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26
ChangeLog
26
ChangeLog
@@ -1,3 +1,29 @@
|
||||
2014-10-20 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 12.6.1 Released.
|
||||
|
||||
* AST-2014-011: Fix POODLE security issues
|
||||
|
||||
There are two aspects to the vulnerability:
|
||||
(1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module
|
||||
to use TLSv1+. At this time, it does not refactor res_jabber/
|
||||
res_xmpp to use the TCP/TLS core, which should be done as an
|
||||
improvement at a latter date.
|
||||
(2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left
|
||||
unspecified, will default to the OpenSSL SSLv23_method. This
|
||||
method allows for all encryption methods, including SSLv2/SSLv3.
|
||||
A MITM can exploit this by forcing a fallback to SSLv3, which
|
||||
leaves the server vulnerable to POODLE. This patch adds WARNINGS
|
||||
if a user uses SSLv2/SSLv3 in their configuration, and explicitly
|
||||
disables SSLv2/SSLv3 if using SSLv23_method.
|
||||
|
||||
For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or
|
||||
SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer
|
||||
support SSLv2 or SSLv3.
|
||||
|
||||
Much thanks to abelbeck for reporting the vulnerability and providing
|
||||
a patch for the res_jabber/res_xmpp modules.
|
||||
|
||||
2014-09-24 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 12.6.0 Released.
|
||||
|
||||
11
UPGRADE.txt
11
UPGRADE.txt
@@ -21,6 +21,17 @@
|
||||
===
|
||||
===========================================================
|
||||
|
||||
From 12.6.0 to 12.6.1:
|
||||
- Due to the POODLE vulnerability (see
|
||||
https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-3566), the
|
||||
default TLS method for TLS clients will no longer allow SSLv3. As
|
||||
SSLv2 was already deprecated, it is no longer allowed by default as
|
||||
well. TLS servers no longer allow SSLv2 or SSLv3 connections. This
|
||||
affects the chan_sip channel driver, AMI, and the Asterisk HTTP server.
|
||||
|
||||
- The res_jabber resource module no longer uses SSLv3 to connect to an
|
||||
XMPP server. It will now only use TLSv1 or later methods.
|
||||
|
||||
From 12.5.0 to 12.6.0:
|
||||
|
||||
ConfBridge:
|
||||
|
||||
@@ -1,518 +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">
|
||||
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-12.6.0</title></head>
|
||||
<body>
|
||||
<h1 align="center"><a name="top">Release Summary</a></h1>
|
||||
<h3 align="center">asterisk-12.6.0</h3>
|
||||
<h3 align="center">Date: 2014-09-24</h3>
|
||||
<h3 align="center"><asteriskteam@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="#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><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series 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-12.5.0.</p>
|
||||
<hr/>
|
||||
<a name="contributors"><h2 align="center">Contributors</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><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 closed by commits that went into this release.</p>
|
||||
<table width="100%" border="0">
|
||||
<tr>
|
||||
<td width="33%"><h3>Coders</h3></td>
|
||||
<td width="33%"><h3>Testers</h3></td>
|
||||
<td width="33%"><h3>Reporters</h3></td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
<td>
|
||||
14 mjordan<br/>
|
||||
12 mmichelson<br/>
|
||||
11 rmudgett<br/>
|
||||
10 gtjoseph<br/>
|
||||
10 jrose<br/>
|
||||
8 file<br/>
|
||||
5 kmoore<br/>
|
||||
4 jcolp<br/>
|
||||
3 wdoekes<br/>
|
||||
2 Jeremy Laine<br/>
|
||||
2 seanbright<br/>
|
||||
1 cloos<br/>
|
||||
1 Elazar Broad<br/>
|
||||
1 elguero<br/>
|
||||
1 newtonr<br/>
|
||||
1 sruffell<br/>
|
||||
</td>
|
||||
<td>
|
||||
2 George Joseph<br/>
|
||||
1 Damien Wedhorn<br/>
|
||||
1 David Herselman<br/>
|
||||
1 Deepak Singh Rawat<br/>
|
||||
1 dimitripietro<br/>
|
||||
1 elguero<br/>
|
||||
1 Kilburn<br/>
|
||||
1 Samuel Galarneau<br/>
|
||||
1 sruffell<br/>
|
||||
1 Tony Lewis<br/>
|
||||
1 wdoekes<br/>
|
||||
</td>
|
||||
<td>
|
||||
7 mjordan<br/>
|
||||
2 mmichelson<br/>
|
||||
2 sharky<br/>
|
||||
2 sruffell<br/>
|
||||
1 amohod<br/>
|
||||
1 ateks<br/>
|
||||
1 bbs2web<br/>
|
||||
1 dimitripietro<br/>
|
||||
1 dsr<br/>
|
||||
1 Each<br/>
|
||||
1 ebroad<br/>
|
||||
1 edvinv<br/>
|
||||
1 falves11<br/>
|
||||
1 jideliov<br/>
|
||||
1 krandonbruse<br/>
|
||||
1 maddog<br/>
|
||||
1 pnlarsson<br/>
|
||||
1 proftech<br/>
|
||||
1 rmudgett<br/>
|
||||
1 RomanSk<br/>
|
||||
1 sgalarneau<br/>
|
||||
1 slavon<br/>
|
||||
1 wdoekes<br/>
|
||||
1 xrobau<br/>
|
||||
</td>
|
||||
</tr>
|
||||
</table>
|
||||
<hr/>
|
||||
<a name="issues"><h2 align="center">Closed Issues</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
|
||||
<h3>Category: . I did not set the category correctly.</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24147">ASTERISK-24147</a>: ARI: channel hangup crashes asterisk process<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421879">421879</a><br/>
|
||||
Reporter: edvinv<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<h3>Category: Applications/app_controlplayback</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421695">421695</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Applications/app_dial</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24225">ASTERISK-24225</a>: Dial option z is broken<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421234">421234</a><br/>
|
||||
Reporter: dimitripietro<br/>
|
||||
Testers: dimitripietro<br/>
|
||||
Coders: rmudgett<br/>
|
||||
<br/>
|
||||
<h3>Category: Applications/app_meetme</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24234">ASTERISK-24234</a>: app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421270">421270</a><br/>
|
||||
Reporter: sruffell<br/>
|
||||
Testers: sruffell<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Applications/app_mixmonitor</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420934">420934</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421186">421186</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<h3>Category: CDR/General</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24237">ASTERISK-24237</a>: CDR: FRACK With PJSIP blonde transfer.<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423525">423525</a><br/>
|
||||
Reporter: rmudgett<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24241">ASTERISK-24241</a>: crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422715">422715</a><br/>
|
||||
Reporter: dsr<br/>
|
||||
Testers: Deepak Singh Rawat<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24254">ASTERISK-24254</a>: CDRs: Application/args/dialplan CEP updated during dial operation<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422718">422718</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Testers: Tony Lewis<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Channels/chan_iax2</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23767">ASTERISK-23767</a>: [patch] Dynamic IAX2 registration stops trying if ever not able to resolve<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422275">422275</a><br/>
|
||||
Reporter: bbs2web<br/>
|
||||
Testers: David Herselman, elguero<br/>
|
||||
Coders: elguero<br/>
|
||||
<br/>
|
||||
<h3>Category: Channels/chan_pjsip</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24143">ASTERISK-24143</a>: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421955">421955</a><br/>
|
||||
Reporter: Each<br/>
|
||||
Coders: jcolp<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422536">422536</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Channels/chan_sip/General</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24178">ASTERISK-24178</a>: [patch]fromdomainport used even if not set<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421719">421719</a><br/>
|
||||
Reporter: ebroad<br/>
|
||||
Coders: Elazar Broad<br/>
|
||||
<br/>
|
||||
<h3>Category: Channels/chan_sip/Messaging</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24301">ASTERISK-24301</a>: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423365">423365</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Channels/chan_sip/WebSocket</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23997">ASTERISK-23997</a>: chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421910">421910</a><br/>
|
||||
Reporter: slavon<br/>
|
||||
Coders: jcolp<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/Configuration</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24231">ASTERISK-24231</a>: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422984">422984</a><br/>
|
||||
Reporter: pnlarsson<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/ManagerInterface</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24331">ASTERISK-24331</a>: Unexpected Errors in Asterisk Manager Interface Output<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423282">423282</a><br/>
|
||||
Reporter: xrobau<br/>
|
||||
Testers: George Joseph<br/>
|
||||
