Bug#860034: unblock: asterisk/1:13.14.1~dfsg-1
Bernhard Schmidt
berni at debian.org
Mon Apr 10 14:09:17 UTC 2017
Package: release.debian.org
Severity: normal
User: release.debian.org at packages.debian.org
Usertags: unblock
Please unblock package asterisk
Compared to 13.14.0~dfsg-1 in Stretch 13.14.1 fixes a security
vulnerability in the CDR code (AST-2017-001 / CVE-2017-7617, see
#859910) and nothing else. Thus I have imported the new upstream
version. The diff is a bit noisy due to the release notes being renamed
and trimmed.
Additionally I have imported two upstream commits to fix Bug#859911.
Basically Asterisk 13 with pjsip has quite some issues regarding source
address selection when running on systems with more than one IP address
(multihomed and/or clustered). It often leaves the source-address
selection to the kernel, sending RTP from a different IP than being
signaled inside SIP. This breaks voice through NAT routers and other
stateful firewalls.
The upstream fix does not handle all issues yet, but is an improvement
over the current state of affairs.
unblock asterisk/1:13.14.1~dfsg-1
diffstat for asterisk-13.14.0~dfsg asterisk-13.14.1~dfsg
.version | 1
ChangeLog | 19
asterisk-13.14.0-summary.html | 336 ---
asterisk-13.14.0-summary.txt | 878 ----------
asterisk-13.14.1-summary.html | 13
asterisk-13.14.1-summary.txt | 83
debian/changelog | 9
debian/patches/859911-pjsip-set-rtp-source-address-part2.patch | 62
debian/patches/859911-pjsip-set-rtp-source-address.patch | 47
debian/patches/series | 4
main/cdr.c | 2
11 files changed, 238 insertions(+), 1216 deletions(-)
Thanks,
Bernhard
-------------- next part --------------
diffstat for asterisk-13.14.0~dfsg asterisk-13.14.1~dfsg
.version | 1
ChangeLog | 19
asterisk-13.14.0-summary.html | 336 ---
asterisk-13.14.0-summary.txt | 878 ----------
asterisk-13.14.1-summary.html | 13
asterisk-13.14.1-summary.txt | 83
debian/changelog | 9
debian/patches/859911-pjsip-set-rtp-source-address-part2.patch | 62
debian/patches/859911-pjsip-set-rtp-source-address.patch | 47
debian/patches/series | 4
main/cdr.c | 2
11 files changed, 238 insertions(+), 1216 deletions(-)
diff -Nru asterisk-13.14.0~dfsg/asterisk-13.14.0-summary.html asterisk-13.14.1~dfsg/asterisk-13.14.0-summary.html
--- asterisk-13.14.0~dfsg/asterisk-13.14.0-summary.html 2017-02-13 20:51:51.000000000 +0100
+++ asterisk-13.14.1~dfsg/asterisk-13.14.0-summary.html 1970-01-01 01:00:00.000000000 +0100
@@ -1,336 +0,0 @@
-<!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-13.14.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.14.0</h3><h3 align="center">Date: 2017-02-13</h3><h3 align="center"><asteriskteam at digium.com></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="#open_issues">Open 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-13.13.0.</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%">35 Richard Mudgett <rmudgett at digium.com><br/>19 George Joseph <gjoseph at digium.com><br/>8 Joshua Colp <jcolp at digium.com><br/>8 Mark Michelson <mmichelson at digium.com><br/>4 Badalyan Vyacheslav <v.badalyan at open-bs.ru><br/>4 Tzafrir Cohen <tzafrir.cohen at xorcom.com><br/>3 Sean Bright <sean.bright at gmail.com><br/>3 Kirill Katsnelson <kkm at smartaction.com><br/>2 Michael Kuron <m.kuron at gmx.de><br/>2 Kevin Harwell <kharwell at digium.com><br/>2 Alexander Traud <pabstraud at compuserve.com><br/>2 Matt Jordan <mjordan at digium.com><br/>2 Timo Ter?s <timo.teras at iki.fi><br/>1 Walter Doekes <walter+asterisk at wjd.nu><br/>1 Alexei Gradinari <alex2grad at gmail.com><br/>1 Martin Tomec <tomec.martin at gmail.com><br/>1 Guido Falsi <madpilot at freebsd.org><br/>1 Joshua Elson <josh at fluentstream.com><br/>1 jassim mishal <jmishal at yahoo.com><br/>1 Aaron An <anjb at ti-net.com.cn><br/>1 nappsoft <infos at nappsoft.com><br/>1 JoshE <josh at fluentstream.com><br/>1 Corey Farrell <git at cfware.com><br/>1 gestoip2 <gestoip2 at ull.edu.es><br/>1 Eduardo Scudeller Libardi <eslibardi at gmail.com><br/>1 David M. Lee <dlee at digium.com><br/>1 Torrey Searle <tsearle at gmail.com><br/></td><td width="33%">1 AaronAn<br/></td><td width="33%">6 Richard Mudgett <rmudgett at digium.com><br/>4 Tzafrir Cohen <tzafrir.cohen at xorcom.com><br/>4 George Joseph <gjoseph at digium.com><br/>3 Michael Kuron <m.kuron at gmx.de><br/>3 Joshua Colp <jcolp at digium.com><br/>3 Kirill Katsnelson <kkm at smartaction.com><br/>3 Tom Pawelek <tompaw at tompaw.net><br/>2 Rusty Newton <rnewton at digium.com><br/>2 Morten Tryfoss <morten at tryfoss.no><br/>2 Alexander Traud <pabstraud at compuserve.com><br/>2 Michael Maier <m1278468 at mailbox.org><br/>1 Walter Doekes <walter+asterisk at wjd.nu><br/>1 Alexei Gradinari <alex2grad at gmail.com><br/>1 Ustinov Artem <artem198315 at gmail.com><br/>1 Dmitriy <codeworld at bk.ru><br/>1 Michael Keuter <lists at mksolutions.info><br/>1 Jonas Kellens <jonas.kellens at telenet.be><br/>1 Guido Falsi <madpilot at freebsd.org><br/>1 Ross Beer <ross.beer at voicehost.co.uk><br/>1 Mikheili Dautashvili <mishadaut at gmail.com><br/>1 Juris Breicis <juris.breicis at gmail.com><br/>1 Aaron An <anjb at ti-net.com.cn><br/>1 nappsoft <infos at nappsoft.com><br/>1 JoshE <josh at fluentstream.com><br/>1 Hector Royo Concepcion <hectorroyo92 at gmail.com><br/>1 Frankie Chin <fchin at biamp.com><br/>1 Eduardo S. Libardi<br/>1 Eduardo Scudeller Libardi <eslibardi at gmail.com><br/>1 Badalian Vyacheslav <slavon.net at gmail.com><br/>1 Doug Lytle <support at drdos.info><br/>1 David Woolley <david.woolley at bts.co.uk><br/>1 David Woolley <forums at david-woolley.me.uk><br/>1 Jonathan Rose <jrose at digium.com><br/>1 Joshua Elson <josh at fluentstream.com><br/>1 Anthony Messina <amessina at messinet.com><br/>1 Marek Cervenka <marek.cervenka at gmail.com><br/>1 Florian Loyau <florian.loyau at astrium-eu-projects.eu><br/>1 cervajs, Inaki Baz Castillo<br/>1 J?rgen H <asterisk.org at hovland.cx><br/>1 Mark Michelson <mmichelson at digium.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>New Feature</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26630">ASTERISK-26630</a>: Make logging PJPROJECT messages a bit easier<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44fe4a576968a61fc3c0f6633e5ea9fc2c104105">[44fe4a5769]</a> Richard Mudgett -- PJPROJECT logging: Made easier to get available logging levels.</li>
-</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26630">ASTERISK-26630</a>: Make logging PJPROJECT messages a bit easier<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44fe4a576968a61fc3c0f6633e5ea9fc2c104105">[44fe4a5769]</a> Richard Mudgett -- PJPROJECT logging: Made easier to get available logging levels.</li>
-</ul><br><h3>Bug</h3><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>Reported by: Rusty Newton<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=91145741885859471c6f07124ffda91d1b2bdd58">[9114574188]</a> Richard Mudgett -- res_pjsip: Add/update ERROR msg if invalid URI.</li>
-</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21094">ASTERISK-21094</a>: MixMonitorMute mutes through stream if already slinear (e.g. Originate)<br/>Reported by: David Woolley<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3aee19991355fc8c4bad6066dd7a0b1cf003c077">[3aee199913]</a> Sean Bright -- audiohooks: Muting a hook can mute underlying frames</li>
-</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26755">ASTERISK-26755</a>: app_queue: Random queues disappear on "core reload queue all"<br/>Reported by: Kirill Katsnelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be92f10a16aaa575a91c00b065abd9562b0cc133">[be92f10a16]</a> Kirill Katsnelson -- app_queue: Fix queues randomly disappearing on reload</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26621">ASTERISK-26621</a>: app_queue: Queue application does not ring members with Local interface<br/>Reported by: Jonas Kellens<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d13be4eff699449172efbd9fed0ee97f6a790b6a">[d13be4eff6]</a> Martin Tomec -- app_queue: Ensure member is removed from pending when hanging up.</li>
-</ul><br><h4>Category: Applications/app_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26716">ASTERISK-26716</a>: ari: Channels with pre-dial handlers cannot be hung up via ARI<br/>Reported by: Tom Pawelek<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43f0ff4b69358687a5f958a57dd8ba4bffc4a5f4">[43f0ff4b69]</a> Richard Mudgett -- channel.c: Fix unbalanced read queue deadlocking local channels.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a199f94908601be10c0b43e3b3b7d67c41c5025f">[a199f94908]</a> Richard Mudgett -- res_agi: Prevent an AGI from eating frames it should not. (Re-do)</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bed318a660c5258925c84f5d3f7012f3183c2de">[6bed318a66]</a> Richard Mudgett -- Frame deferral: Revert API refactoring.</li>
-</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26503">ASTERISK-26503</a>: app_voicemail: Asterisk crashes when MailboxExists is used<br/>Reported by: Doug Lytle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c761b8f45892211b52d2c6655d2641aa4a64cd6">[9c761b8f45]</a> Joshua Colp -- app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.</li>
-</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b285d226d2e9d92f1be53e84435c7d4cbf290be">[4b285d226d]</a> Richard Mudgett -- chan_dahdi.c: Fix bounds check regression.</li>
-</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26673">ASTERISK-26673</a>: chan_pjsip: Crash when using CHANNEL dialplan function around masquerade<br/>Reported by: Joshua Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34e728cfb9f175d85b463547ebf9977305b76c1b">[34e728cfb9]</a> Joshua Colp -- chan_pjsip: Use session for retrieving CHANNEL() information.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26603">ASTERISK-26603</a>: [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no<br/>Reported by: Alexei Gradinari<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf6d13180effc92a2483dccc68f2f188689a40fa">[cf6d13180e]</a> Alexei Gradinari -- chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no</li>
-</ul><br><h4>Category: Channels/chan_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26672">ASTERISK-26672</a>: Crash when setting remote address on RTP instance<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9e459f8ace24f4d0c4febc2eb7fa17162c577c6">[a9e459f8ac]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix uninitialized memory crash.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bcdd282adaae7d47c91172d4e38306fc3b3566c7">[bcdd282ada]</a> Richard Mudgett -- res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac31233dbe996b4e2a1ccf9b7b2b35f6ed19e43c">[ac31233dbe]</a> Richard Mudgett -- acl.c: Improve ast_ouraddrfor() diagnostic messages.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0aa5db4b38e80a059c3cd2c1d70ee9c509c0f488">[0aa5db4b38]</a> Richard Mudgett -- chan_rtp.c: Fix uninitialized memory crash.</li>
-</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26691">ASTERISK-26691</a>: Remember SDP negotiation on SIP_CODEC_INBOUND.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=367128e70be203f1c2a7b5371cd8f72a5dc69905">[367128e70b]</a> Alexander Traud -- chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND.</li>
-</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b285d226d2e9d92f1be53e84435c7d4cbf290be">[4b285d226d]</a> Richard Mudgett -- chan_dahdi.c: Fix bounds check regression.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26523">ASTERISK-26523</a>: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression<br/>Reported by: Michael Keuter<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3614d75f650abd3a9028fccabfcd88649381284">[a3614d75f6]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</li>
