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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-1.4.42</title></head>
-<body>
-<h1 align="center"><a name="top">Release Summary</a></h1>
-<h3 align="center">asterisk-1.4.42</h3>
-<h3 align="center">Date: 2011-06-28</h3>
-<h3 align="center"><asteriskteam@digium.com></h3>
-<hr/>
-<h2 align="center">Table of Contents</h2>
-<ol>
- <li><a href="#summary">Summary</a></li>
- <li><a href="#contributors">Contributors</a></li>
- <li><a href="#issues">Closed Issues</a></li>
- <li><a href="#commits">Other Changes</a></li>
- <li><a href="#diffstat">Diffstat</a></li>
-</ol>
-<hr/>
-<a name="summary"><h2 align="center">Summary</h2></a>
-<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
-<p>The data in this summary reflects changes that have been made since the previous release, asterisk-1.4.41.</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 mnicholson<br/>
-8 rmudgett<br/>
-8 twilson<br/>
-6 alecdavis<br/>
-6 tilghman<br/>
-3 seanbright<br/>
-3 wdoekes<br/>
-2 dvossel<br/>
-2 elguero<br/>
-2 pabelanger<br/>
-2 vrban<br/>
-1 andy11<br/>
-1 enegaard<br/>
-1 FeyFre<br/>
-1 IgorG<br/>
-1 jhirsch<br/>
-1 jrose<br/>
-1 kkm<br/>
-1 nivek<br/>
-1 russell<br/>
-1 tzafrir<br/>
-1 zvision<br/>
-</td>
-<td>
-8 rmudgett<br/>
-4 astmiv<br/>
-3 alecdavis<br/>
-2 globalnetinc<br/>
-2 jde<br/>
-2 nivek<br/>
-2 twilson<br/>
-1 alexandrekeller<br/>
-1 amilcar<br/>
-1 chris-mac<br/>
-1 elguero<br/>
-1 FeyFre<br/>
-1 francesco_r<br/>
-1 IgorG<br/>
-1 irroot<br/>
-1 isis242<br/>
-1 jcromes<br/>
-1 jrose<br/>
-1 kkm<br/>
-1 lefoyer<br/>
-1 lmadsen<br/>
-1 loloski<br/>
-1 mnicholson<br/>
-1 oej<br/>
-1 rymkus<br/>
-1 seanbright<br/>
-1 tilghman<br/>
-1 vrban<br/>
-1 wdoekes<br/>
-</td>
-<td>
-2 alecdavis<br/>
-2 destiny6628<br/>
-2 tzafrir<br/>
-2 vrban<br/>
-1 alexandrekeller<br/>
-1 andy11<br/>
-1 devmod<br/>
-1 docent<br/>
-1 elguero<br/>
-1 feyfre<br/>
-1 igorg<br/>
-1 jamicque<br/>
-1 jasonshugart<br/>
-1 jcromes<br/>
-1 jhirsch<br/>
-1 jmls<br/>
-1 johnz<br/>
-1 jpokorny<br/>
-1 kkm<br/>
-1 kobaz<br/>
-1 lefoyer<br/>
-1 mn3250<br/>
-1 mspuhler<br/>
-1 nivek<br/>
-1 nvitaly<br/>
-1 oej<br/>
-1 pabelanger<br/>
-1 pnlarsson<br/>
-1 pruiz<br/>
-1 sharvanek<br/>
-1 siby<br/>
-1 sysreq<br/>
-1 wdoekes<br/>
-1 zvision<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_dial</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16649">ASTERISK-16649</a>: [patch] Peer does not hang up when caller hangup while app_dial is executing - Deadagi<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313545">313545</a><br/>
-Reporter: mn3250<br/>
-Testers: rmudgett, astmiv<br/>
-Coders: rmudgett<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17874">ASTERISK-17874</a>: [patch] [regression] Revision 315643 app_dial breaks ring groups<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=319527">319527</a><br/>
-Reporter: mspuhler<br/>
-Coders: elguero<br/>
-<br/>
-<h3>Category: Applications/app_externalivr</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17496">ASTERISK-17496</a>: [patch] Small leak in app_externalivr<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=309355">309355</a><br/>
-Reporter: andy11<br/>
-Coders: andy11<br/>
-<br/>
-<h3>Category: Applications/app_meetme</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17896">ASTERISK-17896</a>: [patch] meetme cli cmd completion leaves conferences mutex locked<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=320235">320235</a><br/>
-Reporter: zvision<br/>
-Coders: zvision<br/>
-<br/>
-<h3>Category: Applications/app_mixmonitor</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17038">ASTERISK-17038</a>: [patch] Mixmonitor does not parse file path proper if it contain a . (period)<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=309856">309856</a><br/>
-Reporter: pabelanger<br/>
-Testers: jrose<br/>
-Coders: jrose<br/>
-<br/>
-<h3>Category: Applications/app_voicemail</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16784">ASTERISK-16784</a>: [patch] Message lost when sox fails to re-encode with 'volgain'<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=316707">316707</a><br/>
-Reporter: sysreq<br/>
-Testers: seanbright<br/>
-Coders: seanbright<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17299">ASTERISK-17299</a>: [patch] Compile Error - odbc_storage enabled<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312174">312174</a><br/>
-Reporter: elguero<br/>
-Testers: elguero, nivek, alecdavis<br/>
-Coders: elguero<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17714">ASTERISK-17714</a>: [patch] app_voicemail.c does not compile in 1.4 branch<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=314300">314300</a><br/>
-Reporter: vrban<br/>
-Testers: vrban, alecdavis<br/>
-Coders: vrban<br/>
-<br/>
