Update for 18.17.1

This commit is contained in:
Asterisk Development Team 2023-04-03 10:47:51 -05:00
parent 2227bfbb2b
commit 52f5afc63f
6 changed files with 176 additions and 671 deletions

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18.17.0
18.17.1

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2023-04-03 15:47 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 18.17.1 Released.
2023-03-29 13:49 +0000 [2227bfbb2b] Mike Bradeen <mbradeen@sangoma.com>
* res_pjsip_pubsub: subscription cleanup changes
There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.
First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.
If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.
This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.
That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)
The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.
To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.
Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.
This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.
ASTERISK-30469
Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7
2023-03-19 16:30 +0000 [c740de44f2] Sean Bright <sean@seanbright.com>
* Revert "pbx_ael: Global variables are not expanded."
This reverts commit f67258d17283f245b0d8a34c2288ca290fe679bf.
Reason for revert: Behavior change that breaks existing dialplan.
ASTERISK-30472 #close
Change-Id: Ia03fa7f049f964ed396f6bf6d7ca096b665aa6b3
2023-03-09 17:12 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 18.17.0 Released.

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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-18.17.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.17.0</h3><h3 align="center">Date: 2023-03-09</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>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.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-18.16.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>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.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">13 Naveen Albert <asterisk@phreaknet.org><br/>7 Sean Bright <sean@seanbright.com><br/>6 George Joseph <gjoseph@sangoma.com><br/>6 Mike Bradeen <mbradeen@sangoma.com><br/>2 Asterisk Development Team <asteriskteam@digium.com><br/>2 Igor Goncharovsky <igorg@iqtek.ru><br/>1 Peter Fern <asterisk@obfusc8.org><br/>1 Holger Hans Peter Freyther <holger@moiji-mobile.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Ben Ford <bford@digium.com><br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 cmaj <chris@penguinpbx.com><br/>1 Nick French <nickfrench@gmail.com><br/>1 sungtae kim <sungtae.kim@avoxi.com><br/></td><td width="33%"><td width="33%">10 N A <asterisk@phreaknet.org><br/>5 Michael Bradeen <mbradeen@sangoma.com><br/>3 Sean Bright <sean@seanbright.com><br/>3 George Joseph <gjoseph@digium.com><br/>1 Sebastian Gutierrez<br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Yury Kirsanov <y.kirsanov@gmail.com><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Nick French <nickfrench@gmail.com><br/>1 AvayaXAsterisk <joh.zuerner@yahoo.de><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Joshua C. Colp <jcolp@digium.com><br/>1 Stanislav Abramenkov <stas.abramenkov@gmail.com><br/>1 Julien Alie<br/>1 Oleg <olegsenin@gmail.com><br/>1 cmaj <chris@penguinpbx.com><br/>1 Danila Evgrafov <poooooochta@gmail.com><br/>1 Sebastian Gutierrez <scgm11@gmail.com><br/>1 Julien Alie <jalie@wazo.io><br/>1 Yury Kirsanov<br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>New Feature</h3><h4>Category: Applications/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29810">ASTERISK-29810</a>: app_signal: Add channel signaling applications<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d53c8fc4dc98eb5f403f1817825d259d311dbc66">[d53c8fc4dc]</a> Naveen Albert -- app_signal: Add signaling applications</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30180">ASTERISK-30180</a>: app_broadcast: Add a channel audio multicasting application<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dc3d8c6557095ab6f11b74a6a934bc3602f98c8">[0dc3d8c655]</a> Naveen Albert -- app_broadcast: Add Broadcast application</li>
