asterisk-14.5.0-rc1
Date: 2017-05-22
<asteriskteam@digium.com>
Table of Contents
- Summary
- Contributors
- Closed Issues
- Open 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-14.4.1.
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 |
|---|
24 Sean Bright
17 Richard Mudgett
13 gtjoseph
8 Joshua Colp
4 Kevin Harwell
3 Walter Doekes
3 Alexander Traud
3 Corey Farrell
2 Torrey Searle
2 Mark Michelson
1 Sebastian Gutierrez
1 Roman S.
1 Josh Roberson
1 Joshua Elson
1 Daniel Journo
1 Aaron An
1 Jean Aunis
1 Joshua Colp (license 5000)
1 Ivan Poddubny
1 Thierry Magnien
1 Vitezslav Novy
1 Rodrigo Ramírez Norambuena
| 1 Aaron An
| 3 Joshua Colp
3 Sandro Gauci
3 George Joseph
3 Richard Mudgett
2 Walter Doekes
2 Richard Begg
2 scgm11
2 twisted
2 Alexander Traud
2 Jeremy Kister
2 Matthias Urlichs
2 Ross Beer
2 Matthias Urlichs
2 Jeremy Kister
1 gkloepfer
1 Etienne Lessard
1 Aaron An
1 Jean Aunis - Prescom
1 Frederic LE FOLL
1 Bob Ham
1 Roman S.
1 Andreas Krüger
1 Anthony Critelli
1 Jens Bürger
1 Yaacov Akiba Slama
1 Vitaly K
1 Ksenia
1 Walter Doekes
1 Ivan Myalkin
1 Ove Aursand
1 Anthony Critelli
1 Ksenia
1 abelbeck
1 Vitaly K
1 Joerg Sonnenberger
1 Jens Bürger
1 Sandro Gauci
1 Marcello Ceschia
1 Sébastien Couture
1 Corey Farrell
1 Henning Holtschneider
1 Ivan Myalkin
1 Joel Vandal
1 Badalian Vyacheslav
1 Sean Darcy
1 Ross Beer
1 Alex Villacís Lasso
1 Olle Johansson
1 Richard Kenner
1 Richard Mudgett
1 Etienne Lessard
1 Sean Darcy
1 Joshua Elson
1 Nir Simionovich (GreenfieldTech - Israel)
1 Evers Lab
1 Henning Holtschneider
1 Ove Aursand
1 Frederic LE FOLL
1 Marcello Ceschia
1 Aaron An
1 Gergely Dömsödi
1 Niklas Larsson
1 Torrey Searle
1 Adagio
1 Roman S.
|
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: Applications/app_confbridge
ASTERISK-25506: [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
Reported by: Frederic LE FOLL
- [e9d563c4db] gtjoseph -- app_confbridge: Fix reference to cfg in menu_template_handler
Category: Applications/app_mixmonitor
ASTERISK-26169: format_ogg_vorbis: Memory leak using OGG in MixMonitor
Reported by: Ivan Myalkin
- [07e6f5806a] Sean Bright -- format_ogg_vorbis: Clear ogg/vorbis data structures on close
ASTERISK-26875: app_mixmonitor: Recording out of sync when 183 but no RTP
Reported by: Aaron An
- [7a18dafe53] Aaron An -- audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor.
Category: Applications/app_queue
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove Aursand
- [933c3f028d] Ivan Poddubny -- app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON
Category: CDR/General
ASTERISK-26818: cdr: Problem setting variables in h exten
Reported by: scgm11
- [24420b6676] Sebastian Gutierrez -- cdr: Allow setting of user field from 'h' extension
Category: CDR/cdr_adaptive_odbc
ASTERISK-26818: cdr: Problem setting variables in h exten
Reported by: scgm11
- [24420b6676] Sebastian Gutierrez -- cdr: Allow setting of user field from 'h' extension
Category: CEL/cel_odbc
ASTERISK-25032: [patch]cel_odbc sometimes inserts CEL with wrong eventtime
Reported by: Etienne Lessard
- [247cb7c9f2] gtjoseph -- cel_odbc: Fix timestamp processing for microseconds
Category: CEL/cel_pgsql
ASTERISK-26896: Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
Reported by: twisted
- [c9707febe3] Sean Bright -- cdr_pgsql: Fix buffer overflow calling libpq
- [43c785545b] Josh Roberson -- cel_pgsql.c: Fix buffer overflow calling libpq
Category: Channels/chan_sip/General
ASTERISK-26922: chan_sip: tcpbind uses wrong source address
Reported by: Ksenia
- [5cee143d9c] Thierry Magnien -- channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections
ASTERISK-26951: chan_sip: ACK with SDP does not update a direct media bridge
Reported by: Jean Aunis - Prescom
- [87a24362a2] Jean Aunis -- chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
ASTERISK-26692: res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
Reported by: scgm11
- [f5af1a1f74] Richard Mudgett -- res_rtp_asterisk.c: Fix crash in RTCP DTLS operation.