-<h3>Category: Channels/chan_dahdi</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-14611">ASTERISK-14611</a>: [patch] Stuck channel using FEATD_MF if caller hangs up at the right time<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313188">313188</a><br/>
-Reporter: jcromes<br/>
-Testers: jcromes<br/>
-Coders: pabelanger<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16892">ASTERISK-16892</a>: [patch] Asterisk gets killed during the live calling<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312573">312573</a><br/>
-Reporter: destiny6628<br/>
-Testers: rmudgett<br/>
-Coders: rmudgett<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16964">ASTERISK-16964</a>: Asterisk does not send release message when channel requested during SETUP gets changed during Procedding Message from TELCO<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312573">312573</a><br/>
-Reporter: destiny6628<br/>
-Testers: rmudgett<br/>
-Coders: rmudgett<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17107">ASTERISK-17107</a>: [patch] "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310633">310633</a><br/>
-Reporter: nivek<br/>
-Testers: nivek<br/>
-Coders: nivek<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17120">ASTERISK-17120</a>: Asterisk does not end call properly and stops reacting to following SETUP messages<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312573">312573</a><br/>
-Reporter: jpokorny<br/>
-Testers: rmudgett<br/>
-Coders: rmudgett<br/>
-<br/>
-<h3>Category: Channels/chan_iax2</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15449">ASTERISK-15449</a>: RFC2833 DTMF is not passed correctly when going SIP->IAX2->SIP<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310888">310888</a><br/>
-Reporter: sharvanek<br/>
-Testers: globalnetinc, jde<br/>
-Coders: twilson<br/>
-<br/>
-<h3>Category: Channels/chan_local</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17633">ASTERISK-17633</a>: [patch] Chan_local crashes in fixup<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=316328">316328</a><br/>
-Reporter: oej<br/>
-Testers: oej<br/>
-Coders: dvossel<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17747">ASTERISK-17747</a>: [patch] check_bridge(): misplaced ast_mutex_unlock<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315051">315051</a><br/>
-Reporter: alecdavis<br/>
-Coders: alecdavis<br/>
-<br/>
-<h3>Category: Channels/chan_sip/General</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17124">ASTERISK-17124</a>: Asterisk does not hangup a channel after endpoint hangs up when using FastAGI<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313545">313545</a><br/>
-Reporter: devmod<br/>
-Testers: rmudgett, astmiv<br/>
-Coders: rmudgett<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17682">ASTERISK-17682</a>: [patch] [regression] "sip prune" does not clean the relevant peer objects -> memleak<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=317211">317211</a><br/>
-Reporter: vrban<br/>
-Coders: vrban<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17771">ASTERISK-17771</a>: [patch] switching From-address mid-register breaks channel variables<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=320055">320055</a><br/>
-Reporter: wdoekes<br/>
-Coders: wdoekes<br/>
-<br/>
-<h3>Category: Channels/chan_sip/Registration</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-14953">ASTERISK-14953</a>: [patch] Autocreated peers not deleted when unregister explicitly, become zombies<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315671">315671</a><br/>
-Reporter: kkm<br/>
-Testers: kkm, tilghman, twilson<br/>
-Coders: kkm<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17382">ASTERISK-17382</a>: [patch] Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.)<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=317719">317719</a><br/>
-Reporter: pnlarsson<br/>
-Coders: enegaard<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17535">ASTERISK-17535</a>: [patch] [regression] Cisco phones do not register<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315891">315891</a><br/>
-Reporter: jmls<br/>
-Coders: wdoekes<br/>
-<br/>
-<h3>Category: Core/BuildSystem</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17668">ASTERISK-17668</a>: [patch] fix detection of openssl 1.0<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313277">313277</a><br/>
-Reporter: tzafrir<br/>
-Coders: tzafrir<br/>
-<br/>
-<h3>Category: Core/General</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17370">ASTERISK-17370</a>: [patch] FD 32767 exceeds the maximum size of ast_fdset<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315501">315501</a><br/>
-Reporter: jamicque<br/>
-Testers: chris-mac<br/>
-Coders: tilghman<br/>
-<br/>
-<h3>Category: Core/ManagerInterface</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16837">ASTERISK-16837</a>: [patch] Duplicated event on AMI interface<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=308813">308813</a><br/>
-Reporter: feyfre<br/>
-Testers: FeyFre, twilson<br/>