</ul><br><h4>Category: Resources/res_pjsip_rfc3326</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30319">ASTERISK-30319</a>: Add BYE Reason support for SIP<br/>Reported by: Igor Goncharovsky<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de745157ca80264026f1d0ca35329231884f6286">[de745157ca]</a> Igor Goncharovsky -- res_pjsip_rfc3326: Add SIP causes support for RFC3326</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30262">ASTERISK-30262</a>: res_pjsip_session: Allow a context to be specified for overlap dialing<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec0ca7dcbc23e0d29e047bd6812e9c32f468e041">[ec0ca7dcbc]</a> Naveen Albert -- res_pjsip_session: Add overlap_context option.</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30198">ASTERISK-30198</a>: Error `Too many open files` occurs after about ~8000 calls when using mixmonitor<br/>Reported by: Julien Alie<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1f15569225d959d870fb74098b590a52822b976">[b1f1556922]</a> Peter Fern -- streams: Ensure that stream is closed in ast_stream_and_wait on error</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30417">ASTERISK-30417</a>: Copy/Paste error in UnpauseQueueMember<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=603675f285077b8b229320e6667188b7fd9a8b9e">[603675f285]</a> Sean Bright -- app_queue: Minor docs and logging fixes for UnpauseQueueMember.</li>
</ul><br><h4>Category: Applications/app_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29604">ASTERISK-29604</a>: ari: Segfault with lots of calls<br/>Reported by: Danila Evgrafov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f36ab7978f09f541f04f084a74adcf339beac75">[2f36ab7978]</a> sungtae kim -- res_stasis_snoop: Fix snoop crash</li>
</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30240">ASTERISK-30240</a>: app voicemail odbc build error with gcc 11.1<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1974ebe10956a5dfaeb204c6fa5894cc2d58f53c">[1974ebe109]</a> Naveen Albert -- app_voicemail_odbc: Fix string overflow warning.</li>
</ul><br><h4>Category: Channels/chan_iax2</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30354">ASTERISK-30354</a>: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6aa28346cf9c3e5279cfc9c7e743cf417ecdebb9">[6aa28346cf]</a> Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30162">ASTERISK-30162</a>: when chan_iax is used to relay calls, no ringing indication is played<br/>Reported by: Jaco Kroon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6aa28346cf9c3e5279cfc9c7e743cf417ecdebb9">[6aa28346cf]</a> Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28767">ASTERISK-28767</a>: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late<br/>Reported by: Oleg<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40ed776c409df8bb4b127ea3c6ac7785a856a314">[40ed776c40]</a> Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29604">ASTERISK-29604</a>: ari: Segfault with lots of calls<br/>Reported by: Danila Evgrafov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f36ab7978f09f541f04f084a74adcf339beac75">[2f36ab7978]</a> sungtae kim -- res_stasis_snoop: Fix snoop crash</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27830">ASTERISK-27830</a>: Asterisk crashes on Invalid UTF-8 string<br/>Reported by: AvayaXAsterisk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ddfb7551a42fce6165726b0d5834668c3fdbcf6">[1ddfb7551a]</a> George Joseph -- res_pjsip: Replace invalid UTF-8 sequences in callerid name</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30345">ASTERISK-30345</a>: loader.c: Modules that decline to load cannot be reloaded<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bde2689e1be1cfc451c740011d20b7a75ee58bdd">[bde2689e1b]</a> Naveen Albert -- loader: Allow declined modules to be unloaded.</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30379">ASTERISK-30379</a>: http: fix NULL pointer dereference while enable_status on TLS-only<br/>Reported by: Boris P. Korzun<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e85f23e6e51e0e33a94984788b88df7d578f545f">[e85f23e6e5]</a> Boris P. Korzun -- http.c: Fix NULL pointer dereference bug</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30351">ASTERISK-30351</a>: manager: Originate variables are not added when setvar used in manager.conf<br/>Reported by: Sebastian Gutierrez<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ede683f4ee3f1cb66705bd55d65e44738259783">[9ede683f4e]</a> Naveen Albert -- manager: Fix appending variables.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30367">ASTERISK-30367</a>: pbx: Fix outdated channel snapshots with pbx_exec<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a29f3f864d5d0d665e99ac2975d98323e7b0cb0f">[a29f3f864d]</a> Naveen Albert -- pbx_app: Update outdated pbx_exec channel snapshots.</li>
</ul><br><h4>Category: PBX/pbx_ael</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30406">ASTERISK-30406</a>: pbx_ael: Global variables are not expanded.<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f67258d17283f245b0d8a34c2288ca290fe679bf">[f67258d172]</a> Sean Bright -- pbx_ael: Global variables are not expanded.</li>