-</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26433">ASTERISK-26433</a>: chan_sip: Allows To-tag checks to be bypassed, setting up new calls<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41c6319c4e1261f40813e60017e3b65f4115c94d">[41c6319c4e]</a> Walter Doekes -- chan_sip: Do not allow non-SP/HTAB between header key and colon.</li>
-</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26586">ASTERISK-26586</a>: chan_sip: Segfaults upon reload if client with MWI wasn't registered<br/>Reported by: Michael Kuron<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=493849dcd7224145388fac81bc791d219f17aae1">[493849dcd7]</a> Corey Farrell -- chan_sip: Reorder unload_module to deal with stuck TCP threads.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0cc8351484ffbad5c5d7294f68016a83de00b7c6">[0cc8351484]</a> Michael Kuron -- chan_sip: Fix segfault during module unload</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26604">ASTERISK-26604</a>: chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.<br/>Reported by: Michael Kuron<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e77d6f52039e3ba20374c8a0083ad73bcce9e98">[8e77d6f520]</a> Michael Kuron -- tcptls: Use new certificate upon sip reload</li>
-</ul><br><h4>Category: Channels/chan_sip/WebSocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24330">ASTERISK-24330</a>: Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118<br/>Reported by: Marek Cervenka<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09c36a65356093e2a40cd2b23ff9f945e1028589">[09c36a6535]</a> Matt Jordan -- res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter</li>
-</ul><br><h4>Category: Channels/chan_skinny</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b285d226d2e9d92f1be53e84435c7d4cbf290be">[4b285d226d]</a> Richard Mudgett -- chan_dahdi.c: Fix bounds check regression.</li>
-</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=339c30f2b65b005817b073bc9b7c8ad2e7e3fde9">[339c30f2b6]</a> Sean Bright -- res_rtp_asterisk: Swap byte-order when sending signed linear</li>
-</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26546">ASTERISK-26546</a>: mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'<br/>Reported by: Tzafrir Cohen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bfb8c962c4485815a1e0552113408738fff30eb2">[bfb8c962c4]</a> Tzafrir Cohen -- autoconf: more variants for OSARCH linux-gnu</li>
-</ul><br><h4>Category: Core/DNS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26772">ASTERISK-26772</a>: Crash in srv.c on startup with pjsip<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f2ae7908d265e2c565f1aae5717b4047e2e4504">[1f2ae7908d]</a> nappsoft -- srv: Fix crash when ast_srv_lookup is used and 0 records are returned.</li>
-</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26632">ASTERISK-26632</a>: core: Possibility of a frame "imbalance" leading to stuck channels.<br/>Reported by: Mark Michelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43f0ff4b69358687a5f958a57dd8ba4bffc4a5f4">[43f0ff4b69]</a> Richard Mudgett -- channel.c: Fix unbalanced read queue deadlocking local channels.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26753">ASTERISK-26753</a>: AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe"<br/>Reported by: Kirill Katsnelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=555e8cd2ba3d6e24ce678e7ebc0fb9a8895ead65">[555e8cd2ba]</a> Kirill Katsnelson -- ast_careful_fwrite to support EPIPE gracefully</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25083">ASTERISK-25083</a>: Message.c: Message channel becomes saturated with frames leading to spammy log messages<br/>Reported by: Jonathan Rose<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75a6afbec53141f26ad41c94c1b3499116b23508">[75a6afbec5]</a> Richard Mudgett -- MESSAGE: Flush Message/ast_msg_queue channel alert pipe.</li>
-</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=339c30f2b65b005817b073bc9b7c8ad2e7e3fde9">[339c30f2b6]</a> Sean Bright -- res_rtp_asterisk: Swap byte-order when sending signed linear</li>
-</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26704">ASTERISK-26704</a>: res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.<br/>Reported by: Anthony Messina<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70aff89e5dc7f04235e24382b034cb93b4262321">[70aff89e5d]</a> Sean Bright -- res_odbc: Remove deprecated settings from sample configuration file</li>
-</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26754">ASTERISK-26754</a>: build_tools: make_build_h does not handle \ in user name <br/>Reported by: Kirill Katsnelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c8f84786e7700d678a09d2060838ec2a4815b7d">[3c8f84786e]</a> Kirill Katsnelson -- make_build_h: handle backslashes in external strings</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26546">ASTERISK-26546</a>: mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'<br/>Reported by: Tzafrir Cohen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bfb8c962c4485815a1e0552113408738fff30eb2">[bfb8c962c4]</a> Tzafrir Cohen -- autoconf: more variants for OSARCH linux-gnu</li>
-</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25951">ASTERISK-25951</a>: res_agi: run_agi eats frames it shouldn't<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a199f94908601be10c0b43e3b3b7d67c41c5025f">[a199f94908]</a> Richard Mudgett -- res_agi: Prevent an AGI from eating frames it should not. (Re-do)</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26343">ASTERISK-26343</a>: ASTERISK-25951 causes issues for callerid manipulation through agi<br/>Reported by: Morten Tryfoss<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a199f94908601be10c0b43e3b3b7d67c41c5025f">[a199f94908]</a> Richard Mudgett -- res_agi: Prevent an AGI from eating frames it should not. (Re-do)</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bed318a660c5258925c84f5d3f7012f3183c2de">[6bed318a66]</a> Richard Mudgett -- Frame deferral: Revert API refactoring.</li>
-</ul><br><h4>Category: Resources/res_format_attr_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26579">ASTERISK-26579</a>: codec_opus: Recursiveness when parsing fmtp line<br/>Reported by: J?rgen H<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=888142e89160169220bff1bae742f6ef558a7b11">[888142e891]</a> Joshua Colp -- res_format_attr_opus: Fix crash when fmtp contains spaces.</li>
-</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24330">ASTERISK-24330</a>: Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118<br/>Reported by: Marek Cervenka<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09c36a65356093e2a40cd2b23ff9f945e1028589">[09c36a6535]</a> Matt Jordan -- res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter</li>
-</ul><br><h4>Category: Resources/res_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26704">ASTERISK-26704</a>: res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.<br/>Reported by: Anthony Messina<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70aff89e5dc7f04235e24382b034cb93b4262321">[70aff89e5d]</a> Sean Bright -- res_odbc: Remove deprecated settings from sample configuration file</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26679">ASTERISK-26679</a>: Crash on invalid contact domain (pjsip aor)<br/>Reported by: Dmitriy<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e371e13b9eb6ce1f60bf31a8d25c7d4bdcd01d8d">[e371e13b9e]</a> Joshua Colp -- res_pjsip: Handle invocation of callback on outgoing request when error occurs.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26699">ASTERISK-26699</a>: res_pjsip: Assertion when sending OPTIONS request to endpoint<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e371e13b9eb6ce1f60bf31a8d25c7d4bdcd01d8d">[e371e13b9e]</a> Joshua Colp -- res_pjsip: Handle invocation of callback on outgoing request when error occurs.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26743">ASTERISK-26743</a>: PJPROJECT: Detecting compiled max log level does not work.<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=30cb4eb57fe25e14a14e99008f5706b0f994d10b">[30cb4eb57f]</a> Richard Mudgett -- PJPROJECT logging: Fix detection of max supported log level.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26684">ASTERISK-26684</a>: res_pjsip: Various issues with compact SIP headers<br/>Reported by: Joshua Elson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a398f98b08f98538ec6c70b67f46a5a24fe4d001">[a398f98b08]</a> Joshua Elson -- res_pjsip: Fix known compact header issues</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>Reported by: Rusty Newton<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=91145741885859471c6f07124ffda91d1b2bdd58">[9114574188]</a> Richard Mudgett -- res_pjsip: Add/update ERROR msg if invalid URI.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26490">ASTERISK-26490</a>: res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"<br/>Reported by: Juris Breicis<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d506874477d2b2d64f50fa5d2fa77565934656d6">[d506874477]</a> Richard Mudgett -- Bundled pjproject: Fix finding SIP transactions.</li>
-</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26743">ASTERISK-26743</a>: PJPROJECT: Detecting compiled max log level does not work.<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=30cb4eb57fe25e14a14e99008f5706b0f994d10b">[30cb4eb57f]</a> Richard Mudgett -- PJPROJECT logging: Fix detection of max supported log level.</li>
-</ul><br><h4>Category: Resources/res_pjsip_endpoint_identifier_ip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26735">ASTERISK-26735</a>: res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect<br/>Reported by: Michael Maier<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aae9df06433512f58701daf66ae99c292d6c9908">[aae9df0643]</a> Joshua Colp -- res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d23b2e360789f44923b527fa97564a2ff648268">[6d23b2e360]</a> Joshua Colp -- res_pjsip_endpoint_identifier_ip: Read settings before resolving.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26693">ASTERISK-26693</a>: res_pjsip_endpoint_identifier_ip: Add support for SRV<br/>Reported by: Joshua Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a2f0adccbd0fc3ea18a222783aeecd09cf427815">[a2f0adccbd]</a> Joshua Colp -- res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37aaaa2da24ee7f3631a63b26c96fd3586cedbcb">[37aaaa2da2]</a> Joshua Colp -- res_pjsip_endpoint_identifier_ip: Add support for SRV lookups.</li>
-</ul><br><h4>Category: Resources/res_pjsip_registrar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26644">ASTERISK-26644</a>: PJSIPShowRegistrationsInbound just dumps all aors<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ebc67d305398caf8738dcc4b7c88b9e40fa3d58e">[ebc67d3053]</a> gtjoseph -- res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command</li>
-</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26670">ASTERISK-26670</a>: [patch] Outgoing SIP-URI Dialing via PJSIP<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=569dac8e50dd9213dd6c0f9f060fe697377e6f70">[569dac8e50]</a> Alexander Traud -- res_pjsip_session: Access SIPDOMAIN via Dialplan.</li>