ASTERISK-26897: chan_sip: Security vulnerability with client code header
Reported by: Alex Villacís Lasso
- [b8df68c62d] Corey Farrell -- CDR: Protect from data overflow in ast_cdr_setuserfield.
Category: Channels/chan_sip/Interoperability
ASTERISK-21721: SIP Failed to parse multiple Supported: headers
Reported by: Olle Johansson
- [65c87a8ea9] Alexander Traud -- chan_sip: Session Timers required but refused wrongly.
ASTERISK-26915: chan_sip: Session Timers required but refused wrongly.
Reported by: Alexander Traud
- [65c87a8ea9] Alexander Traud -- chan_sip: Session Timers required but refused wrongly.
Category: Channels/chan_sip/SRTP
ASTERISK-25490: [patch]SDP crypto tag is validated incorrectly
Reported by: Joerg Sonnenberger
- [60ff37fcb8] Alexander Traud -- srtp: Allow zero as tag value for a sRTP Crypto Suite.
Category: Channels/chan_skinny
ASTERISK-26940: Asterisk Skinny memory exhaustion vulnerability leads to DoS
Reported by: Sandro Gauci
- [86333e8079] gtjoseph -- AST-2017-004: chan_skinny: Add EOF check in skinny_session
Category: Core/Bridging
ASTERISK-24529: Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
Reported by: Corey Farrell
- [8cbdd0a5b8] Joshua Colp -- bridge: Fix returning to dialplan when executing Bridge() from AMI.
Category: Core/BuildSystem
ASTERISK-26705: libasteriskssl.so not found when asterisk is installed for the 1st time
Reported by: George Joseph
- [e6087296c3] Walter Doekes -- build: Fix deb build issues with fakeroot
Category: Core/General
ASTERISK-26606: tcptls: Incorrect OpenSSL function call leads to misleading error report
Reported by: Bob Ham
- [ecd371846e] Joshua Colp -- tcptls: Improve error messages for TLS connections.
ASTERISK-26528: [UBSAN] strings.h:signed integer overflow in ast_str_case_hash
Reported by: Badalian Vyacheslav
- [bc9206d539] Torrey Searle -- strings.h: Avoid overflows in the string hash functions
ASTERISK-26903: Listening TCP/TLS sockets stop when temporarily out of open files
Reported by: Walter Doekes
- [866510d85b] Richard Mudgett -- tcptls.c: Cleanup TCP/TLS listener thread on abnormal exit.
Category: Core/RTP
ASTERISK-26515: rtp_engine: Allocate RTP payloads on a per-session basis
Reported by: Joshua Colp
- [d0649254b5] Joshua Colp -- rtp_engine: allocate RTP dynamic payloads per session
Category: Documentation
ASTERISK-26086: res_musiconhold: format option is not documented adequately
Reported by: Jens Bürger
- [63a79e395a] Sean Bright -- res_musiconhold: Document the 'format' option
Category: Formats/format_ogg_vorbis
ASTERISK-26169: format_ogg_vorbis: Memory leak using OGG in MixMonitor
Reported by: Ivan Myalkin
- [07e6f5806a] Sean Bright -- format_ogg_vorbis: Clear ogg/vorbis data structures on close
Category: Formats/format_pcm
ASTERISK-20984: Audible clicks when playing sox encoded au file with STREAM FILE AGI command
Reported by: Roman S.