</ul><br><h4>Category: Resources/res_http_media_cache</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30375">ASTERISK-30375</a>: res_http_media_cache: Crash when URL has no path component.<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6dcb9080255b1adb932048a35f81a3a045a35646">[6dcb908025]</a> Holger Hans Peter Freyther -- res_http_media_cache: Do not crash when there is no extension</li>
</ul><br><h4>Category: Resources/res_phoneprov</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30388">ASTERISK-30388</a>: res_phoneprov: Stale SERVER variable when multi-homed<br/>Reported by: cmaj<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=66f6d9113e5ccc96cc5183a51d98f40fd67e4779">[66f6d9113e]</a> cmaj -- res_phoneprov.c: Multihomed SERVER cache prevention</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30369">ASTERISK-30369</a>: res_pjsip: Websockets from same IP shut down when they shouldn't be<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=db6e9c4e50a6923e23a105209677d34f797681f0">[db6e9c4e50]</a> George Joseph -- res_pjsip_transport_websocket: Add remote port to transport</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30100">ASTERISK-30100</a>: res_pjsip: Path is ignored on INVITE to endpoint<br/>Reported by: Yury Kirsanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=997a2d70b9b95d32f01d79f4555d48931c2d4003">[997a2d70b9]</a> Igor Goncharovsky -- res_pjsip: Fix path usage in case dialing with '@'</li>
</ul><br><h4>Category: Resources/res_pjsip_caller_id</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28767">ASTERISK-28767</a>: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late<br/>Reported by: Oleg<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40ed776c409df8bb4b127ea3c6ac7785a856a314">[40ed776c40]</a> Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30419">ASTERISK-30419</a>: pjsip: Crash when sending NOTIFY in PJSIP 2.13<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=114630279d35eabcef7affdf3c6f9b0b08f6503f">[114630279d]</a> Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30350">ASTERISK-30350</a>: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold<br/>Reported by: Benjamin Keith Ford<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cec98b5f0cfce209b496d35c6433e9a33bada137">[cec98b5f0c]</a> Ben Ford -- res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30391">ASTERISK-30391</a>: res_rtp_asterisk: Issue with transcoding g722 after MES changes<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c75383fd55ca7edac10c7499764097d708b2893">[6c75383fd5]</a> George Joseph -- res_rtp_asterisk: Don't use double math to generate timestamps</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=345ff2d8eea5f6192f8c3d257ac671398079be5f">[345ff2d8ee]</a> George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30424">ASTERISK-30424</a>: pjproject_bundled: cross-compilation broken when ssl autodetected<br/>Reported by: Nick French<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7196eb524bbc0944e9a616e39250b01539b3bc96">[7196eb524b]</a> Nick French -- pjproject_bundled: Fix cross-compilation with SSL libs.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30419">ASTERISK-30419</a>: pjsip: Crash when sending NOTIFY in PJSIP 2.13<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=114630279d35eabcef7affdf3c6f9b0b08f6503f">[114630279d]</a> Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_directory</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30405">ASTERISK-30405</a>: app_directory: Add 's' option to skip channel call<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b75e6d45e3ad7d2ea8fcc47966ff7c8a21b0432">[3b75e6d45e]</a> Mike Bradeen -- app_directory: Add a 'skip call' option.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30404">ASTERISK-30404</a>: app_directory: Add reading directory configuration from custom file<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ef6901e1372b6951ab5ccd0206546413aff95c55">[ef6901e137]</a> Mike Bradeen -- app_directory: add ability to specify configuration file</li>
</ul><br><h4>Category: Applications/app_read</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30411">ASTERISK-30411</a>: app_read: add option to include terminating digit on empty, terminated strings<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc5b7ab4595c9e6625bade5c70d286a9d11e0d31">[fc5b7ab459]</a> Mike Bradeen -- app_read: Add an option to return terminator on empty digits.</li>
