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487 lines
21 KiB
Plaintext
487 lines
21 KiB
Plaintext
Release Summary
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asterisk-16.16.0
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Date: 2021-01-21
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<asteriskteam@digium.com>
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----------------------------------------------------------------------
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Table of Contents
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1. Summary
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2. Contributors
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3. Closed Issues
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4. Open Issues
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5. Other Changes
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6. Diffstat
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----------------------------------------------------------------------
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Summary
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[Back to Top]
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This release is a point release of an existing major version. The changes
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included were made to address problems that have been identified in this
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release series, or are minor, backwards compatible new features or
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improvements. Users should be able to safely upgrade to this version if
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this release series is already in use. Users considering upgrading from a
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previous version are strongly encouraged to review the UPGRADE.txt
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document as well as the CHANGES document for information about upgrading
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to this release series.
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The data in this summary reflects changes that have been made since the
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previous release, asterisk-16.15.0.
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----------------------------------------------------------------------
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Contributors
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[Back to Top]
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This table lists the people who have submitted code, those that have
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tested patches, as well as those that reported issues on the issue tracker
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that were resolved in this release. For coders, the number is how many of
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their patches (of any size) were committed into this release. For testers,
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the number is the number of times their name was listed as assisting with
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testing a patch. Finally, for reporters, the number is the number of
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issues that they reported that were affected by commits that went into
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this release.
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Coders Testers Reporters
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6 Sean Bright 1 Mark Petersen 4 Alexander Traud
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4 Alexander Traud 2 Sean Bright
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3 George Joseph 2 sungtae kim
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3 Jaco Kroon 2 George Joseph
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3 Joshua C. Colp 1 Flole Systems
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2 Asterisk Development Team 1 Michael Maier
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2 Ivan Poddubnyi 1 Ivan Poddubny
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2 Sungtae Kim 1 Julien
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1 Dan Cropp 1 Jaco Kroon
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1 Kevin Harwell 1 Jean Aunis - Prescom
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1 Boris P. Korzun 1 Hendrik Wedhorn
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1 Jean Aunis 1 Robert Sutton
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1 Torrey Searle 1 Alex Hermann
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1 laszlovl 1 Alex Hermann
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1 Richard Mudgett 1 Juan Carlos Castro y Castro
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1 Nathan Bruning 1 Boris P. Korzun
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1 Pirmin Walthert 1 Alexander Greiner-Baer
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1 Stanislav 1 Alexander Traud
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1 Alexander Greiner-Baer 1 Mark Petersen
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1 Dan Cropp
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1 Nathan Bruning
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1 Mark Petersen
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1 Michael Maier
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1 Gant Liu
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1 Schneur Rosenberg
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1 Dan Cropp
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1 Stanislav Abramenkov
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1 Torrey Searle
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1 laszlovl
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1 Mikhail Ivanov
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----------------------------------------------------------------------
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Closed Issues
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[Back to Top]
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This is a list of all issues from the issue tracker that were closed by
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changes that went into this release.