-Coders: FeyFre<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17384">ASTERISK-17384</a>: [patch] Security issue in originate, system permission bypassed if using async<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=314607">314607</a><br/>
-Reporter: kobaz<br/>
-Coders: mnicholson<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17578">ASTERISK-17578</a>: [patch] DoS through manager interface: no timeout for unauthenticated logins<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312761">312761</a><br/>
-Reporter: tzafrir<br/>
-Testers: mnicholson<br/>
-Coders: mnicholson<br/>
-<br/>
-<h3>Category: Core/RTP</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-14589">ASTERISK-14589</a>: [patch] Fix for Sonus DTMF issues<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310888">310888</a><br/>
-Reporter: jasonshugart<br/>
-Testers: globalnetinc, jde<br/>
-Coders: twilson<br/>
-<br/>
-<h3>Category: Features</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17264">ASTERISK-17264</a>: [patch] [regression] Call Pickup Hangs Asterisk (deadlock?)<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=318734">318734</a><br/>
-Reporter: docent<br/>
-Testers: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett<br/>
-Coders: rmudgett, alecdavis<br/>
-<br/>
-<h3>Category: Formats/format_wav</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16957">ASTERISK-16957</a>: [patch] Asterisk does not play wav files with unknown chunk types<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315257">315257</a><br/>
-Reporter: jhirsch<br/>
-Coders: jhirsch<br/>
-<br/>
-<h3>Category: Functions/func_odbc</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16946">ASTERISK-16946</a>: [patch] Call to SQLDescribeCol returns an invalid ColumnSize paramenter on x64 (Patch included)<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310140">310140</a><br/>
-Reporter: pruiz<br/>
-Coders: tilghman<br/>
-<br/>
-<h3>Category: Functions/func_shell</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17630">ASTERISK-17630</a>: [patch] Concatenates uninitialized buffer causes garbage data prior result also may cause crash<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=314822">314822</a><br/>
-Reporter: johnz<br/>
-Coders: russell<br/>
-<br/>
-<h3>Category: General</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17552">ASTERISK-17552</a>: [patch] 'core show locks' should show Thread ID in HEX, then would match up with GDB's backtrace<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310779">310779</a><br/>
-Reporter: alecdavis<br/>
-Coders: alecdavis<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17555">ASTERISK-17555</a>: [patch] Remove extra quote in indications.conf<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=311048">311048</a><br/>
-Reporter: igorg<br/>
-Testers: IgorG<br/>
-Coders: IgorG<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17782">ASTERISK-17782</a>: [patch] bug in contrib/scripts/safe_asterisk<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=317102">317102</a><br/>
-Reporter: lefoyer<br/>
-Testers: wdoekes, lefoyer<br/>
-Coders: wdoekes<br/>
-<br/>
-<h3>Category: PBX/pbx_ael</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17497">ASTERISK-17497</a>: [patch] AELsub() for calling routines that will remain stable between internal changes<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=310435">310435</a><br/>
-Reporter: alexandrekeller<br/>
-Testers: alexandrekeller<br/>
-Coders: tilghman<br/>
-<br/>
-<h3>Category: Resources/res_agi</h3><br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16144">ASTERISK-16144</a>: [patch] AGISTATUS bug in Asterisk 1.6.2.7<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313545">313545</a><br/>
-Reporter: siby<br/>
-Testers: rmudgett, astmiv<br/>
-Coders: rmudgett<br/>
-<br/>
-<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17520">ASTERISK-17520</a>: [patch] HANGUP is not sent to AGI in time<br/>
-Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=313545">313545</a><br/>
-Reporter: nvitaly<br/>
-Testers: rmudgett, astmiv<br/>
-Coders: rmudgett<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/1.4?view=revision&revision=308721">308721</a></td><td>mnicholson</td><td>silence gcc 4.2 compiler warning</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=309947">309947</a></td><td>twilson</td><td>Don't try to free statically allocated memory.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=311199">311199</a></td><td>mnicholson</td><td>Remove the provisional keepalive scheduler entry's reference to the pvt when we remove the scheduler entry.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=311201">311201</a></td><td>mnicholson</td><td>Don't dec the usecount of an eventqent then use it.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=311203">311203</a></td><td>mnicholson</td><td>Don't hold the pvt lock while streaming a file.