[asterisk-commits] bebuild: tag 11.7.0-rc1 r402107 - /tags/11.7.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Oct 28 09:00:47 CDT 2013
Author: bebuild
Date: Mon Oct 28 09:00:45 2013
New Revision: 402107
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402107
Log:
Importing release summary for 11.7.0-rc1 release.
Added:
tags/11.7.0-rc1/asterisk-11.7.0-rc1-summary.html (with props)
tags/11.7.0-rc1/asterisk-11.7.0-rc1-summary.txt (with props)
Added: tags/11.7.0-rc1/asterisk-11.7.0-rc1-summary.html
URL: http://svnview.digium.com/svn/asterisk/tags/11.7.0-rc1/asterisk-11.7.0-rc1-summary.html?view=auto&rev=402107
==============================================================================
--- tags/11.7.0-rc1/asterisk-11.7.0-rc1-summary.html (added)
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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-11.7.0-rc1</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-11.7.0-rc1</h3>
+<h3 align="center">Date: 2013-10-28</h3>
+<h3 align="center"><asteriskteam at digium.com></h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+ <li><a href="#summary">Summary</a></li>
+ <li><a href="#contributors">Contributors</a></li>
+ <li><a href="#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-11.6.0.</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>
+11 jrose<br/>
+8 rmudgett<br/>
+5 elguero<br/>
+4 kmoore<br/>
+4 wdoekes<br/>
+3 mjordan<br/>
+2 kharwell<br/>
+2 NITESH BANSAL<br/>
+2 sgriepentrog<br/>
+1 abelbeck<br/>
+1 adomjan<br/>
+1 coreyfarrell<br/>
+1 Filip Jenicek<br/>
+1 junky<br/>
+1 mmichelson<br/>
+1 nbougues<br/>
+1 Oscar Esteve<br/>
+1 Simone Camporeale<br/>
+1 Torrey Searle<br/>
+1 tsearle<br/>
+1 tzafrir<br/>
+</td>
+<td>
+3 elguero<br/>
+1 Ben Smithurst<br/>
+1 Dalius M.<br/>
+1 rmudgett<br/>
+1 wdoekes<br/>
+</td>
+<td>
+2 gkelleter<br/>
+2 nbansal<br/>
+2 tsearle<br/>
+1 abelbeck<br/>
+1 adomjan<br/>
+1 aragon<br/>
+1 arcanos<br/>
+1 bensmithurst<br/>
+1 brianscott<br/>
+1 coreyfarrell<br/>
+1 dario<br/>
+1 docent<br/>
+1 elguero<br/>
+1 frenk77<br/>
+1 jbigelow<br/>
+1 jkister<br/>
+1 jllull<br/>
+1 jrose<br/>
+1 kharwell<br/>
+1 kmoore<br/>
+1 londonnet<br/>
+1 mdalius<br/>
+1 mjordan<br/>
+1 nbougues<br/>
+1 oscares<br/>
+1 patrickm<br/>
+1 phill<br/>
+1 rudolf<br/>
+1 simone.camporeale<br/>
+1 skavin<br/>
+1 tomaso<br/>
+1 wdoekes<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: Applications/app_confbridge</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19983">ASTERISK-19983</a>: ConfBridge does not expose a mechanism to change the language on the Bridging channel, defaulting to 'en'<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400741">400741</a><br/>
+Reporter: londonnet<br/>
+Testers: rmudgett<br/>
+Coders: junky<br/>
+<br/>
+<h3>Category: Applications/app_queue</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21826">ASTERISK-21826</a>: [patch] Bad queue_log entry when removed member from queue via CLI<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401433">401433</a><br/>
+Reporter: oscares<br/>
+Coders: Oscar Esteve<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22197">ASTERISK-22197</a>: [patch] Queuelog EXITWITHKEY only two of four parameters<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400623">400623</a><br/>
+Reporter: mdalius<br/>
+Testers: Dalius M.<br/>
+Coders: elguero<br/>
+<br/>
+<h3>Category: Applications/app_voicemail</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: Bridges/bridge_native_rtp</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22424">ASTERISK-22424</a>: bridge_native_rtp: Asterisk 12 attempts to remotely bridge on 200OK response to invite when the 200 lacks SDP<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399962">399962</a><br/>
+Reporter: jrose<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: CDR/cdr_adaptive_odbc</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22272">ASTERISK-22272</a>: [patch] Unexepected behaviour with adaptive odbc filter<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401579">401579</a><br/>
+Reporter: jllull<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Channels/chan_dahdi</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: Channels/chan_mgcp</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21190">ASTERISK-21190</a>: chan_mgcp crash on chunked m= sdp line<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401538">401538</a><br/>
+Reporter: adomjan<br/>
+Coders: adomjan<br/>
+<br/>
+<h3>Category: Channels/chan_pjsip</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22667">ASTERISK-22667</a>: crash: directmedia with both phones placing each other on hold<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401446">401446</a><br/>
