asterisk-16.8.0
Date: 2020-02-04
<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-16.7.0.
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 |
|---|
20 Sean Bright
7 George Joseph
6 Richard Mudgett
5 Joshua C. Colp
4 Kevin Harwell
3 Asterisk Development Team
3 Jaco Kroon
2 Pascal Cadotte Michaud
1 Kevin Reeves
1 Frederic LE FOLL
1 Joshua Colp
1 Jean Aunis
1 Rodrigo Ramírez Norambuena
1 Boris P. Korzun
1 Andrew Siplas
1 Corey Farrell
1 snuffy
| | 3 Sean Bright
3 Ross Beer
3 cmaj
2 Pascal Cadotte Michaud
2 Joshua C. Colp
1 Robert Sutton
1 Kevin Flyn
1 Ted G
1 Maciej Michno
1 AvayaXAsterisk
1 Jean Aunis - Prescom
1 George Joseph
1 Jaco Kroon
1 Kevin Reeves
1 candrews
1 Andrew Siplas
1 Frank Matano
1 Cédric Bassaget
1 Kevin Harwell
1 Dan Jenkins
1 kevin@phoneburner.com
1 Maciej Michno
1 Dirk Wendland
1 Jim Van Meggelen
1 Stas Kobzar
1 Jean-Denis Girard
1 Stas Kobzar
1 Boris P. Korzun
1 Cedric BASSAGET
1 Niksa Baldun
1 Ted G
1 Corey Farrell
1 Richard Kenner
1 Frank Matano
1 Joeran Vinzens
1 Frederic LE FOLL
1 Kevin Flyn
1 David M. Lee
1 Dirk Wendland
1 Bryan Nelson
1 Richard Kenner
1 Ross Beer
1 Francois Blackburn
1 Joeran Vinzens
1 Jonathan Harris
1 Jonathan Harris
1 Dan Jenkins
1 AvayaXAsterisk
1 Sean Bright
1 Jean-Denis Girard
1 nappsoft
1 Joshua C. Colp
1 Niksa Baldun
1 Mitch Claborn
1 Robin Leffmann
1 Jim Van Meggelen
1 David Lee
1 Robert Sutton
|
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: Functions/func_curl
ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything
Reported by: candrews
- [f69da94fab] Sean Bright -- func_curl: Add 'followlocation' option to CURLOPT()
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on source port
Reported by: Sean Bright
- [f8b0c2c933] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add port matching support
Bug
Category: Applications/app_chanisavail
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
Reported by: Frederic LE FOLL
- [aa06c6ea29] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.
Category: Applications/app_meetme
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
Reported by: George Joseph
- [7167fd6d46] Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.
Category: Applications/app_queue
ASTERISK-28349: Pause reason not reported in QueueMember AMI event
Reported by: Niksa Baldun
- [5fded77e7f] Sean Bright -- app_queue: Deprecate the QueueMemberPause.Reason field
Category: Applications/app_record
ASTERISK-28682: app_record: Lack of `beep` audio file causes application to return error and hangup
Reported by: Corey Farrell
- [0c07a7ee00] Corey Farrell -- app_record: Do not hang up if beep audio is missing
Category: Applications/app_voicemail
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
- [293600724d] Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail
ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail to leave message
Reported by: Jim Van Meggelen
- [e379fe48e1] Sean Bright -- app_voicemail: Set globals to default values when voicemail.conf missing
Category: Applications/app_voicemail/ODBC
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
- [293600724d] Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail
Category: CDR/General
ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
- [1b452ebb51] George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
Reported by: Frederic LE FOLL
- [aa06c6ea29] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.
Category: Channels/chan_dahdi
ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
Reported by: Andrew Siplas
- [9895e94dba] Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"
Category: Channels/chan_pjsip
ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
Reported by: Jean-Denis Girard
- [992dcdf780] Sean Bright -- res_pjsip_config_wizard: Fix change detection for wizard settings
ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
Reported by: Ross Beer
- [63b8664bfa] George Joseph -- res_pjsip_nat: Restore original contact for REGISTER responses
Category: Channels/chan_sip/General
ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
Reported by: Jean Aunis - Prescom
- [82a870c8c7] Jean Aunis -- chan_sip: voice frames are no longer transmitted after emitting a COLP
ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
Reported by: Jaco Kroon
- [055737d645] Jaco Kroon -- chan_sip: in case of tcp/tls, be less annoying about tx errors.