- [ba30c8b692] Roman S. -- format_pcm: Track actual header size of .au files
Category: Formats/format_wav
ASTERISK-26613: format_wav: wav16 format read file only by 320 - half of frame
Reported by: Vitaly K
- [e24bbf318c] Sean Bright -- format_wav: Read 16khz wav samples properly
Category: Functions/func_cdr
ASTERISK-26173: func_cdr: CDR function does not permit empty values to be assigned
Reported by: gkloepfer
- [fbfce63b30] Joshua Colp -- func_cdr: Allow empty value for CDR dialplan function.
Category: Functions/func_speex
ASTERISK-26926: func_speex: Crash caused by frame with no datalen
Reported by: Richard Kenner
- [4cdf937a2e] Joshua Colp -- frame: Better handle interpolated frames.
Category: General
ASTERISK-26983: Crash in Manager Reload when TLS Config Changes
Reported by: Joshua ElsonASTERISK-26860: Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
Reported by: Evers Lab
- [3a5683d66c] Kevin Harwell -- res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures
Category: Resources/res_hep
ASTERISK-26953: Asterisk crash if hep.conf have some missing parameters
Reported by: Joel Vandal
- [03fb67f027] Sean Bright -- res_hep: Add additional config initialization and validation
Category: Resources/res_musiconhold
ASTERISK-25974: Unused realtime MOH classes not purged on 'moh reload'
Reported by: Sébastien Couture
- [ed3b0cbd2a] Daniel Journo -- Unused realtime MOH classes not purged on 'moh reload'
ASTERISK-26086: res_musiconhold: format option is not documented adequately
Reported by: Jens Bürger
- [63a79e395a] Sean Bright -- res_musiconhold: Document the 'format' option
ASTERISK-23996: No core dumps because of res_musiconhold chdir.
Reported by: Walter Doekes
- [0a8026072b] Sean Bright -- res_musiconhold: Don't chdir() when scanning MoH files
Category: Resources/res_pjsip
ASTERISK-26908: res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
Reported by: Richard Mudgett
- [cd4824ca69] gtjoseph -- res_pjsip_session: Add cleanup to ast_sip_session_terminate
ASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
Reported by: Andreas Krüger
- [6722ec6a6a] Kevin Harwell -- res_pjsip/res_pjsip_callerid: NULL check on caller id name string
ASTERISK-26928: pjsip: Add database tables for PUBLISH support
Reported by: Joshua Colp
- [7942378c55] Joshua Colp -- pjsip: Add Alembic for PUBLISH support.
ASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross Beer
- [e8cace12ba] gtjoseph -- pjproject_bundled: Add 3 upstream patches
ASTERISK-26916: res_pjsip: Excessive refcount reached on transport ao2 object
Reported by: Ross Beer
- [7ee8ec19fd] Richard Mudgett -- res_pjsip: Fix transport ref leak.
ASTERISK-26363: res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
Reported by: Yaacov Akiba Slama
- [e50df8ac81] Joshua Colp -- res_pjsip_session: Allow BYE to be sent on disconnected session.
Category: Resources/res_pjsip/Bundling
ASTERISK-26927: pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
Reported by: Alexander Traud
- [884bc5f76d] Alexander Traud -- pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
ASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross Beer
- [e8cace12ba] gtjoseph -- pjproject_bundled: Add 3 upstream patches
Category: Resources/res_pjsip_caller_id
ASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
Reported by: Andreas Krüger
- [6722ec6a6a] Kevin Harwell -- res_pjsip/res_pjsip_callerid: NULL check on caller id name string
Category: Resources/res_pjsip_pubsub
ASTERISK-26929: pjsip: Add database tables for RLS
Reported by: Joshua Colp
- [1174b566ab] Joshua Colp -- alembic: Add table for 'resource_list' PJSIP RLS type.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-26890: STUN server with non-default-route transport causes INVITE delay
Reported by: George Joseph
- [85bdf33808] Richard Mudgett -- res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions.
- [f592947b98] Richard Mudgett -- res_rtp_asterisk.c: Add stun_blacklist option
ASTERISK-26851: res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
Reported by: Richard Begg
- [797dfc13a9] Richard Mudgett -- res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member.
- [21abdc2d9a] Richard Mudgett -- res_pjsip_sdp_rtp.c: Don't alter global addr variable.
Category: Resources/res_pjsip_session
ASTERISK-26998: res_pjsip_session: INVITE retransmissions could still setup the same call again.