Coders: gtjoseph<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/PBX</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24249">ASTERISK-24249</a>: SIP debugs do not stop<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423068">423068</a><br/>
|
||||
Reporter: amohod<br/>
|
||||
Coders: wdoekes<br/>
|
||||
<br/>
|
||||
<h3>Category: Documentation</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422373">422373</a><br/>
|
||||
Reporter: sharky<br/>
|
||||
Coders: Jeremy Laine<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422378">422378</a><br/>
|
||||
Reporter: sharky<br/>
|
||||
Coders: Jeremy Laine<br/>
|
||||
<br/>
|
||||
<h3>Category: General</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24032">ASTERISK-24032</a>: Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421229">421229</a><br/>
|
||||
Reporter: maddog<br/>
|
||||
Testers: Kilburn, wdoekes<br/>
|
||||
Coders: cloos<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_agi</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420934">420934</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421186">421186</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_ari</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24043">ASTERISK-24043</a>: ARI /continue fails to actually continue into the dialplan<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421416">421416</a><br/>
|
||||
Reporter: krandonbruse<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421695">421695</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24264">ASTERISK-24264</a>: ARI: Adding a channel to a holding bridge automatically starts MOH<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422503">422503</a><br/>
|
||||
Reporter: sgalarneau<br/>
|
||||
Testers: Samuel Galarneau<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_ari_bridges</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24264">ASTERISK-24264</a>: ARI: Adding a channel to a holding bridge automatically starts MOH<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422503">422503</a><br/>
|
||||
Reporter: sgalarneau<br/>
|
||||
Testers: Samuel Galarneau<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_ari_playbacks</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421695">421695</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_fax</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24301">ASTERISK-24301</a>: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423365">423365</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_hep_rtcp</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24236">ASTERISK-24236</a>: res_hep_rtcp: Module incorrectly depends on pjsip<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421064">421064</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Testers: Damien Wedhorn<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_musiconhold</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22252">ASTERISK-22252</a>: res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421779">421779</a><br/>
|
||||
Reporter: wdoekes<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24019">ASTERISK-24019</a>: When a Music On Hold stream starts it restarts at beginning of file.<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421978">421978</a><br/>
|
||||
Reporter: ateks<br/>
|
||||
Coders: rmudgett<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24161">ASTERISK-24161</a>: PJSIPShowEndpoint gives inaccurate count of list items<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423282">423282</a><br/>
|
||||
Reporter: mmichelson<br/>
|
||||
Testers: George Joseph<br/>
|
||||
Coders: gtjoseph<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_endpoint_identifier_ip</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24290">ASTERISK-24290</a>: Endpoint identifier match value fails to parse when CIDR network format is specified<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423417">423417</a><br/>
|
||||
Reporter: proftech<br/>
|
||||
Coders: jrose<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_nat</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23634">ASTERISK-23634</a>: With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423151">423151</a><br/>
|
||||
Reporter: RomanSk<br/>
|
||||
Coders: jcolp<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_pubsub</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24136">ASTERISK-24136</a>: Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423344">423344</a><br/>
|
||||
Reporter: mmichelson<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_sdp_rtp</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23994">ASTERISK-23994</a>: res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421796">421796</a><br/>
|
||||
Reporter: falves11<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_transport_websocket</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24143">ASTERISK-24143</a>: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421955">421955</a><br/>
|
||||
Reporter: Each<br/>
|
||||
Coders: jcolp<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_rtp_asterisk</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23577">ASTERISK-23577</a>: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423151">423151</a><br/>
|
||||
Reporter: jideliov<br/>
|
||||
Coders: jcolp<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422536">422536</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Tests/testsuite</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422536">422536</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mmichelson<br/>
|
||||
<br/>
|
||||
<h3>Category: Utilities/aelparse</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422373">422373</a><br/>
|
||||
Reporter: sharky<br/>
|
||||
Coders: Jeremy Laine<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422378">422378</a><br/>
|
||||
Reporter: sharky<br/>
|
||||
Coders: Jeremy Laine<br/>
|
||||
<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><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
|
||||
<table width="100%" border="1">
|
||||
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420836">420836</a></td><td>rmudgett</td><td>res/stasis/command.c: Fix recent commit using spaces instead of tabs.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420898">420898</a></td><td>wdoekes</td><td>logger: Don't store verbose-magic in the log files.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420949">420949</a></td><td>kmoore</td><td>PJSIP: Prevent crash no-URI contacts</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=420956">420956</a></td><td>rmudgett</td><td>res_pjsip_send_to_voicemail.c: Fix svn file properties.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421009">421009</a></td><td>rmudgett</td><td>ARI: Originate to app local channel subscription code optimization.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421037">421037</a></td><td>mjordan</td><td>cel: Make sure channels in extra fields include their unique IDs as well</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421061">421061</a></td><td>mjordan</td><td>main/file: Move test event to emit PLAYBACK event more consistently</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421165">421165</a></td><td>mjordan</td><td>app_voicemail/app: Remove test events that were duplicated by r421059</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421329">421329</a></td><td>gtjoseph</td><td>func_config: Change 'Not Found' message from ERROR to DEBUG</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421400">421400</a></td><td>rmudgett</td><td>chan_pjsip: Fix attended transfer connected line name update.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421444">421444</a></td><td>kmoore</td><td>AMI Docs: Fix Status channel parameter optionality</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421447">421447</a></td><td>mmichelson</td><td>Fix compilation error on certain versions of GCC.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421485">421485</a></td><td>mmichelson</td><td>Alter documentation for callerid_privacy to use correct values.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421537">421537</a></td><td>kmoore</td><td>Stasis: Add information to blind transfer event</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421565">421565</a></td><td>mmichelson</td><td>Move evaluation of set_var options in pjsip to the end of channel initialization.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421608">421608</a></td><td>rmudgett</td><td>cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421778">421778</a></td><td>mmichelson</td><td>Improve consistency of party ID privacy usage.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421789">421789</a></td><td>mmichelson</td><td>Let's try checking the name and number, instead of the name twice.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421792">421792</a></td><td>mmichelson</td><td>Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421801">421801</a></td><td>rmudgett</td><td>res_musiconhold.c: Remove obsolete REF_DEBUG code.