</ul><br><h4>Category: Applications/app_senddtmf</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30422">ASTERISK-30422</a>: app_senddtmf: add the option for senddtmf to answer<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f015d3e0ccf49010c505834e7065f4b41feeb805">[f015d3e0cc]</a> Mike Bradeen -- app_senddtmf: Add option to answer target channel.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30361">ASTERISK-30361</a>: json.h: Add missing ast_json_object_real_get<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=811584ded3fb482e8bfff1dc950ee756f80008c1">[811584ded3]</a> Naveen Albert -- json.h: Add ast_json_object_real_get.</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29913">ASTERISK-29913</a>: func_json: Adds multi-level and array parsing to JSON_DECODE<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b9ef96173660bafa488be8844dae6816476e1fa">[7b9ef96173]</a> Naveen Albert -- func_json: Enhance parsing capabilities of JSON_DECODE</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30353">ASTERISK-30353</a>: func_frame_trace: Print text for text frames<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e4c01221514d84cf1ac89f5d0b07935fca7110e">[3e4c012215]</a> Naveen Albert -- func_frame_trace: Print text for text frames.</li>
</ul><br><h4>Category: Functions/func_callerid</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30332">ASTERISK-30332</a>: func_callerid: Warn if invalid redirecting reason provided<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=914c8e28c1bb99b8217f93b51ce8f59870350fb6">[914c8e28c1]</a> Naveen Albert -- func_callerid: Warn about invalid redirecting reason.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30325">ASTERISK-30325</a>: Upgrade Asterisk to bundled pjproject 2.13<br/>Reported by: Stanislav Abramenkov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=24e27a3c6ea10ca1d149aaab0df7977790c7dd48">[24e27a3c6e]</a> Mike Bradeen -- res_pjsip: Upgraded bundled pjsip to 2.13</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30280">ASTERISK-30280</a>: Create capability to assign a Media Experience Score to RTP streams<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62745013a4de9bafeda8dad6cab2b98c050ff69d">[62745013a4]</a> George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)</li>
</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a93d16d0f9b0ad1fecef613c3f38624582ab318">7a93d16d0f</a></td><td>Asterisk Development Team</td><td>Update for 18.17.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=69297b59df7131ac90c2003eec98d9e74b66f049">69297b59df</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.17.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a724298da99259373ef2be383b035a8a35800baf">a724298da9</a></td><td>Sean Bright</td><td>test.c: Avoid passing -1 to FD_* family of functions.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4965b9430017f220d7d9ddf62367884bb79d959">f4965b9430</a></td><td>Sean Bright</td><td>test_crypto.c: Fix getcwd(…) build error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7687c75cb6b6216d9525921fd69f1940aec7b73f">7687c75cb6</a></td><td>Sean Bright</td><td>app_queue: Reset all queue defaults before reload.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41d3a57627ef6ca8bf75b0c755c5127136572bc4">41d3a57627</a></td><td>Sean Bright</td><td>doxygen: Fix doxygen errors.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07a850c6dd378de485cba2a919b7951a1e0f27d4">07a850c6dd</a></td><td>Sean Bright</td><td>app_playback.c: Fix PLAYBACKSTATUS regression.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=462133e5e48468096f71866c703c1c0e840bee94">462133e5e4</a></td><td>Alexei Gradinari</td><td>format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8067229418a92cf06385632ebd9dab982588ef26">8067229418</a></td><td>George Joseph</td><td>Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-18.16.0-summary.html | 280
asterisk-18.16.0-summary.txt | 770
b/.version | 2
b/CHANGES | 66
b/ChangeLog | 696
b/UPGRADE.txt | 13
b/apps/app_broadcast.c | 619
b/apps/app_directory.c | 36
b/apps/app_mf.c | 1
b/apps/app_playback.c | 3
b/apps/app_queue.c | 14
b/apps/app_read.c | 23
b/apps/app_senddtmf.c | 31
b/apps/app_signal.c | 471
b/apps/app_voicemail.c | 7
b/asterisk-18.17.0-rc1-summary.html | 186
b/asterisk-18.17.0-rc1-summary.txt | 486
b/channels/chan_iax2.c | 17
b/channels/chan_pjsip.c | 114
b/channels/pjsip/dialplan_functions.c | 67
b/configs/samples/pjsip.conf.sample | 2
b/configs/samples/queues.conf.sample | 10
b/configure |18449 ++++------
b/configure.ac | 13
b/contrib/ast-db-manage/config/versions/f261363a857f_add_overlap_context.py | 21
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/formats/format_wav.c | 2
b/funcs/func_callerid.c | 1
b/funcs/func_frame_trace.c | 1
b/funcs/func_json.c | 232
b/include/asterisk/autoconfig.h.in | 124
b/include/asterisk/channel.h | 4
b/include/asterisk/crypto.h | 12
b/include/asterisk/file.h | 1
