Package: release.debian.org Severity: normal User: release.debian....@packages.debian.org Usertags: unblock
Please unblock package asterisk It's a new upstream patch release fixing a security issue (Bug#923690). Other than that it only contains a change in the Jenkins CI configuration (not relevant for Debian) and the changelog file has been renamed/regenerated. .version | 1 ChangeLog | 35 ++ asterisk-16.2.0-summary.html | 202 --------------- asterisk-16.2.0-summary.txt | 519 ----------------------------------------- asterisk-16.2.1-summary.html | 23 + asterisk-16.2.1-summary.txt | 107 ++++++++ debian/changelog | 9 debian/control | 2 res/res_pjsip_sdp_rtp.c | 2 tests/CI/gates.jenkinsfile | 6 tests/CI/unittests.jenkinsfile | 6 11 files changed, 182 insertions(+), 730 deletions(-) I have also tightened the dependency on libjansson-dev to the one upstream actually needs. Buster has this on all architectures, this is purely necessary for a to-be-done backport to stretch-backports unblock asterisk/1:16.2.1~dfsg-1
diffstat for asterisk-16.2.0~dfsg asterisk-16.2.1~dfsg .version | 1 ChangeLog | 35 ++ asterisk-16.2.0-summary.html | 202 --------------- asterisk-16.2.0-summary.txt | 519 ----------------------------------------- asterisk-16.2.1-summary.html | 23 + asterisk-16.2.1-summary.txt | 107 ++++++++ debian/changelog | 9 debian/control | 2 res/res_pjsip_sdp_rtp.c | 2 tests/CI/gates.jenkinsfile | 6 tests/CI/unittests.jenkinsfile | 6 11 files changed, 182 insertions(+), 730 deletions(-) diff -Nru asterisk-16.2.0~dfsg/asterisk-16.2.0-summary.html asterisk-16.2.1~dfsg/asterisk-16.2.0-summary.html --- asterisk-16.2.0~dfsg/asterisk-16.2.0-summary.html 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/asterisk-16.2.0-summary.html 1970-01-01 01:00:00.000000000 +0100 @@ -1,202 +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-16.2.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.2.0</h3><h3 align="center">Date: 2019-02-15</h3><h3 align="center"><asteriskt...@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="#commits">Other Changes</a></li> -<li><a href="#diffstat">Diffstat</a></li> -</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.1.0.</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%">10 George Joseph <gjos...@digium.com><br/>10 Sean Bright <sean.bri...@gmail.com><br/>4 Kevin Harwell <kharw...@digium.com><br/>3 Alexei Gradinari <alex2g...@gmail.com><br/>2 Joshua C. Colp <jc...@digium.com><br/>2 Asterisk Development Team <asteriskt...@digium.com><br/>2 Jeremy Lainé <jeremy.la...@m4x.org><br/>2 Giuseppe Sucameli <sucam...@netresults.it><br/>2 Joshua Colp <jc...@digium.com><br/>2 Chris-Savinovich <csavinov...@digium.com><br/>2 Richard Mudgett <rmudg...@digium.com><br/>1 Xiemin Chen <chenxie...@gmail.com><br/>1 Mohit Dhiman <mohitdhi...@drishti-soft.com><br/>1 Pirmin Walthert <in...@nappsoft.ch><br/>1 Sungtae Kim <pcher...@gmail.com><br/>1 Diederik de Groot <dkgr...@talon.nl><br/>1 David M. Lee <d...@respoke.io><br/>1 Jean Aunis <jean.au...@prescom.fr><br/>1 Corey Farrell <g...@cfware.com><br/>1 Bryan Boatright <ast-b...@omega71.com><br/>1 Valentin Vidic <vvi...@valentin-vidic.from.hr><br/>1 sungtae kim <sung...@messagebird.com><br/>1 Gerald Schnabel <g...@starface.de><br/>1 Chris Savinovich <csavinov...@digium.com><br/>1 Ben Ford <bf...@digium.com><br/>1 eyalhasson <e...@kolhl.com><br/>1 Sebastian Damm <d...@sipgate.de><br/></td><td width="33%"><td width="33%">4 Joshua C. Colp <jc...@digium.com><br/>3 George Joseph <gjos...@digium.com><br/>2 Alexei Gradinari <alex2g...@gmail.com><br/>2 Giuseppe Sucameli <sucam...@netresults.it><br/>2 Ross Beer <ross.b...@voicehost.co.uk><br/>2 Jeremy Lainé <jeremy.la...@m4x.org><br/>2 David Kuehling <dvdkh...@posteo.de><br/>1 Jean Aunis - Prescom <jean.au...@prescom.fr><br/>1 Andrew Nagy<br/>1 boatright <ast-b...@omega71.com><br/>1 Mohit Dhiman <mohitdhi...@drishti-soft.com><br/>1 sungtae kim <pcher...@gmail.com><br/>1 Ray <rain...@gmail.com><br/>1 Eyal Hasson <e...@kolhl.com><br/>1 abelbeck <lon...@abelbeck.com><br/>1 nappsoft <in...@nappsoft.ch><br/>1 Gianluca Merlo <gianluca.me...@gmail.com><br/>1 Xiemin Chen <chenxie...@gmail.com><br/>1 David Wilcox <david.wil...@cloverbeen.com><br/>1 Andrew Nagy <andrew.n...@the159.com><br/>1 Mark <wie...@woop.la><br/>1 Diederik de Groot <dkgr...@talon.nl><br/>1 Valentin Vidić <vvi...@valentin-vidic.from.hr><br/>1 Gerald Schnabel <g...@starface.de><br/>1 xiemchen<br/>1 David Wilcox<br/>1 Sebastian Damm <d...@sipgate.de><br/>1 David Kuehling<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: . I did not set the category correctly.</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28221";>ASTERISK-28221</a>: Bug in ast_coredumper<br/>Reported by: Andrew Nagy<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3efe5061d5d0aac4c52843e6a3804e9212b10677";>[3efe5061d5]</a> George Joseph -- ast_coredumper: Refactor the pid determination process</li> -</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28201";>ASTERISK-28201</a>: [patch] confbridge: no announce to the marked users when they join an empty conference<br/>Reported by: Alexei Gradinari<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2610379605a48fd43afb1c9d89d9d797a81011df";>[2610379605]</a> Alexei Gradinari -- confbridge: announce to the marked users when they join an empty conference</li> -</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28218";>ASTERISK-28218</a>: app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b)<br/>Reported by: Mark<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d9482695d20e4f7d6d5835dbfc1c11d728fe852";>[2d9482695d]</a> Joshua Colp -- app_queue: Fix crash when using 'b' option on non-ringall queue.</li> -</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28225";>ASTERISK-28225</a>: app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"<br/>Reported by: boatright<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92298434bd9ce591356cda25b556c930a97a75f4";>[92298434bd]</a> Bryan Boatright -- app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28222";>ASTERISK-28222</a>: Regression: MWI polling no longer works<br/>Reported by: abelbeck<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff2ed4eeeee4837820ac23d061f03db5a61f5ec6";>[ff2ed4eeee]</a> George Joseph -- Revert "stasis_cache: Stop caching stasis subscription change messages"</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28215";>ASTERISK-28215</a>: app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs<br/>Reported by: George Joseph<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aebb822d1f5604a376f78e1d8ae555580aad4d64";>[aebb822d1f]</a> George Joseph -- app_voicemail: Don't delete mailbox state unless mailbox is deleted</li> -</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28213";>ASTERISK-28213</a>: res_pjsip: Threads pile up needlessly when AOR is blocked<br/>Reported by: Ross Beer<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28edd2a5cb8b3f0a36f875a2f241fe07deceffc7";>[28edd2a5cb]</a> Kevin Harwell -- res_pjsip_registrar: lock transport monitor when setting 'removing' flag</li> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1fb249132b1554e09e45a8a5f569ecc8b752568";>[f1fb249132]</a> Kevin Harwell -- res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28238";>ASTERISK-28238</a>: PJSIP realtime. getcontext not working with DUNDI<br/>Reported by: Ray<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c3b4dcf807c06447ca4d38c6959b1a2561f60ff";>[9c3b4dcf80]</a> Kevin Harwell -- pjsip/config_global: regcontext context not created</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27095";>ASTERISK-27095</a>: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE<br/>Reported by: George Joseph<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5de36abd5af0b94a9fab1b8b51c5d1d90a95697a";>[5de36abd5a]</a> Pirmin Walthert -- pjproject_bundled: check whether UPDATE is supported on outgoing calls</li> -</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28194";>ASTERISK-28194</a>: chan_sip: Leak using contact ACL<br/>Reported by: Giuseppe Sucameli<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6071ad77f5705a8f4cb3b41847955d92cd265a09";>[6071ad77f5]</a> Giuseppe Sucameli -- chan_sip: Fix leak using contact ACL</li> -</ul><br><h4>Category: Channels/chan_sip/Subscriptions</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28173";>ASTERISK-28173</a>: Deadlock in chan_sip handling subscribe request during res_parking reload<br/>Reported by: Giuseppe Sucameli<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=419db481d1b44091bef52f17cc7acd7957d96c22";>[419db481d1]</a> Giuseppe Sucameli -- Fix deadlock handling subscribe req during res_parking reload</li> -</ul><br><h4>Category: Codecs/codec_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28263";>ASTERISK-28263</a>: codec_opus: errors setting max_playback_rate and bitrate to "sdp"<br/>Reported by: Gianluca Merlo<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6452f9656cb42b588355fcb698ff23db5dba0b7";>[f6452f9656]</a> Kevin Harwell -- codecs.conf.sample: update codec opus docs</li> -</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28271";>ASTERISK-28271</a>: Opensuse Leap 15 --with-jannson-bundled will not compile<br/>Reported by: David Wilcox<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70fa6e6955ea62f9036f775861cc7425c813c050";>[70fa6e6955]</a> George Joseph -- bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28250";>ASTERISK-28250</a>: build: Cross-compilation fails for target arm-linux-gnueabihf<br/>Reported by: Jean Aunis - Prescom<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3a6714158103120aeeffd62df662799858b0654";>[d3a6714158]</a> Jean Aunis -- build : Fix cross-compilation errors</li> -</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28197";>ASTERISK-28197</a>: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases<br/>Reported by: Mohit Dhiman<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b24da607e695897f54f4b21208885fae7ac9158";>[4b24da607e]</a> Mohit Dhiman -- stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.</li> -</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28232";>ASTERISK-28232</a>: core: RAII using clang use-after-scope issue<br/>Reported by: Diederik de Groot<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2c182b6ab63b8c39597f657ff7168c7d3424c8c";>[d2c182b6ab]</a> Diederik de Groot -- RAII: Change order or variables in clang version</li> -</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28252";>ASTERISK-28252</a>: HangupHandler manager events are never thrown<br/>Reported by: Gerald Schnabel<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=735bd4d18576f43134dd17502d9c037bea996e81";>[735bd4d185]</a> Gerald Schnabel -- manager_channels: Fix throwing of HangupHandler manager events</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28244";>ASTERISK-28244</a>: stasis: Filter messages at publishing to AMI/ARI<br/>Reported by: Joshua C. Colp<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fcd07c34fbaa764cee47204a6bea838b8f4a3a27";>[fcd07c34fb]</a> Joshua C. Colp -- stasis / manager / ari: Better filter messages.</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28197";>ASTERISK-28197</a>: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases<br/>Reported by: Mohit Dhiman<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b24da607e695897f54f4b21208885fae7ac9158";>[4b24da607e]</a> Mohit Dhiman -- stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28212";>ASTERISK-28212</a>: stasis: Statistics broke ABI under developer mode<br/>Reported by: Joshua C. Colp<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44a7faca21bf29bd7e09404d5dd200a0c8e95a8f";>[44a7faca21]</a> Corey Farrell -- stasis: Fix ABI between DEVMODE and non-DEVMODE.</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28117";>ASTERISK-28117</a>: stasis: Add statistics for usage when in developer mode<br/>Reported by: Joshua C. Colp<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68ec7d93e82e02a37bacf2e2cc5c4ac0ea4d23c1";>[68ec7d93e8]</a> Joshua C. Colp -- stasis: Add statistics gathering in developer mode.</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28186";>ASTERISK-28186</a>: stasis: Filter messages at publishing based on to_* presence<br/>Reported by: Joshua C. Colp<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79899db740484878166b2d7fa8cbdb41389dd99e";>[79899db740]</a> George Joseph -- stasis: Allow filtering by formatter</li> -</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28104";>ASTERISK-28104</a>: AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps<br/>Reported by: George Joseph<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1051e1dd1876779f82edfbfffa1115455ca2c269";>[1051e1dd18]</a> Ben Ford -- res_stasis: Auto-create context and extens on Stasis app launch.