag
(Reported by David M. Lee)

   - [ASTERISK-20850
   <https://issues.asterisk.org/jira/browse/ASTERISK-20850>] -

[patch]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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24807>] -

Missing mandatory field Max-Forwards
(Reported by Anatoli)

   - [ASTERISK-24808
   <https://issues.asterisk.org/jira/browse/ASTERISK-24808>] -

res_config_odbc: Improper escaping of backslashes occurs with MySQL
(Reported by Javier Acosta)

   - [ASTERISK-23390
   <https://issues.asterisk.org/jira/browse/ASTERISK-23390>] -

NewExten Event with application AGI shows up before and after AGI runs
(Reported by Benjamin Keith Ford)

   - [ASTERISK-24786
   <https://issues.asterisk.org/jira/browse/ASTERISK-24786>] -

[patch] - Asterisk terminates when playing a voicemail stored in LDAP
(Reported by Graham Barnett)

   - [ASTERISK-24739
   <https://issues.asterisk.org/jira/browse/ASTERISK-24739>] -

[patch] - Out of files -- call fails -- numerous files with inodes from
under /usr/share/zoneinfo, mostly posixrules
(Reported by Ed Hynan)

   - [ASTERISK-24755
   <https://issues.asterisk.org/jira/browse/ASTERISK-24755>] -

Asterisk sends unexpected early BYE to transferrer during attended transfer
when using a Stasis bridge
(Reported by John Bigelow)

   - [ASTERISK-24830
   <https://issues.asterisk.org/jira/browse/ASTERISK-24830>] -

res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
(Reported by Stefan Engström)

   - [ASTERISK-24825
   <https://issues.asterisk.org/jira/browse/ASTERISK-24825>] -

Caller ID not recognized using Centrex/Distinctive dialing
(Reported by Richard Mudgett)

   - [ASTERISK-17588
   <https://issues.asterisk.org/jira/browse/ASTERISK-17588>] -

Caller ID on TDM410P *UK* PSTN
(Reported by Daniel Flounders)

   - [ASTERISK-24838
   <https://issues.asterisk.org/jira/browse/ASTERISK-24838>] -

chan_sip: Locking inversion occurs when building a peer causes a peer poke
during request handling
(Reported by Richard Mudgett)

   - [ASTERISK-24751
   <https://issues.asterisk.org/jira/browse/ASTERISK-24751>] -

Integer values in json payload to ARI cause asterisk to crash
(Reported by jeffrey putnam)

   - [ASTERISK-24828
   <https://issues.asterisk.org/jira/browse/ASTERISK-24828>] -

Fix Frame Leaks
(Reported by Kevin Harwell)

   - [ASTERISK-18105
   <https://issues.asterisk.org/jira/browse/ASTERISK-18105>] -

most of asterisk modules are unbuildable in cygwin environment
(Reported by feyfre)

   - [ASTERISK-21845
   <https://issues.asterisk.org/jira/browse/ASTERISK-21845>] -

maxcalls exceeded, Asterisk sends out 480 and also BYE
(Reported by Tony Ching)

   - [ASTERISK-15434
   <https://issues.asterisk.org/jira/browse/ASTERISK-15434>] -

[patch] When ast_pbx_start failed, both an error response and BYE are sent
to the caller
(Reported by Makoto Dei)

   - [ASTERISK-23214
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-17721>] -

Incoming SRTP calls that specify a key lifetime fail
(Reported by Terry Wilson)

   - [ASTERISK-20233
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-22748>] -

SRTP Crypto Offer With Lifetime Not Accepted
(Reported by Alejandro Mejia)

   - [ASTERISK-24800
   <https://issues.asterisk.org/jira/browse/ASTERISK-24800>] -

Crash in __sip_reliable_xmit due to invalid thread ID being passed to
(Reported by JoshE)

   - [ASTERISK-24812
   <https://issues.asterisk.org/jira/browse/ASTERISK-24812>] -

ARI: Creating channels through /channels resource always uses SLIN, which
results in unneeded transcoding
(Reported by Matt Jordan)

   - [ASTERISK-24797
   <https://issues.asterisk.org/jira/browse/ASTERISK-24797>] -

bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)

   - [ASTERISK-24677
   <https://issues.asterisk.org/jira/browse/ASTERISK-24677>] -

ARI GET variable on channel provides unhelpful response on non-existent
(Reported by Joshua C. Colp)

   - [ASTERISK-24785
   <https://issues.asterisk.org/jira/browse/ASTERISK-24785>] -

'Expires' header missing from 200 OK on REGISTER
(Reported by Ross Beer)

   - [ASTERISK-24724
   <https://issues.asterisk.org/jira/browse/ASTERISK-24724>] -

'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)

   - [ASTERISK-24796
   <https://issues.asterisk.org/jira/browse/ASTERISK-24796>] -

Codecs and bucket schema's prevent module unload
(Reported by Corey Farrell)

   - [ASTERISK-24814
   <https://issues.asterisk.org/jira/browse/ASTERISK-24814>] -

asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
(Reported by Corey Farrell)

   - [ASTERISK-24787
   <https://issues.asterisk.org/jira/browse/ASTERISK-24787>] -

[patch] - Microsoft exchange incompatibility for playing back messages
stored in IMAP - play_message: No origtime
(Reported by Graham Barnett)

   - [ASTERISK-22670
   <https://issues.asterisk.org/jira/browse/ASTERISK-22670>] -

Asterisk crashes when processing ISDN AoC Events
(Reported by klaus3000)

   - [ASTERISK-24689
   <https://issues.asterisk.org/jira/browse/ASTERISK-24689>] -

Segfault on hangup after outgoing PRI-Euroisdn call
(Reported by Marcel Manz)

   - [ASTERISK-24740
   <https://issues.asterisk.org/jira/browse/ASTERISK-24740>] -

[patch]Segmentation fault on aoc-e event
(Reported by Panos Gkikakis)

   - [ASTERISK-24799
   <https://issues.asterisk.org/jira/browse/ASTERISK-24799>] -

[patch] make fails with undefined reference to SSLv3_client_method
(Reported by Alexander Traud)

   - [ASTERISK-24451
   <https://issues.asterisk.org/jira/browse/ASTERISK-24451>] -

chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)

   - [ASTERISK-24700
   <https://issues.asterisk.org/jira/browse/ASTERISK-24700>] -

CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
(Reported by Zane Conkle)

   - [ASTERISK-24791
   <https://issues.asterisk.org/jira/browse/ASTERISK-24791>] -

Crash in ast_rtcp_write_report
(Reported by JoshE)

   - [ASTERISK-24085
   <https://issues.asterisk.org/jira/browse/ASTERISK-24085>] -

Documentation - We should remove or further document the 'contact' section
in pjsip.conf
(Reported by Rusty Newton)

   - [ASTERISK-24632
   <https://issues.asterisk.org/jira/browse/ASTERISK-24632>] -

install_prereq script installs pjproject without IPv6 support
(Reported by Rusty Newton)

   - [ASTERISK-24685
   <https://issues.asterisk.org/jira/browse/ASTERISK-24685>] -

"pjsip show version" CLI command
(Reported by Joshua C. Colp)

   - [ASTERISK-24768
   <https://issues.asterisk.org/jira/browse/ASTERISK-24768>] -

res_timing_pthread: file descriptor leak
(Reported by Matthias Urlichs)

   - [ASTERISK-24612
   <https://issues.asterisk.org/jira/browse/ASTERISK-24612>] -

res_pjsip: No information if a required sorcery wizard is not loaded
(Reported by Joshua C. Colp)

   - [ASTERISK-24716
   <https://issues.asterisk.org/jira/browse/ASTERISK-24716>] -

Improve pjsip log messages for presence subscription failure
(Reported by Rusty Newton)

   - [ASTERISK-24771
   <https://issues.asterisk.org/jira/browse/ASTERISK-24771>] -

${CHANNEL(pjsip)} - segfault
(Reported by Niklas Larsson)

   - [ASTERISK-24727
   <https://issues.asterisk.org/jira/browse/ASTERISK-24727>] -

PJSIP: Crash experienced during multi-Asterisk transfer scenario.
(Reported by Mark Michelson)

   - [ASTERISK-24015
   <https://issues.asterisk.org/jira/browse/ASTERISK-24015>] -

app_transfer fails with PJSIP channels
(Reported by Private Name)

   - [ASTERISK-24741
   <https://issues.asterisk.org/jira/browse/ASTERISK-24741>] -

dtls_handler causes Asterisk to crash
(Reported by Zane Conkle)

   - [ASTERISK-24701
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24752>] -

Crash in bridge_manager_service_req when bridge is destroyed by ARI during
(Reported by Richard Mudgett)

   - [ASTERISK-24772
   <https://issues.asterisk.org/jira/browse/ASTERISK-24772>] -

ODBC error in realtime sippeers when device unregisters under MariaDB
(Reported by Richard Miller)

   - [ASTERISK-24479
   <https://issues.asterisk.org/jira/browse/ASTERISK-24479>] -

Enable REF_DEBUG for module references
(Reported by Corey Farrell)

   - [ASTERISK-24742
   <https://issues.asterisk.org/jira/browse/ASTERISK-24742>] -

[patch] Fix ast_odbc_find_table function in res_odbc
(Reported by ibercom)

   - [ASTERISK-24769
   <https://issues.asterisk.org/jira/browse/ASTERISK-24769>] -

res_pjsip_sdp_rtp: Local ICE candidates leaked
(Reported by Matt Jordan)

   - [ASTERISK-24748
   <https://issues.asterisk.org/jira/browse/ASTERISK-24748>] -

res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR
messages may occur
(Reported by Joshua C. Colp)

   - [ASTERISK-24616
   <https://issues.asterisk.org/jira/browse/ASTERISK-24616>] -

Crash in res_format_attr_h264 due to invalid string copy
(Reported by Yura Kocyuba)

   - [ASTERISK-24737
   <https://issues.asterisk.org/jira/browse/ASTERISK-24737>] -

When agent not logged in, agent status shows unavailable, queue status
shows agent invalid
(Reported by Richard Mudgett)

   - [ASTERISK-24635
   <https://issues.asterisk.org/jira/browse/ASTERISK-24635>] -

PJSIP outbound PUBLISH crashes when no response is ever received
(Reported by Marco Paland)

   - [ASTERISK-24736
   <https://issues.asterisk.org/jira/browse/ASTERISK-24736>] -

Memory Leak Fixes
(Reported by Mark Michelson)

   - [ASTERISK-24646
   <https://issues.asterisk.org/jira/browse/ASTERISK-24646>] -

PJSIP changeset 4899 breaks TLS
(Reported by Stephan Eisvogel)