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421859">421859</a></td><td>mjordan</td><td>main/message: Add a new-line to a DEBUG message</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421931">421931</a></td><td>file</td><td>res_pjsip_transport_websocket: Ensure secure Websocket clients can be called.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=421939">421939</a></td><td>file</td><td>res_pjsip_transport_websocket: Fix a progressive memory growth.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422070">422070</a></td><td>mmichelson</td><td>Fix race condition in the scheduler when deleting a running entry.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422090">422090</a></td><td>gtjoseph</td><td>confbridge: Make kick, mute and unmute handle channel targets consistently.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422114">422114</a></td><td>kmoore</td><td>CallerID: Fix parsing of malformed callerid</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422176">422176</a></td><td>gtjoseph</td><td>confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422214">422214</a></td><td>rmudgett</td><td>res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422255">422255</a></td><td>rmudgett</td><td>Added ConfBridge AMI event note to UPGRADE.txt.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422295">422295</a></td><td>mjordan</td><td>LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422441">422441</a></td><td>gtjoseph</td><td>manager: Make WaitEvent action respect eventfilters</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422444">422444</a></td><td>gtjoseph</td><td>confbridge: Add Duration to ConfbridgeList event</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422506">422506</a></td><td>mjordan</td><td>main/cli: Do not attempt to show CDR data for internal channels</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422557">422557</a></td><td>file</td><td>res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422626">422626</a></td><td>jrose</td><td>Manager: Require read permission for SYSTEM in order to send FullyBooted</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422664">422664</a></td><td>jrose</td><td>Call IDs: Fix appearance of call ID in core show channels when NULL</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422746">422746</a></td><td>file</td><td>res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422766">422766</a></td><td>mjordan</td><td>main/rtp_engine: Format NTP timestamps as unsigned ints</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422769">422769</a></td><td>mjordan</td><td>main/cdr: Copy over location information during a fork</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422791">422791</a></td><td>newtonr</td><td>Sounds/BuildSystem: Modifications to include new releases and Japanese language.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422899">422899</a></td><td>seanbright</td><td>pjsip/config_auth.c: Add missing whitespace to log messages.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422904">422904</a></td><td>gtjoseph</td><td>config: bug: fix truncation of included config files on permissions error</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=422964">422964</a></td><td>mmichelson</td><td>Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423128">423128</a></td><td>wdoekes</td><td>contrib: Fix verifyi typo in alembic DB script ps_transport table.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423172">423172</a></td><td>file</td><td>res_pjsip_session: Fix usage of wrong memory pool when creating local SDP.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423208">423208</a></td><td>file</td><td>res_rtp_asterisk: Fix building when pjproject is not used.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423211">423211</a></td><td>file</td><td>res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423254">423254</a></td><td>file</td><td>res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423278">423278</a></td><td>gtjoseph</td><td>config: bug: Fix SEGV in ast_category_insert when matching category isn't found</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423416">423416</a></td><td>rmudgett</td><td>astobj2.c/refcounter.py: Fix to deal with invalid object refs.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423476">423476</a></td><td>gtjoseph</td><td>utils: Create ast_strsep function that ignores separators inside quotes</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423481">423481</a></td><td>seanbright</td><td>res_pjsip: Don't require a password when doing userpass authentication.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/12?view=revision&revision=423503">423503</a></td><td>kmoore</td><td>PJSIP: Prevent T38 framehook being put on wrong channel</td>
|
||||
<td></td></tr></table>
|
||||
<hr/>
|
||||
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><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>
|
||||
LICENSE | 2
|
||||
UPGRADE.txt | 12
|
||||
apps/app_chanspy.c | 2
|
||||
apps/app_confbridge.c | 262 ++-
|
||||
apps/app_dial.c | 2
|
||||
apps/app_macro.c | 7
|
||||
apps/app_meetme.c | 8
|
||||
apps/app_mixmonitor.c | 2
|
||||
apps/app_stack.c | 35
|
||||
apps/app_voicemail.c | 5
|
||||
apps/confbridge/confbridge_manager.c | 81 +
|
||||
channels/chan_iax2.c | 34
|
||||
channels/chan_pjsip.c | 83 -
|
||||
channels/chan_sip.c | 24
|
||||
configs/sip.conf.sample | 4
|
||||
configure.ac | 4
|
||||
contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py | 29
|
||||
contrib/scripts/refcounter.py | 80 -
|
||||
doc/aelparse.8 | 28
|
||||
doc/smsq.8 | 146 ++
|
||||
funcs/func_config.c | 2
|
||||
include/asterisk/channel.h | 43
|
||||
include/asterisk/config.h | 5
|
||||
include/asterisk/framehook.h | 6
|
||||
include/asterisk/res_pjsip.h | 2
|
||||
include/asterisk/res_pjsip_pubsub.h | 21
|
||||
include/asterisk/res_pjsip_session.h | 35
|
||||
include/asterisk/stasis_app_impl.h | 25
|
||||
include/asterisk/stasis_bridges.h | 10
|
||||
include/asterisk/strings.h | 60
|
||||
include/asterisk/utils.h | 9
|
||||
main/app.c | 7
|
||||
main/astobj2.c | 10
|
||||
main/bridge.c | 53
|
||||
main/bridge_after.c | 4
|
||||
main/bridge_channel.c | 4
|
||||
main/callerid.c | 63
|
||||
main/cdr.c | 22
|
||||
main/cel.c | 27
|
||||
main/channel.c | 14
|
||||
main/channel_internal_api.c | 30
|
||||
main/cli.c | 6
|
||||
main/config.c | 132 +
|
||||
main/dns.c | 3
|
||||
main/file.c | 2
|
||||
main/framehook.c | 19
|
||||
main/logger.c | 43
|
||||
main/manager.c | 16
|
||||
main/message.c | 2
|
||||
main/pbx.c | 5
|
||||
main/rtp_engine.c | 4
|
||||
main/sched.c | 43
|
||||
main/stasis_bridges.c | 28
|
||||
main/stasis_channels.c | 219 +++
|
||||
main/utils.c | 83 +
|
||||
res/ari/ari_model_validators.c | 9
|
||||
res/ari/ari_model_validators.h | 1
|
||||
res/ari/resource_channels.c | 13
|
||||
res/res_fax_spandsp.c | 19
|
||||
res/res_hep_rtcp.c | 2
|
||||
res/res_musiconhold.c | 25
|
||||
res/res_pjsip.c | 10
|
||||
res/res_pjsip/config_auth.c | 14
|
||||
res/res_pjsip/config_transport.c | 18
|
||||
res/res_pjsip/location.c | 2
|
||||
res/res_pjsip/pjsip_configuration.c | 6
|
||||
res/res_pjsip/pjsip_options.c | 170 +-
|
||||
res/res_pjsip_caller_id.c | 94 -
|
||||
res/res_pjsip_dialog_info_body_generator.c | 1
|
||||
res/res_pjsip_diversion.c | 1
|
||||
res/res_pjsip_endpoint_identifier_ip.c | 62
|
||||
res/res_pjsip_exten_state.c | 8
|
||||
res/res_pjsip_mwi.c | 7
|
||||
res/res_pjsip_mwi_body_generator.c | 1
|
||||
res/res_pjsip_notify.c | 8
|
||||
res/res_pjsip_pidf_body_generator.c | 1
|
||||
res/res_pjsip_pubsub.c | 25
|
||||
res/res_pjsip_sdp_rtp.c | 2
|
||||
res/res_pjsip_session.c | 72 -
|
||||
res/res_pjsip_t38.c | 13
|
||||
res/res_pjsip_transport_websocket.c | 46
|
||||
res/res_pjsip_xpidf_body_generator.c | 2
|
||||
res/res_rtp_asterisk.c | 703 ++++++----
|
||||
res/res_stasis.c | 30
|
||||
res/res_stasis_answer.c | 2
|
||||
res/res_stasis_playback.c | 20
|
||||
res/res_stasis_recording.c | 20
|
||||
res/stasis/app.c | 31
|
||||
res/stasis/command.c | 41
|
||||
res/stasis/command.h | 9
|
||||
res/stasis/control.c | 81 -
|
||||
res/stasis/messaging.h | 2
|
||||
res/stasis/stasis_bridge.c | 28
|
||||
rest-api/api-docs/events.json | 5
|
||||
sounds/Makefile | 7
|
||||
sounds/sounds.xml | 27
|
||||
tests/test_callerid.c | 165 ++
|
||||
tests/test_cel.c | 21
|
||||
tests/test_strings.c | 80 +
|
||||
tests/test_utils.c | 98 +
|
||||
100 files changed, 3053 insertions(+), 856 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
@@ -1,723 +0,0 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-12.6.0
|
||||
|
||||
Date: 2014-09-24
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Closed Issues
|
||||
4. Other Changes
|
||||
5. Diffstat
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Summary
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This release includes only bug fixes. The changes included were made only
|
||||
to address problems that have been identified in this release series.