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
ASTERISK-26908: res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
Reported by: Richard Mudgett
- [cd4824ca69] gtjoseph -- res_pjsip_session: Add cleanup to ast_sip_session_terminate
Category: Resources/res_pjsip_t38
ASTERISK-26974: res_pjsip: Deadlock in T.38 framehook
Reported by: Richard Mudgett
- [d54ae43cc8] Richard Mudgett -- res_pjsip_t38.c: Fix deadlock in T.38 framehook.
Category: Resources/res_rtp_asterisk
ASTERISK-26143: res_rtp_asterisk: One way audio when transcoding
Reported by: Henning Holtschneider
- [bf19c0e596] Vitezslav Novy -- chan_sip: Change sip_get_codec() to return correct codec list
ASTERISK-26692: res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
Reported by: scgm11
- [f5af1a1f74] Richard Mudgett -- res_rtp_asterisk.c: Fix crash in RTCP DTLS operation.
ASTERISK-26835: res_rtp_asterisk: Crash when freeing RTCP address string
Reported by: Niklas Larsson
- [3adb547069] Richard Mudgett -- rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes.
ASTERISK-26853: res_rtp_asterisk: Crash in pjnath when receiving packet
Reported by: Adagio
- [3adb547069] Richard Mudgett -- rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes.
Category: Resources/res_stun_monitor
ASTERISK-21856: STUN never works when asterisk started without internet access
Reported by: Jeremy Kister
- [22c07f5223] Sean Bright -- res_stun_monitor: Don't fail to load if DNS resolution fails
Category: Resources/res_xmpp
ASTERISK-21009: xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
Reported by: Marcello Ceschia
- [b3fe94e4ca] Sean Bright -- astobj2: Prevent potential deadlocks with ao2_global_obj_release
ASTERISK-24712: xmpp: starttls problem causes connection spew
Reported by: Matthias Urlichs
- [cd93488a82] Sean Bright -- res_xmpp: Use incremental backoff when a read error occurs
- [a48b489de3] Sean Bright -- res_xmpp: Try to provide useful errors messages from OpenSSL
ASTERISK-23510: JABBER_STATUS fails with improper code 7 for unavailable clients
Reported by: Anthony Critelli
- [57fcbb8591] Sean Bright -- res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus
ASTERISK-21855: Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
Reported by: Jeremy Kister
- [eed582888f] Sean Bright -- res_xmpp: Don't crash when trying to send a message without a connection
ASTERISK-25622: WARNING for "JABBER: socket read error" should be more specific
Reported by: Sean Darcy
- [e771bf9787] Sean Bright -- res_xmpp: Include client name in connection related error messages
Category: Third-Party/pjproject
ASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross Beer
- [e8cace12ba] gtjoseph -- pjproject_bundled: Add 3 upstream patches
Category: pjproject/pjsip
ASTERISK-26939: Out of bound memory access in PJSIP multipart parser crashes Asterisk
Reported by: Sandro Gauci
- [04c7bd61ce] Mark Michelson -- AST-2017-003: Handle zero-length body parts correctly.
ASTERISK-26938: Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
Reported by: Sandro Gauci
- [83f6a6a2d6] Mark Michelson -- AST-2017-002: Ensure transaction key buffer is large enough.
ASTERISK-26930: pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
Reported by: abelbeck
- [6b15c7fc05] gtjoseph -- pjproject_bundled: Add --disable-libwebrtc to configure
ASTERISK-26814: pjproject_bundled build fails to download pjproject source when using cURL
Reported by: Gergely Dömsödi
- [8df2f481b6] Kevin Harwell -- pjproject_bundled: raise timeout value used when downloading
Improvement
Category: Core/Sorcery
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
Category: Core/Stasis
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
Category: Resources/res_hep_rtcp
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)
- [097f90220a] Joshua Colp -- res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
Category: Resources/res_pjsip
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
Category: Resources/res_pjsip_pubsub
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
Category: Resources/res_pjsip_registrar
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett
- [d7461f22d9] Richard Mudgett -- res_pjsip_session.c: Process initial INVITE sooner. (key exists)
Open Issues
[Back to Top]This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.