   - [ASTERISK-24711
   <https://issues.asterisk.org/jira/browse/ASTERISK-24711>] -

DTLS handshake broken with latest OpenSSL versions
(Reported by Jared Biel)

   - [ASTERISK-24666
   <https://issues.asterisk.org/jira/browse/ASTERISK-24666>] -

Security Vulnerability: RTP not closed after sip call using unsupported
(Reported by Y Ateya)

   - [ASTERISK-24676
   <https://issues.asterisk.org/jira/browse/ASTERISK-24676>] -

Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
(Reported by Matt Jordan)

   - [ASTERISK-24729
   <https://issues.asterisk.org/jira/browse/ASTERISK-24729>] -

Outbound registration not occuring on new registrations after reload.
(Reported by Richard Mudgett)

   - [ASTERISK-24728
   <https://issues.asterisk.org/jira/browse/ASTERISK-24728>] -

tcptls: Bad file descriptor error when reloading chan_sip
(Reported by Kevin Harwell)

   - [ASTERISK-24721
   <https://issues.asterisk.org/jira/browse/ASTERISK-24721>] -

manager: ModuleLoad action incorrectly reports 'module not found' during a
Reload operation
(Reported by Matt Jordan)

   - [ASTERISK-24715
   <https://issues.asterisk.org/jira/browse/ASTERISK-24715>] -

chan_sip: stale nonce causes failure
(Reported by Kevin Harwell)

   - [ASTERISK-24485
   <https://issues.asterisk.org/jira/browse/ASTERISK-24485>] -

res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)

   - [ASTERISK-24719
   <https://issues.asterisk.org/jira/browse/ASTERISK-24719>] -

ConfBridge recording channels get stuck when recording started/stopped more
than once
(Reported by Richard Mudgett)

   - [ASTERISK-24723
   <https://issues.asterisk.org/jira/browse/ASTERISK-24723>] -

confbridge: CLI command 'confbridge list XXXX' no longer displays user menus
(Reported by Matt Jordan)

   - [ASTERISK-24539
   <https://issues.asterisk.org/jira/browse/ASTERISK-24539>] -

Compile fails on OSX because of sem_timedwait in bridge_channel.c
(Reported by George Joseph)

   - [ASTERISK-24544
   <https://issues.asterisk.org/jira/browse/ASTERISK-24544>] -

Compile fails on OSX Yosemite because of incorrect detection of htonll and
(Reported by George Joseph)

   - [ASTERISK-24231
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24626>] -

Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)

   - [ASTERISK-24693
   <https://issues.asterisk.org/jira/browse/ASTERISK-24693>] -

Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)

   - [ASTERISK-24355
   <https://issues.asterisk.org/jira/browse/ASTERISK-24355>] -

[patch] chan_sip realtime uses case sensitive column comparison for
(Reported by HZMI8gkCvPpom0tM)

   - [ASTERISK-24709
   <https://issues.asterisk.org/jira/browse/ASTERISK-24709>] -

[patch] msg_create_from_file used by MixMonitor m() option does not queue
an MWI event
(Reported by Gareth Palmer)

   - [ASTERISK-24673
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24640>] -

Registration pending stays forever after sip reload
(Reported by Max Man)

   - [ASTERISK-24682
   <https://issues.asterisk.org/jira/browse/ASTERISK-24682>] -

app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT
are an unexpected value
(Reported by Matt Jordan)

   - [ASTERISK-24560
   <https://issues.asterisk.org/jira/browse/ASTERISK-24560>] -

Creating a named ARI bridge twice causes a crash
(Reported by Kinsey Moore)

   - [ASTERISK-24600
   <https://issues.asterisk.org/jira/browse/ASTERISK-24600>] -

Stuck IAX channels, Asterisk stops responding to most traffic, potential
(Reported by Jeff Collell)

   - [ASTERISK-24048
   <https://issues.asterisk.org/jira/browse/ASTERISK-24048>] -

[patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit
(Reported by Ben Klang)

   - [ASTERISK-24288
   <https://issues.asterisk.org/jira/browse/ASTERISK-24288>] -

[patch] - ODBC usage with app_voicemail - voicemail is not deleted after
review, hangup
(Reported by LEI FU)

   - [ASTERISK-24615
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24624>] -

Transfer to invalid extension results in hung channel.
(Reported by Zane Conkle)

   - [ASTERISK-24663
   <https://issues.asterisk.org/jira/browse/ASTERISK-24663>] -

[patch] Unnamed semaphore autoconf check fails on cross compilation
(Reported by abelbeck)

   - [ASTERISK-24655
   <https://issues.asterisk.org/jira/browse/ASTERISK-24655>] -

res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
(Reported by Kevin Harwell)

   - [ASTERISK-23991
   <https://issues.asterisk.org/jira/browse/ASTERISK-23991>] -

[patch]asterisk.pc file contains a small error in the CFlags returned
(Reported by Diederik de Groot)

   - [ASTERISK-23850
   <https://issues.asterisk.org/jira/browse/ASTERISK-23850>] -

Park Application does not respect Return Context Priority
(Reported by Andrew Nagy)

   - [ASTERISK-24665
   <https://issues.asterisk.org/jira/browse/ASTERISK-24665>] -

Configure check required for pjsip_get_dest_info()
(Reported by Mark Michelson)

   - [ASTERISK-24049
   <https://issues.asterisk.org/jira/browse/ASTERISK-24049>] -

Asterisk Manager Interface: A number of list type responses aren't using
(Reported by Jonathan Rose)

   - [ASTERISK-20744
   <https://issues.asterisk.org/jira/browse/ASTERISK-20744>] -

[patch] Security event logging does not work over syslog
(Reported by Michael Keuter)

   - [ASTERISK-24672
   <https://issues.asterisk.org/jira/browse/ASTERISK-24672>] -

[PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)

   - [ASTERISK-24474
   <https://issues.asterisk.org/jira/browse/ASTERISK-24474>] -

sip_to_pjsip.py lacks documentation and does not function
(Reported by John Kiniston)

   - [ASTERISK-24637
   <https://issues.asterisk.org/jira/browse/ASTERISK-24637>] -

Channel re-enters Stasis() when it should not
(Reported by John Bigelow)

   - [ASTERISK-24591
   <https://issues.asterisk.org/jira/browse/ASTERISK-24591>] -

Stasis() side of an ARI originated channel cannot be Redirected
(Reported by Kinsey Moore)

   - [ASTERISK-24376
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24513>] -

Local channel apparently leaked in off-nominal DTMF attended transfer
(Reported by Mark Michelson)

   - [ASTERISK-24367
   <https://issues.asterisk.org/jira/browse/ASTERISK-24367>] -

PJSIP: allow all results in failure to send INVITE
(Reported by Scott Griepentrog)

   - [ASTERISK-24267
   <https://issues.asterisk.org/jira/browse/ASTERISK-24267>] -

Queue variables associated with setinterfacevar, setqueueentryvar,
setqueuevar are not passed to local channel
(Reported by Mitch Claborn)

   - [ASTERISK-24641
   <https://issues.asterisk.org/jira/browse/ASTERISK-24641>] -

Deadlock in Trunk
(Reported by Malcolm Davenport)

   - [ASTERISK-23841
   <https://issues.asterisk.org/jira/browse/ASTERISK-23841>] -

DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.
(Reported by Richard Mudgett)

   - [ASTERISK-24628
   <https://issues.asterisk.org/jira/browse/ASTERISK-24628>] -

[patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes'
(in proxy environment)
(Reported by Karsten Wemheuer)

   - [ASTERISK-23733
   <https://issues.asterisk.org/jira/browse/ASTERISK-23733>] -

'reload acl' fails if acl.conf is not present on startup
(Reported by Richard Kenner)

   - [ASTERISK-24566
   <https://issues.asterisk.org/jira/browse/ASTERISK-24566>] -

Uninit buf in WS write
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24337
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24536>] -

AMI redirect with PJSIP fails to move extra channel
(Reported by Niklas Larsson)

   - [ASTERISK-24619
   <https://issues.asterisk.org/jira/browse/ASTERISK-24619>] -

[patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
(Reported by Walter Doekes)

   - [ASTERISK-24449
   <https://issues.asterisk.org/jira/browse/ASTERISK-24449>] -

Reinvite for T.38 UDPTL fails if SRTP is enabled
(Reported by Andreas Steinmetz)

   - [ASTERISK-22455
   <https://issues.asterisk.org/jira/browse/ASTERISK-22455>] -

Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API
(Reported by David M. Lee)

   - [ASTERISK-24614
   <https://issues.asterisk.org/jira/browse/ASTERISK-24614>] -

Deadlock when DEBUG_THREADS compiler flag enabled
(Reported by Richard Mudgett)

   - [ASTERISK-24604
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24563>] -

Direct Media calls within private network sometimes get one way audio
(Reported by Kevin Harwell)

   - [ASTERISK-24607
   <https://issues.asterisk.org/jira/browse/ASTERISK-24607>] -

res_pjsip_session: re-INVITE with declined media streams results in 488
(Reported by Matt Jordan)

   - [ASTERISK-24472
   <https://issues.asterisk.org/jira/browse/ASTERISK-24472>] -

Asterisk Crash in OpenSSL when calling over WSS from JSSIP
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24514
   <https://issues.asterisk.org/jira/browse/ASTERISK-24514>] -

res_pjsip_outbound_registration: stack overflow when using non-default
sorcery wizard
(Reported by Kevin Harwell)

   - [ASTERISK-24342
   <https://issues.asterisk.org/jira/browse/ASTERISK-24342>] -

PJSIP: Qualifying endpoints attempts to do them all at the same time.
(Reported by Richard Mudgett)

   - [ASTERISK-24556
   <https://issues.asterisk.org/jira/browse/ASTERISK-24556>] -

Asterisk 13 core dumps when calling from pjsip extension to another pjsip
(Reported by Abhay Gupta)

   - [ASTERISK-24537
   <https://issues.asterisk.org/jira/browse/ASTERISK-24537>] -

Stasis: StasisStart/StasisEnd events are not reliably transmitted during
(Reported by Matt Jordan)

   - [ASTERISK-24573
   <https://issues.asterisk.org/jira/browse/ASTERISK-24573>] -

[patch]Out of sync conversation recording when divided in multiple
(Reported by Nuno Borges)

   - [ASTERISK-24572
   <https://issues.asterisk.org/jira/browse/ASTERISK-24572>] -

[patch]App_meetme is loaded without its defaults when the configuration
file is missing
(Reported by Nuno Borges)