|
||||
Users should be able to safely upgrade to this version if this release
|
||||
series is already in use. Users considering upgrading from a previous
|
||||
release series 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-12.5.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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 closed by commits that went into this
|
||||
release.
|
||||
|
||||
Coders Testers Reporters
|
||||
14 mjordan 2 George Joseph 7 mjordan
|
||||
12 mmichelson 1 Damien Wedhorn 2 mmichelson
|
||||
11 rmudgett 1 David Herselman 2 sharky
|
||||
10 gtjoseph 1 Deepak Singh Rawat 2 sruffell
|
||||
10 jrose 1 dimitripietro 1 amohod
|
||||
8 file 1 elguero 1 ateks
|
||||
5 kmoore 1 Kilburn 1 bbs2web
|
||||
4 jcolp 1 Samuel Galarneau 1 dimitripietro
|
||||
3 wdoekes 1 sruffell 1 dsr
|
||||
2 Jeremy Laine 1 Tony Lewis 1 Each
|
||||
2 seanbright 1 wdoekes 1 ebroad
|
||||
1 cloos 1 edvinv
|
||||
1 Elazar Broad 1 falves11
|
||||
1 elguero 1 jideliov
|
||||
1 newtonr 1 krandonbruse
|
||||
1 sruffell 1 maddog
|
||||
1 pnlarsson
|
||||
1 proftech
|
||||
1 rmudgett
|
||||
1 RomanSk
|
||||
1 sgalarneau
|
||||
1 slavon
|
||||
1 wdoekes
|
||||
1 xrobau
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
Category: . I did not set the category correctly.
|
||||
|
||||
ASTERISK-24147: ARI: channel hangup crashes asterisk process
|
||||
Revision: 421879
|
||||
Reporter: edvinv
|
||||
Coders: jrose
|
||||
|
||||
Category: Applications/app_controlplayback
|
||||
|
||||
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
|
||||
preventing early media playback
|
||||
Revision: 421695
|
||||
Reporter: mjordan
|
||||
Coders: mjordan
|
||||
|
||||
Category: Applications/app_dial
|
||||
|
||||
ASTERISK-24225: Dial option z is broken
|
||||
Revision: 421234
|
||||
Reporter: dimitripietro
|
||||
Testers: dimitripietro
|
||||
Coders: rmudgett
|
||||
|
||||
Category: Applications/app_meetme
|
||||
|
||||
ASTERISK-24234: app_meetme: Crash on conference shutdown due to NULL
|
||||
channel passed to meetme_stasis_generate_msg()
|
||||
Revision: 421270
|
||||
Reporter: sruffell
|
||||
Testers: sruffell
|
||||
Coders: mjordan
|
||||
|
||||
Category: Applications/app_mixmonitor
|
||||
|
||||
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
|
||||
bridge feature causes channel to leave AGI has hung up
|
||||
Revision: 420934
|
||||
Reporter: mjordan
|
||||
Coders: jrose
|
||||
|
||||
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
|
||||
bridge feature causes channel to leave AGI has hung up
|
||||
Revision: 421186
|
||||
Reporter: mjordan
|
||||
Coders: jrose
|
||||
|
||||
Category: CDR/General
|
||||
|
||||
ASTERISK-24237: CDR: FRACK With PJSIP blonde transfer.
