asterisk-17.3.0-rc1
Date: 2020-03-05
<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-17.2.0.
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 |
|---|
11 Sean Bright
7 Joshua C. Colp
5 Kevin Harwell
5 Walter Doekes
3 George Joseph
2 Torrey Searle
1 Asterisk Development Team
1 Sebastian Kemper
1 Sylvain Afchain
1 Jaco Kroon
1 Ben Ford
1 lvl
| | 4 Ross Beer
3 Joshua C. Colp
2 Walter Doekes
1 Paul Brooks
1 Martin Zeh
1 Sébastien Duthil
1 Jean Aunis - Prescom
1 Kevin Harwell
1 Sylvain Afchain
1 EDV O-TON
1 Martin Zeh
1 alex
1 Ross Beer
1 Timothy Vanderaerden
1 xrobau
1 Sebastian Kemper
1 Paul Brooks
1 Peter Sokolov
1 Francois Blackburn
1 EDV O-TON
1 Peter Sokolov
1 George Joseph
1 Dmitriy Serov
1 Dmitriy Serov
1 Alex
1 Sébastien Duthil
1 Torrey Searle
1 lvl
1 Benjamin Keith Ford
|
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.
Bug
Category: Channels/chan_pjsip
ASTERISK-28766: PJSIP blind transfer not completed after using Proceeding()
Reported by: lvl
- [88393db74d] lvl -- res_pjsip_refer: ensure refer progress is still sent after Proceeding()
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field
Reported by: Jean Aunis - Prescom
- [c47dbf1fbe] Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI
Category: Channels/chan_sip/Interoperability
ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should return 503
Reported by: Walter Doekes
- [a5ec14970d] Walter Doekes -- chan_sip: Return 503 if we're out of RTP ports
ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed: no audio
Reported by: Walter Doekes
- [be6eec7b25] Walter Doekes -- chan_sip: Always process updated SDP on media source change
Category: Core/Configuration
ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues
Reported by: EDV O-TON
- [646789106f] Sean Bright -- res_config_odbc: Preserve empty strings returned by the database
Category: Core/Stasis
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field
Reported by: Jean Aunis - Prescom
- [c47dbf1fbe] Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI
Category: Resources/res_ari
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
- [7031c3b7bd] Kevin Harwell -- res_stasis: trigger cleanup after update
Category: Resources/res_musiconhold
ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN
Reported by: Ross Beer
- [d1a96fa9ba] Sean Bright -- res_musiconhold: Avoid spurious warning when 'format' is the empty string
Category: Resources/res_pjsip
ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
Reported by: Paul Brooks
- [ace97a3786] Sean Bright -- chan_pjsip: Ignore RTP that we haven't negotiated
Category: Resources/res_pjsip_acl
ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed
Reported by: Timothy Vanderaerden
- [bc3c095e9f] Joshua C. Colp -- pjsip: Update ACLs on named ACL changes.
Category: Resources/res_pjsip_messaging
ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be changed
Reported by: Alex
- [ef5702cef1] Sean Bright -- res_pjsip_messaging: Allow Content-Type to be overridden
ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending
Reported by: Dmitriy Serov
- [69cf67d8c6] Sean Bright -- res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly
Category: Resources/res_pjsip_outbound_registration
ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
Reported by: George Joseph
- [5f36196384] George Joseph -- res_pjsip_outbound_registration: Fix SRV failover on timeout
Category: Resources/res_pjsip_pubsub
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
- [35c9332edf] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold
Reported by: Ross Beer
- [1c8c560f39] Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH transitions
ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used
Reported by: Torrey Searle
- [fb7cff4103] Torrey Searle -- res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough
Category: Resources/res_pjsip_session
ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes
Reported by: Joshua C. Colp
- [3a8da5ece7] Joshua C. Colp -- res_pjsip_session: Fix off-nominal session refreshes.
Category: Resources/res_rtp_asterisk
ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling
Reported by: Joshua C. Colp
- [029c3e49d4] Joshua C. Colp -- res_rtp_asterisk: Improve video performance in certain networks.
ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before allowing sending
Reported by: Benjamin Keith FordASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
Reported by: Kevin Harwell
- [08d8035599] Kevin Harwell -- res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
- [737bd8365e] Kevin Harwell -- stasis/app: don't lock an app before a call to send
Category: Resources/res_stasis_playback
ASTERISK-28713: res_stasis_playback: Error building JSON
Reported by: Sébastien Duthil
- [8b8c1dd07f] Sean Bright -- res_stasis_playback: Prevent media_index from going out of bounds
Category: Utilities/General
ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling troubles
Reported by: Sebastian Kemper
- [61f943d405] Sebastian Kemper -- check_expr2: fix cross-compile/hardening issues
Category: pjproject/pjsip
ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected
Reported by: Peter Sokolov
- [eb33b0fb0b] Sean Bright -- pjproject_bundled: Allow brackets in via parameters
Improvement
Category: Applications/app_mixmonitor
ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
Reported by: xrobau
- [5f28ecc7af] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
Category: Bridges/bridge_native_rtp
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
- [957681e08b] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
Category: Bridges/bridge_simple
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
- [957681e08b] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
Category: Bridges/bridge_softmix
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
- [957681e08b] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
Category: Contrib/General
ASTERISK-28726: install_prereq script uses the interactive mode when installing aptitude
Reported by: Sylvain Afchain
- [cf87d63775] Sylvain Afchain -- install_prereq: Install aptitude non-interactively
Category: Core/HTTP
ASTERISK-28750: TLS/SSL Key too small error
Reported by: Martin Zeh
- [9ef514ae98] Sean Bright -- tcptls.c: Log more informative OpenSSL errors
Category: Core/Streams
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
- [957681e08b] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
Category: Documentation
ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
Reported by: xrobau
- [5f28ecc7af] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
- [957681e08b] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
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 |
|---|
| 1d3256c5e4 | Asterisk Development Team | Update CHANGES and UPGRADE.txt for 17.3.0 |
| a3f5a80a59 | Walter Doekes | say: Remove unused "plural" option from main/say |
| b455d57f7d | Walter Doekes | app_queue: Refactor odd placement of if's around say_position |
| d77a89cb9e | Kevin Harwell | format_cap: make function parameters 'const' |
| 77d4923307 | Jaco Kroon | addons/res_config_mysql: silense warnings about printf format errors. |
| cf26ce5d4f | Sean Bright | ast_tls_cert: Allow private key size to be set on command line |
| 5a0218a960 | Joshua C. Colp | stasis: Use format specifier for size_t. |
| 4b897f6fe4 | Sean Bright | func_odbc: Prevent snprintf() truncation warning |
| 4d6346dc26 | George Joseph | doc: Fix CHANGES entries to have .txt suffix and update READMEs |
| d19dec53b5 | Walter Doekes | chan_sip: Clarify in sample docs how directmediapermit/-acl should be used |
| 325e3f8111 | Joshua C. Colp | res_rtp_asterisk: Don't produce transport-cc if no packets. |
| 54cd865946 | George Joseph | message.c: Add option to suppress the Message channel AMI and ARI events |
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.
.lastclean | 1
.version | 1
ChangeLog |88490 ----------
asterisk-17.2.0-summary.html | 201
asterisk-17.2.0-summary.txt | 620
b/CHANGES | 25
b/UPGRADE.txt | 56
b/addons/res_config_mysql.c | 16
b/apps/app_mixmonitor.c | 29
b/apps/app_queue.c | 58
b/bridges/bridge_native_rtp.c | 168
b/bridges/bridge_simple.c | 198
b/bridges/bridge_softmix.c | 246
b/channels/chan_pjsip.c | 13
b/channels/chan_sip.c | 126
b/channels/sip/include/sip.h | 1
b/configs/samples/asterisk.conf.sample | 5
b/configs/samples/sip.conf.sample | 4
b/configure | 141
b/configure.ac | 22
b/contrib/scripts/ast_tls_cert | 8
b/contrib/scripts/install_prereq | 2
b/doc/CHANGES-staging/README.md | 8
b/doc/CHANGES-staging/hide_messaging_ami_events | 11
b/doc/UPGRADE-staging/README.md | 7
b/funcs/func_odbc.c | 4
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/channel.h | 20
b/include/asterisk/format_cap.h | 4
b/include/asterisk/message.h | 13
b/include/asterisk/options.h | 3
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/say.h | 4
b/include/asterisk/sorcery.h | 27
b/main/asterisk.c | 1
b/main/channel.c | 19
b/main/file.c | 2
b/main/format_cap.c | 4
b/main/message.c | 27
b/main/options.c | 2
b/main/say.c | 12
b/main/sorcery.c | 46
b/main/stasis.c | 4
b/main/stream.c | 22
b/main/tcptls.c | 29
b/makeopts.in | 2
b/menuselect/configure | 22
b/res/ari/ari_model_validators.c | 59
b/res/ari/ari_model_validators.h | 23
b/res/res_config_odbc.c | 2
b/res/res_musiconhold.c | 2
b/res/res_pjsip/pjsip_configuration.c | 19
b/res/res_pjsip_acl.c | 20
b/res/res_pjsip_messaging.c | 54
b/res/res_pjsip_outbound_registration.c | 49
b/res/res_pjsip_refer.c | 7
b/res/res_pjsip_sdp_rtp.c | 56
b/res/res_pjsip_session.c | 107
b/res/res_rtp_asterisk.c | 312
b/res/res_sorcery_config.c | 1
b/res/res_stasis_playback.c | 4
b/res/stasis/messaging.c | 11
b/rest-api/api-docs/endpoints.json | 20
b/rest-api/resources.json | 2
b/third-party/pjproject/configure.m4 | 1
b/third-party/pjproject/patches/0040-ICE-Add-callback-for-finding-valid-pair.patch | 56
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1270
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1382
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
doc/CHANGES-staging/res_fax_negotiate_both | 7
73 files changed, 1670 insertions(+), 92683 deletions(-)