   - [ASTERISK-22367
   <https://issues.asterisk.org/jira/browse/ASTERISK-22367>] -

Rework CEL unit test verification step
(Reported by Kinsey Moore)

   - [ASTERISK-24516
   <https://issues.asterisk.org/jira/browse/ASTERISK-24516>] -

[patch]Asterisk segfaults when playing back voicemail under high
concurrency with an IMAP backend
(Reported by David Duncan Ross Palmer)

   - [ASTERISK-24533
   <https://issues.asterisk.org/jira/browse/ASTERISK-24533>] -

2 threads created per chan_sip entry
(Reported by xrobau)

   - [ASTERISK-24542
   <https://issues.asterisk.org/jira/browse/ASTERISK-24542>] -

[patch]Failure showing codecs via 'core show channeltype '
(Reported by snuffy)

   - [ASTERISK-24469
   <https://issues.asterisk.org/jira/browse/ASTERISK-24469>] -

Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
(Reported by Matt Jordan)

   - [ASTERISK-24534
   <https://issues.asterisk.org/jira/browse/ASTERISK-24534>] -

[patch]Register DB() as escalating to prevent users from writing to astdb
(Reported by Gareth Palmer)

   - [ASTERISK-24531
   <https://issues.asterisk.org/jira/browse/ASTERISK-24531>] -

res_pjsip_acl: ACLs not applied on initial module load
(Reported by Matt Jordan)

   - [ASTERISK-24490
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24528>] -

res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target
causes crash
(Reported by Joshua C. Colp)

   - [ASTERISK-24471
   <https://issues.asterisk.org/jira/browse/ASTERISK-24471>] -

Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from
(Reported by yaron nahum)

   - [ASTERISK-24535
   <https://issues.asterisk.org/jira/browse/ASTERISK-24535>] -

stringfields: Fix regression from fix for unintentional memory retention
and another issue exposed by the fix
(Reported by Corey Farrell)

   - [ASTERISK-24508
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-15242>] -

transmit_refer leaks sip_refer structures
(Reported by David Woolley)

   - [ASTERISK-24522
   <https://issues.asterisk.org/jira/browse/ASTERISK-24522>] -

ConfBridge: delay occurs between kicking all endmarked users when last
marked user leaves
(Reported by Matt Jordan)

   - [ASTERISK-23651
   <https://issues.asterisk.org/jira/browse/ASTERISK-23651>] -

Reloading some modules that are loaded already, results in 'No such module'
before a successful reload
(Reported by Rusty Newton)

   - [ASTERISK-24336
   <https://issues.asterisk.org/jira/browse/ASTERISK-24336>] -

PJSIP timer_min_se value under 90 causes crash
(Reported by Leon Rowland)

   - [ASTERISK-24501
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24498>] -

Segmentation fault in res_hep_rtcp on attended transfer
(Reported by Beppo Mazzucato)

   - [ASTERISK-24281
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24444>] -

PBX: Crash when generating extension for pattern matching hint
(Reported by Leandro Dardini)

   - [ASTERISK-24502
   <https://issues.asterisk.org/jira/browse/ASTERISK-24502>] -

Build fails when dev-mode, dont optimize and coverage are enabled
(Reported by Corey Farrell)

   - [ASTERISK-24505
   <https://issues.asterisk.org/jira/browse/ASTERISK-24505>] -

manager: http connections leak references
(Reported by Corey Farrell)

   - [ASTERISK-24500
   <https://issues.asterisk.org/jira/browse/ASTERISK-24500>] -

Regression introduced in chan_mgcp by SVN revision r227276
(Reported by Xavier Hienne)

   - [ASTERISK-24468
   <https://issues.asterisk.org/jira/browse/ASTERISK-24468>] -

Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly)
national symbols
(Reported by Dmitriy Bubnov)

   - [ASTERISK-24250
   <https://issues.asterisk.org/jira/browse/ASTERISK-24250>] -

[patch] Voicemail with multi-recipients To: header fix
(Reported by abelbeck)

   - [ASTERISK-24504
   <https://issues.asterisk.org/jira/browse/ASTERISK-24504>] -

chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)

   - [ASTERISK-24447
   <https://issues.asterisk.org/jira/browse/ASTERISK-24447>] -

Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
(Reported by Richard Mudgett)

   - [ASTERISK-24257
   <https://issues.asterisk.org/jira/browse/ASTERISK-24257>] -

agent must dial acceptdtmf twice to bridge to queue caller
(Reported by Steve Pitts)

   - [ASTERISK-24492
   <https://issues.asterisk.org/jira/browse/ASTERISK-24492>] -

main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
(Reported by Corey Farrell)

   - [ASTERISK-24491
   <https://issues.asterisk.org/jira/browse/ASTERISK-24491>] -

Memory leak in res_hep
(Reported by Zane Conkle)

   - [ASTERISK-24307
   <https://issues.asterisk.org/jira/browse/ASTERISK-24307>] -

Unintentional memory retention in stringfields
(Reported by Etienne Lessard)

   - [ASTERISK-24438
   <https://issues.asterisk.org/jira/browse/ASTERISK-24438>] -

res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
(Reported by Melissa Shepherd)

   - [ASTERISK-20127
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24487>] -

configuration: sections should be loadable as template even when not marked
(Reported by Scott Griepentrog)

   - [ASTERISK-24482
   <https://issues.asterisk.org/jira/browse/ASTERISK-24482>] -

func_talkdetect: Fix stasis message leak in audiohook callback
(Reported by Corey Farrell)

   - [ASTERISK-24480
   <https://issues.asterisk.org/jira/browse/ASTERISK-24480>] -

res_http_websockets: Module reference decrease below zero
(Reported by Corey Farrell)

   - [ASTERISK-24476
   <https://issues.asterisk.org/jira/browse/ASTERISK-24476>] -

main/app.c / app_voicemail: ast_writestream leaks
(Reported by Corey Farrell)

   - [ASTERISK-22409
   <https://issues.asterisk.org/jira/browse/ASTERISK-22409>] -

Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after
(Reported by Corey Farrell)

   - [ASTERISK-24411
   <https://issues.asterisk.org/jira/browse/ASTERISK-24411>] -

[patch] Status of outbound registration is not changed upon unregistering.
(Reported by John Bigelow)

   - [ASTERISK-24432
   <https://issues.asterisk.org/jira/browse/ASTERISK-24432>] -

Install refcounter.py when REF_DEBUG is enabled
(Reported by Corey Farrell)

   - [ASTERISK-24466
   <https://issues.asterisk.org/jira/browse/ASTERISK-24466>] -

app_queue: fix a couple leaks to struct call_queue
(Reported by Corey Farrell)

   - [ASTERISK-24465
   <https://issues.asterisk.org/jira/browse/ASTERISK-24465>] -

audiohooks list leaks reference to formats
(Reported by Corey Farrell)

   - [ASTERISK-24462
   <https://issues.asterisk.org/jira/browse/ASTERISK-24462>] -

res_pjsip: Stale qualify statistics after disablementation
(Reported by Kevin Harwell)

   - [ASTERISK-24190
   <https://issues.asterisk.org/jira/browse/ASTERISK-24190>] -

IMAP voicemail causes segfault
(Reported by Nick Adams)

   - [ASTERISK-24304
   <https://issues.asterisk.org/jira/browse/ASTERISK-24304>] -

asterisk crashing randomly because of unistim channel
(Reported by dhanapathy sathya)

   - [ASTERISK-24458
   <https://issues.asterisk.org/jira/browse/ASTERISK-24458>] -

chan_phone fails to build on big endian systems
(Reported by Tzafrir Cohen)

   - [ASTERISK-24457
   <https://issues.asterisk.org/jira/browse/ASTERISK-24457>] -

res_fax: fax gateway frames leak
(Reported by Corey Farrell)

   - [ASTERISK-24453
   <https://issues.asterisk.org/jira/browse/ASTERISK-24453>] -

manager: acl_change_sub leaks
(Reported by Corey Farrell)

   - [ASTERISK-24437
   <https://issues.asterisk.org/jira/browse/ASTERISK-24437>] -

Review implementation of ast_bridge_impart for leaks and document proper
(Reported by Scott Griepentrog)

   - [ASTERISK-24430
   <https://issues.asterisk.org/jira/browse/ASTERISK-24430>] -

missing letter "p" in word response in OriginateResponse event documentation
(Reported by Dafi Ni)

   - [ASTERISK-24323
   <https://issues.asterisk.org/jira/browse/ASTERISK-24323>] -

Bug in documentation AGI STREAM FILE CONTROL
(Reported by Martin Cisárik)

   - [ASTERISK-24419
   <https://issues.asterisk.org/jira/browse/ASTERISK-24419>] -

Incorrect syntax for setting language in configs/extensions.conf.sample
(Reported by Ben Klang)

   - [ASTERISK-24454
   <https://issues.asterisk.org/jira/browse/ASTERISK-24454>] -

app_queue: ao2_iterator not destroyed, causing leak
(Reported by Corey Farrell)

   - [ASTERISK-24455
   <https://issues.asterisk.org/jira/browse/ASTERISK-24455>] -

func_cdr: CDR_PROP leaks payload
(Reported by Corey Farrell)

   - [ASTERISK-24435
   <https://issues.asterisk.org/jira/browse/ASTERISK-24435>] -

Asterisk 13 with TC400P segfault
(Reported by Marian Koniuszko)

   - [ASTERISK-24122
   <https://issues.asterisk.org/jira/browse/ASTERISK-24122>] -

Documentaton for res_pjsip option use_avpf needs to be fixed
(Reported by James Van Vleet)

   - [ASTERISK-24381
   <https://issues.asterisk.org/jira/browse/ASTERISK-24381>] -

res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to
erroneous 488 rejections
(Reported by Matt Jordan)

   - [ASTERISK-24063
   <https://issues.asterisk.org/jira/browse/ASTERISK-24063>] -

[patch]Asterisk does not respect outbound proxy when sending qualify
(Reported by Damian Ivereigh)

   - [ASTERISK-24415
   <https://issues.asterisk.org/jira/browse/ASTERISK-24415>] -

Missing AMI VarSet events when channels inherit variables.
(Reported by Richard Mudgett)

   - [ASTERISK-24327
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24426>] -

CDR Batch mode: size used as time value after first expire
(Reported by Shane Blaser)

   - [ASTERISK-24312
   <https://issues.asterisk.org/jira/browse/ASTERISK-24312>] -

SIGABRT when improperly configured realtime pjsip
(Reported by Dafi Ni)