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312070">312070</a></td><td>alecdavis</td><td>app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312119">312119</a></td><td>alecdavis</td><td>app_voicemail:close_mailbox imap_storage doesn't use last_msg_index</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312285">312285</a></td><td>tilghman</td><td>Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17553">ASTERISK-17553</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=312290">312290</a></td><td>alecdavis</td><td>app_voicemail: leave_vociemail doesn't use last_message_index to store next message</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17580">ASTERISK-17580</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=314908">314908</a></td><td>mnicholson</td><td>Prevent the login thread and the app threads from using the asterisk channel at the same time.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315147">315147</a></td><td>mnicholson</td><td>Reverted part of r314607, as it can introduce a regression.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315596">315596</a></td><td>twilson</td><td>Allow transfer loops without allowing forwarding loops</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=315989">315989</a></td><td>seanbright</td><td>Partial revert of r315671 which removed a logging statement and not a manager event.</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-14953">ASTERISK-14953</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=316006">316006</a></td><td>tilghman</td><td>Backport the use of curl from 1.6.2 to make the 1.4 target work on Bamboo.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=316089">316089</a></td><td>tilghman</td><td>Breakage from slightly before the outage; would have fixed sooner but for the outage.</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=317574">317574</a></td><td>twilson</td><td>Re-fix queue round-robin</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=319652">319652</a></td><td>twilson</td><td>Make sure everyone gets an unhold when a transfer succeeds</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=320393">320393</a></td><td>pabelanger</td><td>Solaris compatibility fixes</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=322646">322646</a></td><td>mnicholson</td><td>don't drop any voice frames when checking for T.38 during early media</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17705">ASTERISK-17705</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=322657">322657</a></td><td>mnicholson</td><td>whitespace</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=322698">322698</a></td><td>mnicholson</td><td>unlock pvt when we drop voice frames received in early media when in t.38 mode</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=323559">323559</a></td><td>seanbright</td><td>Resolve a segfault/bus error when we try to map memory that falls on a page</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-15359">ASTERISK-15359</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-16460">ASTERISK-16460</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=323732">323732</a></td><td>twilson</td><td>Fix DYNAMIC_FEATURES</td>
-<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17914">ASTERISK-17914</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=324627">324627</a></td><td>dvossel</td><td>Addresses AST-2011-010, remote crash in IAX2 driver</td>
-<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=325275">325275</a></td><td>twilson</td><td>Don't leak SIP username information</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>
-apps/app_dial.c | 5
-apps/app_directed_pickup.c | 26
-apps/app_meetme.c | 1
-apps/app_mixmonitor.c | 4
-apps/app_voicemail.c | 108 +-
-cdr/cdr_pgsql.c | 8
-channels/chan_agent.c | 181 ++-
-channels/chan_dahdi.c | 2007 ++++++++++++++++++++++++---------------
-channels/chan_iax2.c | 14
-channels/chan_local.c | 4
-channels/chan_sip.c | 106 +-
-channels/chan_skinny.c | 75 +
-configs/http.conf.sample | 7
-configs/indications.conf.sample | 2
-configs/manager.conf.sample | 11
-configs/skinny.conf.sample | 9
-configure.ac | 10
-contrib/scripts/safe_asterisk | 2
-formats/format_wav.c | 84 -
-include/asterisk/autoconfig.h.in | 3
-include/asterisk/select.h | 15
-main/asterisk.c | 6
-main/callerid.c | 19
-main/channel.c | 32
-main/http.c | 25
-main/manager.c | 122 ++
-main/udptl.c | 6
-main/utils.c | 2
-pbx/pbx_ael.c | 30
-res/res_agi.c | 46
-res/res_config_odbc.c | 2
-res/res_features.c | 273 +++--
-32 files changed, 2133 insertions(+), 1112 deletions(-)
-</pre><br/>
-<hr/>
-</body>
-</html>
|
|
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[+]
|
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asterisk-1.4.42.tar.bz2/asterisk-1.4.42-summary.txt
^
|
| @@ -1,527 +0,0 @@
- Release Summary
-
- asterisk-1.4.42
-
- Date: 2011-06-28
-
- <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-1.4.41.
