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Asterisk Autobuilder
d838fa4400 Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.9.0-rc1@411555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28 18:30:13 +00:00
Asterisk Autobuilder
0b76a67c50 Importing release summary for 11.9.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.9.0-rc1@411554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28 18:30:00 +00:00
Asterisk Autobuilder
2875c77537 Importing files for 11.9.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.9.0-rc1@411553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28 18:29:47 +00:00
Asterisk Autobuilder
41b5c02392 Creating tag for the release of asterisk-11.9.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.9.0-rc1@411552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-11.9.0-rc1</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-11.9.0-rc1</h3>
<h3 align="center">Date: 2014-03-28</h3>
<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
<hr/>
<h2 align="center">Table of Contents</h2>
<ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#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-11.8.0.</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>
10 coreyfarrell<br/>
9 kmoore<br/>
8 mjordan<br/>
8 rmudgett<br/>
6 jrose<br/>
5 may<br/>
4 Eugene<br/>
4 jcolp<br/>
4 Jeremy Laine<br/>
4 newtonr<br/>
4 russell<br/>
4 wdoekes<br/>
3 mmichelson<br/>
3 moy<br/>
3 seanbright<br/>
3 tzafrir<br/>
2 dlee<br/>
2 elguero<br/>
2 igorg<br/>
2 kharwell<br/>
2 rmeyerriecks<br/>
2 sgriepentrog<br/>
1 Corey Farrell<br/>
1 Etienne Lessard<br/>
1 file<br/>
1 Guillaume Martres<br/>
1 ibercom<br/>
1 looserouting<br/>
1 marcelloceschia<br/>
1 Michal Rybarik<br/>
1 nbansal<br/>
1 rsw686<br/>
1 sharky<br/>
1 st<br/>
1 Steve Davies<br/>
1 Trevor Peirce<br/>
1 zvision<br/>
</td>
<td>
1 Alec Davis<br/>
1 Dmitry Melekhov<br/>
1 ibercom<br/>
1 Joel Vandal<br/>
1 Michal Rybarik<br/>
1 wushumasters<br/>
</td>
<td>
5 coreyfarrell<br/>
4 sharky<br/>
2 adomjan<br/>
2 rnewton<br/>
2 zvision<br/>
1 alecdavis<br/>
1 aragon<br/>
1 asemych<br/>
1 axonaro<br/>
1 bford<br/>
1 chillman<br/>
1 davidw<br/>
1 fabled<br/>
1 hexanol<br/>
1 ibercom<br/>
1 jamicque<br/>
1 jamuel<br/>
1 joel_vandal<br/>
1 jpsharp<br/>
1 jrose<br/>
1 jvandal<br/>
1 kgoedert<br/>
1 leonroy<br/>
1 looserouting<br/>
1 lordvadr<br/>
1 luke1980<br/>
1 manobela<br/>
1 marcelloceschia<br/>
1 mcargile<br/>
1 mjordan<br/>
1 n8ideas<br/>
1 nbansal<br/>
1 oleke<br/>
1 pz<br/>
1 rmeyerriecks<br/>
1 rsw686<br/>
1 sebmurray<br/>
1 sgriepentrog<br/>
1 shadow431<br/>
1 skycomltd<br/>
1 slesru<br/>
1 StuxForce<br/>
1 supermaxiko<br/>
1 thava<br/>
1 tilt<br/>
1 varnav<br/>
1 wdoekes<br/>
1 xytis<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: Addons/chan_ooh323</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23336">ASTERISK-23336</a>: Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408838">408838</a><br/>
Reporter: asemych<br/>
Coders: may<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23460">ASTERISK-23460</a>: ooh323 channel stuck if call is placed directly and gatekeeper is not available<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411531">411531</a><br/>
Reporter: slesru<br/>
Testers: Dmitry Melekhov<br/>
Coders: may<br/>
<br/>
<h3>Category: Applications/app_chanspy</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22661">ASTERISK-22661</a>: Unable to exit ChanSpy if spied channel does not have a call in progress<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408537">408537</a><br/>
Reporter: chillman<br/>
Coders: elguero<br/>
<br/>
<h3>Category: Applications/app_confbridge</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19499">ASTERISK-19499</a>: ConfBridge MOH is not working for transferee after attended transfer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408643">408643</a><br/>
Reporter: fabled<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23311">ASTERISK-23311</a>: Manager - MoH Stop Event fails to show up when leaving Conference<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410490">410490</a><br/>
Reporter: bford<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23461">ASTERISK-23461</a>: Only first user is muted when joining confbridge with 'startmuted=yes'<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410965">410965</a><br/>
Reporter: manobela<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Applications/app_dial</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23141">ASTERISK-23141</a>: Asterisk crashes on Dial(), in pbx_find_extension at pbx.c<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408786">408786</a><br/>
Reporter: supermaxiko<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Applications/app_forkcdr</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23260">ASTERISK-23260</a>: [patch]ForkCDR v option does not keep CDR variables for subsequent records<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408748">408748</a><br/>
Reporter: zvision<br/>
Coders: zvision<br/>
<br/>
<h3>Category: Applications/app_meetme</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19499">ASTERISK-19499</a>: ConfBridge MOH is not working for transferee after attended transfer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408643">408643</a><br/>
Reporter: fabled<br/>
Coders: kharwell<br/>
<br/>
<h3>Category: Applications/app_stack</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23220">ASTERISK-23220</a>: STACK_PEEK function with no arguments causes crash/core dump<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407103">407103</a><br/>
Reporter: jpsharp<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: CDR/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23046">ASTERISK-23046</a>: Custom CDR fields set during a GoSUB called from app_queue are not inserted<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: StuxForce<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23069">ASTERISK-23069</a>: Custom CDR variable not recorded when set in macro called from app_queue<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408448">408448</a><br/>
Reporter: shadow431<br/>
Coders: newtonr<br/>
<br/>
<h3>Category: CDR/cdr_custom</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23046">ASTERISK-23046</a>: Custom CDR fields set during a GoSUB called from app_queue are not inserted<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: StuxForce<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23069">ASTERISK-23069</a>: Custom CDR variable not recorded when set in macro called from app_queue<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408448">408448</a><br/>
Reporter: shadow431<br/>
Coders: newtonr<br/>
<br/>
<h3>Category: CDR/cdr_radius</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22980">ASTERISK-22980</a>: [patch]Allow building cdr_radius and cel_radius against libfreeradius-client<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406802">406802</a><br/>
Reporter: sharky<br/>
Coders: sharky<br/>
<br/>
<h3>Category: Channels/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23135">ASTERISK-23135</a>: Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409157">409157</a><br/>
Reporter: oleke<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23008">ASTERISK-23008</a>: Local channels loose CALLERID name when DAHDI channel connects<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405927">405927</a><br/>
Reporter: mcargile<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_local</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23008">ASTERISK-23008</a>: Local channels loose CALLERID name when DAHDI channel connects<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405927">405927</a><br/>
Reporter: mcargile<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23232">ASTERISK-23232</a>: LocalBridge AMI Event LocalOptimization value is opposite to what's expected<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407457">407457</a><br/>
Reporter: leonroy<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Channels/chan_mgcp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23100">ASTERISK-23100</a>: [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406038">406038</a><br/>