Bug
Category: Core/Bridging
ASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge
Reported by: Torrey Searle
- [443058f154] Torrey Searle -- bridging: Ensure successful T.38 negotation
Category: Core/Channels
ASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge
Reported by: Torrey Searle
- [443058f154] Torrey Searle -- bridging: Ensure successful T.38 negotation
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 |
|---|
| 0a5265fe10 | Sean Bright | res_hep_rtcp: Add support level to module info |
| fca75547e5 | Rodrigo Ramírez Norambuena | Fix spelling queues.conf.sample file |
| 944f9435bc | gtjoseph | logger: Added logger_queue_limit to the configuration options. |
| fe06758718 | Richard Mudgett | netsock2.c: Made get/set addr port avoid potential uninitialized memory. |
| 7f5ce428ed | Sean Bright | cleanup: Change severity of fread short-read warning |
| d9549bc475 | gtjoseph | res_pjsip_outbound_authenticator_digest: Add context to log messages |
| 00001443dc | Kevin Harwell | vector: defaults and indexes |
| 4cb1458245 | Sean Bright | cleanup: Fix fread() and fwrite() error handling |
| d757a70156 | Sean Bright | core: Use eventfd for alert pipes on Linux when possible |
| 47bc32173f | Richard Mudgett | res_pjsip_session.c: Restructure ast_sip_session_alloc() |
| 6bbccb0dfd | Sean Bright | pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified |
| b0b309d023 | Sean Bright | build: Update config.guess and config.sub |
| 26780e6da4 | gtjoseph | make ari-stubs so doc periodic jobs can run |
| 9464bc3f08 | Richard Mudgett | Revert "bridging: Ensure successful T.38 negotation" |
| 2c5aaeed09 | gtjoseph | modules: change module LOAD_FAILUREs to LOAD_DECLINES (14) |
| cc668bd522 | gtjoseph | modules: change module LOAD_FAILUREs to LOAD_DECLINES |
| 3408561eb7 | Richard Mudgett | stun.c: Fix ast_stun_request() erratic timeout. |
| 5bf425840e | Richard Mudgett | sorcery.c: Speed up ast_sorcery_retrieve_by_id() |
| 63560ae9b2 | Richard Mudgett | res_pjsip: Fix pointer use after unref. |
| 09252e9d59 | Walter Doekes | samples: Undo removal of include from canonicalize-app-names commit. |
| c5533488cc | Walter Doekes | samples: Canonicalize app names in extensions.conf.sample. |
| baf7660c7b | gtjoseph | sample_config: Add samples for pubsub to pjsip.conf.sample |
| 56f9193e6b | Corey Farrell | Forward declare 'struct ast_json' in asterisk.h |
| 451f7e00a6 | Corey Farrell | CEL: Remove header declarations of non-existant functions. |
| 27344675be | gtjoseph | res_pjsip_config_wizard: Add 2 new parameters to help with proxy config |
| b409d49273 | Sean Bright | alembic: Turn off execute bit on non-executable python scripts |
| 11f3b6e17f | Richard Mudgett | Add DTLS sanity check. |
| b033b74118 | Sean Bright | core: Remove embedded module support |
| 701a03738b | Sean Bright | res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts |
| 606103580f | Sean Bright | res_xmpp: Fix ref counting issue |
| 138f2bfa82 | Sean Bright | res_xmpp: Correctly check return value of SSL_connect |
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 |62120 ----------
asterisk-14.4.1-summary.html | 20
asterisk-14.4.1-summary.txt | 101
b/CHANGES | 47
b/Makefile | 39
b/Makefile.moddir_rules | 43
b/Makefile.rules | 12