-</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=339c30f2b65b005817b073bc9b7c8ad2e7e3fde9">[339c30f2b6]</a> Sean Bright -- res_rtp_asterisk: Swap byte-order when sending signed linear</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26710">ASTERISK-26710</a>: [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0<br/>Reported by: Aaron An<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0047b1bc4962b5051608c8f7a02a8b03ee4d7ec5">[0047b1bc49]</a> Aaron An -- res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt)</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26672">ASTERISK-26672</a>: Crash when setting remote address on RTP instance<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9e459f8ace24f4d0c4febc2eb7fa17162c577c6">[a9e459f8ac]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix uninitialized memory crash.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bcdd282adaae7d47c91172d4e38306fc3b3566c7">[bcdd282ada]</a> Richard Mudgett -- res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac31233dbe996b4e2a1ccf9b7b2b35f6ed19e43c">[ac31233dbe]</a> Richard Mudgett -- acl.c: Improve ast_ouraddrfor() diagnostic messages.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0aa5db4b38e80a059c3cd2c1d70ee9c509c0f488">[0aa5db4b38]</a> Richard Mudgett -- chan_rtp.c: Fix uninitialized memory crash.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26617">ASTERISK-26617</a>: res_rtp_asterisk: Can't bind on systems without IPv6<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ceb609edb64e930300806e75dc1a45386f800ef">[2ceb609edb]</a> Guido Falsi -- res_rtp: Fix regression when IPv6 is not available.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26566">ASTERISK-26566</a>: res_rtp_asterisk: RTT miscalculation in RTCP<br/>Reported by: Hector Royo Concepcion<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8756ce64b7874dbcc0cd9a47aa09fc21cb4f9f81">[8756ce64b7]</a> gestoip2 -- res_rtp_asterisk: RTT miscalculation in RTCP</li>
-</ul><br><h4>Category: Resources/res_sorcery_memory_cache</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26731">ASTERISK-26731</a>: res_sorcery_memory_cache: memory leak on every sorcery memory cache populate<br/>Reported by: Ustinov Artem<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75497c33ea4f52d969c0d845ab30d0c152c34c4d">[75497c33ea]</a> Mark Michelson -- Free endpoint ACLs when destroying PJSIP endpoints.</li>
-</ul><br><h4>Category: Tests/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26740">ASTERISK-26740</a>: voicemail API test: uses varlibdir instead of datadir for a sound file<br/>Reported by: Tzafrir Cohen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cd2677f966574980caeb6da2cdcaa2dc5ec1e83f">[cd2677f966]</a> Tzafrir Cohen -- tests: use datadir for sound files</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26739">ASTERISK-26739</a>: voicemail API test: confuses expected and actual values<br/>Reported by: Tzafrir Cohen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b62f84bfb1cec9ad7099aaa6f63ac4cfc08cb829">[b62f84bfb1]</a> Tzafrir Cohen -- test_voicemail_api: order of params to VERIFY macros</li>
-</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26653">ASTERISK-26653</a>: pjproject_bundled doesn't verify already downloaded tarballs<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cd46e86491e3f8946bbe9c97b8a1d7eaf708dda9">[cd46e86491]</a> gtjoseph -- pjproject_bundled: Retry download if previously saved tarball is bad</li>
-</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26655">ASTERISK-26655</a>: [patch]pjsip: Transfers Broken with Compact Headers Enabled<br/>Reported by: JoshE<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0ab9d103f614c6339334c60bc032c1f8f941326c">[0ab9d103f6]</a> JoshE -- res_pjsip_refer: Handle compact Refer-To header.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26490">ASTERISK-26490</a>: res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"<br/>Reported by: Juris Breicis<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d506874477d2b2d64f50fa5d2fa77565934656d6">[d506874477]</a> Richard Mudgett -- Bundled pjproject: Fix finding SIP transactions.</li>
-</ul><br><h3>Improvement</h3><h4>Category: Applications/app_controlplayback</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26562">ASTERISK-26562</a>: app_controlplayback: Transmit Silence on ControlPlayback pause<br/>Reported by: Mikheili Dautashvili<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=12c4e664bc864e4272688fe9ce10a99d0630bc34">[12c4e664bc]</a> Mikheili Dautashvili -- main/app.c: Transmit Silence on ControlPlayback pause</li>
-</ul><br><h4>Category: Resources/res_calendar_caldav</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26624">ASTERISK-26624</a>: res_calendar_caldav: Add support for gmail<br/>Reported by: Eduardo Scudeller Libardi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=53459cdaa958bbbebc3edb18f460e522bbf56b2f">[53459cdaa9]</a> Eduardo S. Libardi -- res_calendar_caldav: Add support reading gmail calendar</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23828">ASTERISK-23828</a>: pjsip - Need a command to list active SIP subscriptions<br/>Reported by: Rusty Newton<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e3dcb9ddd957fb64ddd43a7656463c0fe0c444cf">[e3dcb9ddd9]</a> Richard Mudgett -- res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands.</li>
-</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26527">ASTERISK-26527</a>: Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec<br/>Reported by: Badalian Vyacheslav<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61ba2a014a25a8423f775b61b7ac56f951be3df8">[61ba2a014a]</a> Richard Mudgett -- res_pjsip_outbound_registration.c: Filter redundant statsd reporting.</li>
-</ul><br><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26109">ASTERISK-26109</a>: Asterisk fails building with OpenSSL 1.1.0<br/>Reported by: Tzafrir Cohen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b0c9f07f040dfa1713899f6b5ad1e3321bd56481">[b0c9f07f04]</a> Tzafrir Cohen -- OpenSSL 1.1.0 support</li>
-</ul><br><h4>Category: Core/Jitterbuffer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25270">ASTERISK-25270</a>: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind<br/>Reported by: Florian Loyau<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3614d75f650abd3a9028fccabfcd88649381284">[a3614d75f6]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</li>
-</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25270">ASTERISK-25270</a>: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind<br/>Reported by: Florian Loyau<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3614d75f650abd3a9028fccabfcd88649381284">[a3614d75f6]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</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=ec97c41ac817bce13d8bd9436c35ad3ef95a9f72">ec97c41ac8</a></td><td>gtjoseph</td><td>Update for 13.14.0-rc2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f3290ce4f83c00a8ab4c25378ebf0c0e797453b">3f3290ce4f</a></td><td>Mark Michelson</td><td>Revert "Update qualifies when AOR configuration changes."</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7dbe77d63946ce204611a75d5f79d37f9d624ee1">7dbe77d639</a></td><td>gtjoseph</td><td>Update for 13.14.0-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6492e91392b8fd394193e411c6eb64b45486093f">6492e91392</a></td><td>Mark Michelson</td><td>Update qualifies when AOR configuration changes.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fd28cefdb2a41308b22b6e29d9a10ff38acb2e5">7fd28cefdb</a></td><td>gtjoseph</td><td>debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=456bc3c704ffce7807caa6f5f4bc92ed1720c871">456bc3c704</a></td><td>gtjoseph</td><td>debug_utilities: Add ast_logescalator</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=54b027916a71f2b83b2050cef5ef704ea5de39b2">54b027916a</a></td><td>Torrey Searle</td><td>libastssl/pj: libastssl/pj should have an so_version</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e3150b98ddf4ecfe09e281c73c634b89ba2514e">9e3150b98d</a></td><td>Mark Michelson</td><td>Add reload options to CLI/AMI stale object commands.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c54f9d2bf01ab943c122c5326da52decf6e492cd">c54f9d2bf0</a></td><td>Richard Mudgett</td><td>T.140: Fix format ref and memory leaks.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=607b3ac736e9bdd69acb69f85c2d5a66271a4bb2">607b3ac736</a></td><td>Richard Mudgett</td><td>astobj2.c: Add excessive ref count trap.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab8cb5a7ce227ed2c889d58a04d7167aa49e3f63">ab8cb5a7ce</a></td><td>Richard Mudgett</td><td>main/app.c: Memory corruption from early format destruction.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dcd8e4b1a0fd5159e6799a71d15ba56e16c374a7">dcd8e4b1a0</a></td><td>Richard Mudgett</td><td>frame.c: Fix off-nominal format ref leaks.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=00a227e93dec78170a0637a1216f619f70f73df9">00a227e93d</a></td><td>Richard Mudgett</td><td>stasis_bridge.c: Fix off-nominal stasis control ref leak.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=38a2021c68df5352f601439c7317cac5082d10a8">38a2021c68</a></td><td>Richard Mudgett</td><td>res_musiconhold.c: Fix format ref leak when parsing MOH config class.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab7a9fc5b29c274651159b18f7057c9b150fede3">ab7a9fc5b2</a></td><td>Richard Mudgett</td><td>chan_oss.c: Fix format ref leak in oss_read().</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1484a991e17bb3320fbf8d1cfbe92d7ba5d6ea01">1484a991e1</a></td><td>Richard Mudgett</td><td>Add notes about embedded ast_frame structs holding a format ref.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17f4989d49c7117684c04bff8a3e6fed1c6e6aad">17f4989d49</a></td><td>gtjoseph</td><td>ari: Implement 'debug all' and request/response logging</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=177e81ee47708a30b0a042acf4be19b4b812676e">177e81ee47</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix setting max log level</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a969bf357708514104b67274a6c07d4697ea0eff">a969bf3577</a></td><td>Richard Mudgett</td><td>LISTFILTER: Remove outdated ERROR message.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3890337e7a14bb04bcd2d60b2112de588ec93828">3890337e7a</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix AMI event list counts.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe4801c4f95a89ecc83cec14bec9f90a40994528">fe4801c4f9</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix incorrect message string wrapping.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46484b87300e249b1a8584fe0f45a57e8362f723">46484b8730</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Eliminate trivial SCOPED_LOCK usage.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8160474d7db9b5975172c88d27cc6c1880700831">8160474d7d</a></td><td>Richard Mudgett</td><td>res_pjsip: alloca can never fail.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c628a7acacea0d519e0e5919a2a1db1ba5851ee0">c628a7acac</a></td><td>gtjoseph</td><td>debug_utilities: Create ast_loggrabber</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e335b706ee4e592c168e23dca5a813f84988778f">e335b706ee</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_authenticator_digest.c: Fix spacing in warning messages.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=883e7fde31ab9db886c482f12383f6baa57e9e92">883e7fde31</a></td><td>Kevin Harwell</td><td>abstract/fixed/adpative jitter buffer: disallow frame re-inserts</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=473330983b47618772b1dd414df8e063e5da6a53">473330983b</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Change when high water warning logged.