asterisk-18.17.0
Date: 2023-03-09
<asteriskteam@digium.com>
Table of Contents
- Summary
- Contributors
- Closed Issues
- Other Changes
- Diffstat
Summary
[Back to Top]This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.
The data in this summary reflects changes that have been made since the previous release, asterisk-18.16.0.
Contributors
[Back to Top]This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.
| Coders | Testers | Reporters |
|---|
13 Naveen Albert
7 Sean Bright
6 George Joseph
6 Mike Bradeen
2 Asterisk Development Team
2 Igor Goncharovsky
1 Peter Fern
1 Holger Hans Peter Freyther
1 Boris P. Korzun
1 Ben Ford
1 Alexei Gradinari
1 cmaj
1 Nick French
1 sungtae kim
| | 10 N A
5 Michael Bradeen
3 Sean Bright
3 George Joseph
1 Sebastian Gutierrez
1 Jaco Kroon
1 Yury Kirsanov
1 Benjamin Keith Ford
1 Nick French
1 AvayaXAsterisk
1 Ross Beer
1 Igor Goncharovsky
1 Boris P. Korzun
1 Joshua C. Colp
1 Stanislav Abramenkov
1 Julien Alie
1 Oleg
1 cmaj
1 Danila Evgrafov
1 Sebastian Gutierrez
1 Julien Alie
1 Yury Kirsanov
|
Closed Issues
[Back to Top]This is a list of all issues from the issue tracker that were closed by changes that went into this release.
New Feature
Category: Applications/NewFeature
ASTERISK-29810: app_signal: Add channel signaling applications
Reported by: N A
- [d53c8fc4dc] Naveen Albert -- app_signal: Add signaling applications
ASTERISK-30180: app_broadcast: Add a channel audio multicasting application
Reported by: N A
- [0dc3d8c655] Naveen Albert -- app_broadcast: Add Broadcast application
Category: Resources/res_pjsip_rfc3326
ASTERISK-30319: Add BYE Reason support for SIP
Reported by: Igor Goncharovsky
- [de745157ca] Igor Goncharovsky -- res_pjsip_rfc3326: Add SIP causes support for RFC3326
Category: Resources/res_pjsip_session
ASTERISK-30262: res_pjsip_session: Allow a context to be specified for overlap dialing
Reported by: N A
- [ec0ca7dcbc] Naveen Albert -- res_pjsip_session: Add overlap_context option.
Bug
Category: Applications/app_mixmonitor
ASTERISK-30198: Error `Too many open files` occurs after about ~8000 calls when using mixmonitor
Reported by: Julien Alie
- [b1f1556922] Peter Fern -- streams: Ensure that stream is closed in ast_stream_and_wait on error
Category: Applications/app_queue
ASTERISK-30417: Copy/Paste error in UnpauseQueueMember
Reported by: Sean Bright
- [603675f285] Sean Bright -- app_queue: Minor docs and logging fixes for UnpauseQueueMember.
Category: Applications/app_stasis
ASTERISK-29604: ari: Segfault with lots of calls
Reported by: Danila Evgrafov
- [2f36ab7978] sungtae kim -- res_stasis_snoop: Fix snoop crash
Category: Applications/app_voicemail/ODBC
ASTERISK-30240: app voicemail odbc build error with gcc 11.1
Reported by: Michael Bradeen
- [1974ebe109] Naveen Albert -- app_voicemail_odbc: Fix string overflow warning.
Category: Channels/chan_iax2
ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall
Reported by: N A
- [6aa28346cf] Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.
ASTERISK-30162: when chan_iax is used to relay calls, no ringing indication is played
Reported by: Jaco Kroon
- [6aa28346cf] Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.
Category: Channels/chan_pjsip
ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
Reported by: Oleg
- [40ed776c40] Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.
Category: Channels/chan_sip/General
ASTERISK-29604: ari: Segfault with lots of calls
Reported by: Danila Evgrafov
- [2f36ab7978] sungtae kim -- res_stasis_snoop: Fix snoop crash
Category: Core/BuildSystem
ASTERISK-27830: Asterisk crashes on Invalid UTF-8 string
Reported by: AvayaXAsterisk
- [1ddfb7551a] George Joseph -- res_pjsip: Replace invalid UTF-8 sequences in callerid name
Category: Core/General
ASTERISK-30345: loader.c: Modules that decline to load cannot be reloaded
Reported by: N A
- [bde2689e1b] Naveen Albert -- loader: Allow declined modules to be unloaded.
Category: Core/HTTP
ASTERISK-30379: http: fix NULL pointer dereference while enable_status on TLS-only
Reported by: Boris P. Korzun
- [e85f23e6e5] Boris P. Korzun -- http.c: Fix NULL pointer dereference bug
Category: Core/ManagerInterface
ASTERISK-30351: manager: Originate variables are not added when setvar used in manager.conf
Reported by: Sebastian Gutierrez
- [9ede683f4e] Naveen Albert -- manager: Fix appending variables.