b/UPGRADE.txt | 7
b/addons/Makefile | 2
b/addons/cdr_mysql.c | 43
b/addons/chan_mobile.c | 11
b/addons/format_mp3.c | 8
b/apps/app_adsiprog.c | 2
b/apps/app_agent_pool.c | 5
b/apps/app_alarmreceiver.c | 2
b/apps/app_authenticate.c | 2
b/apps/app_cdr.c | 5
b/apps/app_confbridge.c | 6
b/apps/app_dahdiras.c | 2
b/apps/app_forkcdr.c | 6
b/apps/app_minivm.c | 14
b/apps/app_queue.c | 6
b/apps/app_voicemail.c | 16
b/apps/app_zapateller.c | 2
b/apps/confbridge/conf_config_parser.c | 4
b/build_tools/cflags.xml | 4
b/cdr/cdr_custom.c | 4
b/cdr/cdr_pgsql.c | 56
b/cel/cel_custom.c | 7
b/cel/cel_odbc.c | 90
b/cel/cel_pgsql.c | 27
b/channels/chan_alsa.c | 46
b/channels/chan_dahdi.c | 12
b/channels/chan_iax2.c | 33
b/channels/chan_mgcp.c | 10
b/channels/chan_motif.c | 2
b/channels/chan_nbs.c | 4
b/channels/chan_oss.c | 57
b/channels/chan_phone.c | 6
b/channels/chan_pjsip.c | 11
b/channels/chan_sip.c | 50
b/channels/chan_skinny.c | 6
b/channels/chan_unistim.c | 2
b/codecs/Makefile | 6
b/codecs/codec_a_mu.c | 2
b/codecs/codec_adpcm.c | 2
b/codecs/codec_alaw.c | 2
b/codecs/codec_g722.c | 2
b/codecs/codec_g726.c | 2
b/codecs/codec_gsm.c | 2
b/codecs/codec_ilbc.c | 2
b/codecs/codec_lpc10.c | 2
b/codecs/codec_resample.c | 4
b/codecs/codec_ulaw.c | 2
b/config.guess | 184
b/config.sub | 90
b/configs/samples/asterisk.conf.sample | 4
b/configs/samples/extconfig.conf.sample | 3
b/configs/samples/extensions.conf.sample | 44
b/configs/samples/hep.conf.sample | 6
b/configs/samples/logger.conf.sample | 8
b/configs/samples/musiconhold.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 141
b/configs/samples/pjsip_wizard.conf.sample | 12
b/configs/samples/queues.conf.sample | 4
b/configs/samples/rtp.conf.sample | 19
b/configs/samples/sorcery.conf.sample | 9
b/configure | 218
b/configure.ac | 36
b/contrib/ast-db-manage/config/versions/1d0e332c32af_create_rls_table.py | 39
b/contrib/ast-db-manage/config/versions/2da192dbbc65_add_publish_tables.py | 73
b/formats/format_g719.c | 12
b/formats/format_g723.c | 15
b/formats/format_g726.c | 35
b/formats/format_g729.c | 14
b/formats/format_gsm.c | 18
b/formats/format_h263.c | 18
b/formats/format_h264.c | 18
b/formats/format_ilbc.c | 14
b/formats/format_jpeg.c | 2
b/formats/format_ogg_speex.c | 21
b/formats/format_ogg_vorbis.c | 20
b/formats/format_pcm.c | 101
b/formats/format_siren14.c | 12
b/formats/format_siren7.c | 12
b/formats/format_sln.c | 43
b/formats/format_vox.c | 16
b/formats/format_wav.c | 65
b/formats/format_wav_gsm.c | 16
b/funcs/func_cdr.c | 7
b/funcs/func_holdintercept.c | 2
b/funcs/func_talkdetect.c | 2
b/include/asterisk.h | 1
b/include/asterisk/alertpipe.h | 159
b/include/asterisk/astobj2.h | 4
b/include/asterisk/autoconfig.h.in | 8
b/include/asterisk/cel.h | 23
b/include/asterisk/channel.h | 11
b/include/asterisk/logger.h | 23
b/include/asterisk/module.h | 66
b/include/asterisk/options.h | 1
b/include/asterisk/res_pjsip_session.h | 4
b/include/asterisk/stasis_app.h | 1
b/include/asterisk/stasis_channels.h | 1
b/include/asterisk/stasis_endpoints.h | 1
b/include/asterisk/stasis_system.h | 1
b/include/asterisk/strings.h | 44
b/include/asterisk/vector.h | 62
b/main/Makefile | 19
b/main/alertpipe.c | 166
b/main/asterisk.c | 11
b/main/astobj2.c | 20
b/main/audiohook.c | 11