|
||||
Revision: 423525
|
||||
Reporter: rmudgett
|
||||
Coders: jrose
|
||||
|
||||
ASTERISK-24241: crash: CDRs recursively attempt to update Party B
|
||||
information in a multi-party bridge, overrunning the stack
|
||||
Revision: 422715
|
||||
Reporter: dsr
|
||||
Testers: Deepak Singh Rawat
|
||||
Coders: mjordan
|
||||
|
||||
ASTERISK-24254: CDRs: Application/args/dialplan CEP updated during dial
|
||||
operation
|
||||
Revision: 422718
|
||||
Reporter: mjordan
|
||||
Testers: Tony Lewis
|
||||
Coders: mjordan
|
||||
|
||||
Category: Channels/chan_iax2
|
||||
|
||||
ASTERISK-23767: [patch] Dynamic IAX2 registration stops trying if ever not
|
||||
able to resolve
|
||||
Revision: 422275
|
||||
Reporter: bbs2web
|
||||
Testers: David Herselman, elguero
|
||||
Coders: elguero
|
||||
|
||||
Category: Channels/chan_pjsip
|
||||
|
||||
ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on
|
||||
received 200 OK
|
||||
Revision: 421955
|
||||
Reporter: Each
|
||||
Coders: jcolp
|
||||
|
||||
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
|
||||
engine
|
||||
Revision: 422536
|
||||
Reporter: mjordan
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Channels/chan_sip/General
|
||||
|
||||
ASTERISK-24178: [patch]fromdomainport used even if not set
|
||||
Revision: 421719
|
||||
Reporter: ebroad
|
||||
Coders: Elazar Broad
|
||||
|
||||
Category: Channels/chan_sip/Messaging
|
||||
|
||||
ASTERISK-24301: Security: Out of call MESSAGE requests processed via
|
||||
Message channel driver can crash Asterisk
|
||||
Revision: 423365
|
||||
Reporter: mjordan
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Channels/chan_sip/WebSocket
|
||||
|
||||
ASTERISK-23997: chan_sip: port incorrectly incremented for RTCP ICE
|
||||
candidates in SDP answer
|
||||
Revision: 421910
|
||||
Reporter: slavon
|
||||
Coders: jcolp
|
||||
|
||||
Category: Core/Configuration
|
||||
|
||||
ASTERISK-24231: crash: CLI execution of realtime destroy sippeers id 1
|
||||
causes crash due to NULL name provided to ast_variable
|
||||
Revision: 422984
|
||||
Reporter: pnlarsson
|
||||
Coders: jrose
|
||||
|
||||
Category: Core/ManagerInterface
|
||||
|
||||
ASTERISK-24331: Unexpected Errors in Asterisk Manager Interface Output
|
||||
Revision: 423282
|
||||
Reporter: xrobau
|
||||
Testers: George Joseph
|
||||
Coders: gtjoseph
|
||||
|
||||
Category: Core/PBX
|
||||
|
||||
ASTERISK-24249: SIP debugs do not stop
|
||||
Revision: 423068
|
||||
Reporter: amohod
|
||||
Coders: wdoekes
|
||||
|
||||
Category: Documentation
|
||||
|
||||
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
|
||||
Revision: 422373
|
||||
Reporter: sharky
|
||||
Coders: Jeremy Laine
|
||||
|
||||
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
|
||||
Revision: 422378
|
||||
Reporter: sharky
|
||||
Coders: Jeremy Laine
|
||||
|
||||
Category: General
|
||||
|
||||
ASTERISK-24032: Gentoo compilation emits warning: "_FORTIFY_SOURCE"
|
||||
redefined
|
||||
Revision: 421229
|
||||
Reporter: maddog
|
||||
Testers: Kilburn, wdoekes
|
||||
Coders: cloos
|
||||
|
||||
Category: Resources/res_agi
|
||||
|
||||
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
|
||||
bridge feature causes channel to leave AGI has hung up
|
||||
Revision: 420934
|
||||
Reporter: mjordan
|
||||
Coders: jrose
|
||||
|
||||
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
|
||||
bridge feature causes channel to leave AGI has hung up
|
||||
Revision: 421186
|
||||
Reporter: mjordan
|
||||
Coders: jrose
|
||||
|
||||
Category: Resources/res_ari
|
||||
|
||||
ASTERISK-24043: ARI /continue fails to actually continue into the dialplan
|
||||
Revision: 421416
|
||||
Reporter: krandonbruse
|
||||
Coders: jrose
|
||||
|
||||
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
|
||||
preventing early media playback
|
||||
Revision: 421695
|
||||
Reporter: mjordan
|
||||
Coders: mjordan
|
||||
|
||||
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically
|
||||
starts MOH
|
||||
Revision: 422503
|
||||
Reporter: sgalarneau
|
||||
Testers: Samuel Galarneau
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_ari_bridges
|
||||
|
||||
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically
|
||||
starts MOH
|
||||
Revision: 422503
|
||||
Reporter: sgalarneau
|
||||
Testers: Samuel Galarneau
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_ari_playbacks
|
||||
|
||||
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
|
||||
preventing early media playback
|
||||
Revision: 421695
|
||||
Reporter: mjordan
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_fax
|
||||
|
||||
ASTERISK-24301: Security: Out of call MESSAGE requests processed via
|
||||
Message channel driver can crash Asterisk
|
||||
Revision: 423365
|
||||
Reporter: mjordan
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Resources/res_hep_rtcp
|
||||
|
||||
ASTERISK-24236: res_hep_rtcp: Module incorrectly depends on pjsip
|
||||
Revision: 421064
|
||||
Reporter: mjordan
|
||||
Testers: Damien Wedhorn
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_musiconhold
|
||||
|
||||
ASTERISK-22252: res_musiconhold cleanup - REF_DEBUG reload warnings and
|
||||
ref leaks
|
||||
Revision: 421779
|
||||
Reporter: wdoekes
|
||||
Coders: jrose
|
||||
|
||||
ASTERISK-24019: When a Music On Hold stream starts it restarts at
|
||||
beginning of file.
|
||||
Revision: 421978
|
||||
Reporter: ateks
|
||||
Coders: rmudgett
|
||||
|
||||
Category: Resources/res_pjsip
|
||||
|
||||
ASTERISK-24161: PJSIPShowEndpoint gives inaccurate count of list items
|
||||
Revision: 423282
|
||||
Reporter: mmichelson
|
||||
Testers: George Joseph
|
||||
Coders: gtjoseph
|
||||
|
||||
Category: Resources/res_pjsip_endpoint_identifier_ip
|
||||
|
||||
ASTERISK-24290: Endpoint identifier match value fails to parse when CIDR
|
||||
network format is specified
|
||||
Revision: 423417
|
||||
Reporter: proftech
|
||||
Coders: jrose
|
||||
|
||||
Category: Resources/res_pjsip_nat
|
||||
|
||||
ASTERISK-23634: With TURN Asterisk crashes on multiple (7-10) concurrent
|
||||
WebRTC (avpg/encryption/icesupport) calls
|
||||
Revision: 423151
|
||||
Reporter: RomanSk
|
||||
Coders: jcolp
|
||||
|
||||
Category: Resources/res_pjsip_pubsub
|
||||
|
||||
ASTERISK-24136: Security: Crash in Asterisk's PJSIP code when subscribing
|
||||
to an event with an unexpected body type
|
||||
Revision: 423344
|
||||
Reporter: mmichelson
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Resources/res_pjsip_sdp_rtp
|
||||
|
||||
ASTERISK-23994: res_pjsip_sdp_rtp: owner address in SDP may not be fully
|
||||
qualified domainname
|
||||
Revision: 421796
|
||||
Reporter: falves11
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Resources/res_pjsip_transport_websocket
|
||||
|
||||
ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on
|
||||
received 200 OK
|
||||
Revision: 421955
|
||||
Reporter: Each
|
||||
Coders: jcolp
|
||||
|
||||
Category: Resources/res_rtp_asterisk
|
||||
|
||||
ASTERISK-23577: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when
|
||||
RTP instance is NULL
|
||||
Revision: 423151
|
||||
Reporter: jideliov
|
||||
Coders: jcolp
|
||||
|
||||
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
|
||||
engine
|
||||
Revision: 422536
|
||||
Reporter: mjordan
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Tests/testsuite
|
||||
|
||||
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
|
||||
engine
|
||||
Revision: 422536
|
||||
Reporter: mjordan
|
||||
Coders: mmichelson
|
||||
|
||||
Category: Utilities/aelparse
|
||||
|
||||
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
|
||||
Revision: 422373
|
||||
Reporter: sharky
|
||||
Coders: Jeremy Laine
|
||||
|
||||
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
|
||||
Revision: 422378
|
||||
Reporter: sharky
|
||||
Coders: Jeremy Laine
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Commits Not Associated with an Issue
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all changes that went into this release that did not
|
||||
directly close an issue from the issue tracker. The commits may have been
|
||||
marked as being related to an issue. If that is the case, the issue
|
||||
numbers are listed here, as well.