</li> -</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27959";>ASTERISK-27959</a>: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem<br/>Reported by: David Kuehling<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f60afac587d55860c1cc1f6f2fc2e55f1a0ddfc9";>[f60afac587]</a> Sean Bright -- res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set</li> -</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28257";>ASTERISK-28257</a>: res_http_websocket: PING / PONG opcodes break data reception<br/>Reported by: Jeremy Lainé<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=907d71b5513d17e61b51798cd924fe97b4a5a3b0";>[907d71b551]</a> Jeremy Lainé -- res_http_websocket: ensure control frames do not interfere with data</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28231";>ASTERISK-28231</a>: res_http_websocket: Not responding to Connection Close Frame (opcode 8)<br/>Reported by: Jeremy Lainé<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21a1feece286921c47f9d87dc0260a72721ca5a4";>[21a1feece2]</a> Jeremy Lainé -- res_http_websocket: respond to CLOSE opcode</li> -</ul><br><h4>Category: Resources/res_monitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28249";>ASTERISK-28249</a>: res_monitor: Segfault with Monitor(wav,file,i)<br/>Reported by: Valentin Vidić<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6506c5b1d46184c6d2749261a048d31232868284";>[6506c5b1d4]</a> Valentin Vidic -- channel.c: Fix segfault with Monitor(wav,file,i)</li> -</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28173";>ASTERISK-28173</a>: Deadlock in chan_sip handling subscribe request during res_parking reload<br/>Reported by: Giuseppe Sucameli<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=419db481d1b44091bef52f17cc7acd7957d96c22";>[419db481d1]</a> Giuseppe Sucameli -- Fix deadlock handling subscribe req during res_parking reload</li> -</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28157";>ASTERISK-28157</a>: Asterisk crashes when the res_pjsip_* modules unload<br/>Reported by: sungtae kim<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b6df87816bfe0552b7888bf14efd2a82a6c7dbf";>[1b6df87816]</a> Sungtae Kim -- res_pjsip: Patch for res_pjsip_* module load/reload crash</li> -</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28230";>ASTERISK-28230</a>: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony<br/>Reported by: David Kuehling<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c6271155fb01700eacddd20651ba9765fabce194";>[c6271155fb]</a> Joshua Colp -- res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.</li> -</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28162";>ASTERISK-28162</a>: [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation<br/>Reported by: Alexei Gradinari<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0e57e458bb309720c1c370abd3aa5088a7d7c17";>[c0e57e458b]</a> Alexei Gradinari -- RTP: reset DTMF last seqno/timestamp on RTP renegotiation</li> -</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28182";>ASTERISK-28182</a>: chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE<br/>Reported by: nappsoft<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5de36abd5af0b94a9fab1b8b51c5d1d90a95697a";>[5de36abd5a]</a> Pirmin Walthert -- pjproject_bundled: check whether UPDATE is supported on outgoing calls</li> -</ul><br><h3>Improvement</h3><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28196";>ASTERISK-28196</a>: bridge_softmix: Does not support WebRTC source with multi video tracks.<br/>Reported by: Xiemin Chen<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6cf837aede9dd8d805991d1e065baa699145ef2";>[f6cf837aed]</a> Xiemin Chen -- bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix</li> -</ul><br><h4>Category: Formats/format_g726</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28246";>ASTERISK-28246</a>: Support skipping on the g726 format<br/>Reported by: Eyal Hasson<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1da2e94a38ed0640b5254f236a8b2e9705a922b";>[c1da2e94a3]</a> eyalhasson -- format_g726: add support for seeking</li> -</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28198";>ASTERISK-28198</a>: res_ari: Add new hangup causes for ARI Channel DELETE command<br/>Reported by: Sebastian Damm<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59cf552dd32a55752072439d42461704c64d7167";>[59cf552dd3]</a> Sebastian Damm -- res/res_ari: Add additional hangup reasons</li> -</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28198";>ASTERISK-28198</a>: res_ari: Add new hangup causes for ARI Channel DELETE command<br/>Reported by: Sebastian Damm<ul> -<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59cf552dd32a55752072439d42461704c64d7167";>[59cf552dd3]</a> Sebastian Damm -- res/res_ari: Add additional hangup reasons</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=6a0e6b42eb6d22f2bad4bebb01cd7d6d4dae3a9c";>6a0e6b42eb</a></td><td>Chris Savinovich</td><td>Revert "Test_cel: Fails when DONT_OPTIMIZE is off"</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=246e34cbf40601485b8c870e5bbae48e24814cd5";>246e34cbf4</a></td><td>Asterisk Development Team</td><td>Update for 16.2.0-rc2</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=541d7a52f5b1557b911b85dafd7d6679f8be5b70";>541d7a52f5</a></td><td>Asterisk Development Team</td><td>Update for 16.2.0-rc1</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=19fc99a2fbf899ad5abd7b88b0fc3e225cf7c0ca";>19fc99a2fb</a></td><td>sungtae kim</td><td>Added ARI resource /ari/asterisk/ping</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=603143bd5ae2a2321fa2cd7d714a40c922610039";>603143bd5a</a></td><td>George Joseph</td><td>media_index.c: Refactored so it doesn't cache the index</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05b79d16ab93b038b39412e2570a21205eb499c4";>05b79d16ab</a></td><td>Chris-Savinovich</td><td>Test_cel: Fails when DONT_OPTIMIZE is off</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbef559e0bb24a246558db8b5ddecdd5cf86c857";>dbef559e0b</a></td><td>George Joseph</td><td>app_voicemail: Add Mailbox Aliases</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c11399be3ae48bed620ea5775c435e671495b25";>9c11399be3</a></td><td>George Joseph</td><td>pjproject_bundled: Add patch for double free issue in timer heap</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb6e0df1739b18bae6718e82ab28526054008f01";>fb6e0df173</a></td><td>Sean Bright</td><td>pjsip_transport_management: Shutdown transport immediately on disconnect</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=011e46d5a68a986ac503736208507b23dd071868";>011e46d5a6</a></td><td>Sean Bright</td><td>sched: Make sched_settime() return void because it cannot fail</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44a862fb576cbc76165f2dc07cc443328b10ea09";>44a862fb57</a></td><td>Sean Bright</td><td>res_pjsip_transport_websocket: Don't assert on 0 length payloads</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f22c9f4b7efde0da72c4665cceba380fe1f57b3";>7f22c9f4b7</a></td><td>Alexei Gradinari</td><td>res_pjsip: add option to enable ContactStatus event when contact is updated</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f19607870526952d27e6a806e35535e00c0f5f2a";>f196078705</a></td><td>Richard Mudgett</td><td>stasic.c: Fix printf format type mismatches with arguments.