Reporter: adomjan<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: Channels/chan_sip/DatabaseSupport</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17523">ASTERISK-17523</a>: Qualify for static realtime peers does not work<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410106">410106</a><br/>
Reporter: jamicque<br/>
Testers: wushumasters<br/>
Coders: Trevor Peirce<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21406">ASTERISK-21406</a>: [patch] chan_sip deadlock on monlock between unload_module and do_monitor<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410225">410225</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23061">ASTERISK-23061</a>: [Patch] 'textsupport' setting not mentioned in sip.conf.sample<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: varnav<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23135">ASTERISK-23135</a>: Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409157">409157</a><br/>
Reporter: oleke<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23323">ASTERISK-23323</a>: [patch]chan_sip: missing p->owner checks in handle_response_invite<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409255">409255</a><br/>
Reporter: wdoekes<br/>
Coders: wdoekes<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23373">ASTERISK-23373</a>: [patch]Security: Open FD exhaustion with chan_sip Session-Timers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410311">410311</a><br/>
Reporter: coreyfarrell<br/>
Coders: Corey Farrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23406">ASTERISK-23406</a>: [patch]Fix typo in "sip show peer"<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409473">409473</a><br/>
Reporter: ibercom<br/>
Testers: ibercom<br/>
Coders: ibercom<br/>
<br/>
<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20841">ASTERISK-20841</a>: fromdomain not honored on outbound INVITE request<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411022">411022</a><br/>
Reporter: kgoedert<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Channels/chan_sip/T.38</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22988">ASTERISK-22988</a>: [patch]T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406171">406171</a><br/>
Reporter: adomjan<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: Channels/chan_sip/TCP-TLS</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17727">ASTERISK-17727</a>: [patch] TLS doesn't get all certificate chain<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407273">407273</a><br/>
Reporter: luke1980<br/>
Coders: st, Guillaume Martres<br/>
<br/>
<h3>Category: Channels/chan_sip/Transfers</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19499">ASTERISK-19499</a>: ConfBridge MOH is not working for transferee after attended transfer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408643">408643</a><br/>
Reporter: fabled<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21737">ASTERISK-21737</a>: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP in ast_format_cap_append called from remote bridge loop<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409002">409002</a><br/>
Reporter: alecdavis<br/>
Testers: Alec Davis<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Contrib/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23255">ASTERISK-23255</a>: UUID included for Redhat, but missing for Debian distros in install_prereq script<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408733">408733</a><br/>
Reporter: rnewton<br/>
Coders: kharwell<br/>
<br/>
<h3>Category: Core/Bridging</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21737">ASTERISK-21737</a>: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP in ast_format_cap_append called from remote bridge loop<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409002">409002</a><br/>
Reporter: alecdavis<br/>
Testers: Alec Davis<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Core/CallerID</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23488">ASTERISK-23488</a>: Logic error in callerid checksum processing<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410717">410717</a><br/>
Reporter: rmeyerriecks<br/>
Coders: rmeyerriecks<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23383">ASTERISK-23383</a>: Wrong sense test on stat return code causes unchanged config check to break with include files.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409917">409917</a><br/>
Reporter: davidw<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: Core/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17837">ASTERISK-17837</a>: extconfig.conf - Maximum Include level (1) exceeded<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406644">406644</a><br/>
Reporter: pz<br/>
Coders: russell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19773">ASTERISK-19773</a>: Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407210">407210</a><br/>
Reporter: joel_vandal<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22079">ASTERISK-22079</a>: Segfault: INTERNAL_OBJ (user_data=0x6374652f) at astobj2.c:120<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411089">411089</a><br/>
Reporter: jamuel<br/>
Coders: Steve Davies<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23098">ASTERISK-23098</a>: [patch]possible null pointer dereference in format.c<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408137">408137</a><br/>
Reporter: marcelloceschia<br/>
Coders: marcelloceschia, coreyfarrell<br/>
<br/>
<h3>Category: Core/HTTP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23340">ASTERISK-23340</a>: Security Vulnerability: stack allocation of cookie headers in loop allows for unauthenticated remote denial of service attack<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410381">410381</a><br/>
Reporter: mjordan<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Core/Internationalization</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23509">ASTERISK-23509</a>: [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411244">411244</a><br/>
Reporter: zvision<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/ManagerInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23034">ASTERISK-23034</a>: [patch] manager Originate doesn't abort on failed format_cap allocation<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405745">405745</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23104">ASTERISK-23104</a>: Specifying the SetVar AMI without a Channel cause Asterisk to crash<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409625">409625</a><br/>
Reporter: jvandal<br/>
Testers: Joel Vandal<br/>
Coders: elguero<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23420">ASTERISK-23420</a>: [patch]Memory leak in manager_add_filter function in manager.c<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410609">410609</a><br/>
Reporter: hexanol<br/>
Coders: Etienne Lessard<br/>
<br/>
<h3>Category: Core/RTP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23261">ASTERISK-23261</a>: [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408647">408647</a><br/>
Reporter: rsw686<br/>
Coders: rsw686<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23279">ASTERISK-23279</a>: [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408729">408729</a><br/>
Reporter: nbansal<br/>
Coders: nbansal<br/>
<br/>
<h3>Category: Documentation</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22662">ASTERISK-22662</a>: Documentation fix? - queues.conf says persistentmembers defaults to yes, it appears to lie<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406861">406861</a><br/>
Reporter: rnewton<br/>
Coders: russell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23027">ASTERISK-23027</a>: [patch] Spelling typo "transfered" instead of "transferred"<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23028">ASTERISK-23028</a>: [patch] Asterisk man pages contains unquoted minus signs<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23061">ASTERISK-23061</a>: [Patch] 'textsupport' setting not mentioned in sip.conf.sample<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405792">405792</a><br/>
Reporter: varnav<br/>