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Security
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Category: Resources/res_pjsip_diversion
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ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains
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History-Info
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Reported by: Torrey Searle
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* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
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on tel: uri in History-Info
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Bug
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Category: Applications/app_chanspy
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ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
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Reported by: Hendrik Wedhorn
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* [0a23296834] Sean Bright -- app_chanspy: Spyee information missing in
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ChanSpyStop AMI Event
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Category: Applications/app_mixmonitor
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ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
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Reported by: Robert Sutton
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* [e96f744816] Kevin Harwell -- app_mixmonitor: cleanup datastore when
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monitor thread fails to launch
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Category: Applications/app_queue
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ASTERISK-29155: app_queue: Deadlock between queues container and
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individual queues
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Reported by: George Joseph
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* [8d8c9db618] George Joseph -- app_queue: Fix deadlock between update
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and show queues
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Category: Bridges/bridge_simple
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ASTERISK-29161: Incorrect setup of recall channels
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Reported by: Boris P. Korzun
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* [89d3de37ca] Boris P. Korzun -- bridge_basic: Fixed setup of recall
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channels
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Category: Channels/chan_pjsip
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ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
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instead of a channel variable
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Reported by: Ivan Poddubny
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* [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
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creating a channel
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ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses
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Reported by: George Joseph
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* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
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frames twice on outgoing channels
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ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
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Reported by: Alex Hermann
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* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
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frames twice on outgoing channels
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ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
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Reported by: Julien
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* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
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frames twice on outgoing channels
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ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if
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registration can't be send
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Reported by: Michael Maier
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* [7ed20b9d3b] George Joseph -- Revert
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"res_pjsip_outbound_registration.c: Use our own scheduler and other
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stuff"
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ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the
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Transfer result
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Reported by: Dan Cropp
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* [e127a57761] Dan Cropp -- chan_pjsip: Incorporate channel reference
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count into transfer_refer().
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ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts
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Reported by: Sean Bright
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* [ea744ca7c2] Joshua C. Colp -- pjsip: Match lifetime of INVITE session
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to our session.
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Category: Channels/chan_sip/CodecHandling
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ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
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accepted.
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Reported by: Alexander Traud
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* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
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when its media is disabled.
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ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
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Reported by: Alexander Traud
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* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
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when its media is disabled.
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Category: Channels/chan_sip/SRTP
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ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled
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user-agent.
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Reported by: Alexander Traud
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* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
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when its media is disabled.
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Category: Channels/chan_sip/TCP-TLS
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ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
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Reported by: Alexander Traud
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* [f667c5a781] Alexander Traud -- chan_sip: Remove unused
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sip_socket->port.
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Category: Channels/chan_sip/Video
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ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
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accepted.
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Reported by: Alexander Traud
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* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
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when its media is disabled.
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ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
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Reported by: Alexander Traud
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* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
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when its media is disabled.
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Category: Core/Logging
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ASTERISK-29209: Debug messages printed by scope trace might be missing
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newlines
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Reported by: Alexander Traud
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* [5a2867efa9] George Joseph -- logger.c: Automatically add a newline to
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formats that don't have one
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Category: Functions/func_lock
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ASTERISK-29217: LOCK() can grant the same lock to multiple channels
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spuriously
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Reported by: Jaco Kroon
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* [32e36144c7] Jaco Kroon -- func_lock: fix multiple-channel-grant
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problems.
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Category: General
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ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend
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Reported by: Alexander Traud
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* [4c79bc19d1] Alexander Traud -- loader: Sync load- and build-time
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deps.
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Category: Resources/res_ari_channels
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ASTERISK-29188: null media causing the Asterisk crash
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Reported by: sungtae kim
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* [a47e6965b3] Sungtae Kim -- res_ari: Fix wrong media uri handle for
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channel play
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Category: Resources/res_http_media_cache
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ASTERISK-29173: Media cache URL requests allow infinite redirects
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Reported by: Sean Bright
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* [0c185c9e21] Sean Bright -- res_http_media_cache.c: Set reasonable
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number of redirects
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Category: Resources/res_musiconhold
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ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold
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without entries
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Reported by: Nathan Bruning
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* [bb46595799] Nathan Bruning -- res_musiconhold: Don't crash when
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real-time doesn't return any entries
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Category: Resources/res_pjsip
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ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS
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response
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Reported by: Alexander Greiner-Baer
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* [a8f6238cc8] Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding
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to identity in OPTIONS response
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Category: Resources/res_pjsip_diversion
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ASTERISK-29191: tel: URI in Diversion header causes crash
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Reported by: Mikhail Ivanov
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* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
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on tel: uri in History-Info
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Category: Resources/res_pjsip_outbound_registration
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ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered
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Reported by: Michael Maier
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* [7ed20b9d3b] George Joseph -- Revert
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"res_pjsip_outbound_registration.c: Use our own scheduler and other
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stuff"
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Category: Resources/res_pjsip_session
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ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
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instead of a channel variable
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Reported by: Ivan Poddubny
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* [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
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creating a channel
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Category: Resources/res_stasis
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ASTERISK-29229: Stasis/messaging: text messages not dispatched to all
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subscribers when using generic subscription
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Reported by: Jean Aunis - Prescom
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* [45e1d89135] Jean Aunis -- Stasis/messaging: tech subscriptions
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conflict with endpoint subscriptions.