b/main/bridge_channel.c | 87
b/main/cdr.c | 4
b/main/channel.c | 12
b/main/channel_internal_api.c | 121
b/main/config_options.c | 5
b/main/features.c | 8
b/main/http.c | 23
b/main/loader.c | 95
b/main/logger.c | 192
b/main/manager.c | 14
b/main/netsock2.c | 25
b/main/pbx.c | 76
b/main/rtp_engine.c | 774
b/main/sorcery.c | 10
b/main/stun.c | 11
b/main/tcptls.c | 94
b/main/translate.c | 6
b/main/utils.c | 3
b/makeopts.in | 5
b/res/res_ari.c | 51
b/res/res_ari_applications.c | 19
b/res/res_ari_asterisk.c | 19
b/res/res_ari_bridges.c | 19
b/res/res_ari_channels.c | 19
b/res/res_ari_device_states.c | 19
b/res/res_ari_endpoints.c | 19
b/res/res_ari_events.c | 33
b/res/res_ari_mailboxes.c | 19
b/res/res_ari_model.c | 2
b/res/res_ari_playbacks.c | 19
b/res/res_ari_recordings.c | 19
b/res/res_ari_sounds.c | 19
b/res/res_calendar.c | 6
b/res/res_chan_stats.c | 23
b/res/res_config_sqlite.c | 16
b/res/res_config_sqlite3.c | 6
b/res/res_endpoint_stats.c | 2
b/res/res_hep.c | 34
b/res/res_hep_rtcp.c | 21
b/res/res_http_media_cache.c | 4
b/res/res_http_websocket.c | 2
b/res/res_limit.c | 2
b/res/res_musiconhold.c | 39
b/res/res_pjsip.c | 12
b/res/res_pjsip/config_transport.c | 19
b/res/res_pjsip_caller_id.c | 9
b/res/res_pjsip_config_wizard.c | 38
b/res/res_pjsip_nat.c | 4
b/res/res_pjsip_one_touch_record_info.c | 2
b/res/res_pjsip_outbound_authenticator_digest.c | 80
b/res/res_pjsip_outbound_publish.c | 2
b/res/res_pjsip_outbound_registration.c | 8
b/res/res_pjsip_publish_asterisk.c | 2
b/res/res_pjsip_pubsub.c | 12
b/res/res_pjsip_sdp_rtp.c | 46
b/res/res_pjsip_send_to_voicemail.c | 2
b/res/res_pjsip_session.c | 254
b/res/res_pjsip_t38.c | 32
b/res/res_rtp_asterisk.c | 985
b/res/res_smdi.c | 4
b/res/res_srtp.c | 4
b/res/res_stasis.c | 4
b/res/res_stasis_device_state.c | 5
b/res/res_stasis_playback.c | 5
b/res/res_stasis_recording.c | 5
b/res/res_stasis_test.c | 2
b/res/res_statsd.c | 3
b/res/res_stun_monitor.c | 12
b/res/res_xmpp.c | 158
b/rest-api-templates/res_ari_resource.c.mustache | 41
b/tests/test_bucket.c | 2
b/tests/test_channel_feature_hooks.c | 2
b/tests/test_logger.c | 67
b/tests/test_media_cache.c | 2
b/tests/test_vector.c | 19
b/third-party/Makefile | 2
b/third-party/pjproject/Makefile | 4
b/third-party/pjproject/Makefile.rules | 1
b/third-party/pjproject/configure.m4 | 1
b/third-party/pjproject/patches/0035-r5572-svn-backport-dialog-transaction-deadlock.patch | 32
b/third-party/pjproject/patches/0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch | 119
b/third-party/pjproject/patches/0037-r5576-svn-backport-session-timer-crash.patch | 69
build_tools/embed_modules.xml | 46
contrib/realtime/mssql/mssql_cdr.sql | 58
contrib/realtime/mssql/mssql_config.sql | 1659
contrib/realtime/mssql/mssql_voicemail.sql | 54
contrib/realtime/mysql/mysql_cdr.sql | 40
contrib/realtime/mysql/mysql_config.sql | 1006
contrib/realtime/mysql/mysql_voicemail.sql | 34
contrib/realtime/oracle/oracle_cdr.sql | 52
contrib/realtime/oracle/oracle_config.sql | 1653
contrib/realtime/oracle/oracle_voicemail.sql | 48
contrib/realtime/postgresql/postgresql_cdr.sql | 44
contrib/realtime/postgresql/postgresql_config.sql | 1084
contrib/realtime/postgresql/postgresql_voicemail.sql | 38
210 files changed, 4676 insertions(+), 70184 deletions(-)