</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59717b5e850f428df58c13b6a24d5baadf0e8c40";>59717b5e85</a></td><td>Richard Mudgett</td><td>backtrace.c: Fix casting pointer to/from integral type.</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=970805180e300935148fe14cbbf6e35a09666023";>970805180e</a></td><td>Sean Bright</td><td>res_rtp_asterisk: Remove some unused structure fields.</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=640aac768bb7208e6baece7860753bbdb2b5564c";>640aac768b</a></td><td>Sean Bright</td><td>bridge_builtin_features.c: Set auto(mix)mon variables on both channels</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9febdba05b0767f1938e0261d9f2b5d8f3167902";>9febdba05b</a></td><td>Sean Bright</td><td>Use non-blocking socket() and pipe() wrappers</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16ae8330d2955ef2d7bd8e360b700fa9aaf10e1c";>16ae8330d2</a></td><td>Sean Bright</td><td>utils: Don't set or clear flags that don't need setting or clearing</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c9519796b9d708a4f9e2fd62053436d22b6e78f";>9c9519796b</a></td><td>Sean Bright</td><td>build: Update config.guess and config.sub</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df0b59564e844134ae18be23089ce87aa3f00ca3";>df0b59564e</a></td><td>George Joseph</td><td>Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8a18fb81c1a4885b9b6568d69e56ab40f488d02a";>8a18fb81c1</a></td><td>Sean Bright</td><td>utils: Wrap socket() and pipe() to reduce syscalls</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1657508ddd1e204dddc940876810d40c575066e8";>1657508ddd</a></td><td>David M. Lee</td><td>Removing registrar_expire from basic-pbx config</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6c2662404e6094b8554b49bab672dfc47e9b8c6";>a6c2662404</a></td><td>George Joseph</td><td>CI: Various updates to buildAsterisk.sh</td></tr> -<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60e548ffa5f0eefca1ba23295ae01b97942a5b5c";>60e548ffa5</a></td><td>Chris-Savinovich</td><td>test_websocket_client.c: Disable websocket_client_create_and_connect test.</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-16.1.0-summary.html | 620 -- -asterisk-16.1.0-summary.txt | 1442 ----- -b/.version | 2 -b/CHANGES | 49 -b/ChangeLog | 813 +++ -b/apps/app_confbridge.c | 2 -b/apps/app_queue.c | 2 -b/apps/app_voicemail.c | 335 + -b/apps/confbridge/conf_state_empty.c | 3 -b/apps/confbridge/conf_state_inactive.c | 2 -b/apps/confbridge/include/confbridge.h | 8 -b/asterisk-16.2.0-rc2-summary.html | 11 -b/asterisk-16.2.0-rc2-summary.txt | 81 -b/bridges/bridge_builtin_features.c | 2 -b/bridges/bridge_softmix.c | 16 -b/channels/chan_sip.c | 6 -b/config.guess | 666 +- -b/config.sub | 2535 ++++------ -b/configs/basic-pbx/modules.conf | 1 -b/configs/samples/codecs.conf.sample | 26 -b/configs/samples/pjsip.conf.sample | 5 -b/configs/samples/voicemail.conf.sample | 12 -b/configure | 86 -b/configure.ac | 28 -b/contrib/ast-db-manage/config/versions/0838f8db6a61_pjsip_add_send_contact_status_on_update_.py | 39 -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 | 111 -b/formats/format_g726.c | 35 -b/include/asterisk/autoconfig.h.in | 6 -b/include/asterisk/channel.h | 12 -b/include/asterisk/media_index.h | 20 -b/include/asterisk/res_pjsip.h | 9 -b/include/asterisk/res_pjsip_session.h | 13 -b/include/asterisk/sounds_index.h | 13 -b/include/asterisk/stasis.h | 51 -b/include/asterisk/stasis_internal.h | 5 -b/include/asterisk/stasis_message_router.h | 54 -b/include/asterisk/utils.h | 42 -b/main/alertpipe.c | 11 -b/main/asterisk.c | 4 -b/main/asterisk.exports.in | 1 -b/main/backtrace.c | 10 -b/main/channel.c | 10 -b/main/channel_internal_api.c | 12 -b/main/manager.c | 4 -b/main/manager_channels.c | 10 -b/main/media_index.c | 229 -b/main/pbx.c | 85 -b/main/sched.c | 20 -b/main/sounds.c | 179 -b/main/stasis.c | 877 +++ -b/main/stasis_cache.c | 33 -b/main/stasis_message.c | 16 -b/main/stasis_message_router.c | 71 -b/main/tcptls.c | 3 -b/main/udptl.c | 3 -b/main/utils.c | 44 -b/res/ari/ari_model_validators.c | 70 -b/res/ari/ari_model_validators.h | 22 -b/res/ari/resource_asterisk.c | 18 -b/res/ari/resource_asterisk.h | 11 -b/res/ari/resource_channels.c | 16 -b/res/ari/resource_sounds.c | 28 -b/res/res_agi.c | 7 -b/res/res_ari_asterisk.c | 63 -b/res/res_format_attr_h264.c | 2 -b/res/res_http_websocket.c | 50 -b/res/res_pjsip.c | 3 -b/res/res_pjsip/config_global.c | 72 -b/res/res_pjsip/include/res_pjsip_private.h | 10 -b/res/res_pjsip/pjsip_configuration.c | 35 -b/res/res_pjsip/pjsip_message_filter.c | 1 -b/res/res_pjsip/pjsip_options.c | 55 -b/res/res_pjsip/pjsip_session.c | 85 -b/res/res_pjsip/pjsip_transport_management.c | 77 -b/res/res_pjsip_registrar.c | 27 -b/res/res_pjsip_sdp_rtp.c | 8 -b/res/res_pjsip_session.c | 68 -b/res/res_pjsip_transport_websocket.c | 13 -b/res/res_rtp_asterisk.c | 37 -b/res/res_timing_pthread.c | 7 -b/res/stasis/app.c | 51 -b/rest-api/api-docs/asterisk.json | 33 -b/rest-api/api-docs/channels.json | 8 -b/tests/CI/buildAsterisk.sh | 163 -b/tests/test_stasis.c | 397 + -b/tests/test_websocket_client.c | 1 -b/third-party/jansson/Makefile | 3 -b/third-party/jansson/configure.m4 | 4 -b/third-party/pjproject/configure.m4 | 4 -93 files changed, 5933 insertions(+), 4341 deletions(-)</pre><br></html> \ Kein Zeilenumbruch am Dateiende. diff -Nru asterisk-16.2.0~dfsg/asterisk-16.2.0-summary.txt asterisk-16.2.1~dfsg/asterisk-16.2.0-summary.txt --- asterisk-16.2.0~dfsg/asterisk-16.2.0-summary.txt 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/asterisk-16.2.0-summary.txt 1970-01-01 01:00:00.000000000 +0100 @@ -1,519 +0,0 @@ - Release Summary - - asterisk-16.2.0 - - Date: 2019-02-15 - - <asteriskt...@digium.com> - - ---------------------------------------------------------------------- - - Table of Contents - - 1. Summary - 2. Contributors - 3. Closed Issues - 4. Other Changes - 5. Diffstat - - ---------------------------------------------------------------------- - - Summary - - [Back to Top] - - This release is a point release of an existing major version. The changes - included were made to address problems that have been identified in this - release series, or are minor, backwards compatible new features or - improvements. Users should be able to safely upgrade to this version if - this release series is already in use. Users considering upgrading from a - previous version are strongly encouraged to review the UPGRADE.txt - document as well as the CHANGES document for information about upgrading - to this release series. - - The data in this summary reflects changes that have been made since the - previous release, asterisk-16.1.0. - - ---------------------------------------------------------------------- - - 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 - 10 George Joseph 4 Joshua C. Colp - 10 Sean Bright 3 George Joseph - 4 Kevin Harwell 2 Alexei Gradinari - 3 Alexei Gradinari 2 Giuseppe Sucameli - 2 Joshua C. Colp 2 Ross Beer - 2 Asterisk Development Team 2 Jeremy Lainé - 2 Jeremy Lainé 2 David Kuehling - 2 Giuseppe Sucameli 1 Jean Aunis - Prescom - 2 Joshua Colp 1 Andrew Nagy - 2 Chris-Savinovich 1 boatright - 2 Richard Mudgett 1 Mohit Dhiman - 1 Xiemin Chen 1 sungtae kim - 1 Mohit Dhiman 1 Ray - 1 Pirmin Walthert 1 Eyal Hasson - 1 Sungtae Kim 1 abelbeck - 1 Diederik de Groot 1 nappsoft - 1 David M. Lee 1 Gianluca Merlo - 1 Jean Aunis 1 Xiemin Chen - 1 Corey Farrell 1 David Wilcox - 1 Bryan Boatright 1 Andrew Nagy - 1 Valentin Vidic 1 Mark - 1 sungtae kim 1 Diederik de Groot - 1 Gerald Schnabel 1 Valentin VidiÄ* - 1 Chris Savinovich 1 Gerald Schnabel - 1 Ben Ford 1 xiemchen - 1 eyalhasson 1 David Wilcox - 1 Sebastian Damm 1 Sebastian Damm - 1 David Kuehling - - ---------------------------------------------------------------------- - - 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: . I did not set the category correctly. - - ASTERISK-28221: Bug in ast_coredumper - Reported by: Andrew Nagy - * [3efe5061d5] George Joseph -- ast_coredumper: Refactor the pid - determination process - - Category: Applications/app_confbridge - - ASTERISK-28201: [patch] confbridge: no announce to the marked users when - they join an empty conference - Reported by: Alexei Gradinari - * [2610379605] Alexei Gradinari -- confbridge: announce to the marked - users when they join an empty conference - - Category: Applications/app_queue - - ASTERISK-28218: app_queue: Asterisk crashes when using Queue with a - pre-dial handler (option b) - Reported by: Mark - * [2d9482695d] Joshua Colp -- app_queue: Fix crash when using 'b' option - on non-ringall queue. - - Category: Applications/app_voicemail - - ASTERISK-28225: app_voicemail: Channel variable VM_MESSAGEFILE not updated - correctly if message marked "urgent" - Reported by: boatright - * [92298434bd] Bryan Boatright -- app_voicemail: Fix Channel variable - VM_MESSAGEFILE for "urgent" voicemail - ASTERISK-28222: Regression: MWI polling no longer works - Reported by: abelbeck - * [ff2ed4eeee] George Joseph -- Revert "stasis_cache: Stop caching - stasis subscription change messages" - ASTERISK-28215: app_voicemail: Leaving voicemail sometimes doesn't trigger - NOTIFYs - Reported by: George Joseph - * [aebb822d1f] George Joseph -- app_voicemail: Don't delete mailbox - state unless mailbox is deleted - - Category: Channels/chan_pjsip - - ASTERISK-28213: res_pjsip: Threads pile up needlessly when AOR is blocked - Reported by: Ross Beer - * [28edd2a5cb] Kevin Harwell -- res_pjsip_registrar: lock transport - monitor when setting 'removing' flag - * [f1fb249132] Kevin Harwell -- res_pjsip_registrar: mitigate blocked - threads on reliable transport shutdown - ASTERISK-28238: PJSIP realtime. getcontext not working with DUNDI - Reported by: Ray - * [9c3b4dcf80] Kevin Harwell -- pjsip/config_global: regcontext context - not created - ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite, - we're not trying UPDATE - Reported by: George Joseph - * [5de36abd5a] Pirmin Walthert -- pjproject_bundled: check whether - UPDATE is supported on outgoing calls - - Category: Channels/chan_sip/General - - ASTERISK-28194: chan_sip: Leak using contact ACL - Reported by: Giuseppe Sucameli - * [6071ad77f5] Giuseppe Sucameli -- chan_sip: Fix leak using contact ACL - - Category: Channels/chan_sip/Subscriptions - - ASTERISK-28173: Deadlock in chan_sip handling subscribe request during - res_parking reload - Reported by: Giuseppe Sucameli - * [419db481d1] Giuseppe Sucameli -- Fix deadlock handling subscribe req - during res_parking reload - - Category: Codecs/codec_opus - - ASTERISK-28263: codec_opus: errors setting max_playback_rate and bitrate - to "sdp" - Reported by: Gianluca Merlo - * [f6452f9656] Kevin Harwell -- codecs.conf.sample: update codec opus - docs - - Category: Core/BuildSystem - - ASTERISK-28271: Opensuse Leap 15 --with-jannson-bundled will not compile - Reported by: David Wilcox - * [70fa6e6955] George Joseph -- bundled-jansson: On OpenSuse Leap - libjansson.a was placed in lib64 - ASTERISK-28250: build: Cross-compilation fails for target - arm-linux-gnueabihf - Reported by: Jean Aunis - Prescom - * [d3a6714158] Jean Aunis -- build : Fix cross-compilation errors - - Category: Core/Channels - - ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of - channels past destruction in certain cases - Reported by: Mohit Dhiman - * [4b24da607e] Mohit Dhiman -- stasis/endpoint: Fix memory leak of - channel_ids in ast_endpoint structure. - - Category: Core/General - - ASTERISK-28232: core: RAII using clang use-after-scope issue - Reported by: Diederik de Groot - * [d2c182b6ab] Diederik de Groot -- RAII: Change order or variables in - clang version - - Category: Core/Stasis - - ASTERISK-28252: HangupHandler manager events are never thrown - Reported by: Gerald Schnabel - * [735bd4d185] Gerald Schnabel -- manager_channels: Fix throwing of - HangupHandler manager events - ASTERISK-28244: stasis: Filter messages at publishing to AMI/ARI - Reported by: Joshua C. Colp - * [fcd07c34fb] Joshua C. Colp -- stasis / manager / ari: Better filter - messages. - ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of - channels past destruction in certain cases - Reported by: Mohit Dhiman - * [4b24da607e] Mohit Dhiman -- stasis/endpoint: Fix memory leak of - channel_ids in ast_endpoint structure. - ASTERISK-28212: stasis: Statistics broke ABI under developer mode - Reported by: Joshua C. Colp - * [44a7faca21] Corey Farrell -- stasis: Fix ABI between DEVMODE and - non-DEVMODE. - ASTERISK-28117: stasis: Add statistics for usage when in developer mode - Reported by: Joshua C. Colp - * [68ec7d93e8] Joshua C. Colp -- stasis: Add statistics gathering in - developer mode. - ASTERISK-28186: stasis: Filter messages at publishing based on to_* - presence - Reported by: Joshua C. Colp - * [79899db740] George Joseph -- stasis: Allow filtering by formatter - - Category: Resources/res_ari - - ASTERISK-28104: AstriCon Feedback: Automatically create a 1 line dialplan - context for stasis apps - Reported by: George Joseph - * [1051e1dd18] Ben Ford -- res_stasis: Auto-create context and extens on - Stasis app launch. - - Category: Resources/res_format_attr_h264 - - ASTERISK-27959: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem - Reported by: David Kuehling - * [f60afac587] Sean Bright -- res_format_attr_h264.c: Make sure - profile-level-id fmtp attribute is set - - Category: Resources/res_http_websocket - - ASTERISK-28257: res_http_websocket: PING / PONG opcodes break data - reception - Reported by: Jeremy Lainé - * [907d71b551] Jeremy Lainé -- res_http_websocket: ensure control - frames do not interfere with data - ASTERISK-28231: res_http_websocket: Not responding to Connection Close - Frame (opcode 8) - Reported by: Jeremy Lainé - * [21a1feece2] Jeremy Lainé -- res_http_websocket: respond to CLOSE - opcode - - Category: Resources/res_monitor - - ASTERISK-28249: res_monitor: Segfault with Monitor(wav,file,i) - Reported by: Valentin VidiÄ* - * [6506c5b1d4] Valentin Vidic -- channel.c: Fix segfault with - Monitor(wav,file,i) - - Category: Resources/res_parking - - ASTERISK-28173: Deadlock in chan_sip handling subscribe request during - res_parking reload - Reported by: Giuseppe Sucameli - * [419db481d1] Giuseppe Sucameli -- Fix deadlock handling subscribe req - during res_parking reload - - Category: Resources/res_pjsip_session - - ASTERISK-28157: Asterisk crashes when the res_pjsip_* modules unload - Reported by: sungtae kim - * [1b6df87816] Sungtae Kim -- res_pjsip: Patch for res_pjsip_* module - load/reload crash - - Category: Resources/res_rtp_asterisk - - ASTERISK-28230: res_rtp_asterisk: abs-send-time extension added with - Asterisk 15.5.0 breaks GXV3140 video telephony - Reported by: David Kuehling - * [c6271155fb] Joshua Colp -- res_pjsip_sdp_rtp: Only enable - abs-send-time when WebRTC is enabled. - ASTERISK-28162: [patch] need to reset DTMF last sequence number and - timestamp on RTP renegotiation - Reported by: Alexei Gradinari - * [c0e57e458b] Alexei Gradinari -- RTP: reset DTMF last seqno/timestamp - on RTP renegotiation - - Category: Third-Party/pjproject - - ASTERISK-28182: chan_pjsip: When connected_line_method is set to invite, - asterisk is not trying UPDATE - Reported by: nappsoft - * [5de36abd5a] Pirmin Walthert -- pjproject_bundled: check whether - UPDATE is supported on outgoing calls - - Improvement - - Category: Bridges/bridge_softmix - - ASTERISK-28196: bridge_softmix: Does not support WebRTC source with multi - video tracks. - Reported by: Xiemin Chen - * [f6cf837aed] Xiemin Chen -- bridge_softmix: Use MSID:LABEL metadata as - the cloned stream's appendix - - Category: Formats/format_g726 - - ASTERISK-28246: Support skipping on the g726 format - Reported by: Eyal Hasson - * [c1da2e94a3] eyalhasson -- format_g726: add support for seeking - - Category: Resources/res_ari - - ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE - command - Reported by: Sebastian Damm - * [59cf552dd3] Sebastian Damm -- res/res_ari: Add additional hangup - reasons - - Category: Resources/res_ari_channels - - ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE - command - Reported by: Sebastian Damm - * [59cf552dd3] Sebastian Damm -- res/res_ari: Add additional hangup - reasons - - ---------------------------------------------------------------------- - - 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 | - |------------+------------------+----------------------------------------| - | 6a0e6b42eb | Chris Savinovich | Revert "Test_cel: Fails when | - | | | DONT_OPTIMIZE is off" | - |------------+------------------+----------------------------------------| - | 246e34cbf4 | Asterisk | Update for 16.2.0-rc2 | - | | Development Team | | - |------------+------------------+----------------------------------------| - | 541d7a52f5 | Asterisk | Update for 16.2.0-rc1 | - | | Development Team | | - |------------+------------------+----------------------------------------| - | 19fc99a2fb | sungtae kim | Added ARI resource /ari/asterisk/ping | - |------------+------------------+----------------------------------------| - | 603143bd5a | George Joseph | media_index.c: Refactored so it | - | | | doesn't cache the index | - |------------+------------------+----------------------------------------| - | 05b79d16ab | Chris-Savinovich | Test_cel: Fails when DONT_OPTIMIZE is | - | | | off | - |------------+------------------+----------------------------------------| - | dbef559e0b | George Joseph | app_voicemail: Add Mailbox Aliases | - |------------+------------------+----------------------------------------| - | 9c11399be3 | George Joseph | pjproject_bundled: Add patch for | - | | | double free issue in timer heap | - |------------+------------------+----------------------------------------| - | fb6e0df173 | Sean Bright | pjsip_transport_management: Shutdown | - | | | transport immediately on disconnect | - |------------+------------------+----------------------------------------| - | 011e46d5a6 | Sean Bright | sched: Make sched_settime() return | - | | | void because it cannot fail | - |------------+------------------+----------------------------------------| - | 44a862fb57 | Sean Bright | res_pjsip_transport_websocket: Don't | - | | | assert on 0 length payloads | - |------------+------------------+----------------------------------------| - | | | res_pjsip: add option to enable | - | 7f22c9f4b7 | Alexei Gradinari | ContactStatus event when contact is | - | | | updated | - |------------+------------------+----------------------------------------| - | f196078705 | Richard Mudgett | stasic.c: Fix printf format type | - | | | mismatches with arguments. | - |------------+------------------+----------------------------------------| - | 59717b5e85 | Richard Mudgett | backtrace.c: Fix casting pointer | - | | | to/from integral type. | - |------------+------------------+----------------------------------------| - | 970805180e | Sean Bright | res_rtp_asterisk: Remove some unused | - | | | structure fields. | - |------------+------------------+----------------------------------------| - | | | bridge_builtin_features.c: Set | - | 640aac768b | Sean Bright | auto(mix)mon variables on both | - | | | channels | - |------------+------------------+----------------------------------------| - | 9febdba05b | Sean Bright | Use non-blocking socket() and pipe() | - | | | wrappers | - |------------+------------------+----------------------------------------| - | 16ae8330d2 | Sean Bright | utils: Don't set or clear flags that | - | | | don't need setting or clearing | - |------------+------------------+----------------------------------------| - | 9c9519796b | Sean Bright | build: Update config.guess and | - | | | config.sub | - |------------+------------------+----------------------------------------| - | | | Revert "RTP: reset DTMF last | - | df0b59564e | George Joseph | seqno/timestamp on voice packet with | - | | | marker bit" | - |------------+------------------+----------------------------------------| - | 8a18fb81c1 | Sean Bright | utils: Wrap socket() and pipe() to | - | | | reduce syscalls | - |------------+------------------+----------------------------------------| - | 1657508ddd | David M. Lee | Removing registrar_expire from | - | | | basic-pbx config | - |------------+------------------+----------------------------------------| - | a6c2662404 | George Joseph | CI: Various updates to | - | | | buildAsterisk.sh | - |------------+------------------+----------------------------------------| - | | | test_websocket_client.c: Disable | - | 60e548ffa5 | Chris-Savinovich | websocket_client_create_and_connect | - | | | test. | - +------------------------------------------------------------------------+ - - ---------------------------------------------------------------------- - - 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-16.1.0-summary.html | 620 -- - asterisk-16.1.0-summary.txt | 1442 ----- - b/.version | 2 - b/CHANGES | 49 - b/ChangeLog | 813 +++ - b/apps/app_confbridge.c | 2 - b/apps/app_queue.c | 2 - b/apps/app_voicemail.c | 335 + - b/apps/confbridge/conf_state_empty.c | 3 - b/apps/confbridge/conf_state_inactive.c | 2 - b/apps/confbridge/include/confbridge.h | 8 - b/asterisk-16.2.0-rc2-summary.html | 11 - b/asterisk-16.2.0-rc2-summary.txt | 81 - b/bridges/bridge_builtin_features.c | 2 - b/bridges/bridge_softmix.c | 16 - b/channels/chan_sip.c | 6 - b/config.guess | 666 +- - b/config.sub | 2535 ++++------ - b/configs/basic-pbx/modules.conf | 1 - b/configs/samples/codecs.conf.sample | 26 - b/configs/samples/pjsip.conf.sample | 5 - b/configs/samples/voicemail.conf.sample | 12 - b/configure | 86 - b/configure.ac | 28 - b/contrib/ast-db-manage/config/versions/0838f8db6a61_pjsip_add_send_contact_status_on_update_.py | 39 - 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 | 111 - b/formats/format_g726.c | 35 - b/include/asterisk/autoconfig.h.in | 6 - b/include/asterisk/channel.h | 12 - b/include/asterisk/media_index.h | 20 - b/include/asterisk/res_pjsip.h | 9 - b/include/asterisk/res_pjsip_session.h | 13 - b/include/asterisk/sounds_index.h | 13 - b/include/asterisk/stasis.h | 51 - b/include/asterisk/stasis_internal.h | 5 - b/include/asterisk/stasis_message_router.h | 54 - b/include/asterisk/utils.h | 42 - b/main/alertpipe.c | 11 - b/main/asterisk.c | 4 - b/main/asterisk.exports.in | 1 - b/main/backtrace.c | 10 - b/main/channel.c | 10 - b/main/channel_internal_api.c | 12 - b/main/manager.c | 4 - b/main/manager_channels.c | 10 - b/main/media_index.c | 229 - b/main/pbx.c | 85 - b/main/sched.c | 20 - b/main/sounds.c | 179 - b/main/stasis.c | 877 +++ - b/main/stasis_cache.c | 33 - b/main/stasis_message.c | 16 - b/main/stasis_message_router.c | 71 - b/main/tcptls.c | 3 - b/main/udptl.c | 3 - b/main/utils.c | 44 - b/res/ari/ari_model_validators.c | 70 - b/res/ari/ari_model_validators.h | 22 - b/res/ari/resource_asterisk.c | 18 - b/res/ari/resource_asterisk.h | 11 - b/res/ari/resource_channels.c | 16 - b/res/ari/resource_sounds.c | 28 - b/res/res_agi.c | 7 - b/res/res_ari_asterisk.c | 63 - b/res/res_format_attr_h264.c | 2 - b/res/res_http_websocket.c | 50 - b/res/res_pjsip.c | 3 - b/res/res_pjsip/config_global.c | 72 - b/res/res_pjsip/include/res_pjsip_private.h | 10 - b/res/res_pjsip/pjsip_configuration.c | 35 - b/res/res_pjsip/pjsip_message_filter.c | 1 - b/res/res_pjsip/pjsip_options.c | 55 - b/res/res_pjsip/pjsip_session.c | 85 - b/res/res_pjsip/pjsip_transport_management.c | 77 - b/res/res_pjsip_registrar.c | 27 - b/res/res_pjsip_sdp_rtp.c | 8 - b/res/res_pjsip_session.c | 68 - b/res/res_pjsip_transport_websocket.c | 13 - b/res/res_rtp_asterisk.c | 37 - b/res/res_timing_pthread.c | 7 - b/res/stasis/app.c | 51 - b/rest-api/api-docs/asterisk.json | 33 - b/rest-api/api-docs/channels.json | 8 - b/tests/CI/buildAsterisk.sh | 163 - b/tests/test_stasis.c | 397 + - b/tests/test_websocket_client.c | 1 - b/third-party/jansson/Makefile | 3 - b/third-party/jansson/configure.m4 | 4 - b/third-party/pjproject/configure.m4 | 4 - 93 files changed, 5933 insertions(+), 4341 deletions(-) diff -Nru asterisk-16.2.0~dfsg/asterisk-16.2.1-summary.html asterisk-16.2.1~dfsg/asterisk-16.2.1-summary.html --- asterisk-16.2.0~dfsg/asterisk-16.2.1-summary.html 1970-01-01 01:00:00.000000000 +0100 +++ asterisk-16.2.1~dfsg/asterisk-16.2.1-summary.html 2019-02-28 19:41:32.000000000 +0100 @@ -0,0 +1,23 @@ +<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd";><html xmlns="http://www.w3.org/1999/xhtml";><title>Release Summary - asterisk-16.2.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.2.1</h3><h3 align="center">Date: 2019-02-28</h3><h3 align="center"><asteriskt...@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="#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 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-2019-001.html";>AST-2019-001</a></li> +</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.2.0.</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%">2 George Joseph <gjos...@digium.com><br/></td><td width="33%"><td width="33%">1 Sotiris Ganouris <topg...@gmail.com><br/>1 Sotiris Ganouris<br/></td></tr> +</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28260";>ASTERISK-28260</a>: Asterisk segfault when rtp negotiation is wrong or fails<br/>Reported by: Sotiris Ganouris<ul> +<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85f40b663a2c4713ee972b7cb10a832001e081fe";>[85f40b663a]</a> George Joseph -- res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream</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=8b735172d1bfcbe352a52e5c36b733bd9f7615e0";>8b735172d1</a></td><td>George Joseph</td><td>CI: Update jenkinsfiles with new Gerrit URLs</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-16.2.0-summary.html | 202 -------------- +asterisk-16.2.0-summary.txt | 512 ------------------------------------- +b/.version | 2 +b/ChangeLog | 4 +b/asterisk-16.2.0-rc2-summary.html | 11 +b/asterisk-16.2.0-rc2-summary.txt | 81 +++++ +6 files changed, 93 insertions(+), 719 deletions(-)</pre><br></html> \ Kein Zeilenumbruch am Dateiende. diff -Nru asterisk-16.2.0~dfsg/asterisk-16.2.1-summary.txt asterisk-16.2.1~dfsg/asterisk-16.2.1-summary.txt --- asterisk-16.2.0~dfsg/asterisk-16.2.1-summary.txt 1970-01-01 01:00:00.000000000 +0100 +++ asterisk-16.2.1~dfsg/asterisk-16.2.1-summary.txt 2019-02-28 19:41:32.000000000 +0100 @@ -0,0 +1,107 @@ + Release Summary + + asterisk-16.2.1 + + Date: 2019-02-28 + + <asteriskt...@digium.com> + + ---------------------------------------------------------------------- + + Table of Contents + + 1. Summary + 2. Contributors + 3. Closed Issues + 4. Other Changes + 5. 