   - [ASTERISK-23846
   <https://issues.asterisk.org/jira/browse/ASTERISK-23846>] -

Unistim multilines. Loss of voice after second call drops (on a second
(Reported by Rustam Khankishyiev)

   - [ASTERISK-24413
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24393>] -

rtptimeout=0 doesn't disable rtptimeout
(Reported by Dmitry Melekhov)

   - [ASTERISK-24321
   <https://issues.asterisk.org/jira/browse/ASTERISK-24321>] -

SIP deadlock when running automated queues tests
(Reported by Steve Pitts)

   - [ASTERISK-24392
   <https://issues.asterisk.org/jira/browse/ASTERISK-24392>] -

res_fax: fax gateway sessions leak
(Reported by Corey Farrell)

   - [ASTERISK-24237
   <https://issues.asterisk.org/jira/browse/ASTERISK-24237>] -

CDR: FRACK With PJSIP blonde transfer.
(Reported by Richard Mudgett)

   - [ASTERISK-24394
   <https://issues.asterisk.org/jira/browse/ASTERISK-24394>] -

CDR: FRACK with PJSIP directed pickup.
(Reported by Richard Mudgett)

   - [ASTERISK-18923
   <https://issues.asterisk.org/jira/browse/ASTERISK-18923>] -

res_fax_spandsp usage counter is wrong
(Reported by Grigoriy Puzankin)

   - [ASTERISK-22791
   <https://issues.asterisk.org/jira/browse/ASTERISK-22791>] -

asterisk sends Re-INVITE after receiving a BYE
(Reported by not here)

   - [ASTERISK-13797
   <https://issues.asterisk.org/jira/browse/ASTERISK-13797>] -

[patch] relax badshell tilde test
(Reported by Tzafrir Cohen)

   - [ASTERISK-24325
   <https://issues.asterisk.org/jira/browse/ASTERISK-24325>] -

res_calendar_ews: cannot be used with neon 0.30
(Reported by Tzafrir Cohen)

   - [ASTERISK-24406
   <https://issues.asterisk.org/jira/browse/ASTERISK-24406>] -

Some caller ID strings are parsed differently since 11.13.0
(Reported by Etienne Lessard)

   - [ASTERISK-24387
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-20784>] -

Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
(Reported by NITESH BANSAL)

   - [ASTERISK-15879
   <https://issues.asterisk.org/jira/browse/ASTERISK-15879>] -

[patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP
channel leak
(Reported by Torrey Searle)

   - [ASTERISK-24383
   <https://issues.asterisk.org/jira/browse/ASTERISK-24383>] -

res_rtp_asterisk: Crash if no candidates received for component
(Reported by Kevin Harwell)

   - [ASTERISK-24011
   <https://issues.asterisk.org/jira/browse/ASTERISK-24011>] -

[patch]safe_asterisk tries to set ulimit -n too high on linux systems with
lots of RAM
(Reported by Michael Myles)

   - [ASTERISK-24326
   <https://issues.asterisk.org/jira/browse/ASTERISK-24326>] -

res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted
(Reported by Joshua C. Colp)

   - [ASTERISK-24389
   <https://issues.asterisk.org/jira/browse/ASTERISK-24389>] -

chan_iax2: Unit test on Bamboo failing
(Reported by Kevin Harwell)

   - [ASTERISK-24398
   <https://issues.asterisk.org/jira/browse/ASTERISK-24398>] -

Initialize auth_rejection_permanent on client state to the configuration
parameter value
(Reported by Matt Jordan)

   - [ASTERISK-24354
   <https://issues.asterisk.org/jira/browse/ASTERISK-24354>] -

AMI sendMessage closes AMI connection on error
(Reported by Peter Katzmann)

   - [ASTERISK-24224
   <https://issues.asterisk.org/jira/browse/ASTERISK-24224>] -

When using Bridge() dialplan application, surrogate channel appears in list
and call count is inflated.
(Reported by Mark Michelson)

   - [ASTERISK-24370
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24368>] -

res_pjsip_pubsub: Subscription persistence causes crash when
re-constructing stored subscription
(Reported by Matt Jordan)

   - [ASTERISK-24378
   <https://issues.asterisk.org/jira/browse/ASTERISK-24378>] -

Release AMI connections on shutdown
(Reported by Corey Farrell)

   - [ASTERISK-24384
   <https://issues.asterisk.org/jira/browse/ASTERISK-24384>] -

chan_motif: format capabilities leak on module load error
(Reported by Corey Farrell)

   - [ASTERISK-24199
   <https://issues.asterisk.org/jira/browse/ASTERISK-24199>] -

'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid
(Reported by Joshua C. Colp)

   - [ASTERISK-24195
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24356>] -

PJSIP: Directed pickup causes deadlock
(Reported by Richard Mudgett)

   - [ASTERISK-24262
   <https://issues.asterisk.org/jira/browse/ASTERISK-24262>] -

AMI CoreShowChannel missing several output fields and event documentation
(Reported by Mitch Claborn)

   - [ASTERISK-23781
   <https://issues.asterisk.org/jira/browse/ASTERISK-23781>] -

outgoing missing as enum from contrib/ast-db-manage/config
(Reported by Stephen More)

   - [ASTERISK-24222
   <https://issues.asterisk.org/jira/browse/ASTERISK-24222>] -

PJSIP: Failed assertions when placing a call with no allow= specified
(Reported by Mark Michelson)

   - [ASTERISK-24362
   <https://issues.asterisk.org/jira/browse/ASTERISK-24362>] -

res_hep leaks reference to configuration
(Reported by Corey Farrell)

   - [ASTERISK-22945
   <https://issues.asterisk.org/jira/browse/ASTERISK-22945>] -

[patch] Memory leaks in chan_sip.c with realtime peers
(Reported by ibercom)

   - [ASTERISK-24350
   <https://issues.asterisk.org/jira/browse/ASTERISK-24350>] -

PJSIP shows commands prints unneeded headers
(Reported by snuffy)

   - [ASTERISK-20567
   <https://issues.asterisk.org/jira/browse/ASTERISK-20567>] -

bashism in autosupport
(Reported by Tzafrir Cohen)

   - [ASTERISK-24357
   <https://issues.asterisk.org/jira/browse/ASTERISK-24357>] -

[fax] Out of bounds error in update_modem_bits
(Reported by Jeremy Lainé)

   - [ASTERISK-24348
   <https://issues.asterisk.org/jira/browse/ASTERISK-24348>] -

Built-in editline tab complete segfault with MALLOC_DEBUG
(Reported by Walter Doekes)

   - [ASTERISK-23768
   <https://issues.asterisk.org/jira/browse/ASTERISK-23768>] -

[patch] Asterisk man page contains a (new) unquoted minus sign
(Reported by Jeremy Lainé)

   - [ASTERISK-24295
   <https://issues.asterisk.org/jira/browse/ASTERISK-24295>] -

crash: creating out of dialog OPTIONS request crashes
(Reported by Rogger Padilla)

   - [ASTERISK-24335
   <https://issues.asterisk.org/jira/browse/ASTERISK-24335>] -

[PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of
rejected calls
(Reported by Torrey Searle)

   - [ASTERISK-24339
   <https://issues.asterisk.org/jira/browse/ASTERISK-24339>] -

Swagger API Docs have incorrect basePath
(Reported by Bradley Watkins)

   - [ASTERISK-24265
   <https://issues.asterisk.org/jira/browse/ASTERISK-24265>] -

segfault in asterisk when try to make call to IAX
(Reported by Dafi Ni)

   - [ASTERISK-24290
   <https://issues.asterisk.org/jira/browse/ASTERISK-24290>] -

Endpoint identifier match value fails to parse when CIDR network format is
(Reported by Ray Crumrine)

   - [ASTERISK-24301
   <https://issues.asterisk.org/jira/browse/ASTERISK-24301>] -

Security: Out of call MESSAGE requests processed via Message channel driver
can crash Asterisk
(Reported by Matt Jordan)

   - [ASTERISK-24136
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24161>] -

PJSIPShowEndpoint gives inaccurate count of list items
(Reported by Mark Michelson)

   - [ASTERISK-24331
   <https://issues.asterisk.org/jira/browse/ASTERISK-24331>] -

Unexpected Errors in Asterisk Manager Interface Output
(Reported by xrobau)

   - [ASTERISK-24328
   <https://issues.asterisk.org/jira/browse/ASTERISK-24328>] -

Use of MixMonitor 'm' option results in 0 duration vm description file
(Reported by Scott Griepentrog)

   - [ASTERISK-23577
   <https://issues.asterisk.org/jira/browse/ASTERISK-23577>] -

res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is
(Reported by Jay Jideliov)

   - [ASTERISK-23634
   <https://issues.asterisk.org/jira/browse/ASTERISK-23634>] -

With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC
(avpg/encryption/icesupport) calls
(Reported by Roman Skvirsky)

   - [ASTERISK-24249
   <https://issues.asterisk.org/jira/browse/ASTERISK-24249>] -

SIP debugs do not stop
(Reported by Avinash Mohod)

   - [ASTERISK-24181
   <https://issues.asterisk.org/jira/browse/ASTERISK-24181>] -

RLS: Large lists don't get sent because they exceed the PJSIP message
length limit
(Reported by Jonathan Rose)

   - [ASTERISK-24254
   <https://issues.asterisk.org/jira/browse/ASTERISK-24254>] -

CDRs: Application/args/dialplan CEP updated during dial operation
(Reported by Matt Jordan)

   - [ASTERISK-24241
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24208>] -

Channels with CDR Information Remain Active Even After ConfBrige Is Ended
(Reported by Frankie Chin)

   - [ASTERISK-24223
   <https://issues.asterisk.org/jira/browse/ASTERISK-24223>] -

Gibberish Call-ID on Local channel on origination
(Reported by Mark Michelson)

   - [ASTERISK-24271
   <https://issues.asterisk.org/jira/browse/ASTERISK-24271>] -

Unable to make WebRTC call through chan_PJSIP nor chan_SIP
(Reported by Dafi Ni)

   - [ASTERISK-24212
   <https://issues.asterisk.org/jira/browse/ASTERISK-24212>] -

testsuite: Sporadic crash due to assert on stopping RTP engine
(Reported by Matt Jordan)

   - [ASTERISK-24264
   <https://issues.asterisk.org/jira/browse/ASTERISK-24264>] -

ARI: Adding a channel to a holding bridge automatically starts MOH
(Reported by Samuel Galarneau)

   - [ASTERISK-23767
   <https://issues.asterisk.org/jira/browse/ASTERISK-23767>] -

[patch] Dynamic IAX2 registration stops trying if ever not able to resolve
(Reported by David Herselman)