|
||||
|
||||
+------------------------------------------------------------------------+
|
||||
| Revision | Author | Summary | Issues |
|
||||
| | | | Referenced |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res/stasis/command.c: Fix recent | |
|
||||
| 420836 | rmudgett | commit using spaces instead of | |
|
||||
| | | tabs. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 420898 | wdoekes | logger: Don't store verbose-magic | |
|
||||
| | | in the log files. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 420949 | kmoore | PJSIP: Prevent crash no-URI | |
|
||||
| | | contacts | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 420956 | rmudgett | res_pjsip_send_to_voicemail.c: | |
|
||||
| | | Fix svn file properties. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | ARI: Originate to app local | |
|
||||
| 421009 | rmudgett | channel subscription code | |
|
||||
| | | optimization. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | cel: Make sure channels in extra | |
|
||||
| 421037 | mjordan | fields include their unique IDs | |
|
||||
| | | as well | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | main/file: Move test event to | |
|
||||
| 421061 | mjordan | emit PLAYBACK event more | |
|
||||
| | | consistently | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | app_voicemail/app: Remove test | |
|
||||
| 421165 | mjordan | events that were duplicated by | |
|
||||
| | | r421059 | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421329 | gtjoseph | func_config: Change 'Not Found' | |
|
||||
| | | message from ERROR to DEBUG | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421400 | rmudgett | chan_pjsip: Fix attended transfer | |
|
||||
| | | connected line name update. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421444 | kmoore | AMI Docs: Fix Status channel | |
|
||||
| | | parameter optionality | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421447 | mmichelson | Fix compilation error on certain | |
|
||||
| | | versions of GCC. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Alter documentation for | |
|
||||
| 421485 | mmichelson | callerid_privacy to use correct | |
|
||||
| | | values. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421537 | kmoore | Stasis: Add information to blind | |
|
||||
| | | transfer event | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Move evaluation of set_var | |
|
||||
| 421565 | mmichelson | options in pjsip to the end of | |
|
||||
| | | channel initialization. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | cli.c: Fix tab completion of | |
|
||||
| 421608 | rmudgett | "module load" when MALLOC_DEBUG | |
|
||||
| | | is enabled. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421778 | mmichelson | Improve consistency of party ID | |
|
||||
| | | privacy usage. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Let's try checking the name and | |
|
||||
| 421789 | mmichelson | number, instead of the name | |
|
||||
| | | twice. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Ensure after-bridge behavior is | |
|
||||
| 421792 | mmichelson | correct when moving from Stasis | |
|
||||
| | | to a non-Stasis bridge. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421801 | rmudgett | res_musiconhold.c: Remove | |
|
||||
| | | obsolete REF_DEBUG code. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421859 | mjordan | main/message: Add a new-line to a | |
|
||||
| | | DEBUG message | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_pjsip_transport_websocket: | |
|
||||
| 421931 | file | Ensure secure Websocket clients | |
|
||||
| | | can be called. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 421939 | file | res_pjsip_transport_websocket: | |
|
||||
| | | Fix a progressive memory growth. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Fix race condition in the | |
|
||||
| 422070 | mmichelson | scheduler when deleting a running | |
|
||||
| | | entry. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | confbridge: Make kick, mute and | |
|
||||
| 422090 | gtjoseph | unmute handle channel targets | |
|
||||
| | | consistently. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422114 | kmoore | CallerID: Fix parsing of | |
|
||||
| | | malformed callerid | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | confbridge: Add 'Admin' param to | |
|
||||
| 422176 | gtjoseph | join, leave, mute, unmute and | |
|
||||
| | | talking events | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res/res_pjsip/pjsip_options.c: | |
|
||||
| 422214 | rmudgett | Eliminate excessive RAII_VAR | |
|
||||
| | | usage. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422255 | rmudgett | Added ConfBridge AMI event note | |
|
||||
| | | to UPGRADE.txt. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | LICENSE: Clarify language in | |
|
||||
| 422295 | mjordan | Asterisk's LICENSE to allow for | |
|
||||
| | | linking to UniMRCP | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422441 | gtjoseph | manager: Make WaitEvent action | |
|
||||
| | | respect eventfilters | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422444 | gtjoseph | confbridge: Add Duration to | |
|
||||
| | | ConfbridgeList event | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422506 | mjordan | main/cli: Do not attempt to show | |
|
||||
| | | CDR data for internal channels | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_pjsip_transport_websocket: | |
|
||||
| 422557 | file | Fix crash when the Contact header | |
|
||||
| | | is not a URI. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Manager: Require read permission | |
|
||||
| 422626 | jrose | for SYSTEM in order to send | |
|
||||
| | | FullyBooted | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Call IDs: Fix appearance of call | |
|
||||
| 422664 | jrose | ID in core show channels when | |
|
||||
| | | NULL | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_pjsip_sdp_rtp: Fix retrieval | |
|
||||
| 422746 | file | of "ice-pwd" attribute if in | |
|
||||
| | | session and not media stream. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422766 | mjordan | main/rtp_engine: Format NTP | |
|
||||
| | | timestamps as unsigned ints | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422769 | mjordan | main/cdr: Copy over location | |
|
||||
| | | information during a fork | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Sounds/BuildSystem: Modifications | |
|
||||
| 422791 | newtonr | to include new releases and | |
|
||||
| | | Japanese language. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 422899 | seanbright | pjsip/config_auth.c: Add missing | |
|
||||
| | | whitespace to log messages. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | config: bug: fix truncation of | |
|
||||
| 422904 | gtjoseph | included config files on | |
|
||||
| | | permissions error | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | Remove undocumented default | |
|
||||
| 422964 | mmichelson | behavior of | |
|
||||
| | | ast_play_and_record_full | |
|
||||
| | | acceptdtmf. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | contrib: Fix verifyi typo in | |
|
||||
| 423128 | wdoekes | alembic DB script ps_transport | |
|
||||
| | | table. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_pjsip_session: Fix usage of | |
|
||||
| 423172 | file | wrong memory pool when creating | |
|
||||
| | | local SDP. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 423208 | file | res_rtp_asterisk: Fix building | |
|
||||
| | | when pjproject is not used. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_rtp_asterisk: Fix 100% CPU | |
|
||||
| 423211 | file | usage due to timer heap thread | |
|
||||
| | | spinning. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_rtp_asterisk: Ensure that the | |