Coders: Jeremy Laine, Eugene<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23178">ASTERISK-23178</a>: devicestate.h: device state setting functions are documented with the wrong return values<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407338">407338</a><br/>
Reporter: jrose<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Formats/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23103">ASTERISK-23103</a>: [patch]Crash in ast_format_cmp, in ao2_find<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411310">411310</a><br/>
Reporter: n8ideas<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Functions/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23391">ASTERISK-23391</a>: Audit dialplan function usage of channel variable<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411314">411314</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Functions/func_channel</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23261">ASTERISK-23261</a>: [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408647">408647</a><br/>
Reporter: rsw686<br/>
Coders: rsw686<br/>
<br/>
<h3>Category: PBX/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22861">ASTERISK-22861</a>: [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406245">406245</a><br/>
Reporter: sebmurray<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: PBX/pbx_config</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23297">ASTERISK-23297</a>: Asterisk 12, pbx_config.so segfaults if res_parking.so is not loaded, or if res_parking.conf has no configuration<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408201">408201</a><br/>
Reporter: lordvadr<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_clialiases</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22757">ASTERISK-22757</a>: segfault in res_clialiases.so on reload when mapping "module reload" command<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407210">407210</a><br/>
Reporter: skycomltd<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_fax</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20149">ASTERISK-20149</a>: Crash when faxing SIP to SIP with strictrtp set to yes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409990">409990</a><br/>
Reporter: axonaro<br/>
Testers: Michal Rybarik<br/>
Coders: Michal Rybarik<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22790">ASTERISK-22790</a>: check_modem_rate() may return incorrect rate for V.27<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=405693">405693</a><br/>
Reporter: looserouting<br/>
Coders: looserouting<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23231">ASTERISK-23231</a>: Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409053">409053</a><br/>
Reporter: aragon<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Resources/res_fax_spandsp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20149">ASTERISK-20149</a>: Crash when faxing SIP to SIP with strictrtp set to yes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409990">409990</a><br/>
Reporter: axonaro<br/>
Testers: Michal Rybarik<br/>
Coders: Michal Rybarik<br/>
<br/>
<h3>Category: Resources/res_http_websocket</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21930">ASTERISK-21930</a>: [patch]WebRTC over WSS is not working.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409681">409681</a><br/>
Reporter: tilt<br/>
Coders: moy<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23099">ASTERISK-23099</a>: [patch] WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409681">409681</a><br/>
Reporter: thava<br/>
Coders: moy<br/>
<br/>
<h3>Category: Resources/res_musiconhold</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19499">ASTERISK-19499</a>: ConfBridge MOH is not working for transferee after attended transfer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408643">408643</a><br/>
Reporter: fabled<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23311">ASTERISK-23311</a>: Manager - MoH Stop Event fails to show up when leaving Conference<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410490">410490</a><br/>
Reporter: bford<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Resources/res_parking</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23297">ASTERISK-23297</a>: Asterisk 12, pbx_config.so segfaults if res_parking.so is not loaded, or if res_parking.conf has no configuration<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408201">408201</a><br/>
Reporter: lordvadr<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22911">ASTERISK-22911</a>: [patch]Asterisk fails to resume WebRTC call from hold<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409565">409565</a><br/>
Reporter: xytis<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Resources/res_rtp_asterisk</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22911">ASTERISK-22911</a>: [patch]Asterisk fails to resume WebRTC call from hold<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409565">409565</a><br/>
Reporter: xytis<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23134">ASTERISK-23134</a>: [patch] res_rtp_asterisk port selection cannot handle selinux port restrictions<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406934">406934</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23310">ASTERISK-23310</a>: bridged channel crashes in bridge_p2p_rtp_write<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409524">409524</a><br/>
Reporter: sharky<br/>
Coders: kmoore<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/11?view=revision&revision=405582">405582</a></td><td>file</td><td>cel_manager: Don't crash if configuration file is invalid.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406080">406080</a></td><td>wdoekes</td><td>manager: Clarify eventfilter documentation. Textual changes only.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406217">406217</a></td><td>kmoore</td><td>ConfBridge: Fix channel parameter documentation</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406361">406361</a></td><td>jrose</td><td>res_config_pgsql: Fix a memory leak and use RAII_VAR for cleanup when practical</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406400">406400</a></td><td>rmudgett</td><td>manager: Register atexit shutdown routine only once.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406515">406515</a></td><td>rmudgett</td><td>tcptls.c: Add missing cleanup on off nominal path.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406567">406567</a></td><td>russell</td><td>Protect ast_filestream object when on a channel</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=406918">406918</a></td><td>seanbright</td><td>Make a NOTICE about an invalid channel name more useful.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407074">407074</a></td><td>mjordan</td><td>app_dial: Allow macro/gosub pre-bridge execution to occur on priorities</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23164">ASTERISK-23164</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407456">407456</a></td><td>kmoore</td><td>Logger: Fix handling of absolute paths</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407512">407512</a></td><td>newtonr</td><td>formats/format_wav: enhancing log message "Not a wav file" to be clear on what is supported</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407623">407623</a></td><td>tzafrir</td><td>indications.conf: add stutter tone; end properly</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407765">407765</a></td><td>rmudgett</td><td>chan_iax2: Add some more iaxs[] NULL checks to a routine already full of them.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407818">407818</a></td><td>tzafrir</td><td>chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407857">407857</a></td><td>kmoore</td><td>ConfBridge: Correct prompt playback target</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=407874">407874</a></td><td>wdoekes</td><td>res_config_pgsql: Fix ast_update2_realtime calls.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408021">408021</a></td><td>newtonr</td><td>configs/agents.conf.sample - Remove example for non-functional "goodbye" parameter</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408084">408084</a></td><td>wdoekes</td><td>buildsystem: Don't force main to depend on everything else.