+Reporter: jbigelow<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Channels/chan_sip/CodecHandling</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21131">ASTERISK-21131</a>: [patch] - Asterisk creates SDP with (peer) unsupported audio codec<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401498">401498</a><br/>
+Reporter: nbougues<br/>
+Coders: nbougues<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21204">ASTERISK-21204</a>: Asterisk increments the session version in 2xx message even if a '183 Session in Progress' with SDP has already been sent in response to initial INVITE.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400908">400908</a><br/>
+Reporter: nbansal<br/>
+Coders: NITESH BANSAL<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22477">ASTERISK-22477</a>: array calleridname in the function check_user_full is too short<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400014">400014</a><br/>
+Reporter: docent<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22621">ASTERISK-22621</a>: chan_sip can send two BYEs for a single call<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400971">400971</a><br/>
+Reporter: kmoore<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22667">ASTERISK-22667</a>: crash: directmedia with both phones placing each other on hold<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401446">401446</a><br/>
+Reporter: jbigelow<br/>
+Coders: mjordan<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22729">ASTERISK-22729</a>: [patch] Remove Port Restriction When Checking For NAT<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401182">401182</a><br/>
+Reporter: elguero<br/>
+Testers: elguero<br/>
+Coders: elguero<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22005">ASTERISK-22005</a>: Allow a sip peer to accept both AVP and AVPF calls<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401884">401884</a><br/>
+Reporter: tsearle<br/>
+Coders: tsearle<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17138">ASTERISK-17138</a>: [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication"<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400140">400140</a><br/>
+Reporter: rudolf<br/>
+Coders: kmoore<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22236">ASTERISK-22236</a>: REGISTER reply send to bad port with nat=yes(or force_rport,comedia) in 11.5.0<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401167">401167</a><br/>
+Reporter: frenk77<br/>
+Testers: elguero<br/>
+Coders: elguero<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22428">ASTERISK-22428</a>: [patch] SIP unregister does not fully unregister when using Realtime sip peers and Expires not 0 on 200ok<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399795">399795</a><br/>
+Reporter: bensmithurst<br/>
+Testers: Ben Smithurst, elguero<br/>
+Coders: elguero<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22574">ASTERISK-22574</a>: [patch]Value of expires= is ignored in the Contact header<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400470">400470</a><br/>
+Reporter: phill<br/>
+Coders: Filip Jenicek<br/>
+<br/>
+<h3>Category: Channels/chan_sip/T.38</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17273">ASTERISK-17273</a>: [patch] - Incorrect address specified in SIP re-INVITE with T.38 when directmedia enabled<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399457">399457</a><br/>
+Reporter: dario<br/>
+Coders: jrose<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18706">ASTERISK-18706</a>: UDPTL fail while using directmedia<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399457">399457</a><br/>
+Reporter: jkister<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Codecs/codec_ilbc</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: Core/BuildSystem</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22351">ASTERISK-22351</a>: Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev isn't installed<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401326">401326</a><br/>
+Reporter: arcanos<br/>
+Coders: wdoekes<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22351">ASTERISK-22351</a>: Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev isn't installed<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401326">401326</a><br/>
+Reporter: arcanos<br/>
