Update for 16.5.0-rc1

16.5 16.5.0-rc1
Asterisk Development Team 6 years ago
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16.5.0-rc1

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ChangeLog

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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-16.5.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.5.0-rc1</h3><h3 align="center">Date: 2019-07-18</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-16.4.0.</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%">8 Alexei Gradinari <alex2grad@gmail.com><br/>7 George Joseph <gjoseph@digium.com><br/>2 Asterisk Development Team <asteriskteam@digium.com><br/>2 Joshua Colp <jcolp@digium.com><br/>2 Chris-Savinovich <csavinovich@digium.com><br/>1 Kevin Harwell <kharwell@digium.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Kirsty Tyerman <kirsty.tyerman@boeing.com><br/>1 Abhay Gupta <abhay@avissol.com><br/>1 Francesco Castellano <francesco.castellano@messagenet.it><br/>1 Ben Ford <bford@digium.com><br/>1 Morten Tryfoss <morten@tryfoss.no><br/>1 Guido Falsi <madpilot@FreeBSD.org><br/>1 Nasir Iqbal <nasir@ictinnovations.com><br/></td><td width="33%"><td width="33%">1 Gil Richard <grichard@intertalksystems.com><br/>1 Oleksandr Natalenko<br/>1 Oleksandr Natalenko <oleksandr@natalenko.name><br/>1 Nasir Iqbal <nasir@ictinnovations.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Kirsty Tyerman <kirsty.tyerman@boeing.com><br/>1 Kirsty Tyerman<br/>1 abelbeck <lonnie@abelbeck.com><br/>1 Bernhard Schmidt<br/>1 Abhay Gupta <abhay@avissol.com><br/>1 Gil Richard<br/>1 vijay kumar <vijaykumar@drishti-soft.com><br/>1 Dmitry Svyatogorov <ds@vo-ix.ru><br/>1 pasandev <pasandev@ymail.com><br/>1 Francesco Castellano <francesco.castellano@messagenet.it><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Bernhard Schmidt <berni@birkenwald.de><br/>1 Morten Tryfoss <morten@tryfoss.no><br/>1 Michael Maier <m1278468@mailbox.org><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>Security</h3><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28465">ASTERISK-28465</a>: Broken SDP can cause a segfault in a T.38 reINVITE<br/>Reported by: Francesco Castellano<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c59df17a55a4f91a05df3a833652e04f0853cf6">[6c59df17a5]</a> Francesco Castellano -- chan_sip: Handle invalid SDP answer to T.38 re-invite</li>
</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28447">ASTERISK-28447</a>: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash<br/>Reported by: Gil Richard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2126dc30219b110e5f2870c7cca61676d6848933">[2126dc3021]</a> George Joseph -- res_pjsip_messaging: Check for body in in-dialog message</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28419">ASTERISK-28419</a>: app_amd: Does not work with silence suppression<br/>Reported by: Nasir Iqbal<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52a3d4a761e7ca438a96f58f606b8efb442dce8a">[52a3d4a761]</a> Nasir Iqbal -- app_amd: issue with silence suppression fixed</li>
</ul><br><h4>Category: CDR/cdr_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28435">ASTERISK-28435</a>: cdr_pgsql: Unix socket doesn't work<br/>Reported by: Dmitry Svyatogorov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2621aa190a167f49e9bee9324db9d0540ee4b3a">[c2621aa190]</a> Chris-Savinovich -- cdr_pgsql: fix error in connection string</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28457">ASTERISK-28457</a>: [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317<br/>Reported by: abelbeck<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da1db4f842b097f9c2e6183a5659cda49b2be3ea">[da1db4f842]</a> Chris-Savinovich -- chan_dahdi.c: crash in chan_dahdi</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28427">ASTERISK-28427</a>: new mwi.h include missing from some dahdi source files, causes build failure<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86fb72c4d08961f9db01818a55f0fa499d465d36">[86fb72c4d0]</a> Guido Falsi -- chan_dahdi: add missing include.