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408143">408143</a></td><td>sgriepentrog</td><td>pbx: ast_custom_function_unregister resource leak</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408193">408193</a></td><td>mjordan</td><td>buildsystem: Unbreak the build (infloop) on Asterisk 11+</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408312">408312</a></td><td>may</td><td>Allow different socket and signalling ip on h.323 connection if gk mode is active</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408330">408330</a></td><td>may</td><td>process receiveAndTransmit user input remote caps instead of receive only</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408590">408590</a></td><td>may</td><td>Fix type of roundTripDelay variables</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=408877">408877</a></td><td>newtonr</td><td>configs/voicemail.conf.sample - Make mailcmd sample text more explicit</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409083">409083</a></td><td>dlee</td><td>Fix memory stomping bug in astman.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409129">409129</a></td><td>jrose</td><td>res_rtp_asterisk: Fix checklist creating problems in ICE sessions</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23213">ASTERISK-23213</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409130">409130</a></td><td>jrose</td><td>res_rtp_asterisk: correct build error from r409129</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23213">ASTERISK-23213</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409208">409208</a></td><td>kmoore</td><td>app_queue: Fix documentation generation</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409344">409344</a></td><td>tzafrir</td><td>Makefile: replace -O6 with -O3</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409362">409362</a></td><td>mjordan</td><td>doxygen: Tweak the link back to ye olde Digium website</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409567">409567</a></td><td>kmoore</td><td>AO2: Add an assert for bad objects</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409703">409703</a></td><td>moy</td><td>Fix res/res_http_websocket.c build failure in 32bit due to incorrect print format for uint64_t</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409745">409745</a></td><td>igorg</td><td></td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409761">409761</a></td><td>igorg</td><td>Correct RTP handling in chan_unistim and fix transfer process broken in previous fix:</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409778">409778</a></td><td>seanbright</td><td>Fix references to 'keys' CLI commands in astgenkey</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409834">409834</a></td><td>dlee</td><td>Corrected cross-platform stat nanosecond code</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=409886">409886</a></td><td>mmichelson</td><td>Fix documentation for PRESENCE_STATE to properly illustrate how to create a presence hint.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410044">410044</a></td><td>russell</td><td>moh: fix a refcount error with realtime MOH</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410556">410556</a></td><td>mmichelson</td><td>Prevent delayed astdb syncs.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410606">410606</a></td><td>mmichelson</td><td>Remove an extra ast_cond_wait() that slipped through the patch.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410749">410749</a></td><td>rmeyerriecks</td><td>!fixup: callerid: Logic error in checksum processing</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=410829">410829</a></td><td>seanbright</td><td>res_fax_spandsp: Use g711_free() when available.</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20149">ASTERISK-20149</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411373">411373</a></td><td>mjordan</td><td>chan_sip: Add MESSAGE request to allowed methods</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411408">411408</a></td><td>mjordan</td><td>res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values.</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23459">ASTERISK-23459</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=411458">411458</a></td><td>mjordan</td><td>UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems.</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>
Makefile | 6
Makefile.rules | 2
UPGRADE.txt | 30 +
addons/chan_ooh323.c | 7
addons/ooh323c/src/ooCalls.h | 2
addons/ooh323c/src/ooCapability.c | 16
addons/ooh323c/src/ooCmdChannel.c | 10
addons/ooh323c/src/ooGkClient.c | 16
addons/ooh323c/src/oochannels.c | 3
addons/ooh323c/src/ooh245.c | 8
addons/ooh323c/src/ooh323.c | 12
addons/ooh323c/src/ooq931.c | 2
apps/app_chanspy.c | 8
apps/app_confbridge.c | 51 +-
apps/app_dial.c | 10
apps/app_forkcdr.c | 10
apps/app_jack.c | 5
apps/app_queue.c | 37 +-
apps/app_speech_utils.c | 6
apps/app_stack.c | 21 +
apps/app_transfer.c | 2
apps/app_voicemail.c | 4
apps/confbridge/conf_config_parser.c | 15
apps/confbridge/conf_state.c | 3
apps/confbridge/conf_state_empty.c | 1
apps/confbridge/conf_state_inactive.c | 2
apps/confbridge/conf_state_multi_marked.c | 26 +
apps/confbridge/conf_state_single.c | 3
apps/confbridge/conf_state_single_marked.c | 3
apps/confbridge/include/confbridge.h | 9
cdr/cdr_radius.c | 8
cel/cel_manager.c | 7
cel/cel_radius.c | 8
channels/chan_dahdi.c | 97 +++++
channels/chan_iax2.c | 183 +++++++++-
channels/chan_local.c | 2
channels/chan_mgcp.c | 46 +-
channels/chan_sip.c | 514 ++++++++++++++++-------------
channels/chan_unistim.c | 76 +++-
channels/sig_pri.c | 39 +-
channels/sip/include/sip.h | 2
configs/agents.conf.sample | 5
configs/confbridge.conf.sample | 5
configs/iax.conf.sample | 20 -
configs/indications.conf.sample | 12
configs/manager.conf.sample | 13
configs/queues.conf.sample | 2
configs/res_fax.conf.sample | 10
configs/res_odbc.conf.sample | 8
configs/sip.conf.sample | 12
configs/voicemail.conf.sample | 3
configure.ac | 15
contrib/scripts/astgenkey | 2
contrib/scripts/astgenkey.8 | 2
contrib/scripts/install_prereq | 2
doc/asterisk.8 | 100 ++---
formats/format_wav.c | 2
funcs/func_audiohookinherit.c | 5
funcs/func_blacklist.c | 5
funcs/func_callcompletion.c | 10
funcs/func_callerid.c | 10
funcs/func_channel.c | 28 +
funcs/func_dialplan.c | 12
funcs/func_frame_trace.c | 5
funcs/func_global.c | 6
funcs/func_groupcount.c | 19 -
funcs/func_jitterbuffer.c | 5
funcs/func_math.c | 5
funcs/func_odbc.c | 5
funcs/func_pitchshift.c | 5
funcs/func_presencestate.c | 2
funcs/func_speex.c | 5
funcs/func_strings.c | 15
funcs/func_volume.c | 9
include/asterisk.h | 16
include/asterisk/autoconfig.h.in | 12
include/asterisk/channel.h | 5
include/asterisk/devicestate.h | 1
include/asterisk/frame.h | 26 +
include/asterisk/res_odbc.h | 6
include/asterisk/rtp_engine.h | 4
main/asterisk.c | 25 +
main/astobj2.c | 1
main/callerid.c | 2
main/cel.c | 273 +++++++++++----
main/channel.c | 33 +
main/config.c | 90 ++++-
main/db.c | 23 -
main/devicestate.c | 2
main/features.c | 30 +
main/file.c | 2
main/format.c | 6
main/http.c | 56 +--
main/logger.c | 19 -
main/manager.c | 412 +++++++++++++----------
main/message.c | 20 +
main/pbx.c | 19 -
main/rtp_engine.c | 36 +-
main/say.c | 12
main/tcptls.c | 14
main/udptl.c | 2
res/ael/pval.c | 1
res/res_calendar.c | 5
res/res_clialiases.c | 36 +-
res/res_config_odbc.c | 50 +-
res/res_config_pgsql.c | 43 --
res/res_fax.c | 17
res/res_fax_spandsp.c | 30 +
res/res_http_websocket.c | 211 +++++++----
res/res_jabber.c | 4
res/res_musiconhold.c | 8
res/res_mutestream.c | 5
res/res_odbc.c | 12
res/res_odbc.exports.in | 1
res/res_pktccops.c | 2
res/res_rtp_asterisk.c | 257 ++++++++++----
res/res_xmpp.c | 4
utils/astman.c | 1
utils/conf2ael.c | 2
utils/extconf.c | 41 --
120 files changed, 2460 insertions(+), 1093 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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@@ -0,0 +1,902 @@
Release Summary
asterisk-11.9.0-rc1
Date: 2014-03-28
<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-11.8.0.