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=47474cfd54a9185c1433464ccfd6301427a03957">47474cfd54</a></td><td>gtjoseph</td><td>debug_utilities: Create the ast_coredumper utility</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f8cd73ec3c159f2e6c464952c92d8fdb69394371">f8cd73ec3c</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix compilation with MALLOC_DEBUG</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d7e5a747c312de18647213359103ce6022776864">d7e5a747c3</a></td><td>gtjoseph</td><td>pjproject_bundled: Compile pjsua with max log level = 2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21151408f7d984c7652f479f03ddd452e1d275f7">21151408f7</a></td><td>Richard Mudgett</td><td>bridge_native_rtp.c: Minor code cleanups.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9dcf9e9cea87b7bf7f2bbdbaddd51377a6138bfa">9dcf9e9cea</a></td><td>Richard Mudgett</td><td>bridge_native_rtp.c: Fix native rtp bridge data race.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e2fa3c7edabd52b62830a32930b0b4aaf4bcb9ac">e2fa3c7eda</a></td><td>Richard Mudgett</td><td>res_rtp_asterisk.c: Fix off nominal memory leak.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=815f7551550908c83220196ba08742af0c745772">815f755155</a></td><td>gtjoseph</td><td>pjproject_bundled: Make build single threaded</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab447f8a6a3d05908395ed52ec3b981c1e8943c1">ab447f8a6a</a></td><td>David M. Lee</td><td>configure: fix with-pjproject-bundled</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=35736d419a85b877072fe3688f410376b8ce4613">35736d419a</a></td><td>Richard Mudgett</td><td>autosupport: Add 'pjproject show buildopts'</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=91485734a48e9e44292b80b8993fb464bc3817b0">91485734a4</a></td><td>gtjoseph</td><td>res_sorcery_memory_cache: Change an error to a debug message</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=22820e10fe866df47373bb2ef18d159cea40079b">22820e10fe</a></td><td>Badalyan Vyacheslav</td><td>chan_sip: Delete unneeded check</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6aa2c5e5f967128a89a95e40c8846bbd091eb6fb">6aa2c5e5f9</a></td><td>Badalyan Vyacheslav</td><td>Small code cleanup in chan_sip</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b596fac8384c470af6fc656953609882c629fe7d">b596fac838</a></td><td>Badalyan Vyacheslav</td><td>Fix typo in chan_sip</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=483ed9f1aa2575a94f9ccfd6c490e72d4611f249">483ed9f1aa</a></td><td>Badalyan Vyacheslav</td><td>res_pjsip: Fix 'A = B != C' kind.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b233675d86fef64e6ccce82a0f7b038d3c7e032">4b233675d8</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix missing inclusion of symbols</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=580f83dac713614bd8904f8722a900299ff4f8bc">580f83dac7</a></td><td>Richard Mudgett</td><td>Remove files that got merged in error somehow to the 13 branch.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17b0b91afa1aa6b8b503a75e96b054f13612356a">17b0b91afa</a></td><td>Mark Michelson</td><td>Frame deferral: Re-queue deferred frames one-at-a-time.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a33ed3327aa7530b9c9dbd98bd575f85c15ec8a4">a33ed3327a</a></td><td>Matt Jordan</td><td>res/res_pjsip: Fix documentation whitespace issues</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29e887e9e11d3a60ebc0f177ba3dcddb2e07e8fe">29e887e9e1</a></td><td>gtjoseph</td><td>build_tools: Fix download_externals to handle certified branches</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1fa9090337aeb14cc5e86cd7d212d35cc95b37a">a1fa909033</a></td><td>Timo Ter?s</td><td>codec_dahdi: Fix poll.h include.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86d824b7fff5c9135694dbff6dc2a64b51964752">86d824b7ff</a></td><td>Timo Ter?s</td><td>addons/chan_mobile: do not use strerror_r</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=425da14927f10205911f5db6d6b193b22203c224">425da14927</a></td><td>gtjoseph</td><td>build: Backport addition of librt check to configure.ac</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a40c3a8679f1a02d88c48f0424fa41df09203a8">2a40c3a867</a></td><td>gtjoseph</td><td>pjproject_bundled: Improve reliability of pjproject download</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ee73af1d88c9ff6db90f70f934f5ea57b8ab0625">ee73af1d88</a></td><td>gtjoseph</td><td>Update for 13.12.2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d7b52c434eb23ef470ad51d08ee4029a7078b78">7d7b52c434</a></td><td>Mark Michelson</td><td>Update for 13.12.1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=226a7e36c538de73cee76de4183b1569bd5501e5">226a7e36c5</a></td><td>Mark Michelson</td><td>Update for 13.12.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df75b647da03eba6920020bac0cc950032a1e930">df75b647da</a></td><td>Mark Michelson</td><td>Update for 13.12.0-rc1</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-13.13.0-summary.html | 304 -
-asterisk-13.13.0-summary.txt | 767 ----
-b/.version | 2
-b/CHANGES | 50
-b/ChangeLog | 1614 +++++++++-
-b/addons/chan_mobile.c | 13
-b/apps/app_queue.c | 40
-b/apps/app_voicemail.c | 6
-b/asterisk-13.14.0-rc2-summary.html | 20
-b/asterisk-13.14.0-rc2-summary.txt | 104
-b/bootstrap.sh | 2
-b/bridges/bridge_native_rtp.c | 81
-b/build_tools/download_externals | 4
-b/build_tools/make_build_h | 10
-b/channels/chan_dahdi.c | 4
-b/channels/chan_oss.c | 2
-b/channels/chan_pjsip.c | 9
-b/channels/chan_rtp.c | 7
-b/channels/chan_sip.c | 111
-b/channels/pjsip/dialplan_functions.c | 62
-b/codecs/codec_dahdi.c | 2
-b/configs/samples/ast_debug_tools.conf.sample | 57
-b/configs/samples/pjproject.conf.sample | 25
-b/configs/samples/res_odbc.conf.sample | 6
-b/configure | 12
-b/configure.ac | 10
-b/contrib/Makefile | 14
-b/contrib/ast-db-manage/config/versions/28ab27a7826d_add_srv_lookups_to_identify.py | 31
-b/contrib/realtime/mssql/mssql_config.sql | 14
-b/contrib/realtime/mysql/mysql_config.sql | 6
-b/contrib/realtime/oracle/oracle_config.sql | 14
-b/contrib/realtime/postgresql/postgresql_config.sql | 6
-b/contrib/scripts/ast_coredumper | 533 +++
-b/contrib/scripts/ast_logescalator | 399 ++
-b/contrib/scripts/ast_loggrabber | 255 +
-b/contrib/scripts/autosupport | 4
-b/funcs/func_strings.c | 1
-b/include/asterisk/abstract_jb.h | 3
-b/include/asterisk/ari.h | 24
-b/include/asterisk/channel.h | 72
-b/include/asterisk/frame.h | 2
-b/include/asterisk/mod_format.h | 6
-b/include/asterisk/options.h | 41
-b/include/asterisk/res_pjsip.h | 10
-b/include/asterisk/rtp_engine.h | 4
-b/include/asterisk/stasis_app.h | 49
-b/include/asterisk/tcptls.h | 4
-b/include/asterisk/translate.h | 2
-b/include/asterisk/utils.h | 9
-b/include/jitterbuf.h | 3
-b/main/Makefile | 12
-b/main/abstract_jb.c | 31
-b/main/acl.c | 36
-b/main/app.c | 36
-b/main/asterisk.c | 35
-b/main/astobj2.c | 13
-b/main/audiohook.c | 40
-b/main/autoservice.c | 66
-b/main/channel.c | 217 -
-b/main/channel_internal_api.c | 68
-b/main/fixedjitterbuf.c | 6
-b/main/fixedjitterbuf.h | 3
-b/main/format_compatibility.c | 4
-b/main/frame.c | 69
-b/main/jitterbuf.c | 5
-b/main/libasteriskpj.c | 2
-b/main/libasteriskssl.c | 4
-b/main/manager.c | 6
-b/main/message.c | 9
-b/main/rtp_engine.c | 2
-b/main/srv.c | 3
-b/main/strings.c | 21
-b/main/taskprocessor.c | 8
-b/main/tcptls.c | 94
-b/main/utils.c | 24
-b/makeopts.in | 2
-b/res/ari/ari_websockets.c | 14
-b/res/ari/cli.c | 175 +
-b/res/ari/resource_events.c | 9
-b/res/res_agi.c | 10
-b/res/res_ari.c | 77
-b/res/res_ari_applications.c | 42
-b/res/res_ari_asterisk.c | 120
-b/res/res_ari_bridges.c | 162 -
-b/res/res_ari_channels.c | 318 -
-b/res/res_ari_device_states.c | 27
-b/res/res_ari_endpoints.c | 45
-b/res/res_ari_events.c | 18
-b/res/res_ari_mailboxes.c | 27
-b/res/res_ari_playbacks.c | 24
-b/res/res_ari_recordings.c | 48
-b/res/res_ari_sounds.c | 21
-b/res/res_calendar_caldav.c | 4
-b/res/res_format_attr_opus.c | 38
-b/res/res_musiconhold.c | 2
-b/res/res_pjproject.c | 156
-b/res/res_pjsip.c | 72
-b/res/res_pjsip/pjsip_configuration.c | 6
-b/res/res_pjsip/pjsip_options.c | 17
-b/res/res_pjsip_diversion.c | 3
-b/res/res_pjsip_endpoint_identifier_ip.c | 160
-b/res/res_pjsip_history.c | 91
-b/res/res_pjsip_outbound_authenticator_digest.c | 12
-b/res/res_pjsip_outbound_registration.c | 21
-b/res/res_pjsip_pubsub.c | 635 +++
-b/res/res_pjsip_refer.c | 8
-b/res/res_pjsip_registrar.c | 70
-b/res/res_pjsip_sdp_rtp.c | 7
-b/res/res_pjsip_session.c | 8
-b/res/res_pjsip_t38.c | 7
-b/res/res_pjsip_transport_websocket.c | 5
-b/res/res_rtp_asterisk.c | 117
-b/res/res_sorcery_memory_cache.c | 41
-b/res/res_stasis.c | 23
-b/res/stasis/app.c | 73
-b/res/stasis/app.h | 27
-b/res/stasis/stasis_bridge.c | 6
-b/rest-api-templates/param_parsing.mustache | 15
-b/rest-api-templates/res_ari_resource.c.mustache | 3
-b/tests/test_ari.c | 22
-b/tests/test_substitution.c | 9
-b/tests/test_voicemail_api.c | 52
-b/third-party/Makefile | 2
-b/third-party/Makefile.rules | 4
-b/third-party/pjproject/Makefile | 61
-b/third-party/pjproject/configure.m4 | 8
-b/third-party/pjproject/patches/0000-set_apps_initial_log_level.patch | 13
-res/stasis/cli.c | 216 -
-res/stasis/cli.h | 43
-129 files changed, 5945 insertions(+), 2895 deletions(-)</pre><br></html>
\ Kein Zeilenumbruch am Dateiende.
diff -Nru asterisk-13.14.0~dfsg/asterisk-13.14.0-summary.txt asterisk-13.14.1~dfsg/asterisk-13.14.0-summary.txt
--- asterisk-13.14.0~dfsg/asterisk-13.14.0-summary.txt 2017-02-13 20:51:51.000000000 +0100
+++ asterisk-13.14.1~dfsg/asterisk-13.14.0-summary.txt 1970-01-01 01:00:00.000000000 +0100
@@ -1,878 +0,0 @@
- Release Summary
-
- asterisk-13.14.0
-
- Date: 2017-02-13
-
- <asteriskteam at digium.com>
-
- ----------------------------------------------------------------------
-
- Table of Contents
-
- 1. Summary
- 2. Contributors
- 3. Closed Issues
- 4. Open Issues
- 5. Other Changes
- 6. 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-13.13.0.
-
- ----------------------------------------------------------------------
-
- 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
- 35 Richard Mudgett 1 AaronAn 6 Richard Mudgett
- 19 George Joseph 4 Tzafrir Cohen
- 8 Joshua Colp 4 George Joseph
- 8 Mark Michelson 3 Michael Kuron
- 4 Badalyan Vyacheslav 3 Joshua Colp
- 4 Tzafrir Cohen 3 Kirill Katsnelson
- 3 Sean Bright 3 Tom Pawelek
- 3 Kirill Katsnelson 2 Rusty Newton
- 2 Michael Kuron 2 Morten Tryfoss
- 2 Kevin Harwell 2 Alexander Traud
- 2 Alexander Traud 2 Michael Maier
- 2 Matt Jordan 1 Walter Doekes
- 2 Timo TerACURs 1 Alexei Gradinari
- 1 Walter Doekes 1 Ustinov Artem
- 1 Alexei Gradinari 1 Dmitriy
- 1 Martin Tomec 1 Michael Keuter
- 1 Guido Falsi 1 Jonas Kellens
- 1 Joshua Elson 1 Guido Falsi
- 1 jassim mishal 1 Ross Beer
- 1 Aaron An 1 Mikheili Dautashvili
- 1 nappsoft 1 Juris Breicis
- 1 JoshE 1 Aaron An
- 1 Corey Farrell 1 nappsoft
- 1 gestoip2 1 JoshE
- 1 Eduardo Scudeller Libardi 1 Hector Royo Concepcion
- 1 David M. Lee 1 Frankie Chin
- 1 Torrey Searle 1 Eduardo S. Libardi
- 1 Eduardo Scudeller Libardi
- 1 Badalian Vyacheslav
- 1 Doug Lytle
- 1 David Woolley
- 1 David Woolley
- 1 Jonathan Rose
- 1 Joshua Elson
- 1 Anthony Messina
- 1 Marek Cervenka
- 1 Florian Loyau
- 1 cervajs, Inaki Baz Castillo
- 1 JA,rgen H
- 1 Mark Michelson
-
- ----------------------------------------------------------------------
-
- 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: Core/General
-
- ASTERISK-26630: Make logging PJPROJECT messages a bit easier
- Reported by: Richard Mudgett
- * [44fe4a5769] Richard Mudgett -- PJPROJECT logging: Made easier to get
- available logging levels.