   - [ASTERISK-24280
   <https://issues.asterisk.org/jira/browse/ASTERISK-24280>] -

Add 'rtpbindaddr' setting for chan_sip
(Reported by Paul Belanger)

   - [ASTERISK-24019
   <https://issues.asterisk.org/jira/browse/ASTERISK-24019>] -

When a Music On Hold stream starts it restarts at beginning of file.
(Reported by Jason Richards)

   - [ASTERISK-24143
   <https://issues.asterisk.org/jira/browse/ASTERISK-24143>] -

pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
(Reported by Aleksei Kulakov)

   - [ASTERISK-23997
   <https://issues.asterisk.org/jira/browse/ASTERISK-23997>] -

chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24147
   <https://issues.asterisk.org/jira/browse/ASTERISK-24147>] -

ARI: channel hangup crashes asterisk process
(Reported by Edvin Vidmar)

   - [ASTERISK-23994
   <https://issues.asterisk.org/jira/browse/ASTERISK-23994>] -

res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified
(Reported by Private Name)

   - [ASTERISK-22252
   <https://issues.asterisk.org/jira/browse/ASTERISK-22252>] -

res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks
(Reported by Walter Doekes)

   - [ASTERISK-24178
   <https://issues.asterisk.org/jira/browse/ASTERISK-24178>] -

[patch]fromdomainport used even if not set
(Reported by Elazar Broad)

   - [ASTERISK-24229
   <https://issues.asterisk.org/jira/browse/ASTERISK-24229>] -

ARI: playback of sounds implicitly answers channel, preventing early media
(Reported by Matt Jordan)

   - [ASTERISK-24245
   <https://issues.asterisk.org/jira/browse/ASTERISK-24245>] -

gcc 4.1.2 complains of files that do not end with newlines
(Reported by Shaun Ruffell)

   - [ASTERISK-24246
   <https://issues.asterisk.org/jira/browse/ASTERISK-24246>] -

Quiet warning about type qualifiers ignored on function return type
(Reported by Shaun Ruffell)

   - [ASTERISK-24043
   <https://issues.asterisk.org/jira/browse/ASTERISK-24043>] -

ARI /continue fails to actually continue into the dialplan
(Reported by Krandon Bruse)

   - [ASTERISK-24215
   <https://issues.asterisk.org/jira/browse/ASTERISK-24215>] -

testsuite: ARI Live Dangerously test fails due to wrong response code from
(Reported by Matt Jordan)

   - [ASTERISK-24134
   <https://issues.asterisk.org/jira/browse/ASTERISK-24134>] -

ARI: GET /channels/{channel_id}/variable for channel in dialplan returns
409 conflict
(Reported by Matt Jordan)

   - [ASTERISK-24138
   <https://issues.asterisk.org/jira/browse/ASTERISK-24138>] -

dial: Call forwarding information presented through AMI/ARI is wrong
(Reported by Matt Jordan)

   - [ASTERISK-24234
   <https://issues.asterisk.org/jira/browse/ASTERISK-24234>] -

app_meetme: Crash on conference shutdown due to NULL channel passed to
(Reported by Shaun Ruffell)

   - [ASTERISK-24225
   <https://issues.asterisk.org/jira/browse/ASTERISK-24225>] -

Dial option z is broken
(Reported by dimitripietro)

   - [ASTERISK-24032
   <https://issues.asterisk.org/jira/browse/ASTERISK-24032>] -

Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
(Reported by Kilburn)

   - [ASTERISK-24027
   <https://issues.asterisk.org/jira/browse/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
   <https://issues.asterisk.org/jira/browse/ASTERISK-24236>] -

res_hep_rtcp: Module incorrectly depends on pjsip
(Reported by Matt Jordan)

   - [ASTERISK-23508
   <https://issues.asterisk.org/jira/browse/ASTERISK-23508>] -

Memory Corruption in __ast_string_field_ptr_build_va
(Reported by Arnd Schmitter)

*Improvements made in this release:*

   - [ASTERISK-28658
   <https://issues.asterisk.org/jira/browse/ASTERISK-28658>] -

app_confbridge: Add support for setting maximum sample rate
(Reported by Joshua C. Colp)

   - [ASTERISK-28326
   <https://issues.asterisk.org/jira/browse/ASTERISK-28326>] -

ari: Added timestamp for some ari events.
(Reported by sungtae kim)

   - [ASTERISK-28317
   <https://issues.asterisk.org/jira/browse/ASTERISK-28317>] -

Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL
(Reported by Cirillo Ferreira)

   - [ASTERISK-28279
   <https://issues.asterisk.org/jira/browse/ASTERISK-28279>] -

Added creation timestamp for bridge
(Reported by sungtae kim)

   - [ASTERISK-27483
   <https://issues.asterisk.org/jira/browse/ASTERISK-27483>] -

Allow wrapuptime to be set for each queue member
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-28055
   <https://issues.asterisk.org/jira/browse/ASTERISK-28055>] -

app_queue: Per-member wrapup time missing from AddQueueMember application
(Reported by Niksa Baldun)

   - [ASTERISK-28292
   <https://issues.asterisk.org/jira/browse/ASTERISK-28292>] -

Changed to show all channel stats including wrong media
(Reported by sungtae kim)

   - [ASTERISK-28253
   <https://issues.asterisk.org/jira/browse/ASTERISK-28253>] -

res_pjsip_session: Adding rtcp stats result into the session
(Reported by sungtae kim)

   - [ASTERISK-28246
   <https://issues.asterisk.org/jira/browse/ASTERISK-28246>] -

Support skipping on the g726 format
(Reported by Eyal Hasson)

   - [ASTERISK-28196
   <https://issues.asterisk.org/jira/browse/ASTERISK-28196>] -

bridge_softmix: Does not support WebRTC source with multi video tracks.
(Reported by Xiemin Chen)

   - [ASTERISK-28198
   <https://issues.asterisk.org/jira/browse/ASTERISK-28198>] -

res_ari: Add new hangup causes for ARI Channel DELETE command
(Reported by Sebastian Damm)

   - [ASTERISK-28144
   <https://issues.asterisk.org/jira/browse/ASTERISK-28144>] -

[patch] New function PJSIP_PARSE_URI to parse an URI and return a specified
part of the URI
(Reported by Alexei Gradinari)

   - [ASTERISK-28136
   <https://issues.asterisk.org/jira/browse/ASTERISK-28136>] -

Allow the sip_to_pjsip script to be used in a pipe
(Reported by Pascal Cadotte Michaud)

   - [ASTERISK-28046
   <https://issues.asterisk.org/jira/browse/ASTERISK-28046>] -

Remove stale nonoptreq references
(Reported by Walter Doekes)

   - [ASTERISK-27164
   <https://issues.asterisk.org/jira/browse/ASTERISK-27164>] -

[patch] Add IPv6 Support for DUNDi
(Reported by Adam Secombe)

   - [ASTERISK-28006
   <https://issues.asterisk.org/jira/browse/ASTERISK-28006>] -

PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID
(Reported by Eric Dantie)

   - [ASTERISK-27995
   <https://issues.asterisk.org/jira/browse/ASTERISK-27995>] -

pjproject_bundled: Find shared libraries in root --with-ssl=PATH.
(Reported by Alexander Traud)

   - [ASTERISK-27993
   <https://issues.asterisk.org/jira/browse/ASTERISK-27993>] -

pjsip_wizard example gives wrong info about unsupported SRV records
(Reported by Jonathan Harris)

   - [ASTERISK-27970
   <https://issues.asterisk.org/jira/browse/ASTERISK-27970>] -

res_rtp_asterisk: T.140 packets containing backspace or end of line are
merged with regular text and it causes some UA to break
(Reported by Emmanuel BUU)

   - [ASTERISK-22825
   <https://issues.asterisk.org/jira/browse/ASTERISK-22825>] -

Dialplan Function for Checking Parking Lot Slot
(Reported by JoshE)

   - [ASTERISK-27912
   <https://issues.asterisk.org/jira/browse/ASTERISK-27912>] -

[PATCH] Add predial handler to app_queue
(Reported by Kristian Høgh)

   - [ASTERISK-27929
   <https://issues.asterisk.org/jira/browse/ASTERISK-27929>] -

[patch] BuildSystem: Enable autotools in Solaris 11.
(Reported by Alexander Traud)

   - [ASTERISK-27752
   <https://issues.asterisk.org/jira/browse/ASTERISK-27752>] -

Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)

   - [ASTERISK-27910
   <https://issues.asterisk.org/jira/browse/ASTERISK-27910>] -

[patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated.
(Reported by Alexander Traud)

   - [ASTERISK-27906
   <https://issues.asterisk.org/jira/browse/ASTERISK-27906>] -

[patch] res_crypto: Allow OpenSSL configured with no-deprecated.
(Reported by Alexander Traud)

   - [ASTERISK-27877
   <https://issues.asterisk.org/jira/browse/ASTERISK-27877>] -

app_confbridge: Add talking indicator for ConfBridgeList AMI response
(Reported by William McCall)

   - [ASTERISK-27873
   <https://issues.asterisk.org/jira/browse/ASTERISK-27873>] -

documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event
(Reported by Alessandro Polidori)

   - [ASTERISK-27846
   <https://issues.asterisk.org/jira/browse/ASTERISK-27846>] -

ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)

   - [ASTERISK-27867
   <https://issues.asterisk.org/jira/browse/ASTERISK-27867>] -

[patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated.
(Reported by Alexander Traud)

   - [ASTERISK-27796
   <https://issues.asterisk.org/jira/browse/ASTERISK-27796>] -

res_hep: Allow create_address to resolve a provided hostname
(Reported by Sebastian Gutierrez)

   - [ASTERISK-27820
   <https://issues.asterisk.org/jira/browse/ASTERISK-27820>] -

[patch] Add DragonFly BSD.
(Reported by Alexander Traud)

   - [ASTERISK-25129
   <https://issues.asterisk.org/jira/browse/ASTERISK-25129>] -

wrong automatic ras address assignment if multihomed
(Reported by Dmitry Melekhov)

   - [ASTERISK-27793
   <https://issues.asterisk.org/jira/browse/ASTERISK-27793>] -

cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)

   - [ASTERISK-27697
   <https://issues.asterisk.org/jira/browse/ASTERISK-27697>] -

Enable in-dialog NOTIFY on chan_pjsip channels
(Reported by Nathan Bruning)

   - [ASTERISK-27770
   <https://issues.asterisk.org/jira/browse/ASTERISK-27770>] -

[patch] install_prereq: Add Slackware (somehow).
(Reported by Alexander Traud)