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-2019-001 + + The data in this summary reflects changes that have been made since the + previous release, asterisk-16.2.0. + + ---------------------------------------------------------------------- + + 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 + 2 George Joseph 1 Sotiris Ganouris + 1 Sotiris Ganouris + + ---------------------------------------------------------------------- + + Closed Issues + + [Back to Top] + + This is a list of all issues from the issue tracker that were closed by + changes that went into this release. + + Security + + Category: Channels/chan_pjsip + + ASTERISK-28260: Asterisk segfault when rtp negotiation is wrong or fails + Reported by: Sotiris Ganouris + * [85f40b663a] George Joseph -- res_pjsip_sdp_rtp: Fix return code from + apply_negotiated_sdp_stream + + ---------------------------------------------------------------------- + + 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 | + |------------+---------------+-------------------------------------------| + | 8b735172d1 | George Joseph | CI: Update jenkinsfiles with new Gerrit | + | | | URLs | + +------------------------------------------------------------------------+ + + ---------------------------------------------------------------------- + + 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-16.2.0-summary.html | 202 -------------- + asterisk-16.2.0-summary.txt | 512 ------------------------------------- + b/.version | 2 + b/ChangeLog | 4 + b/asterisk-16.2.0-rc2-summary.html | 11 + b/asterisk-16.2.0-rc2-summary.txt | 81 +++++ + 6 files changed, 93 insertions(+), 719 deletions(-) diff -Nru asterisk-16.2.0~dfsg/ChangeLog asterisk-16.2.1~dfsg/ChangeLog --- asterisk-16.2.0~dfsg/ChangeLog 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/ChangeLog 2019-02-28 19:41:32.000000000 +0100 @@ -1,3 +1,38 @@ +2019-02-28 18:41 +0000 Asterisk Development Team <asteriskt...@digium.com> + + * asterisk 16.2.1 Released. + +2019-01-30 13:25 +0000 [85f40b663a] George Joseph <gjos...@digium.com> + + * res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream + + apply_negotiated_sdp_stream was returning a "1" when no joint + capabilities were found on an outgoing call instead of a "-1". + This indicated to res_pjsip_session that the handler DID handle + the sdp when in fact it didn't. Without the appropriate setup, + a subsequent media frame coming in would have an invalid stream_num + and cause a seg fault when the stream was attempted to be retrieved. + + apply_negotiated_sdp_stream now returns the correct "-1" and any + media is now discarded before it reaches the core stream processing. + + ASTERISK-28260 + Reported by: Sotiris Ganouris + + Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f + +2019-02-27 10:37 +0000 [8b735172d1] George Joseph <gjos...@digium.com> + + * CI: Update jenkinsfiles with new Gerrit URLs + + The recent upgrade of Gerrit to 2.16 elimiated referencing a + repository in a way the jenkinsfiles were relying on so + the URL references were changed to a more consistent and supported + format. + + Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e + (cherry picked from commit 5ce084579f897096163b4e0c2ed4e8e1a8558cca) + 2019-02-15 16:31 +0000 Asterisk Development Team <asteriskt...@digium.com> * asterisk 16.2.0 Released. diff -Nru asterisk-16.2.0~dfsg/debian/changelog asterisk-16.2.1~dfsg/debian/changelog --- asterisk-16.2.0~dfsg/debian/changelog 2019-02-20 23:49:31.000000000 +0100 +++ asterisk-16.2.1~dfsg/debian/changelog 2019-03-07 23:13:24.000000000 +0100 @@ -1,3 +1,12 @@ +asterisk (1:16.2.1~dfsg-1) unstable; urgency=medium + + * New upstream version 16.2.1~dfsg + - CVE-2019-7251 / AST-2019-001 (Closes: #923690) + Remote crash vulnerability with SDP protocol violation + * Bump dependency on libjansson-dev to >= 2.11 (required by upstream) + + -- Bernhard Schmidt <be...@debian.org> Thu, 07 Mar 2019 23:13:24 +0100 + asterisk (1:16.2.0~dfsg-1) unstable; urgency=medium * New upstream version 16.2.0~dfsg diff -Nru asterisk-16.2.0~dfsg/debian/control asterisk-16.2.1~dfsg/debian/control --- asterisk-16.2.0~dfsg/debian/control 2019-02-20 23:49:31.000000000 +0100 +++ asterisk-16.2.1~dfsg/debian/control 2019-03-07 23:13:24.000000000 +0100 @@ -31,7 +31,7 @@ libical-dev, libiksemel-dev, libjack-dev, - libjansson-dev, + libjansson-dev (>= 2.11), libldap-dev, liblua5.1-0-dev, libncurses-dev, diff -Nru asterisk-16.2.0~dfsg/res/res_pjsip_sdp_rtp.c asterisk-16.2.1~dfsg/res/res_pjsip_sdp_rtp.c --- asterisk-16.2.0~dfsg/res/res_pjsip_sdp_rtp.c 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/res/res_pjsip_sdp_rtp.c 2019-02-28 19:41:32.000000000 +0100 @@ -1945,7 +1945,7 @@ } if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) { - return 1; + return -1; } /* Set the channel uniqueid on the RTP instance now that it is becoming active */ diff -Nru asterisk-16.2.0~dfsg/tests/CI/gates.jenkinsfile asterisk-16.2.1~dfsg/tests/CI/gates.jenkinsfile --- asterisk-16.2.0~dfsg/tests/CI/gates.jenkinsfile 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/tests/CI/gates.jenkinsfile 2019-02-28 19:41:32.000000000 +0100 @@ -79,7 +79,7 @@ stage ("Checkout") { sh "sudo chown -R jenkins:users ." - env.GERRIT_PROJECT_URL = env.GERRIT_CHANGE_URL.replaceAll(/\/[0-9]+$/, "/${env.GERRIT_PROJECT}") + env.GERRIT_PROJECT_URL = env.GIT_URL.replaceAll(/[^\/]+$/, env.GERRIT_PROJECT) /* * Jenkins has already automatically checked out the base branch @@ -102,10 +102,10 @@ checkout scm: [$class: 'GitSCM', branches: [[name: env.GERRIT_BRANCH ]], extensions: [ - [$class: 'ScmName', name: 'gerrit-public'], + [$class: 'ScmName', name: env.GERRIT_NAME], [$class: 'CleanBeforeCheckout'], [$class: 'PreBuildMerge', options: [ - mergeRemote: 'gerrit-public', + mergeRemote: env.GERRIT_NAME, fastForwardMode: 'NO_FF', mergeStrategy: 'RECURSIVE', mergeTarget: env.GERRIT_BRANCH]], diff -Nru asterisk-16.2.0~dfsg/tests/CI/unittests.jenkinsfile asterisk-16.2.1~dfsg/tests/CI/unittests.jenkinsfile --- asterisk-16.2.0~dfsg/tests/CI/unittests.jenkinsfile 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/tests/CI/unittests.jenkinsfile 2019-02-28 19:41:32.000000000 +0100 @@ -80,7 +80,7 @@ stage ("Checkout") { sh "sudo chown -R jenkins:users ." - env.GERRIT_PROJECT_URL = env.GERRIT_CHANGE_URL.replaceAll(/\/[0-9]+$/, "/${env.GERRIT_PROJECT}") + env.GERRIT_PROJECT_URL = env.GIT_URL.replaceAll(/[^\/]+$/, env.GERRIT_PROJECT) /* * Jenkins has already automatically checked out the base branch @@ -103,10 +103,10 @@ checkout scm: [$class: 'GitSCM', branches: [[name: env.GERRIT_BRANCH ]], extensions: [ - [$class: 'ScmName', name: 'gerrit-public'], + [$class: 'ScmName', name: env.GERRIT_NAME], [$class: 'CleanBeforeCheckout'], [$class: 'PreBuildMerge', options: [ - mergeRemote: 'gerrit-public', + mergeRemote: env.GERRIT_NAME, fastForwardMode: 'NO_FF', mergeStrategy: 'RECURSIVE', mergeTarget: env.GERRIT_BRANCH]], diff -Nru asterisk-16.2.0~dfsg/.version asterisk-16.2.1~dfsg/.version --- asterisk-16.2.0~dfsg/.version 2019-02-15 17:31:09.000000000 +0100 +++ asterisk-16.2.1~dfsg/.version 2019-02-28 19:41:32.000000000 +0100 @@ -1 +1 @@ -16.2.0 \ Kein Zeilenumbruch am Dateiende. +16.2.1 \ Kein Zeilenumbruch am Dateiende.