-
- Category: Resources/res_pjsip/Bundling
-
- ASTERISK-26630: Make logging PJPROJECT messages a bit easier
- Reported by: Richard Mudgett
- * [44fe4a5769] Richard Mudgett -- PJPROJECT logging: Made easier to get
- available logging levels.
-
- Bug
-
- Category: Applications/app_dial
-
- ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
- Reported by: Rusty Newton
- * [9114574188] Richard Mudgett -- res_pjsip: Add/update ERROR msg if
- invalid URI.
-
- Category: Applications/app_mixmonitor
-
- ASTERISK-21094: MixMonitorMute mutes through stream if already slinear
- (e.g. Originate)
- Reported by: David Woolley
- * [3aee199913] Sean Bright -- audiohooks: Muting a hook can mute
- underlying frames
-
- Category: Applications/app_queue
-
- ASTERISK-26755: app_queue: Random queues disappear on "core reload queue
- all"
- Reported by: Kirill Katsnelson
- * [be92f10a16] Kirill Katsnelson -- app_queue: Fix queues randomly
- disappearing on reload
- ASTERISK-26621: app_queue: Queue application does not ring members with
- Local interface
- Reported by: Jonas Kellens
- * [d13be4eff6] Martin Tomec -- app_queue: Ensure member is removed from
- pending when hanging up.
-
- Category: Applications/app_stasis
-
- ASTERISK-26716: ari: Channels with pre-dial handlers cannot be hung up via
- ARI
- Reported by: Tom Pawelek
- * [43f0ff4b69] Richard Mudgett -- channel.c: Fix unbalanced read queue
- deadlocking local channels.
- * [a199f94908] Richard Mudgett -- res_agi: Prevent an AGI from eating
- frames it should not. (Re-do)
- * [6bed318a66] Richard Mudgett -- Frame deferral: Revert API
- refactoring.
-
- Category: Applications/app_voicemail
-
- ASTERISK-26503: app_voicemail: Asterisk crashes when MailboxExists is used
- Reported by: Doug Lytle
- * [9c761b8f45] Joshua Colp -- app_voicemail: Clear voice mailbox in
- MailboxExists and MAILBOX_EXISTS.
-
- Category: Channels/chan_dahdi
-
- ASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and
- missing paren issues
- Reported by: George Joseph
- * [4b285d226d] Richard Mudgett -- chan_dahdi.c: Fix bounds check
- regression.
-
- Category: Channels/chan_pjsip
-
- ASTERISK-26673: chan_pjsip: Crash when using CHANNEL dialplan function
- around masquerade
- Reported by: Joshua Colp
- * [34e728cfb9] Joshua Colp -- chan_pjsip: Use session for retrieving
- CHANNEL() information.
- ASTERISK-26603: [patch] chan_pjsip: not switching sending codec to
- receiving codec when asymmetric_rtp_codec=no
- Reported by: Alexei Gradinari
- * [cf6d13180e] Alexei Gradinari -- chan_pjsip: fix switching sending
- codec when asymmetric_rtp_codec=no
-
- Category: Channels/chan_rtp
-
- ASTERISK-26672: Crash when setting remote address on RTP instance
- Reported by: Richard Mudgett
- * [a9e459f8ac] Richard Mudgett -- res_rtp_asterisk.c: Fix uninitialized
- memory crash.
- * [bcdd282ada] Richard Mudgett -- res_rtp_asterisk.c: Initialize ourip
- passed to ast_find_ourip().
- * [ac31233dbe] Richard Mudgett -- acl.c: Improve ast_ouraddrfor()
- diagnostic messages.
- * [0aa5db4b38] Richard Mudgett -- chan_rtp.c: Fix uninitialized memory
- crash.
-
- Category: Channels/chan_sip/CodecHandling
-
- ASTERISK-26691: Remember SDP negotiation on SIP_CODEC_INBOUND.
- Reported by: Alexander Traud
- * [367128e70b] Alexander Traud -- chan_sip: Remember SDP negotiation on
- SIP_CODEC_INBOUND.
-
- Category: Channels/chan_sip/General
-
- ASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and
- missing paren issues
- Reported by: George Joseph
- * [4b285d226d] Richard Mudgett -- chan_dahdi.c: Fix bounds check
- regression.
- ASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls
- after 2 minutes - rtptimeout behaving badly - regression
- Reported by: Michael Keuter
- * [a3614d75f6] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
- updated"
-
- Category: Channels/chan_sip/Interoperability
-
- ASTERISK-26433: chan_sip: Allows To-tag checks to be bypassed, setting up
- new calls
- Reported by: Walter Doekes
- * [41c6319c4e] Walter Doekes -- chan_sip: Do not allow non-SP/HTAB
- between header key and colon.
-
- Category: Channels/chan_sip/TCP-TLS
-
- ASTERISK-26586: chan_sip: Segfaults upon reload if client with MWI wasn't
- registered
- Reported by: Michael Kuron
- * [493849dcd7] Corey Farrell -- chan_sip: Reorder unload_module to deal
- with stuck TCP threads.
- * [0cc8351484] Michael Kuron -- chan_sip: Fix segfault during module
- unload
- ASTERISK-26604: chan_sip: sip reload doesn't apply changes to tlscertfile,
- tlsciphers, etc.
- Reported by: Michael Kuron
- * [8e77d6f520] Michael Kuron -- tcptls: Use new certificate upon sip
- reload
-
- Category: Channels/chan_sip/WebSocket
-
- ASTERISK-24330: Requirement for 'wss' value in Contact header transport
- parameter on inbound traffic violates RFC7118
- Reported by: Marek Cervenka
- * [09c36a6535] Matt Jordan -- res_pjsip/chan_sip: Advertise 'ws' in the
- SIP URI transport parameter
-
- Category: Channels/chan_skinny
-
- ASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and
- missing paren issues
- Reported by: George Joseph
- * [4b285d226d] Richard Mudgett -- chan_dahdi.c: Fix bounds check
- regression.
-
- Category: Codecs/General
-
- ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
- order on Intel platform when using slin codec
- Reported by: Frankie Chin
- * [339c30f2b6] Sean Bright -- res_rtp_asterisk: Swap byte-order when
- sending signed linear
-
- Category: Core/BuildSystem
-
- ASTERISK-26546: mips64el and x32 - undefined reference to symbol
- 'dlopen@@GLIBC_2.2'
- Reported by: Tzafrir Cohen
- * [bfb8c962c4] Tzafrir Cohen -- autoconf: more variants for OSARCH
- linux-gnu
-
- Category: Core/DNS
-
- ASTERISK-26772: Crash in srv.c on startup with pjsip
- Reported by: nappsoft
- * [1f2ae7908d] nappsoft -- srv: Fix crash when ast_srv_lookup is used
- and 0 records are returned.
-
- Category: Core/General
-
- ASTERISK-26632: core: Possibility of a frame "imbalance" leading to stuck
- channels.
- Reported by: Mark Michelson
- * [43f0ff4b69] Richard Mudgett -- channel.c: Fix unbalanced read queue
- deadlocking local channels.
- ASTERISK-26753: AMI disconnect causes "ast_careful_fwrite: fwrite()
- returned error: Broken pipe"
- Reported by: Kirill Katsnelson
- * [555e8cd2ba] Kirill Katsnelson -- ast_careful_fwrite to support EPIPE
- gracefully
- ASTERISK-25083: Message.c: Message channel becomes saturated with frames
- leading to spammy log messages
- Reported by: Jonathan Rose
- * [75a6afbec5] Richard Mudgett -- MESSAGE: Flush Message/ast_msg_queue
- channel alert pipe.
-
- Category: Core/RTP
-
- ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
- order on Intel platform when using slin codec
- Reported by: Frankie Chin
- * [339c30f2b6] Sean Bright -- res_rtp_asterisk: Swap byte-order when
- sending signed linear
-
- Category: Documentation
-
- ASTERISK-26704: res_odbc.conf contains deprecated configuration:
- 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were
- replaced by 'max_connections'.
- Reported by: Anthony Messina
- * [70aff89e5d] Sean Bright -- res_odbc: Remove deprecated settings from
- sample configuration file
-
- Category: General
-
- ASTERISK-26754: build_tools: make_build_h does not handle \ in user name
- Reported by: Kirill Katsnelson
- * [3c8f84786e] Kirill Katsnelson -- make_build_h: handle backslashes in
- external strings
- ASTERISK-26546: mips64el and x32 - undefined reference to symbol
- 'dlopen@@GLIBC_2.2'
- Reported by: Tzafrir Cohen
- * [bfb8c962c4] Tzafrir Cohen -- autoconf: more variants for OSARCH
- linux-gnu
-
- Category: Resources/res_agi
-
- ASTERISK-25951: res_agi: run_agi eats frames it shouldn't
- Reported by: George Joseph
- * [a199f94908] Richard Mudgett -- res_agi: Prevent an AGI from eating
- frames it should not. (Re-do)
- ASTERISK-26343: ASTERISK-25951 causes issues for callerid manipulation
- through agi
- Reported by: Morten Tryfoss
- * [a199f94908] Richard Mudgett -- res_agi: Prevent an AGI from eating
- frames it should not. (Re-do)
- * [6bed318a66] Richard Mudgett -- Frame deferral: Revert API
- refactoring.
-
- Category: Resources/res_format_attr_opus
-
- ASTERISK-26579: codec_opus: Recursiveness when parsing fmtp line
- Reported by: JA,rgen H
- * [888142e891] Joshua Colp -- res_format_attr_opus: Fix crash when fmtp
- contains spaces.
-
- Category: Resources/res_http_websocket
-
- ASTERISK-24330: Requirement for 'wss' value in Contact header transport
- parameter on inbound traffic violates RFC7118
- Reported by: Marek Cervenka
- * [09c36a6535] Matt Jordan -- res_pjsip/chan_sip: Advertise 'ws' in the
- SIP URI transport parameter
-
- Category: Resources/res_odbc
-
- ASTERISK-26704: res_odbc.conf contains deprecated configuration:
- 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were
- replaced by 'max_connections'.
- Reported by: Anthony Messina
- * [70aff89e5d] Sean Bright -- res_odbc: Remove deprecated settings from
- sample configuration file
-
- Category: Resources/res_pjsip
-
- ASTERISK-26679: Crash on invalid contact domain (pjsip aor)
- Reported by: Dmitriy
- * [e371e13b9e] Joshua Colp -- res_pjsip: Handle invocation of callback
- on outgoing request when error occurs.
- ASTERISK-26699: res_pjsip: Assertion when sending OPTIONS request to
- endpoint
- Reported by: Ross Beer
- * [e371e13b9e] Joshua Colp -- res_pjsip: Handle invocation of callback
- on outgoing request when error occurs.
- ASTERISK-26743: PJPROJECT: Detecting compiled max log level does not work.
- Reported by: Richard Mudgett
- * [30cb4eb57f] Richard Mudgett -- PJPROJECT logging: Fix detection of
- max supported log level.