+Coders: wdoekes<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: Core/Jitterbuffer</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: Core/Logging</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22456">ASTERISK-22456</a>: Logger.conf: Logging types ignored after specifying a verbose level<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401833">401833</a><br/>
+Reporter: kharwell<br/>
+Coders: kharwell<br/>
+<br/>
+<h3>Category: Core/ManagerInterface</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22578">ASTERISK-22578</a>: Invalid manager logins aren't reported via security events: Invalid IE Specified ERROR<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400421">400421</a><br/>
+Reporter: mjordan<br/>
+Coders: kmoore<br/>
+<br/>
+<h3>Category: Core/RTP</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21464">ASTERISK-21464</a>: with directrtpsetup some payload type identifiers from A party's INVITE are not copied to the INVITE for B party<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=402042">402042</a><br/>
+Reporter: skavin<br/>
+Coders: sgriepentrog<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22424">ASTERISK-22424</a>: bridge_native_rtp: Asterisk 12 attempts to remotely bridge on 200OK response to invite when the 200 lacks SDP<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399962">399962</a><br/>
+Reporter: jrose<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Core/UDPTL</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18706">ASTERISK-18706</a>: UDPTL fail while using directmedia<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399457">399457</a><br/>
+Reporter: jkister<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Functions/func_config</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22483">ASTERISK-22483</a>: AST_LIST_INSERT_TAIL doesn't set field.next on added entry<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400697">400697</a><br/>
+Reporter: brianscott<br/>
+Coders: kmoore<br/>
+<br/>
+<h3>Category: Functions/func_math</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<br/>
+<br/>
+<h3>Category: General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22540">ASTERISK-22540</a>: WARNING[2324] asterisk.c: Fork failed: Cannot allocate memory in /var/log/asterisk/messages then segfault<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399513">399513</a><br/>
+Reporter: aragon<br/>
+Coders: kharwell<br/>
+<br/>
+<h3>Category: Resources/res_jabber</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22410">ASTERISK-22410</a>: [patch] Change "Error isn't a PubSub error ..." error log to a debug log<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401120">401120</a><br/>
+Reporter: abelbeck<br/>
+Coders: abelbeck<br/>
+<br/>
+<h3>Category: Resources/res_odbc</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22459">ASTERISK-22459</a>: Compiling res_odbc against iODBC instead of unixodbc produces runtime errors<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400768">400768</a><br/>
+Reporter: patrickm<br/>
+Testers: wdoekes<br/>
+Coders: wdoekes<br/>
+<br/>
+<h3>Category: Resources/res_rtp_asterisk</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21170">ASTERISK-21170</a>: DTMF timestamp issue<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401620">401620</a><br/>
+Reporter: nbansal<br/>
+Coders: NITESH BANSAL<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21917">ASTERISK-21917</a>: [patch] STUN crashes when SIP is bound to ipv4 and ipv6<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400681">400681</a><br/>
+Reporter: tsearle<br/>
+Coders: Torrey Searle<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22667">ASTERISK-22667</a>: crash: directmedia with both phones placing each other on hold<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401446">401446</a><br/>
+Reporter: jbigelow<br/>
+Coders: mjordan<br/>
+<br/>
+<h3>Category: Resources/res_rtp_multicast</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22567">ASTERISK-22567</a>: [patch]MutlicastRTP does not set SSRC. SSRC is always set to 0<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400394">400394</a><br/>
+Reporter: simone.camporeale<br/>
+Coders: Simone Camporeale<br/>
+<br/>
+<h3>Category: Resources/res_xmpp</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22410">ASTERISK-22410</a>: [patch] Change "Error isn't a PubSub error ..." error log to a debug log<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401120">401120</a><br/>
+Reporter: abelbeck<br/>
+Coders: abelbeck<br/>
+<br/>