Category: Core/PBX
ASTERISK-30367: pbx: Fix outdated channel snapshots with pbx_exec
Reported by: N A
- [a29f3f864d] Naveen Albert -- pbx_app: Update outdated pbx_exec channel snapshots.
Category: PBX/pbx_ael
ASTERISK-30406: pbx_ael: Global variables are not expanded.
Reported by: Sean Bright
- [f67258d172] Sean Bright -- pbx_ael: Global variables are not expanded.
Category: Resources/res_http_media_cache
ASTERISK-30375: res_http_media_cache: Crash when URL has no path component.
Reported by: Sean Bright
- [6dcb908025] Holger Hans Peter Freyther -- res_http_media_cache: Do not crash when there is no extension
Category: Resources/res_phoneprov
ASTERISK-30388: res_phoneprov: Stale SERVER variable when multi-homed
Reported by: cmaj
- [66f6d9113e] cmaj -- res_phoneprov.c: Multihomed SERVER cache prevention
Category: Resources/res_pjsip
ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they shouldn't be
Reported by: Joshua C. Colp
- [db6e9c4e50] George Joseph -- res_pjsip_transport_websocket: Add remote port to transport
ASTERISK-30100: res_pjsip: Path is ignored on INVITE to endpoint
Reported by: Yury Kirsanov
- [997a2d70b9] Igor Goncharovsky -- res_pjsip: Fix path usage in case dialing with '@'
Category: Resources/res_pjsip_caller_id
ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
Reported by: Oleg
- [40ed776c40] Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.
Category: Resources/res_pjsip_pubsub
ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13
Reported by: Ross Beer
- [114630279d] Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold
Reported by: Benjamin Keith Ford
- [cec98b5f0c] Ben Ford -- res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
Category: Resources/res_rtp_asterisk
ASTERISK-30391: res_rtp_asterisk: Issue with transcoding g722 after MES changes
Reported by: George Joseph
- [6c75383fd5] George Joseph -- res_rtp_asterisk: Don't use double math to generate timestamps
- [345ff2d8ee] George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)
Category: pjproject/pjsip
ASTERISK-30424: pjproject_bundled: cross-compilation broken when ssl autodetected
Reported by: Nick French
- [7196eb524b] Nick French -- pjproject_bundled: Fix cross-compilation with SSL libs.
ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13
Reported by: Ross Beer
- [114630279d] Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request
Improvement
Category: Applications/app_directory
ASTERISK-30405: app_directory: Add 's' option to skip channel call
Reported by: Michael Bradeen
- [3b75e6d45e] Mike Bradeen -- app_directory: Add a 'skip call' option.
ASTERISK-30404: app_directory: Add reading directory configuration from custom file
Reported by: Michael Bradeen
- [ef6901e137] Mike Bradeen -- app_directory: add ability to specify configuration file
Category: Applications/app_read
ASTERISK-30411: app_read: add option to include terminating digit on empty, terminated strings
Reported by: Michael Bradeen
- [fc5b7ab459] Mike Bradeen -- app_read: Add an option to return terminator on empty digits.
Category: Applications/app_senddtmf
ASTERISK-30422: app_senddtmf: add the option for senddtmf to answer
Reported by: Michael Bradeen
- [f015d3e0cc] Mike Bradeen -- app_senddtmf: Add option to answer target channel.
Category: Core/General
ASTERISK-30361: json.h: Add missing ast_json_object_real_get
Reported by: N A
- [811584ded3] Naveen Albert -- json.h: Add ast_json_object_real_get.
Category: Functions/General
ASTERISK-29913: func_json: Adds multi-level and array parsing to JSON_DECODE
Reported by: N A
- [7b9ef96173] Naveen Albert -- func_json: Enhance parsing capabilities of JSON_DECODE
ASTERISK-30353: func_frame_trace: Print text for text frames
Reported by: N A
- [3e4c012215] Naveen Albert -- func_frame_trace: Print text for text frames.
Category: Functions/func_callerid
ASTERISK-30332: func_callerid: Warn if invalid redirecting reason provided
Reported by: N A
- [914c8e28c1] Naveen Albert -- func_callerid: Warn about invalid redirecting reason.