-
- ----------------------------------------------------------------------
-
- 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 mnicholson 8 rmudgett 2 alecdavis
- 8 rmudgett 4 astmiv 2 destiny6628
- 8 twilson 3 alecdavis 2 tzafrir
- 6 alecdavis 2 globalnetinc 2 vrban
- 6 tilghman 2 jde 1 alexandrekeller
- 3 seanbright 2 nivek 1 andy11
- 3 wdoekes 2 twilson 1 devmod
- 2 dvossel 1 alexandrekeller 1 docent
- 2 elguero 1 amilcar 1 elguero
- 2 pabelanger 1 chris-mac 1 feyfre
- 2 vrban 1 elguero 1 igorg
- 1 andy11 1 FeyFre 1 jamicque
- 1 enegaard 1 francesco_r 1 jasonshugart
- 1 FeyFre 1 IgorG 1 jcromes
- 1 IgorG 1 irroot 1 jhirsch
- 1 jhirsch 1 isis242 1 jmls
- 1 jrose 1 jcromes 1 johnz
- 1 kkm 1 jrose 1 jpokorny
- 1 nivek 1 kkm 1 kkm
- 1 russell 1 lefoyer 1 kobaz
- 1 tzafrir 1 lmadsen 1 lefoyer
- 1 zvision 1 loloski 1 mn3250
- 1 mnicholson 1 mspuhler
- 1 oej 1 nivek
- 1 rymkus 1 nvitaly
- 1 seanbright 1 oej
- 1 tilghman 1 pabelanger
- 1 vrban 1 pnlarsson
- 1 wdoekes 1 pruiz
- 1 sharvanek
- 1 siby
- 1 sysreq
- 1 wdoekes
- 1 zvision
-
- ----------------------------------------------------------------------
-
- 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_dial
-
- ASTERISK-16649: [patch] Peer does not hang up when caller hangup while
- app_dial is executing - Deadagi
- Revision: 313545
- Reporter: mn3250
- Testers: rmudgett, astmiv
- Coders: rmudgett
-
- ASTERISK-17874: [patch] [regression] Revision 315643 app_dial breaks ring
- groups
- Revision: 319527
- Reporter: mspuhler
- Coders: elguero
-
- Category: Applications/app_externalivr
-
- ASTERISK-17496: [patch] Small leak in app_externalivr
- Revision: 309355
- Reporter: andy11
- Coders: andy11
-
- Category: Applications/app_meetme
-
- ASTERISK-17896: [patch] meetme cli cmd completion leaves conferences mutex
- locked
- Revision: 320235
- Reporter: zvision
- Coders: zvision
-
- Category: Applications/app_mixmonitor
-
- ASTERISK-17038: [patch] Mixmonitor does not parse file path proper if it
- contain a . (period)
- Revision: 309856
- Reporter: pabelanger
- Testers: jrose
- Coders: jrose
-
- Category: Applications/app_voicemail
-
- ASTERISK-16784: [patch] Message lost when sox fails to re-encode with
- 'volgain'
- Revision: 316707
- Reporter: sysreq
- Testers: seanbright
- Coders: seanbright
-
- ASTERISK-17299: [patch] Compile Error - odbc_storage enabled
- Revision: 312174
- Reporter: elguero
- Testers: elguero, nivek, alecdavis
- Coders: elguero
-
- ASTERISK-17714: [patch] app_voicemail.c does not compile in 1.4 branch
- Revision: 314300
- Reporter: vrban
- Testers: vrban, alecdavis
- Coders: vrban
-
- Category: Channels/chan_dahdi
-
- ASTERISK-14611: [patch] Stuck channel using FEATD_MF if caller hangs up at
- the right time
- Revision: 313188
- Reporter: jcromes
- Testers: jcromes
- Coders: pabelanger
-
- ASTERISK-16892: [patch] Asterisk gets killed during the live calling
- Revision: 312573
- Reporter: destiny6628
- Testers: rmudgett
- Coders: rmudgett
-
- ASTERISK-16964: Asterisk does not send release message when channel
- requested during SETUP gets changed during Procedding Message from TELCO
- Revision: 312573
- Reporter: destiny6628
- Testers: rmudgett
- Coders: rmudgett
-
- ASTERISK-17107: [patch] "Caller*ID failed checksum" on Wildcard TDM2400P
- and TDM410
- Revision: 310633
- Reporter: nivek
- Testers: nivek
- Coders: nivek
-
- ASTERISK-17120: Asterisk does not end call properly and stops reacting to
- following SETUP messages
- Revision: 312573
- Reporter: jpokorny
- Testers: rmudgett
- Coders: rmudgett
-
- Category: Channels/chan_iax2
-
- ASTERISK-15449: RFC2833 DTMF is not passed correctly when going
- SIP->IAX2->SIP
- Revision: 310888
- Reporter: sharvanek
- Testers: globalnetinc, jde
- Coders: twilson
-
- Category: Channels/chan_local
-
- ASTERISK-17633: [patch] Chan_local crashes in fixup
- Revision: 316328
- Reporter: oej
- Testers: oej
- Coders: dvossel
-
- ASTERISK-17747: [patch] check_bridge(): misplaced ast_mutex_unlock
- Revision: 315051
- Reporter: alecdavis
- Coders: alecdavis
-
- Category: Channels/chan_sip/General
-
- ASTERISK-17124: Asterisk does not hangup a channel after endpoint hangs up
- when using FastAGI
- Revision: 313545
- Reporter: devmod
- Testers: rmudgett, astmiv
- Coders: rmudgett
-
- ASTERISK-17682: [patch] [regression] "sip prune" does not clean the
- relevant peer objects -> memleak
- Revision: 317211
- Reporter: vrban
- Coders: vrban
-
- ASTERISK-17771: [patch] switching From-address mid-register breaks channel
- variables
- Revision: 320055
- Reporter: wdoekes
- Coders: wdoekes
-
- Category: Channels/chan_sip/Registration
-
- ASTERISK-14953: [patch] Autocreated peers not deleted when unregister
- explicitly, become zombies
- Revision: 315671
- Reporter: kkm
- Testers: kkm, tilghman, twilson
- Coders: kkm
-
- ASTERISK-17382: [patch] Regression after r297603 (Improve handling of
- REGISTER requests with multiple contact headers.)