Category: Channels/chan_sip/Messaging
ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
Reported by: Frank Matano
- [31027f33db] Sean Bright -- chan_sip.c: Stop handling continuation lines after reading headers
Category: Channels/chan_sip/Transfers
ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
- [1b452ebb51] George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge
Category: Codecs/codec_silk
ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation' output
Reported by: Sean Bright
- [efecc9d139] Sean Bright -- translate.c: Fix silk 24kHz truncation in 'core show translation'
Category: Configs/Basic-PBX
ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order mark is present
Reported by: Robin Leffmann
- [a78758d0a2] Sean Bright -- config.c: Skip UTF-8 BOMs if present when reading config files
Category: Contrib/General
ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax
Reported by: Richard Kenner
- [b9b50774f5] snuffy -- contrib/valgrind: Fix use of frame-level suppression
ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
Reported by: Pascal Cadotte Michaud
- [b8e635916f] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid typo
Category: Core/Streams
ASTERISK-28625: Playback of local files impacted by large media cache
Reported by: Kevin Reeves
- [e013f502b1] Kevin Reeves -- main/file.c: Limit media cache usage to remote files.
Category: Documentation
ASTERISK-24484: Update documentation for statsd module - usage requirements unclear
Reported by: Dan Jenkins
- [04c81f9748] Sean Bright -- res_statsd: Document that res_statsd does nothing on its own
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames
Reported by: Joshua C. Colp
- [8d87fef5a1] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames
ASTERISK-28507: Wiki docs missing for MessageWaiting
Reported by: David M. Lee
- [4cf32f2578] George Joseph -- CI: Update buildAsterisk.sh to do a "make full"
Category: Functions/General
ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
Reported by: Pascal Cadotte Michaud
- [2d2b28bfa4] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation
Category: Functions/func_odbc
ASTERISK-28497: func_odbc: truncating Unicode string on readsql
Reported by: Boris P. Korzun
- [e54299cd3e] Boris P. Korzun -- func_odbc: acf_odbc_read() and cli_odbc_read() unicode support
Category: General
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
Reported by: Ted G
- [8af0dea0c7] George Joseph -- res_rtp_asterisk: Add frame list cleanups to ast_rtp_read
Category: PBX/General
ASTERISK-28695: core: minmemfree watermark uses free RAM, not available RAM
Reported by: Kevin Flyn
- [f5a1e8b04d] Sean Bright -- pbx.c: Include filesystem cache in free memory calculation
ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
Reported by: Dirk Wendland
- [64692a3c72] George Joseph -- sig_pri: Fix deadlock caused by sig_pri_queue_hangup
Category: Resources/res_ari
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
- [1627e8eddc] Kevin Harwell -- res_stasis: trigger cleanup after update
Category: Resources/res_fax
ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config option
Reported by: Kevin Harwell
- [d17bbcb9f1] Kevin Harwell -- res_fax: wrap v21 detected Asterisk initiated negotiation with config option
Category: Resources/res_http_websocket
ASTERISK-28562: SIP WSS message not processed until next frame arrives
Reported by: Robert Sutton
- [47ba42f4a0] Sean Bright -- websocket: Consider pending SSL data when waiting for socket input
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames
Reported by: Joshua C. Colp
- [8d87fef5a1] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames
Category: Resources/res_pjsip_notify
ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present instead of just one
Reported by: AvayaXAsterisk
- [0a56edca4d] Sean Bright -- res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY
Category: Resources/res_pjsip_pubsub
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
- [939e18d63e] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.
ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not preserve XML version number
Reported by: Bryan Nelson
- [8318b05f25] Joshua C. Colp -- res_pjsip_pubsub: Add ability to persist generator state information.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
- [186c4e9b36] Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.
Category: Resources/res_pjsip_session
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
- [186c4e9b36] Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.
Category: Resources/res_realtime
ASTERISK-21794: CLI command 'realtime update2' syntax failure when using according to usage help
Reported by: Cedric BASSAGET
- [fbe18165d5] Sean Bright -- res_realtime: Fix 'realtime update2' argument handling
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
- [42c51263b9] Kevin Harwell -- stasis/app: don't lock an app before a call to send
ASTERISK-28633: stasis bridge topic leak
Reported by: Joeran Vinzens
- [dd82ebecd3] George Joseph -- stasis.c: Use correct topic name in stasis_topic_pool_delete_topic
Category: Resources/res_statsd
ASTERISK-24484: Update documentation for statsd module - usage requirements unclear
Reported by: Dan Jenkins
- [04c81f9748] Sean Bright -- res_statsd: Document that res_statsd does nothing on its own
Improvement
Category: Applications/app_confbridge
ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. Colp
- [5622df0a94] Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.
Category: Bridges/bridge_softmix
ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. Colp
- [5622df0a94] Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.
Category: Channels/chan_pjsip
ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
Reported by: cmaj
- [a7692ce2f4] Richard Mudgett -- app_chanisavail.c: Simplify dialplan using ChanIsAvail.
- [144b774b85] Richard Mudgett -- app_dial.c: Simplify dialplan using Dial.
- [2780be334d] Richard Mudgett -- app_page.c: Simplify dialplan using Page.