|
||||
| 423254 | file | thread terminating pj stuff is | |
|
||||
| | | registered. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | config: bug: Fix SEGV in | |
|
||||
| 423278 | gtjoseph | ast_category_insert when matching | |
|
||||
| | | category isn't found | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 423416 | rmudgett | astobj2.c/refcounter.py: Fix to | |
|
||||
| | | deal with invalid object refs. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | utils: Create ast_strsep function | |
|
||||
| 423476 | gtjoseph | that ignores separators inside | |
|
||||
| | | quotes | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| | | res_pjsip: Don't require a | |
|
||||
| 423481 | seanbright | password when doing userpass | |
|
||||
| | | authentication. | |
|
||||
|----------+------------+-----------------------------------+------------|
|
||||
| 423503 | kmoore | PJSIP: Prevent T38 framehook | |
|
||||
| | | being put on wrong channel | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
LICENSE | 2
|
||||
UPGRADE.txt | 12
|
||||
apps/app_chanspy.c | 2
|
||||
apps/app_confbridge.c | 262 ++-
|
||||
apps/app_dial.c | 2
|
||||
apps/app_macro.c | 7
|
||||
apps/app_meetme.c | 8
|
||||
apps/app_mixmonitor.c | 2
|
||||
apps/app_stack.c | 35
|
||||
apps/app_voicemail.c | 5
|
||||
apps/confbridge/confbridge_manager.c | 81 +
|
||||
channels/chan_iax2.c | 34
|
||||
channels/chan_pjsip.c | 83 -
|
||||
channels/chan_sip.c | 24
|
||||
configs/sip.conf.sample | 4
|
||||
configure.ac | 4
|
||||
contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py | 29
|
||||
contrib/scripts/refcounter.py | 80 -
|
||||
doc/aelparse.8 | 28
|
||||
doc/smsq.8 | 146 ++
|
||||
funcs/func_config.c | 2
|
||||
include/asterisk/channel.h | 43
|
||||
include/asterisk/config.h | 5
|
||||
include/asterisk/framehook.h | 6
|
||||
include/asterisk/res_pjsip.h | 2
|
||||
include/asterisk/res_pjsip_pubsub.h | 21
|
||||
include/asterisk/res_pjsip_session.h | 35
|
||||
include/asterisk/stasis_app_impl.h | 25
|
||||
include/asterisk/stasis_bridges.h | 10
|
||||
include/asterisk/strings.h | 60
|
||||
include/asterisk/utils.h | 9
|
||||
main/app.c | 7
|
||||
main/astobj2.c | 10
|
||||
main/bridge.c | 53
|
||||
main/bridge_after.c | 4
|
||||
main/bridge_channel.c | 4
|
||||
main/callerid.c | 63
|
||||
main/cdr.c | 22
|
||||
main/cel.c | 27
|
||||
main/channel.c | 14
|
||||
main/channel_internal_api.c | 30
|
||||
main/cli.c | 6
|
||||
main/config.c | 132 +
|
||||
main/dns.c | 3
|
||||
main/file.c | 2
|
||||
main/framehook.c | 19
|
||||
main/logger.c | 43
|
||||
main/manager.c | 16
|
||||
main/message.c | 2
|
||||
main/pbx.c | 5
|
||||
main/rtp_engine.c | 4
|
||||
main/sched.c | 43
|
||||
main/stasis_bridges.c | 28
|
||||
main/stasis_channels.c | 219 +++
|
||||
main/utils.c | 83 +
|
||||
res/ari/ari_model_validators.c | 9
|
||||
res/ari/ari_model_validators.h | 1
|
||||
res/ari/resource_channels.c | 13
|
||||
res/res_fax_spandsp.c | 19
|
||||
res/res_hep_rtcp.c | 2
|
||||
res/res_musiconhold.c | 25
|
||||
res/res_pjsip.c | 10
|
||||
res/res_pjsip/config_auth.c | 14
|
||||
res/res_pjsip/config_transport.c | 18
|
||||
res/res_pjsip/location.c | 2
|
||||
res/res_pjsip/pjsip_configuration.c | 6
|
||||
res/res_pjsip/pjsip_options.c | 170 +-
|
||||
res/res_pjsip_caller_id.c | 94 -
|
||||
res/res_pjsip_dialog_info_body_generator.c | 1
|
||||
res/res_pjsip_diversion.c | 1
|
||||
res/res_pjsip_endpoint_identifier_ip.c | 62
|
||||
res/res_pjsip_exten_state.c | 8
|
||||
res/res_pjsip_mwi.c | 7
|
||||
res/res_pjsip_mwi_body_generator.c | 1
|
||||
res/res_pjsip_notify.c | 8
|
||||
res/res_pjsip_pidf_body_generator.c | 1
|
||||
res/res_pjsip_pubsub.c | 25
|
||||
res/res_pjsip_sdp_rtp.c | 2
|
||||
res/res_pjsip_session.c | 72 -
|
||||
res/res_pjsip_t38.c | 13
|
||||
res/res_pjsip_transport_websocket.c | 46
|
||||
res/res_pjsip_xpidf_body_generator.c | 2
|
||||
res/res_rtp_asterisk.c | 703 ++++++----
|
||||
res/res_stasis.c | 30
|
||||
res/res_stasis_answer.c | 2
|
||||
res/res_stasis_playback.c | 20
|
||||
res/res_stasis_recording.c | 20
|
||||
res/stasis/app.c | 31
|
||||
res/stasis/command.c | 41
|
||||
res/stasis/command.h | 9
|
||||
res/stasis/control.c | 81 -
|
||||
res/stasis/messaging.h | 2
|
||||
res/stasis/stasis_bridge.c | 28
|
||||
rest-api/api-docs/events.json | 5
|
||||
sounds/Makefile | 7
|
||||
sounds/sounds.xml | 27
|
||||
tests/test_callerid.c | 165 ++
|
||||
tests/test_cel.c | 21
|
||||
tests/test_strings.c | 80 +
|
||||
tests/test_utils.c | 98 +
|
||||
100 files changed, 3053 insertions(+), 856 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
65
asterisk-12.6.1-summary.html
Normal file
65
asterisk-12.6.1-summary.html
Normal file
@@ -0,0 +1,65 @@
|
||||
<!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">
|
||||
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-12.6.1</title></head>
|
||||
<body>
|
||||
<h1 align="center"><a name="top">Release Summary</a></h1>
|
||||
<h3 align="center">asterisk-12.6.1</h3>
|
||||
<h3 align="center">Date: 2014-10-20</h3>
|
||||
<h3 align="center"><asteriskteam@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="#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><br/><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: <a href="http://downloads.asterisk.org/pub/security/AST-2014-011.html">AST-2014-011</a></p>
|
||||
<p>The data in this summary reflects changes that have been made since the previous release, asterisk-12.6.0.</p>
|
||||
<hr/>
|
||||
<a name="contributors"><h2 align="center">Contributors</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><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 closed by commits that went into this release.</p>
|
||||
<table width="100%" border="0">
|
||||
<tr>
|
||||
<td width="33%"><h3>Coders</h3></td>
|
||||
<td width="33%"><h3>Testers</h3></td>
|
||||
<td width="33%"><h3>Reporters</h3></td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
<td>
|
||||
3 bebuild<br/>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
</tr>
|
||||
</table>
|
||||
<hr/>
|
||||
<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
|
||||
<table width="100%" border="1">
|
||||
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/tags/12.6.1?view=revision&revision=425990">425990</a></td><td>bebuild</td><td>Create 12.6.1</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/tags/12.6.1?view=revision&revision=426000">426000</a></td><td>bebuild</td><td>Update .version, remove summaries</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/tags/12.6.1?view=revision&revision=426064">426064</a></td><td>bebuild</td><td>Merge 425987</td>
|
||||
<td></td></tr></table>
|
||||
<hr/>
|
||||
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><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>
|
||||
.version | 2
|
||||
ChangeLog | 26 +
|
||||
UPGRADE.txt | 11
|
||||
asterisk-12.6.0-summary.html | 518 ------------------------------
|
||||
asterisk-12.6.0-summary.txt | 723 -------------------------------------------
|
||||
main/tcptls.c | 22 +
|
||||
res/res_jabber.c | 5
|
||||
res/res_xmpp.c | 6
|
||||
8 files changed, 64 insertions(+), 1249 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
95
asterisk-12.6.1-summary.txt
Normal file
95
asterisk-12.6.1-summary.txt
Normal file
@@ -0,0 +1,95 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-12.6.1
|
||||
|
||||
Date: 2014-10-20
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Other Changes
|
||||
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-2014-011
|
||||
|
||||
The data in this summary reflects changes that have been made since the
|
||||
previous release, asterisk-12.6.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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 closed by commits that went into this
|
||||
release.