b/include/asterisk/json.h | 9
b/include/asterisk/pbx.h | 6
b/include/asterisk/res_aeap.h | 8
b/include/asterisk/res_aeap_message.h | 3
b/include/asterisk/res_geolocation.h | 4
b/include/asterisk/res_pjsip.h | 66
b/include/asterisk/res_stir_shaken.h | 2
b/include/asterisk/rtp_engine.h | 54
b/include/asterisk/time.h | 88
b/include/asterisk/utf8.h | 53
b/include/asterisk/xml.h | 18
b/main/bridge_basic.c | 2
b/main/file.c | 4
b/main/http.c | 10
b/main/loader.c | 25
b/main/manager.c | 6
b/main/pbx_app.c | 2
b/main/rtp_engine.c | 74
b/main/stasis_channels.c | 33
b/main/test.c | 13
b/main/utf8.c | 544
b/menuselect/autoconfig.h.in | 22
b/menuselect/configure | 3476 -
b/res/ael/pval.c | 14
b/res/res_aeap/transaction.h | 4
b/res/res_aeap/transport.h | 2
b/res/res_geolocation/geoloc_eprofile.c | 14
b/res/res_http_media_cache.c | 9
b/res/res_phoneprov.c | 20
b/res/res_pjsip.c | 337
b/res/res_pjsip/pjsip_config.xml | 10
b/res/res_pjsip/pjsip_configuration.c | 5
b/res/res_pjsip/pjsip_manager.xml | 3
b/res/res_pjsip/pjsip_transport_events.c | 2
b/res/res_pjsip_caller_id.c | 227
b/res/res_pjsip_path.c | 73
b/res/res_pjsip_pubsub.c | 101
b/res/res_pjsip_rfc3326.c | 31
b/res/res_pjsip_sdp_rtp.c | 4
b/res/res_pjsip_session.c | 24
b/res/res_rtp_asterisk.c | 547
b/res/res_speech_aeap.c | 51
b/res/res_stasis_snoop.c | 10
b/res/res_stir_shaken.c | 2
b/tests/test_crypto.c | 32
b/tests/test_res_rtp.c | 189
b/third-party/pjproject/configure.m4 | 7
b/third-party/pjproject/patches/0000-remove-third-party.patch | 6
b/third-party/pjproject/patches/0010-Make-sure-that-NOTIFY-tdata-is-set-before-sending-it_new-129fb323a66dd1fd16880fe5ba5e6a57.patch | 46
third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch | 413
third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch | 287
86 files changed, 15235 insertions(+), 14543 deletions(-)</pre><br></html>

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

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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-18.17.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.17.1</h3><h3 align="center">Date: 2023-04-03</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>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.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-18.17.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>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.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">1 Sean Bright <sean@seanbright.com><br/>1 Mike Bradeen <mbradeen@sangoma.com><br/></td><td width="33%"><td width="33%">1 N A <asterisk@phreaknet.org><br/>1 isrl <isrlgb@gmail.com><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: PBX/pbx_ael</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30472">ASTERISK-30472</a>: pbx_ael: Literal usage for variables broken<br/>Reported by: isrl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c740de44f25b213e1cba7c70e512a094c35ce3c3">[c740de44f2]</a> Sean Bright -- Revert "pbx_ael: Global variables are not expanded."</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30469">ASTERISK-30469</a>: res_pjsip_pubsub: Regression for subscription shutdowns<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2227bfbb2b7cb631c8b6c3f6d2b5a696e46c253f">[2227bfbb2b]</a> Mike Bradeen -- res_pjsip_pubsub: subscription cleanup changes</li>
</ul><br><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>ael/pval.c | 14 +--------
res_pjsip_pubsub.c | 81 +++++++++++++++++++++++++++++++----------------------
2 files changed, 51 insertions(+), 44 deletions(-)</pre><br></html>

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Release Summary
asterisk-18.17.1
Date: 2023-04-03
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-18.17.0.
----------------------------------------------------------------------
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
1 Sean Bright 1 N A
1 Mike Bradeen 1 isrl
----------------------------------------------------------------------
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: PBX/pbx_ael
ASTERISK-30472: pbx_ael: Literal usage for variables broken
Reported by: isrl
* [c740de44f2] Sean Bright -- Revert "pbx_ael: Global variables are not
expanded."
Category: Resources/res_pjsip_pubsub
ASTERISK-30469: res_pjsip_pubsub: Regression for subscription shutdowns
Reported by: N A
* [2227bfbb2b] Mike Bradeen -- res_pjsip_pubsub: subscription cleanup
changes
----------------------------------------------------------------------
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.
ael/pval.c | 14 +--------
res_pjsip_pubsub.c | 81 +++++++++++++++++++++++++++++++----------------------
2 files changed, 51 insertions(+), 44 deletions(-)