----------------------------------------------------------------------
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
10 coreyfarrell 1 Alec Davis 5 coreyfarrell
9 kmoore 1 Dmitry Melekhov 4 sharky
8 mjordan 1 ibercom 2 adomjan
8 rmudgett 1 Joel Vandal 2 rnewton
6 jrose 1 Michal Rybarik 2 zvision
5 may 1 wushumasters 1 alecdavis
4 Eugene 1 aragon
4 jcolp 1 asemych
4 Jeremy Laine 1 axonaro
4 newtonr 1 bford
4 russell 1 chillman
4 wdoekes 1 davidw
3 mmichelson 1 fabled
3 moy 1 hexanol
3 seanbright 1 ibercom
3 tzafrir 1 jamicque
2 dlee 1 jamuel
2 elguero 1 joel_vandal
2 igorg 1 jpsharp
2 kharwell 1 jrose
2 rmeyerriecks 1 jvandal
2 sgriepentrog 1 kgoedert
1 Corey Farrell 1 leonroy
1 Etienne Lessard 1 looserouting
1 file 1 lordvadr
1 Guillaume Martres 1 luke1980
1 ibercom 1 manobela
1 looserouting 1 marcelloceschia
1 marcelloceschia 1 mcargile
1 Michal Rybarik 1 mjordan
1 nbansal 1 n8ideas
1 rsw686 1 nbansal
1 sharky 1 oleke
1 st 1 pz
1 Steve Davies 1 rmeyerriecks
1 Trevor Peirce 1 rsw686
1 zvision 1 sebmurray
1 sgriepentrog
1 shadow431
1 skycomltd
1 slesru
1 StuxForce
1 supermaxiko
1 thava
1 tilt
1 varnav
1 wdoekes
1 xytis
----------------------------------------------------------------------
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: Addons/chan_ooh323
ASTERISK-23336: Asterisk warning "Don't know how to indicate condition 33
on ooh323c" on outgoing calls from H323 to SIP peer
Revision: 408838
Reporter: asemych
Coders: may
ASTERISK-23460: ooh323 channel stuck if call is placed directly and
gatekeeper is not available
Revision: 411531
Reporter: slesru
Testers: Dmitry Melekhov
Coders: may
Category: Applications/app_chanspy
ASTERISK-22661: Unable to exit ChanSpy if spied channel does not have a
call in progress
Revision: 408537
Reporter: chillman
Coders: elguero
Category: Applications/app_confbridge
ASTERISK-19499: ConfBridge MOH is not working for transferee after
attended transfer
Revision: 408643
Reporter: fabled
Coders: kharwell
ASTERISK-23311: Manager - MoH Stop Event fails to show up when leaving
Conference
Revision: 410490
Reporter: bford
Coders: rmudgett
ASTERISK-23461: Only first user is muted when joining confbridge with
'startmuted=yes'
Revision: 410965
Reporter: manobela
Coders: jrose
Category: Applications/app_dial
ASTERISK-23141: Asterisk crashes on Dial(), in pbx_find_extension at pbx.c
Revision: 408786
Reporter: supermaxiko
Coders: coreyfarrell
Category: Applications/app_forkcdr
ASTERISK-23260: [patch]ForkCDR v option does not keep CDR variables for
subsequent records
Revision: 408748
Reporter: zvision
Coders: zvision
Category: Applications/app_meetme
ASTERISK-19499: ConfBridge MOH is not working for transferee after
attended transfer
Revision: 408643
Reporter: fabled
Coders: kharwell
Category: Applications/app_stack
ASTERISK-23220: STACK_PEEK function with no arguments causes crash/core
dump
Revision: 407103
Reporter: jpsharp
Coders: coreyfarrell
Category: CDR/General
ASTERISK-23046: Custom CDR fields set during a GoSUB called from app_queue
are not inserted
Revision: 405792
Reporter: StuxForce
Coders: Jeremy Laine, Eugene
ASTERISK-23069: Custom CDR variable not recorded when set in macro called
from app_queue
Revision: 408448
Reporter: shadow431
Coders: newtonr
Category: CDR/cdr_custom
ASTERISK-23046: Custom CDR fields set during a GoSUB called from app_queue
are not inserted
Revision: 405792
Reporter: StuxForce
Coders: Jeremy Laine, Eugene
ASTERISK-23069: Custom CDR variable not recorded when set in macro called
from app_queue
Revision: 408448
Reporter: shadow431
Coders: newtonr
Category: CDR/cdr_radius
ASTERISK-22980: [patch]Allow building cdr_radius and cel_radius against
libfreeradius-client
Revision: 406802
Reporter: sharky
Coders: sharky
Category: Channels/General
ASTERISK-23135: Crash - segfault in ast_channel_hangupcause_set - probably
introduced in 11.7.0
Revision: 409157
Reporter: oleke
Coders: rmudgett
Category: Channels/chan_dahdi
ASTERISK-23008: Local channels loose CALLERID name when DAHDI channel
connects
Revision: 405927
Reporter: mcargile
Coders: rmudgett
Category: Channels/chan_local
ASTERISK-23008: Local channels loose CALLERID name when DAHDI channel
connects
Revision: 405927
Reporter: mcargile
Coders: rmudgett
ASTERISK-23232: LocalBridge AMI Event LocalOptimization value is opposite
to what's expected
Revision: 407457
Reporter: leonroy
Coders: jrose
Category: Channels/chan_mgcp
ASTERISK-23100: [patch] In chan_mgcp the ident in transmitted request and
request queue may differ - fix for locking
Revision: 406038
Reporter: adomjan
Coders: kmoore
Category: Channels/chan_sip/DatabaseSupport
ASTERISK-17523: Qualify for static realtime peers does not work
Revision: 410106
Reporter: jamicque
Testers: wushumasters
Coders: Trevor Peirce
Category: Channels/chan_sip/General
ASTERISK-21406: [patch] chan_sip deadlock on monlock between unload_module
and do_monitor
Revision: 410225
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-23061: [Patch] 'textsupport' setting not mentioned in
sip.conf.sample
Revision: 405792
Reporter: varnav
Coders: Jeremy Laine, Eugene
ASTERISK-23135: Crash - segfault in ast_channel_hangupcause_set - probably
introduced in 11.7.0
Revision: 409157
Reporter: oleke
Coders: rmudgett
ASTERISK-23323: [patch]chan_sip: missing p->owner checks in
handle_response_invite
Revision: 409255
Reporter: wdoekes
Coders: wdoekes
ASTERISK-23373: [patch]Security: Open FD exhaustion with chan_sip
Session-Timers
Revision: 410311
Reporter: coreyfarrell
Coders: Corey Farrell
ASTERISK-23406: [patch]Fix typo in "sip show peer"
Revision: 409473
Reporter: ibercom
Testers: ibercom
Coders: ibercom
Category: Channels/chan_sip/Interoperability
ASTERISK-20841: fromdomain not honored on outbound INVITE request
Revision: 411022
Reporter: kgoedert
Coders: jcolp
Category: Channels/chan_sip/T.38
ASTERISK-22988: [patch]T38 , SIP 488 after Rejecting image media offer due
to invalid or unsupported syntax
Revision: 406171
Reporter: adomjan
Coders: kmoore
Category: Channels/chan_sip/TCP-TLS
ASTERISK-17727: [patch] TLS doesn't get all certificate chain
Revision: 407273
Reporter: luke1980
Coders: st, Guillaume Martres
Category: Channels/chan_sip/Transfers
ASTERISK-19499: ConfBridge MOH is not working for transferee after
attended transfer
Revision: 408643
Reporter: fabled
Coders: kharwell
ASTERISK-21737: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP
in ast_format_cap_append called from remote bridge loop
Revision: 409002
Reporter: alecdavis
Testers: Alec Davis
Coders: mjordan
Category: Contrib/General
ASTERISK-23255: UUID included for Redhat, but missing for Debian distros
in install_prereq script
Revision: 408733
Reporter: rnewton
Coders: kharwell
Category: Core/Bridging
ASTERISK-21737: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP
in ast_format_cap_append called from remote bridge loop
Revision: 409002
Reporter: alecdavis
Testers: Alec Davis
Coders: mjordan
Category: Core/CallerID
ASTERISK-23488: Logic error in callerid checksum processing
Revision: 410717
Reporter: rmeyerriecks
Coders: rmeyerriecks
Category: Core/Configuration
ASTERISK-23383: Wrong sense test on stat return code causes unchanged
config check to break with include files.