- Revision: 317719
- Reporter: pnlarsson
- Coders: enegaard
-
- ASTERISK-17535: [patch] [regression] Cisco phones do not register
- Revision: 315891
- Reporter: jmls
- Coders: wdoekes
-
- Category: Core/BuildSystem
-
- ASTERISK-17668: [patch] fix detection of openssl 1.0
- Revision: 313277
- Reporter: tzafrir
- Coders: tzafrir
-
- Category: Core/General
-
- ASTERISK-17370: [patch] FD 32767 exceeds the maximum size of ast_fdset
- Revision: 315501
- Reporter: jamicque
- Testers: chris-mac
- Coders: tilghman
-
- Category: Core/ManagerInterface
-
- ASTERISK-16837: [patch] Duplicated event on AMI interface
- Revision: 308813
- Reporter: feyfre
- Testers: FeyFre, twilson
- Coders: FeyFre
-
- ASTERISK-17384: [patch] Security issue in originate, system permission
- bypassed if using async
- Revision: 314607
- Reporter: kobaz
- Coders: mnicholson
-
- ASTERISK-17578: [patch] DoS through manager interface: no timeout for
- unauthenticated logins
- Revision: 312761
- Reporter: tzafrir
- Testers: mnicholson
- Coders: mnicholson
-
- Category: Core/RTP
-
- ASTERISK-14589: [patch] Fix for Sonus DTMF issues
- Revision: 310888
- Reporter: jasonshugart
- Testers: globalnetinc, jde
- Coders: twilson
-
- Category: Features
-
- ASTERISK-17264: [patch] [regression] Call Pickup Hangs Asterisk
- (deadlock?)
- Revision: 318734
- Reporter: docent
- Testers: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot,
- rymkus, loloski, rmudgett
- Coders: rmudgett, alecdavis
-
- Category: Formats/format_wav
-
- ASTERISK-16957: [patch] Asterisk does not play wav files with unknown
- chunk types
- Revision: 315257
- Reporter: jhirsch
- Coders: jhirsch
-
- Category: Functions/func_odbc
-
- ASTERISK-16946: [patch] Call to SQLDescribeCol returns an invalid
- ColumnSize paramenter on x64 (Patch included)
- Revision: 310140
- Reporter: pruiz
- Coders: tilghman
-
- Category: Functions/func_shell
-
- ASTERISK-17630: [patch] Concatenates uninitialized buffer causes garbage
- data prior result also may cause crash
- Revision: 314822
- Reporter: johnz
- Coders: russell
-
- Category: General
-
- ASTERISK-17552: [patch] 'core show locks' should show Thread ID in HEX,
- then would match up with GDB's backtrace
- Revision: 310779
- Reporter: alecdavis
- Coders: alecdavis
-
- ASTERISK-17555: [patch] Remove extra quote in indications.conf
- Revision: 311048
- Reporter: igorg
- Testers: IgorG
- Coders: IgorG
-
- ASTERISK-17782: [patch] bug in contrib/scripts/safe_asterisk
- Revision: 317102
- Reporter: lefoyer
- Testers: wdoekes, lefoyer
- Coders: wdoekes
-
- Category: PBX/pbx_ael
-
- ASTERISK-17497: [patch] AELsub() for calling routines that will remain
- stable between internal changes
- Revision: 310435
- Reporter: alexandrekeller
- Testers: alexandrekeller
- Coders: tilghman
-
- Category: Resources/res_agi
-
- ASTERISK-16144: [patch] AGISTATUS bug in Asterisk 1.6.2.7
- Revision: 313545
- Reporter: siby
- Testers: rmudgett, astmiv
- Coders: rmudgett
-
- ASTERISK-17520: [patch] HANGUP is not sent to AGI in time
- Revision: 313545
- Reporter: nvitaly
- Testers: rmudgett, astmiv
- Coders: rmudgett
-
- ----------------------------------------------------------------------
-
- 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 |
- |----------+------------+------------------------------+-----------------|
- | 308721 | mnicholson | silence gcc 4.2 compiler | |
- | | | warning | |
- |----------+------------+------------------------------+-----------------|
- | 309947 | twilson | Don't try to free statically | |
- | | | allocated memory. | |
- |----------+------------+------------------------------+-----------------|
- | | | Remove the provisional | |
- | 311199 | mnicholson | keepalive scheduler entry's | |
- | | | reference to the pvt when we | |
- | | | remove the scheduler entry. | |
- |----------+------------+------------------------------+-----------------|
- | 311201 | mnicholson | Don't dec the usecount of an | |
- | | | eventqent then use it. | |
- |----------+------------+------------------------------+-----------------|
- | 311203 | mnicholson | Don't hold the pvt lock | |
- | | | while streaming a file. | |
- |----------+------------+------------------------------+-----------------|
- | | | app_voicemail: close_mailbox | |
- | 312070 | alecdavis | needs to respect additional | |
- | | | messages while mailbox is | |
- | | | open. | |
- |----------+------------+------------------------------+-----------------|
- | | | app_voicemail:close_mailbox | |
- | 312119 | alecdavis | imap_storage doesn't use | |
- | | | last_msg_index | |
- |----------+------------+------------------------------+-----------------|
- | | | Found some leaking file | |
- | 312285 | tilghman | descriptors while looking at | ASTERISK-17553 |
- | | | ast_FD_SETSIZE dead code. | |
- |----------+------------+------------------------------+-----------------|
- | | | app_voicemail: | |
- | 312290 | alecdavis | leave_vociemail doesn't use | ASTERISK-17580 |
- | | | last_message_index to store | |
- | | | next message | |