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28444">ASTERISK-28444</a>: chan_pjsip: Peer IP for SSL handshake errors not logged<br/>Reported by: Bernhard Schmidt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2db5173b88aca5138ec6949d1806c13ecaada922">[2db5173b88]</a> George Joseph -- pjproject_bundled: Add peer information to most SSL/TLS errors</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25371">ASTERISK-25371</a>: Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event<br/>Reported by: Abhay Gupta<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72f26aa8eb9ec6b07ac790e83df41a0ffdf1af0e">[72f26aa8eb]</a> Abhay Gupta -- chan_pjsip.c: Check for channel and session to not be NULL in hangup</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27994">ASTERISK-27994</a>: PJSIP: Early media ringback not indicated after Progress()<br/>Reported by: Gregory Massel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de82bdd746f20df959caf04cc6d5cb160cfadb35">[de82bdd746]</a> Alexei Gradinari -- pjsip: replace 180 by 183 if SDP negotiation has completed</li>
</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26006">ASTERISK-26006</a>: Show offending IP for TLS setup failures in logs<br/>Reported by: Oleksandr Natalenko<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dc61e41fa93ae9da0299ba24aa2b27f187e9a18">[0dc61e41fa]</a> George Joseph -- tcptls.c: Add peer hostname and port to some error messages</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26006">ASTERISK-26006</a>: Show offending IP for TLS setup failures in logs<br/>Reported by: Oleksandr Natalenko<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dc61e41fa93ae9da0299ba24aa2b27f187e9a18">[0dc61e41fa]</a> George Joseph -- tcptls.c: Add peer hostname and port to some error messages</li>
</ul><br><h4>Category: Resources/res_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27981">ASTERISK-27981</a>: res_fax: Fax session leak with fax gatewaying<br/>Reported by: pasandev<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0a574253e9ac2f91cf2860ae2a18adf32784e56">[e0a574253e]</a> Alexei Gradinari -- res_fax: fix segfault on inactive "reserved" fax session</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28458">ASTERISK-28458</a>: res_pjsip_sdp_rtp: Remove unused variable<br/>Reported by: Michael Maier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=83aba363fe57fd5192205586828bada78384dd9b">[83aba363fe]</a> Kevin Harwell -- res_pjsip_sdp_rtp: Remove unused variable</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28018">ASTERISK-28018</a>: IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate<br/>Reported by: vijay kumar<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82789aafd6bd428b5112aa3d028d8ce4098acc1e">[82789aafd6]</a> Joshua Colp -- res_rtp_asterisk: Add support for DTLS packet fragmentation.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28421">ASTERISK-28421</a>: Wrong type used for timestamp in res_rtp_asterisk<br/>Reported by: Morten Tryfoss<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9351aa3f0e2c45db7c2379a94688531b6e674765">[9351aa3f0e]</a> Morten Tryfoss -- res_rtp_asterisk: timestamp should be unsigned instead of signed int</li>
</ul><br><h3>Improvement</h3><h4>Category: PBX/pbx_dundi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28234">ASTERISK-28234</a>: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi<br/>Reported by: Kirsty Tyerman<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1c84709b868583d5d030b1ba093c1228edc0779">[a1c84709b8]</a> Kirsty Tyerman -- pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi</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=cb831a18d15a9c1434df757a23d34bd3f488af99">cb831a18d1</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.5.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17560292371766a4dc0ba236f84c94a1e5d6212c">1756029237</a></td><td>Joshua Colp</td><td>res_rtp_asterisk: Move where DTLS MTU variable is defined.