   - [ASTERISK-27769
   <https://issues.asterisk.org/jira/browse/ASTERISK-27769>] -

[patch] install_prereq: Add Gentoo Linux.
(Reported by Alexander Traud)

   - [ASTERISK-27738
   <https://issues.asterisk.org/jira/browse/ASTERISK-27738>] -

[patch] install_prereq: Add Arch Linux.
(Reported by Alexander Traud)

   - [ASTERISK-27736
   <https://issues.asterisk.org/jira/browse/ASTERISK-27736>] -

[patch] install_prereq: Add SUSE.
(Reported by Alexander Traud)

   - [ASTERISK-27253
   <https://issues.asterisk.org/jira/browse/ASTERISK-27253>] -

[patch] libsrtp-2.1.x support
(Reported by Alexander Traud)

   - [ASTERISK-27728
   <https://issues.asterisk.org/jira/browse/ASTERISK-27728>] -

[patch] BuildSystem: Add NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27730
   <https://issues.asterisk.org/jira/browse/ASTERISK-27730>] -

PJSIP: Update bundled PJPROJECT to version 2.7.2
(Reported by Richard Mudgett)

   - [ASTERISK-27729
   <https://issues.asterisk.org/jira/browse/ASTERISK-27729>] -

[patch] install_prereq: Add NetBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27683
   <https://issues.asterisk.org/jira/browse/ASTERISK-27683>] -

[patch] BuildSystem: Allow newer autotools on OpenBSD.
(Reported by Alexander Traud)

   - [ASTERISK-27348
   <https://issues.asterisk.org/jira/browse/ASTERISK-27348>] -

[patch]contrib/scripts: add a way to migrate from chan_sip to chan_pjsip
(Reported by Torrey Searle)

   - [ASTERISK-27661
   <https://issues.asterisk.org/jira/browse/ASTERISK-27661>] -

Add new AMI Event for Load, Unload
(Reported by sungtae kim)

   - [ASTERISK-27651
   <https://issues.asterisk.org/jira/browse/ASTERISK-27651>] -

app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to
ConfbridgeList AMI events
(Reported by Richard Mudgett)

   - [ASTERISK-27647
   <https://issues.asterisk.org/jira/browse/ASTERISK-27647>] -

app_confbridge/bridge_softmix: When channel muted report talking stopped if
was talking.
(Reported by Richard Mudgett)

   - [ASTERISK-27084
   <https://issues.asterisk.org/jira/browse/ASTERISK-27084>] -

Reduce verbosity while loading PBX extensions.
(Reported by Ludovic Gasc (Eyepea))

   - [ASTERISK-24372
   <https://issues.asterisk.org/jira/browse/ASTERISK-24372>] -

[patch] Add config option to play a prompt to the "winner" in app_followme
(Reported by Graham Mainwaring)

   - [ASTERISK-27537
   <https://issues.asterisk.org/jira/browse/ASTERISK-27537>] -

res_pjsip: Add new AMI Action for PJSIPShowAors
(Reported by sungtae kim)

   - [ASTERISK-24297
   <https://issues.asterisk.org/jira/browse/ASTERISK-24297>] -

cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)

   - [ASTERISK-27470
   <https://issues.asterisk.org/jira/browse/ASTERISK-27470>] -

Add new object for VoicemailUserEntry
(Reported by sungtae kim)

   - [ASTERISK-27461
   <https://issues.asterisk.org/jira/browse/ASTERISK-27461>] -

3PCC patch for AMI "SIPnotify"
(Reported by Yasuhiko Kamata)

   - [ASTERISK-27449
   <https://issues.asterisk.org/jira/browse/ASTERISK-27449>] -

[PATCH] When failing to acquire target during attended transfer, display
wanted extension
(Reported by Niklas Larsson)

   - [ASTERISK-27456
   <https://issues.asterisk.org/jira/browse/ASTERISK-27456>] -

app_voicemail: Add new object for VoicemailUserEntry
(Reported by sungtae kim)

   - [ASTERISK-27380
   <https://issues.asterisk.org/jira/browse/ASTERISK-27380>] -

ast_coredumper: allow pointing out the asterisk binary explicitly
(Reported by Tzafrir Cohen)

   - [ASTERISK-23556
   <https://issues.asterisk.org/jira/browse/ASTERISK-23556>] -

Compilation warning for invert.c (array subscript is above array bounds)
(Reported by Marcello Ceschia)

   - [ASTERISK-27359
   <https://issues.asterisk.org/jira/browse/ASTERISK-27359>] -

pjproject bundled: Don't disable assertions when --enable-dev-mode is used.
(Reported by Corey Farrell)

   - [ASTERISK-27355
   <https://issues.asterisk.org/jira/browse/ASTERISK-27355>] -

Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)

   - [ASTERISK-27335
   <https://issues.asterisk.org/jira/browse/ASTERISK-27335>] -

CDR performance needs improvement.
(Reported by Richard Mudgett)

   - [ASTERISK-27278
   <https://issues.asterisk.org/jira/browse/ASTERISK-27278>] -

[patch] chan_sip: Provide access to read the full SIP Request-URI from
(Reported by David J. Pryke)

   - [ASTERISK-27255
   <https://issues.asterisk.org/jira/browse/ASTERISK-27255>] -

alembic: Add support for Microsoft SQL server
(Reported by Florian Floimair)

   - [ASTERISK-27220
   <https://issues.asterisk.org/jira/browse/ASTERISK-27220>] -

Enable CHANNEL function to get from and to tag from SIP Headers
(Reported by Andre Nazario)

   - [ASTERISK-27169
   <https://issues.asterisk.org/jira/browse/ASTERISK-27169>] -

Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)

   - [ASTERISK-27173
   <https://issues.asterisk.org/jira/browse/ASTERISK-27173>] -

Support for GMIME 3.0
(Reported by Tzafrir Cohen)

   - [ASTERISK-27085
   <https://issues.asterisk.org/jira/browse/ASTERISK-27085>] -

[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
(Reported by Torrey Searle)

   - [ASTERISK-27066
   <https://issues.asterisk.org/jira/browse/ASTERISK-27066>] -

res_pjsip: Add DTMF INFO Failback mode
(Reported by Torrey Searle)

   - [ASTERISK-27092
   <https://issues.asterisk.org/jira/browse/ASTERISK-27092>] -

[patch] app_queue: Add Priority to AMI QueueStatus
(Reported by Niklas Larsson)

   - [ASTERISK-27068
   <https://issues.asterisk.org/jira/browse/ASTERISK-27068>] -

app_voicemail: Add global option "imap_poll_logout" to specify post-polling
(Reported by Alexei Gradinari)

   - [ASTERISK-26230
   <https://issues.asterisk.org/jira/browse/ASTERISK-26230>] -

[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on
(Reported by Alexei Gradinari)

   - [ASTERISK-27043
   <https://issues.asterisk.org/jira/browse/ASTERISK-27043>] -

Core/BuildSystem: Add defines to fix build with LibreSSL
(Reported by Guido Falsi)

   - [ASTERISK-27042
   <https://issues.asterisk.org/jira/browse/ASTERISK-27042>] -

Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h
(Reported by Guido Falsi)

   - [ASTERISK-26419
   <https://issues.asterisk.org/jira/browse/ASTERISK-26419>] -

audiohooks: Remove redundant codec translations when using audiohooks
(Reported by Michael Walton)

   - [ASTERISK-26976
   <https://issues.asterisk.org/jira/browse/ASTERISK-26976>] -

libsrtp-2.x.x support
(Reported by Alex)

   - [ASTERISK-27014
   <https://issues.asterisk.org/jira/browse/ASTERISK-27014>] -

configurable busy_timeout in sqlite backends
(Reported by Marek Cervenka)

   - [ASTERISK-26124
   <https://issues.asterisk.org/jira/browse/ASTERISK-26124>] -

res_agi: Set audio format for EAGI audio stream
(Reported by John Fawcett)

   - [ASTERISK-26088
   <https://issues.asterisk.org/jira/browse/ASTERISK-26088>] -

Investigate heavy memory utilization by res_pjsip_pubsub
(Reported by Richard Mudgett)

   - [ASTERISK-26427
   <https://issues.asterisk.org/jira/browse/ASTERISK-26427>] -

res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp
when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))

   - [ASTERISK-26932
   <https://issues.asterisk.org/jira/browse/ASTERISK-26932>] -

[patch] SIP/SDP: No rtpmap for static RTP payload IDs
(Reported by Alexander Traud)

   - [ASTERISK-26864
   <https://issues.asterisk.org/jira/browse/ASTERISK-26864>] -

res_pjsip_session: Add support for overlap dialling
(Reported by Richard Begg)

   - [ASTERISK-26846
   <https://issues.asterisk.org/jira/browse/ASTERISK-26846>] -

chan_sip: Add rtcp-mux support
(Reported by Sean Bright)

   - [ASTERISK-26568
   <https://issues.asterisk.org/jira/browse/ASTERISK-26568>] -

pbx_spool: OUTGOING_RETRY variable
(Reported by Roman Shubovich)

   - [ASTERISK-26292
   <https://issues.asterisk.org/jira/browse/ASTERISK-26292>] -

app_confbridge: 3D-Conferencing via Binaural Synthesis
(Reported by Dennis Guse)

   - [ASTERISK-23828
   <https://issues.asterisk.org/jira/browse/ASTERISK-23828>] -

pjsip - Need a command to list active SIP subscriptions
(Reported by Rusty Newton)

   - [ASTERISK-26559
   <https://issues.asterisk.org/jira/browse/ASTERISK-26559>] -

app_queue: New service level calculation
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26658
   <https://issues.asterisk.org/jira/browse/ASTERISK-26658>] -

Add ability for dialplan show to display filenames/line numbers of
registered extensions
(Reported by Jonathan R. Rose)

   - [ASTERISK-26527
   <https://issues.asterisk.org/jira/browse/ASTERISK-26527>] -

Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
(Reported by Badalian Vyacheslav)

   - [ASTERISK-22992
   <https://issues.asterisk.org/jira/browse/ASTERISK-22992>] -

[patch]Asterisk app_originate doesn't allow setting Caller*ID on the
originating channel
(Reported by Anthony Messina)

   - [ASTERISK-26624
   <https://issues.asterisk.org/jira/browse/ASTERISK-26624>] -

res_calendar_caldav: Add support for gmail
(Reported by Eduardo Scudeller Libardi)

   - [ASTERISK-26562
   <https://issues.asterisk.org/jira/browse/ASTERISK-26562>] -

app_controlplayback: Transmit Silence on ControlPlayback pause
(Reported by Mikheili Dautashvili)