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Category: Resources/res_stir_shaken
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ASTERISK-29175: res_pjsip_stir_shaken: Fix module description
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Reported by: Stanislav Abramenkov
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* [159522003a] Stanislav -- res_pjsip_stir_shaken: Fix module
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description
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Category: pjproject/pjsip
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ASTERISK-29191: tel: URI in Diversion header causes crash
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Reported by: Mikhail Ivanov
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* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
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on tel: uri in History-Info
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ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set
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Reported by: Flole Systems
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* [11def974a8] Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of
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strings when appropriate
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Improvement
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Category: Applications/app_voicemail/NewFeature
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ASTERISK-29118: VoiceMail() should have an option to play greetings as
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Early Media
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Reported by: Juan Carlos Castro y Castro
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* [15566494f9] Joshua C. Colp -- voicemail: add option 'e' to play
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greetings as early media
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Category: Channels/chan_pjsip
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ASTERISK-28549: Two repeated 183
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Reported by: Gant Liu
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* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
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frames twice on outgoing channels
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Category: Contrib/General
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ASTERISK-29216: contrib: systemd asterisk service for centos8 or other
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newer linux versions
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Reported by: Mark Petersen
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* [7df88c98d0] Jaco Kroon -- contrib/systemd: Added note on common
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issues with systemd and asterisk
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Category: Resources/res_http_media_cache
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ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in
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/tmp
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Reported by: laszlovl
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* [8d2558209b] laszlovl -- Introduce astcachedir, to be used for
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temporary bucket files
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Category: Resources/res_pjsip_session
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ASTERISK-28549: Two repeated 183
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Reported by: Gant Liu
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* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
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frames twice on outgoing channels
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----------------------------------------------------------------------
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Open Issues
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[Back to Top]
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This is a list of all open issues from the issue tracker that were
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referenced by changes that went into this release.
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Bug
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Category: Applications/app_voicemail/ODBC
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ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
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Reported by: Schneur Rosenberg
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* [2b7af3eb27] Sean Bright -- app_voicemail: Prevent deadlocks when out
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of ODBC database connections
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Category: Resources/res_pjsip_session
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ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
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asterisk crash
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Reported by: sungtae kim
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* [ab3f57d88f] Sungtae Kim -- res_pjsip_session: Fixed NULL active media
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topology handle
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----------------------------------------------------------------------
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Commits Not Associated with an Issue
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[Back to Top]
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This is a list of all changes that went into this release that did not
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reference a JIRA issue.