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31d755e8050f0dbe2b8a214625674f04767c1f87">31d755e805</a></td><td>George Joseph</td><td>sig_pri: Address gcc9 issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=01712bbdc950f4d1efce4251c360e7db793809e4">01712bbdc9</a></td><td>George Joseph</td><td>CI: New way to determnine libdir</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ee2f01f62572aeec728e5413a89527e24ba8cfb">1ee2f01f62</a></td><td>George Joseph</td><td>chan_dahdi: Address gcc9 issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b77318a2cc4746529380271cd9da4c62b819b90">8b77318a2c</a></td><td>Alexei Gradinari</td><td>translate.c do not log WARNING on empty audio frame</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ccc92b6ecb6f40df20cceff2d1e2703bf7230c04">ccc92b6ecb</a></td><td>George Joseph</td><td>app_confbridge: Attended transfer event fixup</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=694097ee68c25e8cb58f44e68e09e19283870010">694097ee68</a></td><td>Sean Bright</td><td>pjproject: Update to 2.9 release</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86cd77ec0a22b380a73b9d859779181ae6e13bff">86cd77ec0a</a></td><td>Alexei Gradinari</td><td>app_attended_transfer: new application AttendedTransfer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6321b559b9ef0881690233907320294a9beea3ce">6321b559b9</a></td><td>Alexei Gradinari</td><td>res_fax: gateway sends T.38 request to both endpoints if V.21 detected</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2c07acecaa34434fd989c972a2432b084d1789d">d2c07aceca</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.4.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e77704f45c829c605f2eb28ce72ef49892cfe393">e77704f45c</a></td><td>Alexei Gradinari</td><td>res_fax: add channel name to CLI 'fax show session'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec74fd56a7c22c858c4b7f8876b5610aab09dd3e">ec74fd56a7</a></td><td>Ben Ford</td><td>build: Fix file format in CHANGES-staging.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=db5bc0fabfb68c87e04770ba2a5ea57be7bfb117">db5bc0fabf</a></td><td>Alexei Gradinari</td><td>app_blind_transfer: new application BlindTransfer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9516fb64c9d59437c6527a23f8d5168e0d7eaf0e">9516fb64c9</a></td><td>Alexei Gradinari</td><td>app_readexten: new option 'p' to stop reading on '#' key</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>.lastclean | 1
.version | 1
ChangeLog |84424 ----------
asterisk-16.4.0-summary.html | 229
asterisk-16.4.0-summary.txt | 568
b/CHANGES | 27
b/apps/app_amd.c | 2
b/apps/app_attended_transfer.c | 143
b/apps/app_blind_transfer.c | 137
b/apps/app_confbridge.c | 157
b/apps/app_readexten.c | 11
b/apps/confbridge/confbridge_manager.c | 27
b/apps/confbridge/include/confbridge.h | 11
b/cdr/cdr_pgsql.c | 24
b/channels/chan_dahdi.c | 16
b/channels/chan_pjsip.c | 33
b/channels/chan_sip.c | 8
b/channels/sig_analog.c | 7
b/channels/sig_pri.c | 7
b/configs/samples/dundi.conf.sample | 6
b/configs/samples/rtp.conf.sample | 4
b/configure | 112
b/configure.ac | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/stasis_bridges.h | 23
b/main/stasis_bridges.c | 36
b/main/tcptls.c | 24
b/main/translate.c | 5
b/menuselect/configure | 14
b/menuselect/example_menuselect-tree | 4
b/menuselect/test/menuselect-tree | 4
b/pbx/pbx_dundi.c | 131
b/res/res_fax.c | 57
b/res/res_pjsip_messaging.c | 9
b/res/res_pjsip_sdp_rtp.c | 1
b/res/res_rtp_asterisk.c | 201
b/tests/CI/buildAsterisk.sh | 2
b/tests/CI/findLibdir.sh | 14
b/third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1237
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1335
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch | 33
third-party/pjproject/patches/0020-Fixed-2172-Avoid-double-reference-counter-decrements.patch | 42
third-party/pjproject/patches/0030-Re-2176-Removed-pop_freelist-push_freelist-after-rem.patch | 37
third-party/pjproject/patches/0031-Re-2191-transport-timer-cleanup.patch | 372
third-party/pjproject/patches/0032-Re-2191-Fixed-crash-in-SIP-transport-destroy-due-to-.patch | 96
50 files changed, 1289 insertions(+), 88665 deletions(-)</pre><br></html>