- ASTERISK-26684: res_pjsip: Various issues with compact SIP headers
- Reported by: Joshua Elson
- * [a398f98b08] Joshua Elson -- res_pjsip: Fix known compact header
- issues
- ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
- Reported by: Rusty Newton
- * [9114574188] Richard Mudgett -- res_pjsip: Add/update ERROR msg if
- invalid URI.
- ASTERISK-26490: res_pjsip: sends 481 Call/Transaction Does Not Exist when
- transaction branch parameter contains "_"
- Reported by: Juris Breicis
- * [d506874477] Richard Mudgett -- Bundled pjproject: Fix finding SIP
- transactions.
-
- Category: Resources/res_pjsip/Bundling
-
- ASTERISK-26743: PJPROJECT: Detecting compiled max log level does not work.
- Reported by: Richard Mudgett
- * [30cb4eb57f] Richard Mudgett -- PJPROJECT logging: Fix detection of
- max supported log level.
-
- Category: Resources/res_pjsip_endpoint_identifier_ip
-
- ASTERISK-26735: res_pjsip_endpoint_identifier_ip: "srv_lookups" after
- match in .conf has no effect
- Reported by: Michael Maier
- * [aae9df0643] Joshua Colp -- res_pjsip_endpoint_identifier_ip: Fix
- memory leak of hosts when resolving.
- * [6d23b2e360] Joshua Colp -- res_pjsip_endpoint_identifier_ip: Read
- settings before resolving.
- ASTERISK-26693: res_pjsip_endpoint_identifier_ip: Add support for SRV
- Reported by: Joshua Colp
- * [a2f0adccbd] Joshua Colp -- res_pjsip_endpoint_identifier_ip: Ensure
- error defaults to 0.
- * [37aaaa2da2] Joshua Colp -- res_pjsip_endpoint_identifier_ip: Add
- support for SRV lookups.
-
- Category: Resources/res_pjsip_registrar
-
- ASTERISK-26644: PJSIPShowRegistrationsInbound just dumps all aors
- Reported by: George Joseph
- * [ebc67d3053] gtjoseph -- res_pjsip_registrar: AMI Add
- RegistrationInboundContactStatuses command
-
- Category: Resources/res_pjsip_session
-
- ASTERISK-26670: [patch] Outgoing SIP-URI Dialing via PJSIP
- Reported by: Alexander Traud
- * [569dac8e50] Alexander Traud -- res_pjsip_session: Access SIPDOMAIN
- via Dialplan.
-
- Category: Resources/res_rtp_asterisk
-
- ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
- order on Intel platform when using slin codec
- Reported by: Frankie Chin
- * [339c30f2b6] Sean Bright -- res_rtp_asterisk: Swap byte-order when
- sending signed linear
- ASTERISK-26710: [patch] res_rtp_asterisk: CHANNEL arguments,
- (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- Reported by: Aaron An
- * [0047b1bc49] Aaron An -- res_rtp_asterisk: Fix bug in function
- CHANNEL(rtcp, all_rtt)
- ASTERISK-26672: Crash when setting remote address on RTP instance
- Reported by: Richard Mudgett
- * [a9e459f8ac] Richard Mudgett -- res_rtp_asterisk.c: Fix uninitialized
- memory crash.
- * [bcdd282ada] Richard Mudgett -- res_rtp_asterisk.c: Initialize ourip
- passed to ast_find_ourip().
- * [ac31233dbe] Richard Mudgett -- acl.c: Improve ast_ouraddrfor()
- diagnostic messages.
- * [0aa5db4b38] Richard Mudgett -- chan_rtp.c: Fix uninitialized memory
- crash.
- ASTERISK-26617: res_rtp_asterisk: Can't bind on systems without IPv6
- Reported by: Guido Falsi
- * [2ceb609edb] Guido Falsi -- res_rtp: Fix regression when IPv6 is not
- available.
- ASTERISK-26566: res_rtp_asterisk: RTT miscalculation in RTCP
- Reported by: Hector Royo Concepcion
- * [8756ce64b7] gestoip2 -- res_rtp_asterisk: RTT miscalculation in RTCP
-
- Category: Resources/res_sorcery_memory_cache
-
- ASTERISK-26731: res_sorcery_memory_cache: memory leak on every sorcery
- memory cache populate
- Reported by: Ustinov Artem
- * [75497c33ea] Mark Michelson -- Free endpoint ACLs when destroying
- PJSIP endpoints.
-
- Category: Tests/General
-
- ASTERISK-26740: voicemail API test: uses varlibdir instead of datadir for
- a sound file
- Reported by: Tzafrir Cohen
- * [cd2677f966] Tzafrir Cohen -- tests: use datadir for sound files
- ASTERISK-26739: voicemail API test: confuses expected and actual values
- Reported by: Tzafrir Cohen
- * [b62f84bfb1] Tzafrir Cohen -- test_voicemail_api: order of params to
- VERIFY macros
-
- Category: Third-Party/pjproject
-
- ASTERISK-26653: pjproject_bundled doesn't verify already downloaded
- tarballs
- Reported by: George Joseph
- * [cd46e86491] gtjoseph -- pjproject_bundled: Retry download if
- previously saved tarball is bad
-
- Category: pjproject/pjsip
-
- ASTERISK-26655: [patch]pjsip: Transfers Broken with Compact Headers
- Enabled
- Reported by: JoshE
- * [0ab9d103f6] JoshE -- res_pjsip_refer: Handle compact Refer-To header.
- ASTERISK-26490: res_pjsip: sends 481 Call/Transaction Does Not Exist when
- transaction branch parameter contains "_"
- Reported by: Juris Breicis
- * [d506874477] Richard Mudgett -- Bundled pjproject: Fix finding SIP
- transactions.
-
- Improvement
-
- Category: Applications/app_controlplayback
-
- ASTERISK-26562: app_controlplayback: Transmit Silence on ControlPlayback
- pause
- Reported by: Mikheili Dautashvili
- * [12c4e664bc] Mikheili Dautashvili -- main/app.c: Transmit Silence on
- ControlPlayback pause
-
- Category: Resources/res_calendar_caldav
-
- ASTERISK-26624: res_calendar_caldav: Add support for gmail
- Reported by: Eduardo Scudeller Libardi
- * [53459cdaa9] Eduardo S. Libardi -- res_calendar_caldav: Add support
- reading gmail calendar
-
- Category: Resources/res_pjsip
-
- ASTERISK-23828: pjsip - Need a command to list active SIP subscriptions
- Reported by: Rusty Newton
- * [e3dcb9ddd9] Richard Mudgett -- res_pjsip_pubsub.c: Implement "pjsip
- show subscriptions" commands.
-
- Category: Tests/testsuite
-
- ASTERISK-26527: Testsuite: increase timeout to check "core fullybooted
- wait" up to 30 sec
- Reported by: Badalian Vyacheslav
- * [61ba2a014a] Richard Mudgett -- res_pjsip_outbound_registration.c:
- Filter redundant statsd reporting.
-
- ----------------------------------------------------------------------
-
- Open Issues
-
- [Back to Top]
-
- This is a list of all open issues from the issue tracker that were
- referenced by changes that went into this release.
-
- Bug
-
- Category: Core/BuildSystem
-
- ASTERISK-26109: Asterisk fails building with OpenSSL 1.1.0
- Reported by: Tzafrir Cohen
- * [b0c9f07f04] Tzafrir Cohen -- OpenSSL 1.1.0 support
-
- Category: Core/Jitterbuffer
-
- ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using
- JitterBuffers of any kind
- Reported by: Florian Loyau
- * [a3614d75f6] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
- updated"
-
- Category: Core/RTP
-
- ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using
- JitterBuffers of any kind
- Reported by: Florian Loyau
- * [a3614d75f6] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
- updated"
-
- ----------------------------------------------------------------------
-
- 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 |
- |------------+-------------+---------------------------------------------|
- | ec97c41ac8 | gtjoseph | Update for 13.14.0-rc2 |
- |------------+-------------+---------------------------------------------|
- | 3f3290ce4f | Mark | Revert "Update qualifies when AOR |
- | | Michelson | configuration changes." |
- |------------+-------------+---------------------------------------------|
- | 7dbe77d639 | gtjoseph | Update for 13.14.0-rc1 |
- |------------+-------------+---------------------------------------------|
- | 6492e91392 | Mark | Update qualifies when AOR configuration |
- | | Michelson | changes. |
- |------------+-------------+---------------------------------------------|
- | 7fd28cefdb | gtjoseph | debug_utilities: Install ast_logescalator |
- | | | to /var/lib/asterisk/scripts |
- |------------+-------------+---------------------------------------------|
- | 456bc3c704 | gtjoseph | debug_utilities: Add ast_logescalator |
- |------------+-------------+---------------------------------------------|
- | 54b027916a | Torrey | libastssl/pj: libastssl/pj should have an |
- | | Searle | so_version |
- |------------+-------------+---------------------------------------------|
- | 9e3150b98d | Mark | Add reload options to CLI/AMI stale object |
- | | Michelson | commands. |
- |------------+-------------+---------------------------------------------|
- | c54f9d2bf0 | Richard | T.140: Fix format ref and memory leaks. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | 607b3ac736 | Richard | astobj2.c: Add excessive ref count trap. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | ab8cb5a7ce | Richard | main/app.c: Memory corruption from early |
- | | Mudgett | format destruction. |
- |------------+-------------+---------------------------------------------|
- | dcd8e4b1a0 | Richard | frame.c: Fix off-nominal format ref leaks. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | 00a227e93d | Richard | stasis_bridge.c: Fix off-nominal stasis |
- | | Mudgett | control ref leak. |
- |------------+-------------+---------------------------------------------|
- | 38a2021c68 | Richard | res_musiconhold.c: Fix format ref leak when |
- | | Mudgett | parsing MOH config class. |
- |------------+-------------+---------------------------------------------|
- | ab7a9fc5b2 | Richard | chan_oss.c: Fix format ref leak in |
- | | Mudgett | oss_read(). |
- |------------+-------------+---------------------------------------------|
- | 1484a991e1 | Richard | Add notes about embedded ast_frame structs |
- | | Mudgett | holding a format ref. |
- |------------+-------------+---------------------------------------------|
- | 17f4989d49 | gtjoseph | ari: Implement 'debug all' and |
- | | | request/response logging |
- |------------+-------------+---------------------------------------------|
- | 177e81ee47 | gtjoseph | pjproject_bundled: Fix setting max log |
- | | | level |
- |------------+-------------+---------------------------------------------|
- | a969bf3577 | Richard | LISTFILTER: Remove outdated ERROR message. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | 3890337e7a | Richard | res_pjsip_pubsub.c: Fix AMI event list |
- | | Mudgett | counts. |
- |------------+-------------+---------------------------------------------|
- | fe4801c4f9 | Richard | res_pjsip_pubsub.c: Fix incorrect message |
- | | Mudgett | string wrapping. |
- |------------+-------------+---------------------------------------------|
- | 46484b8730 | Richard | res_pjsip_pubsub.c: Eliminate trivial |
- | | Mudgett | SCOPED_LOCK usage. |
- |------------+-------------+---------------------------------------------|
- | 8160474d7d | Richard | res_pjsip: alloca can never fail. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | c628a7acac | gtjoseph | debug_utilities: Create ast_loggrabber |
- |------------+-------------+---------------------------------------------|
- | e335b706ee | Richard | res_pjsip_outbound_authenticator_digest.c: |
- | | Mudgett | Fix spacing in warning messages. |
- |------------+-------------+---------------------------------------------|