+<h3>Category: Tests/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a>: [patch] memory leaks 1.8+<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401830">401830</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: coreyfarrell<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=399564">399564</a></td><td>kmoore</td><td>Ensure global types in the config framework are initialized</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399708">399708</a></td><td>rmudgett</td><td>chan_iax2: Prevent some needless breaking of the native IAX2 bridge.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=399834">399834</a></td><td>rmudgett</td><td>chan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400279">400279</a></td><td>tzafrir</td><td>man pages for astdb2bdb and astdb2sqlite3</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400315">400315</a></td><td>elguero</td><td>Cast Integer Argument To Unsigned Char</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400723">400723</a></td><td>rmudgett</td><td>app_confbridge: Fix duplicate default_user profile.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=400909">400909</a></td><td>rmudgett</td><td>chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22429">ASTERISK-22429</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401016">401016</a></td><td>rmudgett</td><td>chan_iax2: Fix channel left locked in off nominal code path.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401076">401076</a></td><td>wdoekes</td><td>Don't check all realtime queues when doing "queue show some_queue".</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401379">401379</a></td><td>rmudgett</td><td>chan_dahdi: Fix unable to get index warning when transferring an analog call.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401661">401661</a></td><td>jrose</td><td>memory leaks: Memory leak cleanup patch by Corey Farrell (first set)</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401705">401705</a></td><td>jrose</td><td>memory leaks: Memory leak cleanup patch by Corey Farrell (second set)</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401744">401744</a></td><td>jrose</td><td>app_voicemail: Memory Leaks against tests</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401783">401783</a></td><td>jrose</td><td>astobj2: Unregister debug CLI commands at exit</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401787">401787</a></td><td>jrose</td><td>jitterbuf: Fix memory leak on jitter buffer reset</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401791">401791</a></td><td>jrose</td><td>test_linkedlists: Fix memory leak</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401896">401896</a></td><td>jrose</td><td>revert clicompat-r2.patch from r401704</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=401935">401935</a></td><td>jrose</td><td>Put clicompat-r2.patch back in</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22467">ASTERISK-22467</a></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 | 1
+UPGRADE.txt | 10 ++
+apps/app_confbridge.c | 3
+apps/app_queue.c | 37 +++++---
+apps/app_voicemail.c | 6 -
+apps/confbridge/conf_config_parser.c | 4
+apps/confbridge/include/confbridge.h | 1
+cdr/cdr_adaptive_odbc.c | 8 +
+channels/chan_dahdi.c | 55 +++++++++----
+channels/chan_iax2.c | 49 ++++++-----
+channels/chan_mgcp.c | 6 -
+channels/chan_sip.c | 147 +++++++++++++++++++++++------------
+channels/sig_analog.c | 52 +++++-------
+channels/sig_ss7.c | 9 +-
+channels/sip/include/sip.h | 1
+channels/sip/reqresp_parser.c | 6 +
+codecs/ilbc/doCPLC.c | 2
+configs/confbridge.conf.sample | 3
+configs/sip.conf.sample | 3
+configure.ac | 14 +++
+doc/astdb2bdb.8 | 46 ++++++++++
+doc/astdb2sqlite3.8 | 39 +++++++++
+funcs/func_config.c | 4
+funcs/func_math.c | 3
+include/asterisk/pbx.h | 10 ++
+include/asterisk/rtp_engine.h | 6 -
+main/app.c | 4
+main/asterisk.c | 6 +
+main/astobj2.c | 6 +
+main/channel.c | 16 +++
+main/config_options.c | 26 +++++-
+main/data.c | 3
+main/editline/readline.c | 7 +
+main/editline/term.c | 6 -
+main/jitterbuf.c | 6 -
+main/logger.c | 3
+main/pbx.c | 63 ++++++++-------
+main/rtp_engine.c | 90 ++++++++++-----------
+main/security_events.c | 6 -
+main/test.c | 6 +
+main/translate.c | 4
+main/utils.c | 20 +++-
+res/res_jabber.c | 2
+res/res_rtp_asterisk.c | 69 +++++++++++-----
+res/res_rtp_multicast.c | 8 -
+res/res_xmpp.c | 2
+tests/test_dlinklists.c | 2
+tests/test_linkedlists.c | 7 -
+utils/clicompat.c | 10 ++
+49 files changed, 628 insertions(+), 269 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
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@@ -1,0 +1,544 @@
+ Release Summary
+
+ asterisk-11.7.0-rc1
+
+ Date: 2013-10-28
+
+ <asteriskteam at 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-11.6.0.