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Release Summary
asterisk-16.5.0-rc1
Date: 2019-07-18
<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-16.4.0.
----------------------------------------------------------------------
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
8 Alexei Gradinari 1 Gil Richard
7 George Joseph 1 Oleksandr Natalenko
2 Asterisk Development Team 1 Oleksandr Natalenko
2 Joshua Colp 1 Nasir Iqbal
2 Chris-Savinovich 1 Guido Falsi
1 Kevin Harwell 1 Kirsty Tyerman
1 Sean Bright 1 Kirsty Tyerman
1 Kirsty Tyerman 1 abelbeck
1 Abhay Gupta 1 Bernhard Schmidt
1 Francesco Castellano 1 Abhay Gupta
1 Ben Ford 1 Gil Richard
1 Morten Tryfoss 1 vijay kumar
1 Guido Falsi 1 Dmitry Svyatogorov
1 Nasir Iqbal 1 pasandev
1 Francesco Castellano
1 Gregory Massel
1 Bernhard Schmidt
1 Morten Tryfoss
1 Michael Maier
----------------------------------------------------------------------
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.
Security
Category: Channels/chan_sip/Interoperability
ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE
Reported by: Francesco Castellano
* [6c59df17a5] Francesco Castellano -- chan_sip: Handle invalid SDP
answer to T.38 re-invite
Category: Resources/res_pjsip_messaging
ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes
crash
Reported by: Gil Richard
* [2126dc3021] George Joseph -- res_pjsip_messaging: Check for body in
in-dialog message
Bug
Category: Applications/app_amd
ASTERISK-28419: app_amd: Does not work with silence suppression
Reported by: Nasir Iqbal
* [52a3d4a761] Nasir Iqbal -- app_amd: issue with silence suppression
fixed
Category: CDR/cdr_pgsql
ASTERISK-28435: cdr_pgsql: Unix socket doesn't work
Reported by: Dmitry Svyatogorov
* [c2621aa190] Chris-Savinovich -- cdr_pgsql: fix error in connection
string
Category: Channels/chan_dahdi
ASTERISK-28457: [patch] Fix crash in chan_dahdi on 32-bit systems caused
by ASTERISK-28317
Reported by: abelbeck
* [da1db4f842] Chris-Savinovich -- chan_dahdi.c: crash in chan_dahdi
ASTERISK-28427: new mwi.h include missing from some dahdi source files,
causes build failure
Reported by: Guido Falsi
* [86fb72c4d0] Guido Falsi -- chan_dahdi: add missing include.
Category: Channels/chan_pjsip
ASTERISK-28444: chan_pjsip: Peer IP for SSL handshake errors not logged
Reported by: Bernhard Schmidt
* [2db5173b88] George Joseph -- pjproject_bundled: Add peer information
to most SSL/TLS errors
ASTERISK-25371: Crash in hangup at chan_pjsip.c:1749 when Asterisk
attempts to generate hangup event
Reported by: Abhay Gupta
* [72f26aa8eb] Abhay Gupta -- chan_pjsip.c: Check for channel and
session to not be NULL in hangup
ASTERISK-27994: PJSIP: Early media ringback not indicated after Progress()
Reported by: Gregory Massel
* [de82bdd746] Alexei Gradinari -- pjsip: replace 180 by 183 if SDP
negotiation has completed
Category: Channels/chan_sip/TCP-TLS
ASTERISK-26006: Show offending IP for TLS setup failures in logs
Reported by: Oleksandr Natalenko
* [0dc61e41fa] George Joseph -- tcptls.c: Add peer hostname and port to
some error messages
Category: Core/Logging
ASTERISK-26006: Show offending IP for TLS setup failures in logs
Reported by: Oleksandr Natalenko
* [0dc61e41fa] George Joseph -- tcptls.c: Add peer hostname and port to
some error messages
Category: Resources/res_fax
ASTERISK-27981: res_fax: Fax session leak with fax gatewaying
Reported by: pasandev
* [e0a574253e] Alexei Gradinari -- res_fax: fix segfault on inactive
"reserved" fax session
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28458: res_pjsip_sdp_rtp: Remove unused variable
Reported by: Michael Maier
* [83aba363fe] Kevin Harwell -- res_pjsip_sdp_rtp: Remove unused
variable
Category: Resources/res_rtp_asterisk
ASTERISK-28018: IP Fragmentation happening instead of DTLS fragmentation
on handshake server hello certificate
Reported by: vijay kumar
* [82789aafd6] Joshua Colp -- res_rtp_asterisk: Add support for DTLS
packet fragmentation.
ASTERISK-28421: Wrong type used for timestamp in res_rtp_asterisk
Reported by: Morten Tryfoss
* [9351aa3f0e] Morten Tryfoss -- res_rtp_asterisk: timestamp should be
unsigned instead of signed int
Improvement
Category: PBX/pbx_dundi
ASTERISK-28234: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
Reported by: Kirsty Tyerman
* [a1c84709b8] Kirsty Tyerman -- pbx_dundi: added IPv4/IPv6 dual bind
support to DUNDi
----------------------------------------------------------------------
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 |
|------------+------------------+----------------------------------------|
| cb831a18d1 | Asterisk | Update CHANGES and UPGRADE.txt for |
| | Development Team | 16.5.0 |
|------------+------------------+----------------------------------------|