Category: Resources/res_pjsip
ASTERISK-30325: Upgrade Asterisk to bundled pjproject 2.13
Reported by: Stanislav Abramenkov
- [24e27a3c6e] Mike Bradeen -- res_pjsip: Upgraded bundled pjsip to 2.13
Category: Resources/res_rtp_asterisk
ASTERISK-30280: Create capability to assign a Media Experience Score to RTP streams
Reported by: George Joseph
- [62745013a4] George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)
Commits Not Associated with an Issue
[Back to Top]This is a list of all changes that went into this release that did not reference a JIRA issue.
| Revision | Author | Summary |
|---|
| 7a93d16d0f | Asterisk Development Team | Update for 18.17.0-rc1 |
| 69297b59df | Asterisk Development Team | Update CHANGES and UPGRADE.txt for 18.17.0 |
| a724298da9 | Sean Bright | test.c: Avoid passing -1 to FD_* family of functions. |
| f4965b9430 | Sean Bright | test_crypto.c: Fix getcwd(…) build error. |
| 7687c75cb6 | Sean Bright | app_queue: Reset all queue defaults before reload. |
| 41d3a57627 | Sean Bright | doxygen: Fix doxygen errors. |
| 07a850c6dd | Sean Bright | app_playback.c: Fix PLAYBACKSTATUS regression. |
| 462133e5e4 | Alexei Gradinari | format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...) |
| 8067229418 | George Joseph | Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" |
Diffstat Results
[Back to Top]This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.
asterisk-18.16.0-summary.html | 280
asterisk-18.16.0-summary.txt | 770
b/.version | 2
b/CHANGES | 66
b/ChangeLog | 696
b/UPGRADE.txt | 13
b/apps/app_broadcast.c | 619
b/apps/app_directory.c | 36
b/apps/app_mf.c | 1
b/apps/app_playback.c | 3
b/apps/app_queue.c | 14
b/apps/app_read.c | 23
b/apps/app_senddtmf.c | 31
b/apps/app_signal.c | 471
b/apps/app_voicemail.c | 7
b/asterisk-18.17.0-rc1-summary.html | 186
b/asterisk-18.17.0-rc1-summary.txt | 486
b/channels/chan_iax2.c | 17
b/channels/chan_pjsip.c | 114
b/channels/pjsip/dialplan_functions.c | 67
b/configs/samples/pjsip.conf.sample | 2
b/configs/samples/queues.conf.sample | 10
b/configure |18449 ++++------
b/configure.ac | 13
b/contrib/ast-db-manage/config/versions/f261363a857f_add_overlap_context.py | 21
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/formats/format_wav.c | 2
b/funcs/func_callerid.c | 1
b/funcs/func_frame_trace.c | 1
b/funcs/func_json.c | 232
b/include/asterisk/autoconfig.h.in | 124
b/include/asterisk/channel.h | 4
b/include/asterisk/crypto.h | 12
b/include/asterisk/file.h | 1
b/include/asterisk/json.h | 9
b/include/asterisk/pbx.h | 6
b/include/asterisk/res_aeap.h | 8
b/include/asterisk/res_aeap_message.h | 3
b/include/asterisk/res_geolocation.h | 4
b/include/asterisk/res_pjsip.h | 66
b/include/asterisk/res_stir_shaken.h | 2
b/include/asterisk/rtp_engine.h | 54
b/include/asterisk/time.h | 88
b/include/asterisk/utf8.h | 53
b/include/asterisk/xml.h | 18
b/main/bridge_basic.c | 2
b/main/file.c | 4
b/main/http.c | 10
b/main/loader.c | 25
b/main/manager.c | 6
b/main/pbx_app.c | 2
b/main/rtp_engine.c | 74
b/main/stasis_channels.c | 33
b/main/test.c | 13
b/main/utf8.c | 544
b/menuselect/autoconfig.h.in | 22
b/menuselect/configure | 3476 -
b/res/ael/pval.c | 14
b/res/res_aeap/transaction.h | 4
b/res/res_aeap/transport.h | 2
b/res/res_geolocation/geoloc_eprofile.c | 14
b/res/res_http_media_cache.c | 9
b/res/res_phoneprov.c | 20
b/res/res_pjsip.c | 337
b/res/res_pjsip/pjsip_config.xml | 10
b/res/res_pjsip/pjsip_configuration.c | 5
b/res/res_pjsip/pjsip_manager.xml | 3
b/res/res_pjsip/pjsip_transport_events.c | 2
b/res/res_pjsip_caller_id.c | 227
b/res/res_pjsip_path.c | 73
b/res/res_pjsip_pubsub.c | 101
b/res/res_pjsip_rfc3326.c | 31
b/res/res_pjsip_sdp_rtp.c | 4
b/res/res_pjsip_session.c | 24
b/res/res_rtp_asterisk.c | 547
b/res/res_speech_aeap.c | 51
b/res/res_stasis_snoop.c | 10
b/res/res_stir_shaken.c | 2
b/tests/test_crypto.c | 32
b/tests/test_res_rtp.c | 189
b/third-party/pjproject/configure.m4 | 7
b/third-party/pjproject/patches/0000-remove-third-party.patch | 6
b/third-party/pjproject/patches/0010-Make-sure-that-NOTIFY-tdata-is-set-before-sending-it_new-129fb323a66dd1fd16880fe5ba5e6a57.patch | 46
third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch | 413
third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch | 287
86 files changed, 15235 insertions(+), 14543 deletions(-)