- |----------+------------+------------------------------+-----------------|
- | | | Prevent the login thread and | |
- | 314908 | mnicholson | the app threads from using | |
- | | | the asterisk channel at the | |
- | | | same time. | |
- |----------+------------+------------------------------+-----------------|
- | | | Reverted part of r314607, as | |
- | 315147 | mnicholson | it can introduce a | |
- | | | regression. | |
- |----------+------------+------------------------------+-----------------|
- | 315596 | twilson | Allow transfer loops without | |
- | | | allowing forwarding loops | |
- |----------+------------+------------------------------+-----------------|
- | | | Partial revert of r315671 | |
- | 315989 | seanbright | which removed a logging | ASTERISK-14953 |
- | | | statement and not a manager | |
- | | | event. | |
- |----------+------------+------------------------------+-----------------|
- | | | Backport the use of curl | |
- | 316006 | tilghman | from 1.6.2 to make the 1.4 | |
- | | | target work on Bamboo. | |
- |----------+------------+------------------------------+-----------------|
- | | | Breakage from slightly | |
- | 316089 | tilghman | before the outage; would | |
- | | | have fixed sooner but for | |
- | | | the outage. | |
- |----------+------------+------------------------------+-----------------|
- | 317574 | twilson | Re-fix queue round-robin | |
- |----------+------------+------------------------------+-----------------|
- | | | Make sure everyone gets an | |
- | 319652 | twilson | unhold when a transfer | |
- | | | succeeds | |
- |----------+------------+------------------------------+-----------------|
- | 320393 | pabelanger | Solaris compatibility fixes | |
- |----------+------------+------------------------------+-----------------|
- | | | don't drop any voice frames | |
- | 322646 | mnicholson | when checking for T.38 | ASTERISK-17705 |
- | | | during early media | |
- |----------+------------+------------------------------+-----------------|
- | 322657 | mnicholson | whitespace | |
- |----------+------------+------------------------------+-----------------|
- | | | unlock pvt when we drop | |
- | 322698 | mnicholson | voice frames received in | |
- | | | early media when in t.38 | |
- | | | mode | |
- |----------+------------+------------------------------+-----------------|
- | | | Resolve a segfault/bus error | ASTERISK-15359, |
- | 323559 | seanbright | when we try to map memory | ASTERISK-16460 |
- | | | that falls on a page | |
- |----------+------------+------------------------------+-----------------|
- | 323732 | twilson | Fix DYNAMIC_FEATURES | ASTERISK-17914 |
- |----------+------------+------------------------------+-----------------|
- | 324627 | dvossel | Addresses AST-2011-010, | |
- | | | remote crash in IAX2 driver | |
- |----------+------------+------------------------------+-----------------|
- | 325275 | twilson | Don't leak SIP username | |
- | | | information | |
- +------------------------------------------------------------------------+
-
- ----------------------------------------------------------------------
-
- 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.
-
- apps/app_dial.c | 5
- apps/app_directed_pickup.c | 26
- apps/app_meetme.c | 1
- apps/app_mixmonitor.c | 4
- apps/app_voicemail.c | 108 +-
- cdr/cdr_pgsql.c | 8
- channels/chan_agent.c | 181 ++-
- channels/chan_dahdi.c | 2007 ++++++++++++++++++++++++---------------
- channels/chan_iax2.c | 14
- channels/chan_local.c | 4
- channels/chan_sip.c | 106 +-
- channels/chan_skinny.c | 75 +
- configs/http.conf.sample | 7
- configs/indications.conf.sample | 2
- configs/manager.conf.sample | 11
- configs/skinny.conf.sample | 9
- configure.ac | 10
- contrib/scripts/safe_asterisk | 2
- formats/format_wav.c | 84 -
- include/asterisk/autoconfig.h.in | 3
- include/asterisk/select.h | 15
- main/asterisk.c | 6
- main/callerid.c | 19
- main/channel.c | 32
- main/http.c | 25
- main/manager.c | 122 ++
- main/udptl.c | 6
- main/utils.c | 2
- pbx/pbx_ael.c | 30
- res/res_agi.c | 46
- res/res_config_odbc.c | 2
- res/res_features.c | 273 +++--
- 32 files changed, 2133 insertions(+), 1112 deletions(-)
-
- ----------------------------------------------------------------------
|
|
[-]
[+]
|
Changed |
asterisk-1.4.43.tar.bz2/.version
^
|
| @@ -1 +1 @@
-1.4.42
+1.4.43
|
|
[-]
[+]
|
Changed |
asterisk-1.4.43.tar.bz2/CHANGES
^
|
| @@ -1,3 +1,11 @@
+Changes since Asterisk 1.4.42
+
+ * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
+ now defaults to yes. It is very important that phones requiring nat=no be
+ specifically set as such instead of relying on the default setting. If at all
+ possible, all devices should have nat settings configured in the general section as
+ opposed to configuring nat per-device.
+
Changes since Asterisk 1.2:
* over 4,000 commits since 1.2
|
|
[-]
[+]
|
Changed |
asterisk-1.4.43.tar.bz2/ChangeLog
^
|
| @@ -1,3 +1,44 @@
+2011-12-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.4.43 Released.