|
||||
|
||||
Coders Testers Reporters
|
||||
3 bebuild
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Commits Not Associated with an Issue
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all changes that went into this release that did not
|
||||
directly close an issue from the issue tracker. The commits may have been
|
||||
marked as being related to an issue. If that is the case, the issue
|
||||
numbers are listed here, as well.
|
||||
|
||||
+------------------------------------------------------------------------+
|
||||
| Revision | Author | Summary | Issues Referenced |
|
||||
|----------+---------+-------------------------------+-------------------|
|
||||
| 425990 | bebuild | Create 12.6.1 | |
|
||||
|----------+---------+-------------------------------+-------------------|
|
||||
| 426000 | bebuild | Update .version, remove | |
|
||||
| | | summaries | |
|
||||
|----------+---------+-------------------------------+-------------------|
|
||||
| 426064 | bebuild | Merge 425987 | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
.version | 2
|
||||
ChangeLog | 26 +
|
||||
UPGRADE.txt | 11
|
||||
asterisk-12.6.0-summary.html | 518 ------------------------------
|
||||
asterisk-12.6.0-summary.txt | 723 -------------------------------------------
|
||||
main/tcptls.c | 22 +
|
||||
res/res_jabber.c | 5
|
||||
res/res_xmpp.c | 6
|
||||
8 files changed, 64 insertions(+), 1249 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
@@ -747,6 +747,8 @@ static int __ssl_setup(struct ast_tls_config *cfg, int client)
|
||||
cfg->enabled = 0;
|
||||
return 0;
|
||||
#else
|
||||
int disable_ssl = 0;
|
||||
|
||||
if (!cfg->enabled) {
|
||||
return 0;
|
||||
}
|
||||
@@ -762,22 +764,21 @@ static int __ssl_setup(struct ast_tls_config *cfg, int client)
|
||||
if (client) {
|
||||
#ifndef OPENSSL_NO_SSL2
|
||||
if (ast_test_flag(&cfg->flags, AST_SSL_SSLV2_CLIENT)) {
|
||||
ast_log(LOG_WARNING, "Usage of SSLv2 is discouraged due to known vulnerabilities. Please use 'tlsv1' or leave the TLS method unspecified!\n");
|
||||
cfg->ssl_ctx = SSL_CTX_new(SSLv2_client_method());
|
||||
} else
|
||||
#endif
|
||||
if (ast_test_flag(&cfg->flags, AST_SSL_SSLV3_CLIENT)) {
|
||||
ast_log(LOG_WARNING, "Usage of SSLv3 is discouraged due to known vulnerabilities. Please use 'tlsv1' or leave the TLS method unspecified!\n");
|
||||
cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method());
|
||||
} else if (ast_test_flag(&cfg->flags, AST_SSL_TLSV1_CLIENT)) {
|
||||
cfg->ssl_ctx = SSL_CTX_new(TLSv1_client_method());
|
||||
} else {
|
||||
/* SSLv23_client_method() sends SSLv2, this was the original
|
||||
* default for ssl clients before the option was given to
|
||||
* pick what protocol a client should use. In order not
|
||||
* to break expected behavior it remains the default. */
|
||||
disable_ssl = 1;
|
||||
cfg->ssl_ctx = SSL_CTX_new(SSLv23_client_method());
|
||||
}
|
||||
} else {
|
||||
/* SSLv23_server_method() supports TLSv1, SSLv2, and SSLv3 inbound connections. */
|
||||
disable_ssl = 1;
|
||||
cfg->ssl_ctx = SSL_CTX_new(SSLv23_server_method());
|
||||
}
|
||||
|
||||
@@ -787,6 +788,17 @@ static int __ssl_setup(struct ast_tls_config *cfg, int client)
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Due to the POODLE vulnerability, completely disable
|
||||
* SSLv2 and SSLv3 if we are not explicitly told to use
|
||||
* them. SSLv23_*_method supports TLSv1+.
|
||||
*/
|
||||
if (disable_ssl) {
|
||||
long ssl_opts;
|
||||
|
||||
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
|
||||
SSL_CTX_set_options(cfg->ssl_ctx, ssl_opts);
|
||||
}
|
||||
|
||||
SSL_CTX_set_verify(cfg->ssl_ctx,
|
||||
ast_test_flag(&cfg->flags, AST_SSL_VERIFY_CLIENT) ? SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT : SSL_VERIFY_NONE,
|
||||
NULL);
|
||||
|
||||
@@ -1290,14 +1290,17 @@ static int aji_start_tls(struct aji_client *client)
|
||||
static int aji_tls_handshake(struct aji_client *client)
|
||||
{
|
||||
int sock;
|
||||
long ssl_opts;
|
||||
|
||||
ast_debug(1, "Starting TLS handshake\n");
|
||||
|
||||
/* Choose an SSL/TLS protocol version, create SSL_CTX */
|
||||
client->ssl_method = SSLv3_method();
|
||||
client->ssl_method = SSLv23_method();
|
||||
if (!(client->ssl_context = SSL_CTX_new((SSL_METHOD *) client->ssl_method))) {
|
||||
return IKS_NET_TLSFAIL;
|
||||
}
|
||||
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
|
||||
SSL_CTX_set_options(client->ssl_context, ssl_opts);
|
||||
|
||||
/* Create new SSL session */
|
||||
if (!(client->ssl_session = SSL_new(client->ssl_context))) {
|
||||
|
||||
@@ -2637,6 +2637,7 @@ static int xmpp_client_requested_tls(struct ast_xmpp_client *client, struct ast_
|
||||
{
|
||||
#ifdef HAVE_OPENSSL
|
||||
int sock;
|
||||
long ssl_opts;
|
||||
#endif
|
||||
|
||||
if (!strcmp(iks_name(node), "success")) {
|
||||
@@ -2655,11 +2656,14 @@ static int xmpp_client_requested_tls(struct ast_xmpp_client *client, struct ast_
|
||||
ast_log(LOG_ERROR, "Somehow we managed to try to start TLS negotiation on client '%s' without OpenSSL support, disconnecting\n", client->name);
|
||||
return -1;
|
||||
#else
|
||||
client->ssl_method = SSLv3_method();
|
||||
client->ssl_method = SSLv23_method();
|
||||
if (!(client->ssl_context = SSL_CTX_new((SSL_METHOD *) client->ssl_method))) {
|
||||
goto failure;
|
||||
}
|
||||
|
||||
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
|
||||
SSL_CTX_set_options(client->ssl_context, ssl_opts);
|
||||
|
||||
if (!(client->ssl_session = SSL_new(client->ssl_context))) {
|
||||
goto failure;
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user