Category: Core/HTTP
ASTERISK-28710: Should be able to disable the /httpstatus URI in the built-in HTTP server
Reported by: Sean Bright
- [a2a4e1026c] Sean Bright -- http: Add ability to disable /httpstatus URI
Category: Documentation
ASTERISK-28673: GET FULL VARIABLE documentation clarification
Reported by: Jonathan Harris
- [60fd1322d7] Sean Bright -- res_agi: Improve GET FULL VARIABLE documentation
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 |
|---|
| 126beb3e6c | Joshua Colp | REVERT: Add option to suppress the Message channel AMI and ARI events |
| bfe9e1b2e7 | George Joseph | message.c: Add option to suppress the Message channel AMI and ARI events |
| c92e2bb09f | Asterisk Development Team | Update for 16.8.0-rc2 |
| b7b813eb34 | Asterisk Development Team | Update for 16.8.0-rc1 |
| eb1ec0498d | Asterisk Development Team | Update CHANGES and UPGRADE.txt for 16.8.0 |
| a7aaca9eaa | Sean Bright | func_odbc.conf.sample: Add example lookup |
| f49517efb9 | Rodrigo Ramírez Norambuena | queue_log: Add alembic script for generate db table for queue_log |
| 13fa33588f | Sean Bright | app_voicemail, say: Fix various leading whitespace problems |
| b92b0469ff | Jaco Kroon | netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr. |
| de078debab | Kevin Harwell | app_agent_pool: Update XML docs for AgentLogin |
| 11753d94d8 | Richard Mudgett | features.c: Make Bridge application tolerate unspecified channel. |
| 00e745066c | Richard Mudgett | app_chanspy.c: Reduce log message level from notice to verbose. |
| 198f4cbdbf | Richard Mudgett | app_softhangup.c: Reduce unnecessary warning to verbose message. |
| efa13eb0a0 | Sean Bright | db: Initialize condition primitive before use |
| 77941efad9 | Jaco Kroon | ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. |
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-16.7.0-summary.html | 309 ---
asterisk-16.7.0-summary.txt | 763 -------
b/.version | 2
b/CHANGES | 55
b/ChangeLog | 834 ++++++++
b/apps/app_agent_pool.c | 4
b/apps/app_chanisavail.c | 140 -
b/apps/app_chanspy.c | 3
b/apps/app_confbridge.c | 2
b/apps/app_dial.c | 51
b/apps/app_page.c | 30
b/apps/app_queue.c | 2
b/apps/app_record.c | 3
b/apps/app_softhangup.c | 2
b/apps/app_voicemail.c | 407 ++--
b/apps/confbridge/conf_config_parser.c | 17
b/apps/confbridge/include/confbridge.h | 1
b/asterisk-16.8.0-rc2-summary.html | 15
b/asterisk-16.8.0-rc2-summary.txt | 96 +
b/bridges/bridge_softmix.c | 18
b/channels/chan_dahdi.c | 2
b/channels/chan_sip.c | 30
b/channels/sig_pri.c | 23
b/configs/samples/confbridge.conf.sample | 4
b/configs/samples/func_odbc.conf.sample | 8
b/configs/samples/http.conf.sample | 10
b/configs/samples/pjsip.conf.sample | 1
b/contrib/ast-db-manage/README.md | 1
b/contrib/ast-db-manage/queue_log.ini.sample | 58
b/contrib/ast-db-manage/queue_log/env.py | 1
b/contrib/ast-db-manage/queue_log/script.py.mako | 24
b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py | 38
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/contrib/valgrind.supp | 14
b/doc/CHANGES-staging/res_fax_negotiate_both | 7
b/doc/appdocsxml.dtd | 2
b/funcs/func_curl.c | 11
b/funcs/func_odbc.c | 22
b/funcs/func_pjsip_contact.c | 6
b/include/asterisk/acl.h | 37
b/include/asterisk/bridge.h | 21
b/include/asterisk/http_websocket.h | 14
b/include/asterisk/iostream.h | 14
b/include/asterisk/netsock2.h | 42
b/include/asterisk/res_fax.h | 3
b/include/asterisk/res_pjsip_pubsub.h | 23
b/include/asterisk/stasis.h | 3
b/main/acl.c | 74
b/main/bridge.c | 7
b/main/cdr.c | 15
b/main/config.c | 12
b/main/db.c | 3
b/main/features.c | 28
b/main/file.c | 7
b/main/http.c | 56
b/main/iostream.c | 14
b/main/pbx.c | 12
b/main/say.c | 956 +++++-----
b/main/stasis.c | 17
b/main/translate.c | 8
b/res/res_agi.c | 20
b/res/res_fax.c | 26
b/res/res_http_websocket.c | 11
b/res/res_pjsip/pjsip_message_filter.c | 40
b/res/res_pjsip_config_wizard.c | 7
b/res/res_pjsip_dialog_info_body_generator.c | 80
b/res/res_pjsip_endpoint_identifier_ip.c | 86
b/res/res_pjsip_nat.c | 84
b/res/res_pjsip_notify.c | 22
b/res/res_pjsip_pubsub.c | 87
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_realtime.c | 56
b/res/res_stasis.c | 8
b/res/res_statsd.c | 23
74 files changed, 2842 insertions(+), 2094 deletions(-)