Revision: 409917
Reporter: davidw
Coders: kmoore
Category: Core/General
ASTERISK-17837: extconfig.conf - Maximum Include level (1) exceeded
Revision: 406644
Reporter: pz
Coders: russell
ASTERISK-19773: Asterisk crash on issuing Asterisk-CLI 'reload' command
multiple times on cli_aliases
Revision: 407210
Reporter: joel_vandal
Coders: jcolp
ASTERISK-22079: Segfault: INTERNAL_OBJ (user_data=0x6374652f) at
astobj2.c:120
Revision: 411089
Reporter: jamuel
Coders: Steve Davies
ASTERISK-23098: [patch]possible null pointer dereference in format.c
Revision: 408137
Reporter: marcelloceschia
Coders: marcelloceschia, coreyfarrell
Category: Core/HTTP
ASTERISK-23340: Security Vulnerability: stack allocation of cookie headers
in loop allows for unauthenticated remote denial of service attack
Revision: 410381
Reporter: mjordan
Coders: rmudgett
Category: Core/Internationalization
ASTERISK-23509: [patch]SayNumber for Polish language tries to play empty
files for numbers divisible by 100
Revision: 411244
Reporter: zvision
Coders: jcolp
Category: Core/ManagerInterface
ASTERISK-23034: [patch] manager Originate doesn't abort on failed
format_cap allocation
Revision: 405745
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-23104: Specifying the SetVar AMI without a Channel cause Asterisk
to crash
Revision: 409625
Reporter: jvandal
Testers: Joel Vandal
Coders: elguero
ASTERISK-23420: [patch]Memory leak in manager_add_filter function in
manager.c
Revision: 410609
Reporter: hexanol
Coders: Etienne Lessard
Category: Core/RTP
ASTERISK-23261: [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}
Revision: 408647
Reporter: rsw686
Coders: rsw686
ASTERISK-23279: [patch]Asterisk doesn't support the dynamic payload change
in rtp mapping in the 200 OK response
Revision: 408729
Reporter: nbansal
Coders: nbansal
Category: Documentation
ASTERISK-22662: Documentation fix? - queues.conf says persistentmembers
defaults to yes, it appears to lie
Revision: 406861
Reporter: rnewton
Coders: russell
ASTERISK-23027: [patch] Spelling typo "transfered" instead of
"transferred"
Revision: 405792
Reporter: sharky
Coders: Jeremy Laine, Eugene
ASTERISK-23028: [patch] Asterisk man pages contains unquoted minus signs
Revision: 405792
Reporter: sharky
Coders: Jeremy Laine, Eugene
ASTERISK-23061: [Patch] 'textsupport' setting not mentioned in
sip.conf.sample
Revision: 405792
Reporter: varnav
Coders: Jeremy Laine, Eugene
ASTERISK-23178: devicestate.h: device state setting functions are
documented with the wrong return values
Revision: 407338
Reporter: jrose
Coders: rmudgett
Category: Formats/General
ASTERISK-23103: [patch]Crash in ast_format_cmp, in ao2_find
Revision: 411310
Reporter: n8ideas
Coders: coreyfarrell
Category: Functions/General
ASTERISK-23391: Audit dialplan function usage of channel variable
Revision: 411314
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Functions/func_channel
ASTERISK-23261: [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}
Revision: 408647
Reporter: rsw686
Coders: rsw686
Category: PBX/General
ASTERISK-22861: [patch]Specifying a null time as parameter to GotoIfTime
or ExecIfTime causes segmentation fault
Revision: 406245
Reporter: sebmurray
Coders: coreyfarrell
Category: PBX/pbx_config
ASTERISK-23297: Asterisk 12, pbx_config.so segfaults if res_parking.so is
not loaded, or if res_parking.conf has no configuration
Revision: 408201
Reporter: lordvadr
Coders: mjordan
Category: Resources/res_clialiases
ASTERISK-22757: segfault in res_clialiases.so on reload when mapping
"module reload" command
Revision: 407210
Reporter: skycomltd
Coders: jcolp
Category: Resources/res_fax
ASTERISK-20149: Crash when faxing SIP to SIP with strictrtp set to yes
Revision: 409990
Reporter: axonaro
Testers: Michal Rybarik
Coders: Michal Rybarik
ASTERISK-22790: check_modem_rate() may return incorrect rate for V.27
Revision: 405693
Reporter: looserouting
Coders: looserouting
ASTERISK-23231: Since 405693 If we have res_fax.conf file set to
minrate=2400, then res_fax refuse to load
Revision: 409053
Reporter: aragon
Coders: coreyfarrell
Category: Resources/res_fax_spandsp
ASTERISK-20149: Crash when faxing SIP to SIP with strictrtp set to yes
Revision: 409990
Reporter: axonaro
Testers: Michal Rybarik
Coders: Michal Rybarik
Category: Resources/res_http_websocket
ASTERISK-21930: [patch]WebRTC over WSS is not working.