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+------------------------------------------------------------------------+
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| Revision | Author | Summary |
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|------------+-------------+---------------------------------------------|
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| | Asterisk | |
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| e08d866884 | Development | Update for 16.16.0-rc1 |
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| | Team | |
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|------------+-------------+---------------------------------------------|
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| | Asterisk | |
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| 6056818467 | Development | Update CHANGES and UPGRADE.txt for 16.16.0 |
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| | Team | |
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|------------+-------------+---------------------------------------------|
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| aca435dfe7 | Jaco Kroon | pbx_lua: Add LUA_VERSIONS environment |
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| | | variable to ./configure. |
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|------------+-------------+---------------------------------------------|
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| 4c5bffb217 | Sean Bright | asterisk: Export additional manager |
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| | | functions |
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|------------+-------------+---------------------------------------------|
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| 89cf7899be | Richard | res_pjsip_session.c: Fix compiler warnings. |
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| | Mudgett | |
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|------------+-------------+---------------------------------------------|
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| 7e4bb4ed11 | Joshua C. | res_pjsip_pidf_digium_body_supplement: |
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| | Colp | Support Sangoma user agent. |
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|------------+-------------+---------------------------------------------|
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| ddbf3a7f73 | Sean Bright | media_cache: Fix reference leak with bucket |
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| | | file metadata |
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|------------+-------------+---------------------------------------------|
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| a360150ee0 | Sean Bright | CHANGES: Remove already applied CHANGES |
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| | | update |
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|------------+-------------+---------------------------------------------|
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| d1a78e047d | Alexander | modules.conf: Align the comments for more |
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| | Traud | conclusiveness. |
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+------------------------------------------------------------------------+
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----------------------------------------------------------------------
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Diffstat Results
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[Back to Top]
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This is a summary of the changes to the source code that went into this
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release that was generated using the diffstat utility.
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asterisk-16.15.0-summary.html | 213 ----------
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asterisk-16.15.0-summary.txt | 543 --------------------------
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b/.version | 2
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b/CHANGES | 18
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b/ChangeLog | 507 ++++++++++++++++++++++++
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b/Makefile | 6
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b/apps/app_chanspy.c | 6
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b/apps/app_mixmonitor.c | 23 +
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b/apps/app_queue.c | 245 ++++++-----
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b/apps/app_voicemail.c | 36 +
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b/asterisk-16.16.0-rc1-summary.html | 164 +++++++
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b/asterisk-16.16.0-rc1-summary.txt | 492 +++++++++++++++++++++++
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b/build_tools/install_subst | 1
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b/build_tools/make_defaults_h | 1
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b/build_tools/mkpkgconfig | 1
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b/channels/chan_pjsip.c | 214 +++-------
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b/channels/chan_sip.c | 32 -
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b/channels/sip/include/sip.h | 2
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b/configs/basic-pbx/modules.conf | 8
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b/configs/samples/asterisk.conf.sample | 1
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b/configs/samples/modules.conf.sample | 9
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b/configure | 11
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b/configure.ac | 9
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b/contrib/systemd/asterisk.service | 7
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b/funcs/func_lock.c | 163 ++-----
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b/funcs/func_odbc.c | 1
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b/funcs/func_periodic_hook.c | 1
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b/include/asterisk/manager.h | 4
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b/include/asterisk/paths.h | 1
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b/main/asterisk.c | 4
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b/main/bridge_basic.c | 2
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b/main/bucket.c | 3
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b/main/logger.c | 5
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b/main/manager.c | 6
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b/main/manager_channels.c | 18
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b/main/media_cache.c | 1
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b/main/options.c | 7
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b/main/pbx_variables.c | 2
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b/makeopts.in | 1
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b/res/res_hep_pjsip.c | 2
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b/res/res_http_media_cache.c | 1
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b/res/res_musiconhold.c | 21 -
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b/res/res_odbc.c | 1
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b/res/res_pjproject.c | 2
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b/res/res_pjsip.c | 2
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b/res/res_pjsip/pjsip_options.c | 2
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b/res/res_pjsip_diversion.c | 11
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b/res/res_pjsip_dlg_options.c | 2
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b/res/res_pjsip_nat.c | 10
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b/res/res_pjsip_outbound_registration.c | 296 +++++---------
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b/res/res_pjsip_pidf_digium_body_supplement.c | 8
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b/res/res_pjsip_session.c | 66 +--
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b/res/res_pjsip_stir_shaken.c | 4
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b/res/res_pjsip_transport_websocket.c | 2
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b/res/res_stasis_playback.c | 7
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b/res/res_stasis_snoop.c | 12
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b/res/stasis/messaging.c | 33 +
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doc/CHANGES-staging/hide_messaging_ami_events | 11
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58 files changed, 1826 insertions(+), 1437 deletions(-)
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