- | 883e7fde31 | Kevin | abstract/fixed/adpative jitter buffer: |
- | | Harwell | disallow frame re-inserts |
- |------------+-------------+---------------------------------------------|
- | 473330983b | Richard | taskprocessor.c: Change when high water |
- | | Mudgett | warning logged. |
- |------------+-------------+---------------------------------------------|
- | 47474cfd54 | gtjoseph | debug_utilities: Create the ast_coredumper |
- | | | utility |
- |------------+-------------+---------------------------------------------|
- | f8cd73ec3c | gtjoseph | pjproject_bundled: Fix compilation with |
- | | | MALLOC_DEBUG |
- |------------+-------------+---------------------------------------------|
- | d7e5a747c3 | gtjoseph | pjproject_bundled: Compile pjsua with max |
- | | | log level = 2 |
- |------------+-------------+---------------------------------------------|
- | 21151408f7 | Richard | bridge_native_rtp.c: Minor code cleanups. |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | 9dcf9e9cea | Richard | bridge_native_rtp.c: Fix native rtp bridge |
- | | Mudgett | data race. |
- |------------+-------------+---------------------------------------------|
- | e2fa3c7eda | Richard | res_rtp_asterisk.c: Fix off nominal memory |
- | | Mudgett | leak. |
- |------------+-------------+---------------------------------------------|
- | 815f755155 | gtjoseph | pjproject_bundled: Make build single |
- | | | threaded |
- |------------+-------------+---------------------------------------------|
- | ab447f8a6a | David M. | configure: fix with-pjproject-bundled |
- | | Lee | |
- |------------+-------------+---------------------------------------------|
- | 35736d419a | Richard | autosupport: Add 'pjproject show buildopts' |
- | | Mudgett | |
- |------------+-------------+---------------------------------------------|
- | 91485734a4 | gtjoseph | res_sorcery_memory_cache: Change an error |
- | | | to a debug message |
- |------------+-------------+---------------------------------------------|
- | 22820e10fe | Badalyan | chan_sip: Delete unneeded check |
- | | Vyacheslav | |
- |------------+-------------+---------------------------------------------|
- | 6aa2c5e5f9 | Badalyan | Small code cleanup in chan_sip |
- | | Vyacheslav | |
- |------------+-------------+---------------------------------------------|
- | b596fac838 | Badalyan | Fix typo in chan_sip |
- | | Vyacheslav | |
- |------------+-------------+---------------------------------------------|
- | 483ed9f1aa | Badalyan | res_pjsip: Fix 'A = B != C' kind. |
- | | Vyacheslav | |
- |------------+-------------+---------------------------------------------|
- | 4b233675d8 | gtjoseph | pjproject_bundled: Fix missing inclusion of |
- | | | symbols |
- |------------+-------------+---------------------------------------------|
- | 580f83dac7 | Richard | Remove files that got merged in error |
- | | Mudgett | somehow to the 13 branch. |
- |------------+-------------+---------------------------------------------|
- | 17b0b91afa | Mark | Frame deferral: Re-queue deferred frames |
- | | Michelson | one-at-a-time. |
- |------------+-------------+---------------------------------------------|
- | a33ed3327a | Matt Jordan | res/res_pjsip: Fix documentation whitespace |
- | | | issues |
- |------------+-------------+---------------------------------------------|
- | 29e887e9e1 | gtjoseph | build_tools: Fix download_externals to |
- | | | handle certified branches |
- |------------+-------------+---------------------------------------------|
- | a1fa909033 | Timo | codec_dahdi: Fix poll.h include. |
- | | TerACURs | |
- |------------+-------------+---------------------------------------------|
- | 86d824b7ff | Timo | addons/chan_mobile: do not use strerror_r |
- | | TerACURs | |
- |------------+-------------+---------------------------------------------|
- | 425da14927 | gtjoseph | build: Backport addition of librt check to |
- | | | configure.ac |
- |------------+-------------+---------------------------------------------|
- | 2a40c3a867 | gtjoseph | pjproject_bundled: Improve reliability of |
- | | | pjproject download |
- |------------+-------------+---------------------------------------------|
- | ee73af1d88 | gtjoseph | Update for 13.12.2 |
- |------------+-------------+---------------------------------------------|
- | 7d7b52c434 | Mark | Update for 13.12.1 |
- | | Michelson | |
- |------------+-------------+---------------------------------------------|
- | 226a7e36c5 | Mark | Update for 13.12.0 |
- | | Michelson | |
- |------------+-------------+---------------------------------------------|
- | df75b647da | Mark | Update for 13.12.0-rc1 |
- | | Michelson | |
- +------------------------------------------------------------------------+
-
- ----------------------------------------------------------------------
-
- 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-13.13.0-summary.html | 304 -
- asterisk-13.13.0-summary.txt | 767 ----
- b/.version | 2
- b/CHANGES | 50
- b/ChangeLog | 1614 +++++++++-
- b/addons/chan_mobile.c | 13
- b/apps/app_queue.c | 40
- b/apps/app_voicemail.c | 6
- b/asterisk-13.14.0-rc2-summary.html | 20
- b/asterisk-13.14.0-rc2-summary.txt | 104
- b/bootstrap.sh | 2
- b/bridges/bridge_native_rtp.c | 81
- b/build_tools/download_externals | 4
- b/build_tools/make_build_h | 10
- b/channels/chan_dahdi.c | 4
- b/channels/chan_oss.c | 2
- b/channels/chan_pjsip.c | 9
- b/channels/chan_rtp.c | 7
- b/channels/chan_sip.c | 111
- b/channels/pjsip/dialplan_functions.c | 62
- b/codecs/codec_dahdi.c | 2
- b/configs/samples/ast_debug_tools.conf.sample | 57
- b/configs/samples/pjproject.conf.sample | 25
- b/configs/samples/res_odbc.conf.sample | 6
- b/configure | 12
- b/configure.ac | 10
- b/contrib/Makefile | 14
- b/contrib/ast-db-manage/config/versions/28ab27a7826d_add_srv_lookups_to_identify.py | 31
- b/contrib/realtime/mssql/mssql_config.sql | 14
- b/contrib/realtime/mysql/mysql_config.sql | 6
- b/contrib/realtime/oracle/oracle_config.sql | 14
- b/contrib/realtime/postgresql/postgresql_config.sql | 6
- b/contrib/scripts/ast_coredumper | 533 +++
- b/contrib/scripts/ast_logescalator | 399 ++
- b/contrib/scripts/ast_loggrabber | 255 +
- b/contrib/scripts/autosupport | 4
- b/funcs/func_strings.c | 1
- b/include/asterisk/abstract_jb.h | 3
- b/include/asterisk/ari.h | 24
- b/include/asterisk/channel.h | 72
- b/include/asterisk/frame.h | 2
- b/include/asterisk/mod_format.h | 6
- b/include/asterisk/options.h | 41
- b/include/asterisk/res_pjsip.h | 10
- b/include/asterisk/rtp_engine.h | 4
- b/include/asterisk/stasis_app.h | 49
- b/include/asterisk/tcptls.h | 4
- b/include/asterisk/translate.h | 2
- b/include/asterisk/utils.h | 9
- b/include/jitterbuf.h | 3
- b/main/Makefile | 12
- b/main/abstract_jb.c | 31
- b/main/acl.c | 36
- b/main/app.c | 36
- b/main/asterisk.c | 35
- b/main/astobj2.c | 13
- b/main/audiohook.c | 40
- b/main/autoservice.c | 66
- b/main/channel.c | 217 -
- b/main/channel_internal_api.c | 68
- b/main/fixedjitterbuf.c | 6
- b/main/fixedjitterbuf.h | 3
- b/main/format_compatibility.c | 4
- b/main/frame.c | 69
- b/main/jitterbuf.c | 5
- b/main/libasteriskpj.c | 2
- b/main/libasteriskssl.c | 4
- b/main/manager.c | 6
- b/main/message.c | 9
- b/main/rtp_engine.c | 2
- b/main/srv.c | 3
- b/main/strings.c | 21
- b/main/taskprocessor.c | 8
- b/main/tcptls.c | 94
- b/main/utils.c | 24
- b/makeopts.in | 2
- b/res/ari/ari_websockets.c | 14
- b/res/ari/cli.c | 175 +
- b/res/ari/resource_events.c | 9
- b/res/res_agi.c | 10
- b/res/res_ari.c | 77
- b/res/res_ari_applications.c | 42
- b/res/res_ari_asterisk.c | 120
- b/res/res_ari_bridges.c | 162 -
- b/res/res_ari_channels.c | 318 -
- b/res/res_ari_device_states.c | 27
- b/res/res_ari_endpoints.c | 45
- b/res/res_ari_events.c | 18
- b/res/res_ari_mailboxes.c | 27
- b/res/res_ari_playbacks.c | 24
- b/res/res_ari_recordings.c | 48
- b/res/res_ari_sounds.c | 21
- b/res/res_calendar_caldav.c | 4
- b/res/res_format_attr_opus.c | 38
- b/res/res_musiconhold.c | 2
- b/res/res_pjproject.c | 156
- b/res/res_pjsip.c | 72
- b/res/res_pjsip/pjsip_configuration.c | 6
- b/res/res_pjsip/pjsip_options.c | 17
- b/res/res_pjsip_diversion.c | 3
- b/res/res_pjsip_endpoint_identifier_ip.c | 160
- b/res/res_pjsip_history.c | 91
- b/res/res_pjsip_outbound_authenticator_digest.c | 12
- b/res/res_pjsip_outbound_registration.c | 21
- b/res/res_pjsip_pubsub.c | 635 +++
- b/res/res_pjsip_refer.c | 8
- b/res/res_pjsip_registrar.c | 70
- b/res/res_pjsip_sdp_rtp.c | 7
- b/res/res_pjsip_session.c | 8
- b/res/res_pjsip_t38.c | 7
- b/res/res_pjsip_transport_websocket.c | 5
- b/res/res_rtp_asterisk.c | 117
- b/res/res_sorcery_memory_cache.c | 41
- b/res/res_stasis.c | 23
- b/res/stasis/app.c | 73
- b/res/stasis/app.h | 27
- b/res/stasis/stasis_bridge.c | 6
- b/rest-api-templates/param_parsing.mustache | 15
- b/rest-api-templates/res_ari_resource.c.mustache | 3
- b/tests/test_ari.c | 22
- b/tests/test_substitution.c | 9
- b/tests/test_voicemail_api.c | 52
- b/third-party/Makefile | 2
- b/third-party/Makefile.rules | 4
- b/third-party/pjproject/Makefile | 61
- b/third-party/pjproject/configure.m4 | 8
- b/third-party/pjproject/patches/0000-set_apps_initial_log_level.patch | 13
- res/stasis/cli.c | 216 -
- res/stasis/cli.h | 43
- 129 files changed, 5945 insertions(+), 2895 deletions(-)
diff -Nru asterisk-13.14.0~dfsg/asterisk-13.14.1-summary.html asterisk-13.14.1~dfsg/asterisk-13.14.1-summary.html
--- asterisk-13.14.0~dfsg/asterisk-13.14.1-summary.html 1970-01-01 01:00:00.000000000 +0100
+++ asterisk-13.14.1~dfsg/asterisk-13.14.1-summary.html 2017-04-04 14:38:46.000000000 +0200
@@ -0,0 +1,13 @@
+<!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-13.14.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.14.1</h3><h3 align="center">Date: 2017-04-04</h3><h3 align="center"><asteriskteam at digium.com></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 has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
+<li><a href="http://downloads.asterisk.org/pub/security/AST-2017-001.html">AST-2017-001</a></li>
+</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.14.0.</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 Corey Farrell <git at cfware.com><br/></td><td width="33%"><td width="33%">1 Alex Villac?s Lasso <a_villacis at palosanto.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: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26897">ASTERISK-26897</a>: chan_sip: Security vulnerability with client code header<br/>Reported by: Alex Villac?s Lasso<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a667d8f8b1a3f57f43a439b793eb6993817dd82b">[a667d8f8b1]</a> Corey Farrell -- CDR: Protect from data overflow in ast_cdr_setuserfield.</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>0 files changed</pre><br></html>
\ Kein Zeilenumbruch am Dateiende.