Revision: 409681
Reporter: tilt
Coders: moy
ASTERISK-23099: [patch] WSS: enable ast_websocket_read() function to read
the whole available data at first and then wait for any fragmented packets
Revision: 409681
Reporter: thava
Coders: moy
Category: Resources/res_musiconhold
ASTERISK-19499: ConfBridge MOH is not working for transferee after
attended transfer
Revision: 408643
Reporter: fabled
Coders: kharwell
ASTERISK-23311: Manager - MoH Stop Event fails to show up when leaving
Conference
Revision: 410490
Reporter: bford
Coders: rmudgett
Category: Resources/res_parking
ASTERISK-23297: Asterisk 12, pbx_config.so segfaults if res_parking.so is
not loaded, or if res_parking.conf has no configuration
Revision: 408201
Reporter: lordvadr
Coders: mjordan
Category: Resources/res_pjsip
ASTERISK-22911: [patch]Asterisk fails to resume WebRTC call from hold
Revision: 409565
Reporter: xytis
Coders: jrose
Category: Resources/res_rtp_asterisk
ASTERISK-22911: [patch]Asterisk fails to resume WebRTC call from hold
Revision: 409565
Reporter: xytis
Coders: jrose
ASTERISK-23134: [patch] res_rtp_asterisk port selection cannot handle
selinux port restrictions
Revision: 406934
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-23310: bridged channel crashes in bridge_p2p_rtp_write
Revision: 409524
Reporter: sharky
Coders: kmoore
----------------------------------------------------------------------
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 |
|--------+------------+-----------------------------------+--------------|
|405582 |file |cel_manager: Don't crash if | |
| | |configuration file is invalid. | |
|--------+------------+-----------------------------------+--------------|
| | |manager: Clarify eventfilter | |
|406080 |wdoekes |documentation. Textual changes | |
| | |only. | |
|--------+------------+-----------------------------------+--------------|
|406217 |kmoore |ConfBridge: Fix channel parameter | |
| | |documentation | |
|--------+------------+-----------------------------------+--------------|
| | |res_config_pgsql: Fix a memory leak| |
|406361 |jrose |and use RAII_VAR for cleanup when | |
| | |practical | |
|--------+------------+-----------------------------------+--------------|
|406400 |rmudgett |manager: Register atexit shutdown | |
| | |routine only once. | |
|--------+------------+-----------------------------------+--------------|
|406515 |rmudgett |tcptls.c: Add missing cleanup on | |
| | |off nominal path. | |
|--------+------------+-----------------------------------+--------------|
|406567 |russell |Protect ast_filestream object when | |
| | |on a channel | |
|--------+------------+-----------------------------------+--------------|
|406918 |seanbright |Make a NOTICE about an invalid | |
| | |channel name more useful. | |
|--------+------------+-----------------------------------+--------------|
| | |app_dial: Allow macro/gosub | |
|407074 |mjordan |pre-bridge execution to occur on |ASTERISK-23164|
| | |priorities | |
|--------+------------+-----------------------------------+--------------|
|407456 |kmoore |Logger: Fix handling of absolute | |
| | |paths | |
|--------+------------+-----------------------------------+--------------|
| | |formats/format_wav: enhancing log | |
|407512 |newtonr |message "Not a wav file" to be | |
| | |clear on what is supported | |
|--------+------------+-----------------------------------+--------------|
|407623 |tzafrir |indications.conf: add stutter tone;| |
| | |end properly | |
|--------+------------+-----------------------------------+--------------|
| | |chan_iax2: Add some more iaxs[] | |
|407765 |rmudgett |NULL checks to a routine already | |
| | |full of them. | |
|--------+------------+-----------------------------------+--------------|
| | |chan_dahdi: handle | |
|407818 |tzafrir |DAHDI_EVENT_REMOVED on a pri | |
| | |D-Channel | |
|--------+------------+-----------------------------------+--------------|
|407857 |kmoore |ConfBridge: Correct prompt playback| |
| | |target | |
|--------+------------+-----------------------------------+--------------|
|407874 |wdoekes |res_config_pgsql: Fix | |
| | |ast_update2_realtime calls. | |
|--------+------------+-----------------------------------+--------------|
| | |configs/agents.conf.sample - Remove| |
|408021 |newtonr |example for non-functional | |
| | |"goodbye" parameter | |
|--------+------------+-----------------------------------+--------------|
|408084 |wdoekes |buildsystem: Don't force main to | |
| | |depend on everything else. | |
|--------+------------+-----------------------------------+--------------|
|408143 |sgriepentrog|pbx: ast_custom_function_unregister| |
| | |resource leak | |
|--------+------------+-----------------------------------+--------------|
|408193 |mjordan |buildsystem: Unbreak the build | |
| | |(infloop) on Asterisk 11+ | |
|--------+------------+-----------------------------------+--------------|
| | |Allow different socket and | |
|408312 |may |signalling ip on h.323 connection | |
| | |if gk mode is active | |
|--------+------------+-----------------------------------+--------------|
| | |process receiveAndTransmit user | |
|408330 |may |input remote caps instead of | |
| | |receive only | |
|--------+------------+-----------------------------------+--------------|
|408590 |may |Fix type of roundTripDelay | |
| | |variables | |
|--------+------------+-----------------------------------+--------------|
| | |configs/voicemail.conf.sample - | |
|408877 |newtonr |Make mailcmd sample text more | |
| | |explicit | |
|--------+------------+-----------------------------------+--------------|
|409083 |dlee |Fix memory stomping bug in astman. | |
|--------+------------+-----------------------------------+--------------|
|409129 |jrose |res_rtp_asterisk: Fix checklist |ASTERISK-23213|
| | |creating problems in ICE sessions | |
|--------+------------+-----------------------------------+--------------|
|409130 |jrose |res_rtp_asterisk: correct build |ASTERISK-23213|
| | |error from r409129 | |
|--------+------------+-----------------------------------+--------------|
|409208 |kmoore |app_queue: Fix documentation | |
| | |generation | |
|--------+------------+-----------------------------------+--------------|