   - [ASTERISK-24517
   <https://issues.asterisk.org/jira/browse/ASTERISK-24517>] -

TLS support for Solaris, Ming and non-glibc Linux systems
(Reported by Timo Teräs)

   - [ASTERISK-26540
   <https://issues.asterisk.org/jira/browse/ASTERISK-26540>] -

cdr_radius: use radcli instead of freeradius-client
(Reported by Tzafrir Cohen)

   - [ASTERISK-26558
   <https://issues.asterisk.org/jira/browse/ASTERISK-26558>] -

app_queue: add variable to know if the call is not answered after a queue
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26176
   <https://issues.asterisk.org/jira/browse/ASTERISK-26176>] -

chan_sip: Add AccountCode to AMI PeerEntry
(Reported by Sebastian Gutierrez)

   - [ASTERISK-26217
   <https://issues.asterisk.org/jira/browse/ASTERISK-26217>] -

[patch] Codec 2 Mode 2400
(Reported by Alexander Traud)

   - [ASTERISK-26538
   <https://issues.asterisk.org/jira/browse/ASTERISK-26538>] -

codec_opus: Add sample to configs/samples/codecs.conf.sample
(Reported by Kevin Harwell)

   - [ASTERISK-26488
   <https://issues.asterisk.org/jira/browse/ASTERISK-26488>] -

ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
(Reported by Matt Jordan)

   - [ASTERISK-26418
   <https://issues.asterisk.org/jira/browse/ASTERISK-26418>] -

res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that
are not involved in RTP
(Reported by Michael Walton)

   - [ASTERISK-26422
   <https://issues.asterisk.org/jira/browse/ASTERISK-26422>] -

[patch] Force calendars to do new fetch after module reload
(Reported by Ludovic Gasc (Eyepea))

   - [ASTERISK-26398
   <https://issues.asterisk.org/jira/browse/ASTERISK-26398>] -

core: Remove ABI differences of LOW_MEMORY
(Reported by Corey Farrell)

   - [ASTERISK-26409
   <https://issues.asterisk.org/jira/browse/ASTERISK-26409>] -

codec_opus: Update Asterisk to support the translation codec.
(Reported by Kevin Harwell)

   - [ASTERISK-26289
   <https://issues.asterisk.org/jira/browse/ASTERISK-26289>] -

Announcer channels in ConfBridges cause inefficiencies
(Reported by Mark Michelson)

   - [ASTERISK-26321
   <https://issues.asterisk.org/jira/browse/ASTERISK-26321>] -

ARI : Add reason answered_elsewhere to channel hangup
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-25980
   <https://issues.asterisk.org/jira/browse/ASTERISK-25980>] -

[patch]res_fax: set FAXMODE variable to let dialplan know what fax
transport was used
(Reported by Alexei Gradinari)

   - [ASTERISK-26229
   <https://issues.asterisk.org/jira/browse/ASTERISK-26229>] -

[patch] app_voicemail: Add taskprocessor alert level options.
(Reported by Alexei Gradinari)

   - [ASTERISK-26218
   <https://issues.asterisk.org/jira/browse/ASTERISK-26218>] -

[patch] iLBC 20
(Reported by Alexander Traud)

   - [ASTERISK-26190
   <https://issues.asterisk.org/jira/browse/ASTERISK-26190>] -

[patch] SRTP: Enable AES-256 and AES-GCM.
(Reported by Alexander Traud)

   - [ASTERISK-26220
   <https://issues.asterisk.org/jira/browse/ASTERISK-26220>] -

Add support for noreturn function attributes.
(Reported by Corey Farrell)

   - [ASTERISK-22131
   <https://issues.asterisk.org/jira/browse/ASTERISK-22131>] -

Update the make dependencies script to pull, build, and install the correct
(Reported by Matt Jordan)

   - [ASTERISK-25471
   <https://issues.asterisk.org/jira/browse/ASTERISK-25471>] -

[patch]Add subscribe_context to res_pjsip
(Reported by JoshE)

   - [ASTERISK-26159
   <https://issues.asterisk.org/jira/browse/ASTERISK-26159>] -

res_hep: enabled by default and information sent to default address
(Reported by Ross Beer)

   - [ASTERISK-25578
   <https://issues.asterisk.org/jira/browse/ASTERISK-25578>] -

[patch] SIP/SDP: No rtpmap for static RTP payload IDs
(Reported by Alexander Traud)

   - [ASTERISK-26059
   <https://issues.asterisk.org/jira/browse/ASTERISK-26059>] -

[patch]core: New channel variable FORWARDERNAME
(Reported by Alexei Gradinari)

   - [ASTERISK-20527
   <https://issues.asterisk.org/jira/browse/ASTERISK-20527>] -

AuthID cannot be set for registrations when callbackexten is used
(Reported by Timo Teräs)

   - [ASTERISK-26011
   <https://issues.asterisk.org/jira/browse/ASTERISK-26011>] -

[patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts
(Reported by Alexei Gradinari)

   - [ASTERISK-26055
   <https://issues.asterisk.org/jira/browse/ASTERISK-26055>] -

[patch]res_pjsip: chatty verbose messages
(Reported by Alexei Gradinari)

   - [ASTERISK-26064
   <https://issues.asterisk.org/jira/browse/ASTERISK-26064>] -

followme: allow disabling callee prompt
(Reported by Tzafrir Cohen)

   - [ASTERISK-26010
   <https://issues.asterisk.org/jira/browse/ASTERISK-26010>] -

[patch]func_odbc: single database connection should be optional
(Reported by Alexei Gradinari)

   - [ASTERISK-25965
   <https://issues.asterisk.org/jira/browse/ASTERISK-25965>] -

res_pjsip_outbound_publish: Allow multiple clients per configuration
(Reported by Kevin Harwell)

   - [ASTERISK-25994
   <https://issues.asterisk.org/jira/browse/ASTERISK-25994>] -

[patch]res_pjsip: module load priority
(Reported by Alexei Gradinari)

   - [ASTERISK-25931
   <https://issues.asterisk.org/jira/browse/ASTERISK-25931>] -

PJSIP: add "reg_server" to contacts.
(Reported by Alexei Gradinari)

   - [ASTERISK-25835
   <https://issues.asterisk.org/jira/browse/ASTERISK-25835>] -

Authentication using 'Username' field from Digest
(Reported by Ross Beer)

   - [ASTERISK-25930
   <https://issues.asterisk.org/jira/browse/ASTERISK-25930>] -

PJSIP: disable multi domain to improve realtime performace
(Reported by Alexei Gradinari)

   - [ASTERISK-25865
   <https://issues.asterisk.org/jira/browse/ASTERISK-25865>] -

Message-Account Missing From PJSIP MWI
(Reported by Ross Beer)

   - [ASTERISK-25444
   <https://issues.asterisk.org/jira/browse/ASTERISK-25444>] -

[patch]Music On Hold Warning misleading
(Reported by Conrad de Wet)

   - [ASTERISK-25846
   <https://issues.asterisk.org/jira/browse/ASTERISK-25846>] -

Gracefully deal with Absent Stasis Apps
(Reported by Andrew Nagy)

   - [ASTERISK-25791
   <https://issues.asterisk.org/jira/browse/ASTERISK-25791>] -

res_pjsip_caller_id: Lack of support for Anonymous
(Reported by Anthony Messina)

   - [ASTERISK-25767
   <https://issues.asterisk.org/jira/browse/ASTERISK-25767>] -

[patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)

   - [ASTERISK-25068
   <https://issues.asterisk.org/jira/browse/ASTERISK-25068>] -

Move commonly used FreePBX extra sounds to the core set
(Reported by Rusty Newton)

   - [ASTERISK-25627
   <https://issues.asterisk.org/jira/browse/ASTERISK-25627>] -

Easily Preventable Compile Warning
(Reported by Diederik de Groot)

   - [ASTERISK-25558
   <https://issues.asterisk.org/jira/browse/ASTERISK-25558>] -

[patch]chan_sip option 'notifyringing' doc fix and addition of
(Reported by Ward van Wanrooij)

   - [ASTERISK-25618
   <https://issues.asterisk.org/jira/browse/ASTERISK-25618>] -

res_pjsip: Check for readability of TLS files at startup
(Reported by George Joseph)

   - [ASTERISK-25581
   <https://issues.asterisk.org/jira/browse/ASTERISK-25581>] -

[patch]Add value reason a pause on CLI
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25572
   <https://issues.asterisk.org/jira/browse/ASTERISK-25572>] -

Endpoints: Add StatsD stats for Asterisk endpoints
(Reported by Matt Jordan)

   - [ASTERISK-25571
   <https://issues.asterisk.org/jira/browse/ASTERISK-25571>] -

PJSIP: Add StatsD stats for some common PJSIP objects
(Reported by Matt Jordan)

   - [ASTERISK-25518
   <https://issues.asterisk.org/jira/browse/ASTERISK-25518>] -

taskprocessor: Add high water mark
(Reported by Jonathan Rose)

   - [ASTERISK-25495
   <https://issues.asterisk.org/jira/browse/ASTERISK-25495>] -

[patch] Prevent old-update packages on repository Debian systems
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25477
   <https://issues.asterisk.org/jira/browse/ASTERISK-25477>] -

pjsip show "command" like [criteria]
(Reported by Bryant Zimmerman)

   - [ASTERISK-24718
   <https://issues.asterisk.org/jira/browse/ASTERISK-24718>] -

[patch]Add inital support of "sanitize" to configure
(Reported by Badalian Vyacheslav)

   - [ASTERISK-24870
   <https://issues.asterisk.org/jira/browse/ASTERISK-24870>] -

ARI: Subscriptions to bridges generally not super useful
(Reported by Matt Jordan)

   - [ASTERISK-25376
   <https://issues.asterisk.org/jira/browse/ASTERISK-25376>] -

Scripts: check file versions for Asterisk and dependencies
(Reported by Scott Griepentrog)

   - [ASTERISK-25405
   <https://issues.asterisk.org/jira/browse/ASTERISK-25405>] -

[patch] CLI: core show fd: add timestamp
(Reported by Alexander Traud)

   - [ASTERISK-25310
   <https://issues.asterisk.org/jira/browse/ASTERISK-25310>] -

[patch]on FreeBSD also pthread_attr_init() defaults to
(Reported by Guido Falsi)

   - [ASTERISK-25256
   <https://issues.asterisk.org/jira/browse/ASTERISK-25256>] -

[patch]Post AMI VarSet to empty string events when Asterisk deletes a
dialplan variable.
(Reported by Richard Mudgett)