+
+ ----------------------------------------------------------------------
+
+ 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
+ 11 jrose 3 elguero 2 gkelleter
+ 8 rmudgett 1 Ben Smithurst 2 nbansal
+ 5 elguero 1 Dalius M. 2 tsearle
+ 4 kmoore 1 rmudgett 1 abelbeck
+ 4 wdoekes 1 wdoekes 1 adomjan
+ 3 mjordan 1 aragon
+ 2 kharwell 1 arcanos
+ 2 NITESH BANSAL 1 bensmithurst
+ 2 sgriepentrog 1 brianscott
+ 1 abelbeck 1 coreyfarrell
+ 1 adomjan 1 dario
+ 1 coreyfarrell 1 docent
+ 1 Filip Jenicek 1 elguero
+ 1 junky 1 frenk77
+ 1 mmichelson 1 jbigelow
+ 1 nbougues 1 jkister
+ 1 Oscar Esteve 1 jllull
+ 1 Simone Camporeale 1 jrose
+ 1 Torrey Searle 1 kharwell
+ 1 tsearle 1 kmoore
+ 1 tzafrir 1 londonnet
+ 1 mdalius
+ 1 mjordan
+ 1 nbougues
+ 1 oscares
+ 1 patrickm
+ 1 phill
+ 1 rudolf
+ 1 simone.camporeale
+ 1 skavin
+ 1 tomaso
+ 1 wdoekes
+
+ ----------------------------------------------------------------------
+
+ 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: Applications/app_confbridge
+
+ ASTERISK-19983: ConfBridge does not expose a mechanism to change the
+ language on the Bridging channel, defaulting to 'en'
+ Revision: 400741
+ Reporter: londonnet
+ Testers: rmudgett
+ Coders: junky
+
+ Category: Applications/app_queue
+
+ ASTERISK-21826: [patch] Bad queue_log entry when removed member from queue
+ via CLI
+ Revision: 401433
+ Reporter: oscares
+ Coders: Oscar Esteve
+
+ ASTERISK-22197: [patch] Queuelog EXITWITHKEY only two of four parameters
+ Revision: 400623
+ Reporter: mdalius
+ Testers: Dalius M.
+ Coders: elguero
+
+ Category: Applications/app_voicemail
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ Category: Bridges/bridge_native_rtp
+
+ ASTERISK-22424: bridge_native_rtp: Asterisk 12 attempts to remotely bridge
+ on 200OK response to invite when the 200 lacks SDP
+ Revision: 399962
+ Reporter: jrose
+ Coders: jrose
+
+ Category: CDR/cdr_adaptive_odbc
+
+ ASTERISK-22272: [patch] Unexepected behaviour with adaptive odbc filter
+ Revision: 401579
+ Reporter: jllull
+ Coders: rmudgett
+
+ Category: Channels/chan_dahdi
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ Category: Channels/chan_mgcp
+
+ ASTERISK-21190: chan_mgcp crash on chunked m= sdp line
+ Revision: 401538
+ Reporter: adomjan
+ Coders: adomjan
+
+ Category: Channels/chan_pjsip
+
+ ASTERISK-22667: crash: directmedia with both phones placing each other on
+ hold
+ Revision: 401446
+ Reporter: jbigelow
+ Coders: mjordan
+
+ Category: Channels/chan_sip/CodecHandling
+
+ ASTERISK-21131: [patch] - Asterisk creates SDP with (peer) unsupported
+ audio codec
+ Revision: 401498
+ Reporter: nbougues
+ Coders: nbougues
+
+ Category: Channels/chan_sip/General
+
+ ASTERISK-21204: Asterisk increments the session version in 2xx message
+ even if a '183 Session in Progress' with SDP has already been sent in
+ response to initial INVITE.