| 1756029237 | Joshua Colp | res_rtp_asterisk: Move where DTLS MTU |
| | | variable is defined. |
|------------+------------------+----------------------------------------|
| 31d755e805 | George Joseph | sig_pri: Address gcc9 issues |
|------------+------------------+----------------------------------------|
| 01712bbdc9 | George Joseph | CI: New way to determnine libdir |
|------------+------------------+----------------------------------------|
| 1ee2f01f62 | George Joseph | chan_dahdi: Address gcc9 issues |
|------------+------------------+----------------------------------------|
| 8b77318a2c | Alexei Gradinari | translate.c do not log WARNING on |
| | | empty audio frame |
|------------+------------------+----------------------------------------|
| ccc92b6ecb | George Joseph | app_confbridge: Attended transfer |
| | | event fixup |
|------------+------------------+----------------------------------------|
| 694097ee68 | Sean Bright | pjproject: Update to 2.9 release |
|------------+------------------+----------------------------------------|
| 86cd77ec0a | Alexei Gradinari | app_attended_transfer: new application |
| | | AttendedTransfer |
|------------+------------------+----------------------------------------|
| 6321b559b9 | Alexei Gradinari | res_fax: gateway sends T.38 request to |
| | | both endpoints if V.21 detected |
|------------+------------------+----------------------------------------|
| d2c07aceca | Asterisk | Update CHANGES and UPGRADE.txt for |
| | Development Team | 16.4.0 |
|------------+------------------+----------------------------------------|
| e77704f45c | Alexei Gradinari | res_fax: add channel name to CLI 'fax |
| | | show session' |
|------------+------------------+----------------------------------------|
| ec74fd56a7 | Ben Ford | build: Fix file format in |
| | | CHANGES-staging. |
|------------+------------------+----------------------------------------|
| db5bc0fabf | Alexei Gradinari | app_blind_transfer: new application |
| | | BlindTransfer |
|------------+------------------+----------------------------------------|
| 9516fb64c9 | Alexei Gradinari | app_readexten: new option 'p' to stop |
| | | reading on '#' key |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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 |84424 ----------
asterisk-16.4.0-summary.html | 229
asterisk-16.4.0-summary.txt | 568
b/CHANGES | 27
b/apps/app_amd.c | 2
b/apps/app_attended_transfer.c | 143
b/apps/app_blind_transfer.c | 137
b/apps/app_confbridge.c | 157
b/apps/app_readexten.c | 11
b/apps/confbridge/confbridge_manager.c | 27
b/apps/confbridge/include/confbridge.h | 11
b/cdr/cdr_pgsql.c | 24
b/channels/chan_dahdi.c | 16
b/channels/chan_pjsip.c | 33
b/channels/chan_sip.c | 8
b/channels/sig_analog.c | 7
b/channels/sig_pri.c | 7
b/configs/samples/dundi.conf.sample | 6
b/configs/samples/rtp.conf.sample | 4
b/configure | 112
b/configure.ac | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/stasis_bridges.h | 23
b/main/stasis_bridges.c | 36
b/main/tcptls.c | 24
b/main/translate.c | 5
b/menuselect/configure | 14
b/menuselect/example_menuselect-tree | 4
b/menuselect/test/menuselect-tree | 4
b/pbx/pbx_dundi.c | 131
b/res/res_fax.c | 57
b/res/res_pjsip_messaging.c | 9
b/res/res_pjsip_sdp_rtp.c | 1
b/res/res_rtp_asterisk.c | 201
b/tests/CI/buildAsterisk.sh | 2
b/tests/CI/findLibdir.sh | 14
b/third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1237
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1335
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch | 33
third-party/pjproject/patches/0020-Fixed-2172-Avoid-double-reference-counter-decrements.patch | 42
third-party/pjproject/patches/0030-Re-2176-Removed-pop_freelist-push_freelist-after-rem.patch | 37
third-party/pjproject/patches/0031-Re-2191-transport-timer-cleanup.patch | 372
third-party/pjproject/patches/0032-Re-2191-Fixed-crash-in-SIP-transport-destroy-due-to-.patch | 96
50 files changed, 1289 insertions(+), 88665 deletions(-)

@ -0,0 +1,41 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start DATETIME,
answer DATETIME,
end DATETIME,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';

File diff suppressed because it is too large Load Diff

@ -0,0 +1,35 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BLOB,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';

@ -0,0 +1,45 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start TIMESTAMP WITHOUT TIME ZONE,
answer TIMESTAMP WITHOUT TIME ZONE,
"end" TIMESTAMP WITHOUT TIME ZONE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
COMMIT;

File diff suppressed because it is too large Load Diff

@ -0,0 +1,39 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BYTEA,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
COMMIT;
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