   - [ASTERISK-25040
   <https://issues.asterisk.org/jira/browse/ASTERISK-25040>] -

pbx: Improve performance of reloads by making hint destruction more
(Reported by Matt Jordan)

   - [ASTERISK-25067
   <https://issues.asterisk.org/jira/browse/ASTERISK-25067>] -

Sorcery Caching: Implement a new caching module
(Reported by Matt Jordan)

   - [ASTERISK-25114
   <https://issues.asterisk.org/jira/browse/ASTERISK-25114>] -

res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
(Reported by George Joseph)

   - [ASTERISK-25132
   <https://issues.asterisk.org/jira/browse/ASTERISK-25132>] -

escaping manually
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-25072
   <https://issues.asterisk.org/jira/browse/ASTERISK-25072>] -

res_pjsip_outbound_registration: line functionality. Additional check for
using the request URI
(Reported by Dmitriy Serov)

   - [ASTERISK-25109
   <https://issues.asterisk.org/jira/browse/ASTERISK-25109>] -

[patch] CEL and CDR - Assigned separator for column name and values.
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-24815
   <https://issues.asterisk.org/jira/browse/ASTERISK-24815>] -

[patch] Enable TLS Dual-Certificates (ECC+RSA)
(Reported by Alexander Traud)

   - [ASTERISK-25063
   <https://issues.asterisk.org/jira/browse/ASTERISK-25063>] -

[patch]add X.509 subject alternative name support to Asterisk TLS support
(Reported by Maciej Szmigiero)

   - [ASTERISK-25044
   <https://issues.asterisk.org/jira/browse/ASTERISK-25044>] -

sorcery: Add ability to insert a new wizard into an object type's list
(Reported by George Joseph)

   - [ASTERISK-24892
   <https://issues.asterisk.org/jira/browse/ASTERISK-24892>] -

Super Awesome Company sound prompts
(Reported by Rusty Newton)

   - [ASTERISK-24744
   <https://issues.asterisk.org/jira/browse/ASTERISK-24744>] -

Swedish Core Voice prompts
(Reported by Tove Hjelm)

   - [ASTERISK-25049
   <https://issues.asterisk.org/jira/browse/ASTERISK-25049>] -

CLI: Enable automatic references to modules
(Reported by Corey Farrell)

   - [ASTERISK-25056
   <https://issues.asterisk.org/jira/browse/ASTERISK-25056>] -

Modules: Make ast_module_info->self available to auxiliary sources.
(Reported by Corey Farrell)

   - [ASTERISK-25045
   <https://issues.asterisk.org/jira/browse/ASTERISK-25045>] -

vector: Add new capabilities and unit tests
(Reported by George Joseph)

   - [ASTERISK-25043
   <https://issues.asterisk.org/jira/browse/ASTERISK-25043>] -

[patch] Avoiding ERR_remove_state in OpenSSL
(Reported by Alexander Traud)

   - [ASTERISK-24706
   <https://issues.asterisk.org/jira/browse/ASTERISK-24706>] -

[patch]add auto-dtmf mode for pjsip
(Reported by yaron nahum)

   - [ASTERISK-24917
   <https://issues.asterisk.org/jira/browse/ASTERISK-24917>] -

[patch] clang compilation warnings
(Reported by Diederik de Groot)

   - [ASTERISK-25051
   <https://issues.asterisk.org/jira/browse/ASTERISK-25051>] -

Remove unneeded uses of optional_api providers.
(Reported by Corey Farrell)

   - [ASTERISK-24974
   <https://issues.asterisk.org/jira/browse/ASTERISK-24974>] -

Astobj2: Allow reference debugging to be enabled/disabled by config.
(Reported by Corey Farrell)

   - [ASTERISK-24730
   <https://issues.asterisk.org/jira/browse/ASTERISK-24730>] -

[patch] Add blank line between headers and output for Command action
(Reported by Gareth Palmer)

   - [ASTERISK-24980
   <https://issues.asterisk.org/jira/browse/ASTERISK-24980>] -

cdr_adaptive_odbc: refactor lines to concatenate of columns name
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-24947
   <https://issues.asterisk.org/jira/browse/ASTERISK-24947>] -

res_pjsip: Add a PJSIP resolver using core DNS
(Reported by Joshua C. Colp)

   - [ASTERISK-24965
   <https://issues.asterisk.org/jira/browse/ASTERISK-24965>] -

cel_pgsql - log_error string references CDR instead of CEL
(Reported by Rodrigo Ramirez Norambuena)

   - [ASTERISK-24960
   <https://issues.asterisk.org/jira/browse/ASTERISK-24960>] -

Build System: Create MOD_ADD_SOURCE macro for module Makefiles
(Reported by Corey Farrell)

   - [ASTERISK-24939
   <https://issues.asterisk.org/jira/browse/ASTERISK-24939>] -

[patch]IAX make calltoken expiration time configurable
(Reported by Y Ateya)

   - [ASTERISK-24918
   <https://issues.asterisk.org/jira/browse/ASTERISK-24918>] -

pjsip: add CLI options to display global and system configuration
(Reported by Scott Griepentrog)

   - [ASTERISK-24862
   <https://issues.asterisk.org/jira/browse/ASTERISK-24862>] -

[patch] Support in-dialog OPTIONS
(Reported by yaron nahum)

   - [ASTERISK-24802
   <https://issues.asterisk.org/jira/browse/ASTERISK-24802>] -

stasis: set a channel variable on websocket disconnect error
(Reported by Kevin Harwell)

   - [ASTERISK-24133
   <https://issues.asterisk.org/jira/browse/ASTERISK-24133>] -

[patch]Please support Clang; Allow no-exec stacks
(Reported by Jeffrey Walton)

   - [ASTERISK-24790
   <https://issues.asterisk.org/jira/browse/ASTERISK-24790>] -

Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in
(Reported by Graham Barnett)

   - [ASTERISK-24813
   <https://issues.asterisk.org/jira/browse/ASTERISK-24813>] -

asterisk.c: #if statement in listener() confuses code folding editors
(Reported by Corey Farrell)

   - [ASTERISK-24811
   <https://issues.asterisk.org/jira/browse/ASTERISK-24811>] -

asterisk-publication sorcery object does not use realtime
(Reported by Matt Hoskins)

   - [ASTERISK-24745
   <https://issues.asterisk.org/jira/browse/ASTERISK-24745>] -

[patch]Add no_answer to ARI hangup causes
(Reported by Ben Merrills)

   - [ASTERISK-24316
   <https://issues.asterisk.org/jira/browse/ASTERISK-24316>] -

For httpd server, need option to define server name for security purposes
(Reported by Andrew Nagy)

   - [ASTERISK-24671
   <https://issues.asterisk.org/jira/browse/ASTERISK-24671>] -

Missing docs for the CDR AMI Event
(Reported by Dan Jenkins)

   - [ASTERISK-24575
   <https://issues.asterisk.org/jira/browse/ASTERISK-24575>] -

[patch]Make capath work for res_pjsip
(Reported by cloos)

   - [ASTERISK-24678
   <https://issues.asterisk.org/jira/browse/ASTERISK-24678>] -

[PATCH] Added atxfer* settings to features.conf.sample
(Reported by Niklas Larsson)

   - [ASTERISK-24412
   <https://issues.asterisk.org/jira/browse/ASTERISK-24412>] -

[patch]Incomplete channel originate/continue handling with ARI
(Reported by Nir Simionovich (GreenfieldTech - Israel))

   - [ASTERISK-24351
   <https://issues.asterisk.org/jira/browse/ASTERISK-24351>] -

[patch] Allow passing options and command to MixMonitor when recording in
(Reported by Gareth Palmer)

   - [ASTERISK-24553
   <https://issues.asterisk.org/jira/browse/ASTERISK-24553>] -

ARI/AMI: Include language in standard channel snapshot output
(Reported by Matt Jordan)

   - [ASTERISK-24552
   <https://issues.asterisk.org/jira/browse/ASTERISK-24552>] -

ARI: Allow associating a channel as an initiator of an Origination for
record keeping purposes
(Reported by Matt Jordan)

   - [ASTERISK-24577
   <https://issues.asterisk.org/jira/browse/ASTERISK-24577>] -

Speed up loopback switches by avoiding unneeded lookups
(Reported by Birger "WIMPy" Harzenetter)

   - [ASTERISK-24530
   <https://issues.asterisk.org/jira/browse/ASTERISK-24530>] -

[patch] app_record stripping 1/4 second from recordings
(Reported by Ben Smithurst)

   - [ASTERISK-24283
   <https://issues.asterisk.org/jira/browse/ASTERISK-24283>] -

[patch]Microseconds precision in the eventtime column in the cel_odbc module
(Reported by Etienne Lessard)

   - [ASTERISK-24128
   <https://issues.asterisk.org/jira/browse/ASTERISK-24128>] -

[Patch] Adding default dtls settings
(Reported by Michael K.)

   - [ASTERISK-24279
   <https://issues.asterisk.org/jira/browse/ASTERISK-24279>] -

Documentation: Clarify the behaviour of the CDR property 'unanswered'
(Reported by Matt Jordan)

   - [ASTERISK-23512
   <https://issues.asterisk.org/jira/browse/ASTERISK-23512>] -

Inaccurate comment in manager.conf.sample
(Reported by Richard Miller)

   - [ASTERISK-24365
   <https://issues.asterisk.org/jira/browse/ASTERISK-24365>] -

[Patch] Dialplan function to get first/head caller channel on queue
(Reported by Kristian Høgh)

   - [ASTERISK-23324
   <https://issues.asterisk.org/jira/browse/ASTERISK-23324>] -

[patch] - QLOOG commiting Japanese translated prompts
(Reported by Kevin McCoy)

   - [ASTERISK-24038
   <https://issues.asterisk.org/jira/browse/ASTERISK-24038>] -

device state: Report ONHOLD device state if channel driver defers device
state calculation to core
(Reported by Matt Jordan)

   - [ASTERISK-24171
   <https://issues.asterisk.org/jira/browse/ASTERISK-24171>] -

[patch] Provide a manpage for the aelparse utility
(Reported by Jeremy Lainé)

   - [ASTERISK-23953
   <https://issues.asterisk.org/jira/browse/ASTERISK-23953>] -

Testsuite: Off-nominal Authenticate test
(Reported by Matt Jordan)

   - [ASTERISK-24045
   <https://issues.asterisk.org/jira/browse/ASTERISK-24045>] -

[patch]Voicemail to email at multiple email addresses
(Reported by Jacob Barber)

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

*Thank you for your continued support of Asterisk!*
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