|409344 |tzafrir |Makefile: replace -O6 with -O3 | |
|--------+------------+-----------------------------------+--------------|
|409362 |mjordan |doxygen: Tweak the link back to ye | |
| | |olde Digium website | |
|--------+------------+-----------------------------------+--------------|
|409567 |kmoore |AO2: Add an assert for bad objects | |
|--------+------------+-----------------------------------+--------------|
| | |Fix res/res_http_websocket.c build | |
|409703 |moy |failure in 32bit due to incorrect | |
| | |print format for uint64_t | |
|--------+------------+-----------------------------------+--------------|
|409745 |igorg | | |
|--------+------------+-----------------------------------+--------------|
| | |Correct RTP handling in | |
|409761 |igorg |chan_unistim and fix transfer | |
| | |process broken in previous fix: | |
|--------+------------+-----------------------------------+--------------|
|409778 |seanbright |Fix references to 'keys' CLI | |
| | |commands in astgenkey | |
|--------+------------+-----------------------------------+--------------|
|409834 |dlee |Corrected cross-platform stat | |
| | |nanosecond code | |
|--------+------------+-----------------------------------+--------------|
| | |Fix documentation for | |
|409886 |mmichelson |PRESENCE_STATE to properly | |
| | |illustrate how to create a presence| |
| | |hint. | |
|--------+------------+-----------------------------------+--------------|
|410044 |russell |moh: fix a refcount error with | |
| | |realtime MOH | |
|--------+------------+-----------------------------------+--------------|
|410556 |mmichelson |Prevent delayed astdb syncs. | |
|--------+------------+-----------------------------------+--------------|
|410606 |mmichelson |Remove an extra ast_cond_wait() | |
| | |that slipped through the patch. | |
|--------+------------+-----------------------------------+--------------|
|410749 |rmeyerriecks|!fixup: callerid: Logic error in | |
| | |checksum processing | |
|--------+------------+-----------------------------------+--------------|
|410829 |seanbright |res_fax_spandsp: Use g711_free() |ASTERISK-20149|
| | |when available. | |
|--------+------------+-----------------------------------+--------------|
|411373 |mjordan |chan_sip: Add MESSAGE request to | |
| | |allowed methods | |
|--------+------------+-----------------------------------+--------------|
| | |res_config_odbc/res_odbc: Fix | |
|411408 |mjordan |handling of non-text columns |ASTERISK-23459|
| | |updates with empty values. | |
|--------+------------+-----------------------------------+--------------|
|411458 |mjordan |UPGRADE: Note IAX2 compatibility | |
| | |issue between 1.4 and 1.8+ systems.| |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
Makefile | 6
Makefile.rules | 2
UPGRADE.txt | 30 +
addons/chan_ooh323.c | 7
addons/ooh323c/src/ooCalls.h | 2
addons/ooh323c/src/ooCapability.c | 16
addons/ooh323c/src/ooCmdChannel.c | 10
addons/ooh323c/src/ooGkClient.c | 16
addons/ooh323c/src/oochannels.c | 3
addons/ooh323c/src/ooh245.c | 8
addons/ooh323c/src/ooh323.c | 12
addons/ooh323c/src/ooq931.c | 2
apps/app_chanspy.c | 8
apps/app_confbridge.c | 51 +-
apps/app_dial.c | 10
apps/app_forkcdr.c | 10
apps/app_jack.c | 5
apps/app_queue.c | 37 +-
apps/app_speech_utils.c | 6
apps/app_stack.c | 21 +
apps/app_transfer.c | 2
apps/app_voicemail.c | 4
apps/confbridge/conf_config_parser.c | 15
apps/confbridge/conf_state.c | 3
apps/confbridge/conf_state_empty.c | 1
apps/confbridge/conf_state_inactive.c | 2
apps/confbridge/conf_state_multi_marked.c | 26 +
apps/confbridge/conf_state_single.c | 3
apps/confbridge/conf_state_single_marked.c | 3
apps/confbridge/include/confbridge.h | 9
cdr/cdr_radius.c | 8
cel/cel_manager.c | 7
cel/cel_radius.c | 8
channels/chan_dahdi.c | 97 +++++
channels/chan_iax2.c | 183 +++++++++-
channels/chan_local.c | 2
channels/chan_mgcp.c | 46 +-
channels/chan_sip.c | 514 ++++++++++++++++-------------
channels/chan_unistim.c | 76 +++-
channels/sig_pri.c | 39 +-
channels/sip/include/sip.h | 2
configs/agents.conf.sample | 5
configs/confbridge.conf.sample | 5
configs/iax.conf.sample | 20 -
configs/indications.conf.sample | 12
configs/manager.conf.sample | 13
configs/queues.conf.sample | 2
configs/res_fax.conf.sample | 10
configs/res_odbc.conf.sample | 8
configs/sip.conf.sample | 12
configs/voicemail.conf.sample | 3
configure.ac | 15
contrib/scripts/astgenkey | 2
contrib/scripts/astgenkey.8 | 2
contrib/scripts/install_prereq | 2
doc/asterisk.8 | 100 ++---
formats/format_wav.c | 2
funcs/func_audiohookinherit.c | 5
funcs/func_blacklist.c | 5
funcs/func_callcompletion.c | 10
funcs/func_callerid.c | 10
funcs/func_channel.c | 28 +
funcs/func_dialplan.c | 12
funcs/func_frame_trace.c | 5
funcs/func_global.c | 6
funcs/func_groupcount.c | 19 -
funcs/func_jitterbuffer.c | 5
funcs/func_math.c | 5
funcs/func_odbc.c | 5
funcs/func_pitchshift.c | 5
funcs/func_presencestate.c | 2
funcs/func_speex.c | 5
funcs/func_strings.c | 15
funcs/func_volume.c | 9
include/asterisk.h | 16
include/asterisk/autoconfig.h.in | 12
include/asterisk/channel.h | 5
include/asterisk/devicestate.h | 1
include/asterisk/frame.h | 26 +
include/asterisk/res_odbc.h | 6
include/asterisk/rtp_engine.h | 4
main/asterisk.c | 25 +
main/astobj2.c | 1
main/callerid.c | 2
main/cel.c | 273 +++++++++++----
main/channel.c | 33 +
main/config.c | 90 ++++-
main/db.c | 23 -
main/devicestate.c | 2
main/features.c | 30 +
main/file.c | 2
main/format.c | 6
main/http.c | 56 +--
main/logger.c | 19 -
main/manager.c | 412 +++++++++++++----------
main/message.c | 20 +
main/pbx.c | 19 -
main/rtp_engine.c | 36 +-
main/say.c | 12
main/tcptls.c | 14
main/udptl.c | 2
res/ael/pval.c | 1
res/res_calendar.c | 5
res/res_clialiases.c | 36 +-
res/res_config_odbc.c | 50 +-
res/res_config_pgsql.c | 43 --
res/res_fax.c | 17
res/res_fax_spandsp.c | 30 +
res/res_http_websocket.c | 211 +++++++----
res/res_jabber.c | 4
res/res_musiconhold.c | 8
res/res_mutestream.c | 5
res/res_odbc.c | 12
res/res_odbc.exports.in | 1
res/res_pktccops.c | 2
res/res_rtp_asterisk.c | 257 ++++++++++----
res/res_xmpp.c | 4
utils/astman.c | 1
utils/conf2ael.c | 2
utils/extconf.c | 41 --
120 files changed, 2460 insertions(+), 1093 deletions(-)
----------------------------------------------------------------------