+ Revision: 400908
+ Reporter: nbansal
+ Coders: NITESH BANSAL
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ ASTERISK-22477: array calleridname in the function check_user_full is too
+ short
+ Revision: 400014
+ Reporter: docent
+ Coders: rmudgett
+
+ ASTERISK-22621: chan_sip can send two BYEs for a single call
+ Revision: 400971
+ Reporter: kmoore
+ Coders: mmichelson
+
+ ASTERISK-22667: crash: directmedia with both phones placing each other on
+ hold
+ Revision: 401446
+ Reporter: jbigelow
+ Coders: mjordan
+
+ ASTERISK-22729: [patch] Remove Port Restriction When Checking For NAT
+ Revision: 401182
+ Reporter: elguero
+ Testers: elguero
+ Coders: elguero
+
+ Category: Channels/chan_sip/Interoperability
+
+ ASTERISK-22005: Allow a sip peer to accept both AVP and AVPF calls
+ Revision: 401884
+ Reporter: tsearle
+ Coders: tsearle
+
+ Category: Channels/chan_sip/Registration
+
+ ASTERISK-17138: [patch] Asterisk not re-registering after it receives
+ "Forbidden - wrong password on authentication"
+ Revision: 400140
+ Reporter: rudolf
+ Coders: kmoore
+
+ ASTERISK-22236: REGISTER reply send to bad port with nat=yes(or
+ force_rport,comedia) in 11.5.0
+ Revision: 401167
+ Reporter: frenk77
+ Testers: elguero
+ Coders: elguero
+
+ ASTERISK-22428: [patch] SIP unregister does not fully unregister when
+ using Realtime sip peers and Expires not 0 on 200ok
+ Revision: 399795
+ Reporter: bensmithurst
+ Testers: Ben Smithurst, elguero
+ Coders: elguero
+
+ ASTERISK-22574: [patch]Value of expires= is ignored in the Contact header
+ Revision: 400470
+ Reporter: phill
+ Coders: Filip Jenicek
+
+ Category: Channels/chan_sip/T.38
+
+ ASTERISK-17273: [patch] - Incorrect address specified in SIP re-INVITE
+ with T.38 when directmedia enabled
+ Revision: 399457
+ Reporter: dario
+ Coders: jrose
+
+ ASTERISK-18706: UDPTL fail while using directmedia
+ Revision: 399457
+ Reporter: jkister
+ Coders: jrose
+
+ Category: Codecs/codec_ilbc
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ Category: Core/BuildSystem
+
+ ASTERISK-22351: Segfault in LIBEDIT_INTERNAL after tgetstr(), when
+ libncurses5-dev isn't installed
+ Revision: 401326
+ Reporter: arcanos
+ Coders: wdoekes
+
+ Category: Core/General
+
+ ASTERISK-22351: Segfault in LIBEDIT_INTERNAL after tgetstr(), when
+ libncurses5-dev isn't installed
+ Revision: 401326
+ Reporter: arcanos
+ Coders: wdoekes
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ Category: Core/Jitterbuffer
+
+ ASTERISK-22467: [patch] memory leaks 1.8+
+ Revision: 401830
+ Reporter: coreyfarrell
+ Coders: coreyfarrell
+
+ Category: Core/Logging
+
+ ASTERISK-22456: Logger.conf: Logging types ignored after specifying a
+ verbose level
+ Revision: 401833
+ Reporter: kharwell
+ Coders: kharwell
+
+ Category: Core/ManagerInterface
+
+ ASTERISK-22578: Invalid manager logins aren't reported via security
+ events: Invalid IE Specified ERROR
+ Revision: 400421
+ Reporter: mjordan
+ Coders: kmoore
+
+ Category: Core/RTP
+
+ ASTERISK-21464: with directrtpsetup some payload type identifiers from A
+ party's INVITE are not copied to the INVITE for B party
+ Revision: 402042
+ Reporter: skavin
+ Coders: sgriepentrog
+
+ ASTERISK-22424: bridge_native_rtp: Asterisk 12 attempts to remotely bridge
+ on 200OK response to invite when the 200 lacks SDP
+ Revision: 399962
+ Reporter: jrose
+ Coders: jrose
+
+ Category: Core/UDPTL
+
+ ASTERISK-18706: UDPTL fail while using directmedia
+ Revision: 399457
+ Reporter: jkister
+ Coders: jrose
+
+ Category: Functions/func_config
+
+ ASTERISK-22483: AST_LIST_INSERT_TAIL doesn't set field.next on added entry
+ Revision: 400697
[... 229 lines stripped ...]
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