diff -Nru asterisk-13.14.0~dfsg/asterisk-13.14.1-summary.txt asterisk-13.14.1~dfsg/asterisk-13.14.1-summary.txt
--- asterisk-13.14.0~dfsg/asterisk-13.14.1-summary.txt 1970-01-01 01:00:00.000000000 +0100
+++ asterisk-13.14.1~dfsg/asterisk-13.14.1-summary.txt 2017-04-04 14:38:46.000000000 +0200
@@ -0,0 +1,83 @@
+ Release Summary
+
+ asterisk-13.14.1
+
+ Date: 2017-04-04
+
+ <asteriskteam at digium.com>
+
+ ----------------------------------------------------------------------
+
+ Table of Contents
+
+ 1. Summary
+ 2. Contributors
+ 3. Closed Issues
+ 4. Diffstat
+
+ ----------------------------------------------------------------------
+
+ Summary
+
+ [Back to Top]
+
+ This release has been made to address one or more security vulnerabilities
+ that have been identified. A security advisory document has been published
+ for each vulnerability that includes additional information. Users of
+ versions of Asterisk that are affected are strongly encouraged to review
+ the advisories and determine what action they should take to protect their
+ systems from these issues.
+
+ Security Advisories:
+
+ * AST-2017-001
+
+ The data in this summary reflects changes that have been made since the
+ previous release, asterisk-13.14.0.
+
+ ----------------------------------------------------------------------
+
+ 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 Corey Farrell 1 Alex VillacAs Lasso
+
+ ----------------------------------------------------------------------
+
+ 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: Channels/chan_sip/General
+
+ ASTERISK-26897: chan_sip: Security vulnerability with client code header
+ Reported by: Alex VillacAs Lasso
+ * [a667d8f8b1] Corey Farrell -- CDR: Protect from data overflow in
+ ast_cdr_setuserfield.
+
+ ----------------------------------------------------------------------
+
+ 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.
+
+ 0 files changed
diff -Nru asterisk-13.14.0~dfsg/ChangeLog asterisk-13.14.1~dfsg/ChangeLog
--- asterisk-13.14.0~dfsg/ChangeLog 2017-02-13 20:51:51.000000000 +0100
+++ asterisk-13.14.1~dfsg/ChangeLog 2017-04-04 14:38:46.000000000 +0200
@@ -1,3 +1,22 @@
+2017-04-04 12:38 +0000 Asterisk Development Team <asteriskteam at digium.com>
+
+ * asterisk 13.14.1 Released.
+
+2017-03-27 09:03 +0000 [a667d8f8b1] Corey Farrell <git at cfware.com>
+
+ * CDR: Protect from data overflow in ast_cdr_setuserfield.
+
+ ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could
+ result in a buffer overrun when called from chan_sip or func_cdr. This patch
+ adds a maximum bytes written to the field by using ast_copy_string instead.
+
+ ASTERISK-26897 #close
+ patches:
+ 0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted
+ by Corey Farrell (license #5909)
+
+ Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c
+
2017-02-13 19:51 +0000 Asterisk Development Team <asteriskteam at digium.com>
* asterisk 13.14.0 Released.
diff -Nru asterisk-13.14.0~dfsg/debian/changelog asterisk-13.14.1~dfsg/debian/changelog
--- asterisk-13.14.0~dfsg/debian/changelog 2017-02-14 21:54:29.000000000 +0100
+++ asterisk-13.14.1~dfsg/debian/changelog 2017-04-10 12:53:03.000000000 +0200
@@ -1,3 +1,12 @@
+asterisk (1:13.14.1~dfsg-1) unstable; urgency=medium
+
+ * New upstream version 13.14.1
+ - Fixes AST-2017-001 (Buffer overflow in CDR's set user) (Closes: #859910)
+ * Import upstream fix to set the RTP source address to the address bound by
+ the PJSIP transport (Closes: #859911)
+
+ -- Bernhard Schmidt <berni at debian.org> Mon, 10 Apr 2017 12:53:03 +0200
+
asterisk (1:13.14.0~dfsg-1) unstable; urgency=medium
[ Bernhard Schmidt ]
diff -Nru asterisk-13.14.0~dfsg/debian/patches/859911-pjsip-set-rtp-source-address-part2.patch asterisk-13.14.1~dfsg/debian/patches/859911-pjsip-set-rtp-source-address-part2.patch
--- asterisk-13.14.0~dfsg/debian/patches/859911-pjsip-set-rtp-source-address-part2.patch 1970-01-01 01:00:00.000000000 +0100
+++ asterisk-13.14.1~dfsg/debian/patches/859911-pjsip-set-rtp-source-address-part2.patch 2017-04-10 12:53:03.000000000 +0200
@@ -0,0 +1,62 @@
+From 2634f48fb57e1358ad1de0683e9779cc05efe2c4 Mon Sep 17 00:00:00 2001
+From: Richard Mudgett <rmudgett at digium.com>
+Date: Mon, 3 Apr 2017 15:38:06 -0500
+Subject: [PATCH] res_pjsip_sdp_rtp.c: Don't alter global addr variable.
+
+* create_rtp(): Fix unexpected alteration of global address_rtp if a
+transport is bound to an address.
+
+* create_rtp(): Fix use of uninitialized memory if the endpoint RTP media
+address is invalid or the transport has an invalid address.
+
+ASTERISK-26851
+
+Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7
+---
+ res/res_pjsip_sdp_rtp.c | 23 ++++++++++++++++++-----
+ 1 file changed, 18 insertions(+), 5 deletions(-)
+
+diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
+index d44171cf8f..4cfd4d5b4a 100644
+--- a/res/res_pjsip_sdp_rtp.c
++++ b/res/res_pjsip_sdp_rtp.c
+@@ -194,8 +194,16 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
+ struct ast_sockaddr *media_address = &address_rtp;
+
+ if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
+- ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
+- media_address = &temp_media_address;
++ if (ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0)) {
++ ast_debug(1, "Endpoint %s: Binding RTP media to %s\n",
++ ast_sorcery_object_get_id(session->endpoint),
++ session->endpoint->media.address);
++ media_address = &temp_media_address;
++ } else {
++ ast_debug(1, "Endpoint %s: RTP media address invalid: %s\n",
++ ast_sorcery_object_get_id(session->endpoint),
++ session->endpoint->media.address);
++ }
+ } else {
+ struct ast_sip_transport *transport =
+ ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
+@@ -205,9 +213,14 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
+ char hoststr[PJ_INET6_ADDRSTRLEN];
+
+ pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
+- ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
+- session->endpoint->transport, hoststr);
+- ast_sockaddr_parse(media_address, hoststr, 0);
++ if (ast_sockaddr_parse(&temp_media_address, hoststr, 0)) {
++ ast_debug(1, "Transport %s bound to %s: Using it for RTP media.\n",
++ session->endpoint->transport, hoststr);
++ media_address = &temp_media_address;
++ } else {
++ ast_debug(1, "Transport %s bound to %s: Invalid for RTP media.\n",
++ session->endpoint->transport, hoststr);
++ }
+ }
+ ao2_cleanup(transport);
+ }
+--
+2.11.0
+
diff -Nru asterisk-13.14.0~dfsg/debian/patches/859911-pjsip-set-rtp-source-address.patch asterisk-13.14.1~dfsg/debian/patches/859911-pjsip-set-rtp-source-address.patch
--- asterisk-13.14.0~dfsg/debian/patches/859911-pjsip-set-rtp-source-address.patch 1970-01-01 01:00:00.000000000 +0100
+++ asterisk-13.14.1~dfsg/debian/patches/859911-pjsip-set-rtp-source-address.patch 2017-04-10 12:53:03.000000000 +0200
@@ -0,0 +1,47 @@
+From 139bc3495f8f46201b4aac6c84a5f0d04822f9da Mon Sep 17 00:00:00 2001
+From: Richard Begg <asterisk at meric.id.au>
+Date: Wed, 15 Mar 2017 08:22:42 +1100
+Subject: [PATCH] res_pjsip_sdp_rtp: RTP instance does not use same IP as
+ explicit transport
+
+Currently a wildcard address is used for the local RTP socket, which
+will not always result in the same address as used by the SIP socket
+(e.g. if explicit transport addresses are configured).
+Use the transport's host address when binding new local RTP sockets if
+available.
+
+ASTERISK-26851
+
+Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a
+---
+ res/res_pjsip_sdp_rtp.c | 14 ++++++++++++++
+ 1 file changed, 14 insertions(+)
+
+diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
+index b27050ed86..2b31d146ea 100644
+--- a/res/res_pjsip_sdp_rtp.c
++++ b/res/res_pjsip_sdp_rtp.c
+@@ -179,6 +179,20 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
+ if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
+ ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
+ media_address = &temp_media_address;
++ } else {
++ struct ast_sip_transport *transport =
++ ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
++ session->endpoint->transport);
++
++ if (transport && transport->state) {
++ char hoststr[PJ_INET6_ADDRSTRLEN];
++
++ pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
++ ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
++ session->endpoint->transport, hoststr);
++ ast_sockaddr_parse(media_address, hoststr, 0);
++ }
++ ao2_cleanup(transport);
+ }
+
+ if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
+--
+2.11.0
+
diff -Nru asterisk-13.14.0~dfsg/debian/patches/series asterisk-13.14.1~dfsg/debian/patches/series
--- asterisk-13.14.0~dfsg/debian/patches/series 2017-02-14 21:54:29.000000000 +0100
+++ asterisk-13.14.1~dfsg/debian/patches/series 2017-04-10 12:53:03.000000000 +0200
@@ -38,3 +38,7 @@
OpenSSL-1.1.0-support-3.patch
pjsip_unresolved_symbol.patch
+
+# Bug#859911 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
+859911-pjsip-set-rtp-source-address.patch
+859911-pjsip-set-rtp-source-address-part2.patch
diff -Nru asterisk-13.14.0~dfsg/main/cdr.c asterisk-13.14.1~dfsg/main/cdr.c
--- asterisk-13.14.0~dfsg/main/cdr.c 2017-02-13 20:51:51.000000000 +0100
+++ asterisk-13.14.1~dfsg/main/cdr.c 2017-04-04 14:38:46.000000000 +0200
@@ -3254,7 +3254,7 @@
if (it_cdr->fn_table == &finalized_state_fn_table) {
continue;
}
- strcpy(it_cdr->party_a.userfield, userfield);
+ ast_copy_string(it_cdr->party_a.userfield, userfield, AST_MAX_USER_FIELD);
}
ao2_unlock(cdr);
}
diff -Nru asterisk-13.14.0~dfsg/.version asterisk-13.14.1~dfsg/.version
--- asterisk-13.14.0~dfsg/.version 2017-02-13 20:51:51.000000000 +0100
+++ asterisk-13.14.1~dfsg/.version 2017-04-04 14:38:46.000000000 +0200
@@ -1 +1 @@
-13.14.0
\ Kein Zeilenumbruch am Dateiende.
+13.14.1
\ Kein Zeilenumbruch am Dateiende.
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