+
+ * AST-2011-013
+
+2011-12-08 20:54 +0000 [r347657] Leif Madsen <lmadsen@digium.com>
+
+ * /: Update svn:externals to use menuselect from 1.4.42 and not
+ later. This change is required because when making security
+ releases, if you pull from menuselect/trunk you'll get changes
+ meant for later versions of Asterisk.
+
+2011-11-21 19:54 +0000 [r345776] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default to
+ nat=yes; warn when nat in general and peer differ It is possible
+ to enumerate SIP usernames when the general and user/peer nat
+ settings differ in whether to respond to the port a request is
+ sent from or the port listed for responses in the Via header. In
+ 1.4 and 1.6.2, this would mean if one setting was nat=yes or
+ nat=route and the other was either nat=no or nat=never. In 1.8
+ and 10, this would mean when one was nat=force_rport and the
+ other was nat=no. In order to address this problem, it was
+ decided to switch the default behavior to nat=yes/force_rport as
+ it is the most commonly used option and to strongly discourage
+ setting nat per-peer/user when at all possible. For more
+ discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+ (closes issue ASTERISK-18862) Review:
+ https://reviewboard.asterisk.org/r/1591/
+
+2011-07-08 22:26 +0000 [r327251] Jason Parker <jparker@digium.com>
+
+ * formats, codecs/gsm/src, funcs, codecs/lpc10, main/db1-ast/btree,
+ main, main/db1-ast/recno, res, pbx, pbx/ael, channels,
+ main/stdtime, utils, agi, codecs, main/db1-ast/hash, apps,
+ main/db1-ast/db, main/db1-ast/mpool, cdr: Add .o files to
+ svn:ignore property, since it's only ignored if locally
+ configured to do so.
+
2011-06-28 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.4.42 Released.
|
|
[-]
[+]
|
Added |
asterisk-1.4.43.tar.bz2/asterisk-1.4.43-summary.html
^
|
| @@ -0,0 +1,55 @@
+<!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-1.4.43</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-1.4.43</h3>
+<h3 align="center">Date: 2011-12-08</h3>
+<h3 align="center"><asteriskteam@digium.com></h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+ <li><a href="#summary">Summary</a></li>
+ <li><a href="#contributors">Contributors</a></li>
+ <li><a href="#commits">Other Changes</a></li>
+ <li><a href="#diffstat">Diffstat</a></li>
+</ol>
+<hr/>
+<a name="summary"><h2 align="center">Summary</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This release has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p>
+<p>Security Advisories: <a href="http://downloads.asterisk.org/pub/security/AST-2011-013.html">AST-2011-013</a></p>
+<p>The data in this summary reflects changes that have been made since the previous release, asterisk-1.4.42.</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>
+1 bebuild<br/>
+</td>
+<td>
+</td>
+<td>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
+<table width="100%" border="1">
+<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/tags/1.4.43?view=revision&revision=347664">347664</a></td><td>bebuild</td><td>Creating tag for the release of asterisk-1.4.43</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>
+0 files changed
+</pre><br/>
+<hr/>
+</body>
+</html>
|
|
[-]
[+]
|
Added |
asterisk-1.4.43.tar.bz2/asterisk-1.4.43-summary.txt
^
|
| @@ -0,0 +1,83 @@
+ Release Summary
+
+ asterisk-1.4.43
+
+ Date: 2011-12-08
+
+ <asteriskteam@digium.com>
+
+ ----------------------------------------------------------------------
+
+ Table of Contents
+
+ 1. Summary
+ 2. Contributors
+ 3. Other Changes
+ 4. Diffstat
+
+ ----------------------------------------------------------------------
+
+ Summary
+
+ [Back to Top]
+
+ This release has been made to address one or more security vulnerabilities
+ that have been identified. A security advisory document has been published
+ for each vulnerability that includes additional information. Users of
+ versions of Asterisk that are affected are strongly encouraged to review
+ the advisories and determine what action they should take to protect their
+ systems from these issues.
+
+ Security Advisories: AST-2011-013
+
+ The data in this summary reflects changes that have been made since the
+ previous release, asterisk-1.4.42.
+
+ ----------------------------------------------------------------------
+
+ 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
+ 1 bebuild
+
+ ----------------------------------------------------------------------
+
+ Commits Not Associated with an Issue
+
+ [Back to Top]
+
+ This is a list of all changes that went into this release that did not
+ directly close an issue from the issue tracker. The commits may have been
+ marked as being related to an issue. If that is the case, the issue
+ numbers are listed here, as well.
+
+ +------------------------------------------------------------------------+
+ | Revision | Author | Summary | Issues Referenced |
+ |----------+---------+-------------------------------+-------------------|
+ | 347664 | bebuild | Creating tag for the release | |
+ | | | of asterisk-1.4.43 | |
+ +------------------------------------------------------------------------+
+
+ ----------------------------------------------------------------------
+
+ Diffstat Results
+
+ [Back to Top]
+
+ This is a summary of the changes to the source code that went into this
+ release that was generated using the diffstat utility.
+
+ 0 files changed
+
+ ----------------------------------------------------------------------
|
|
[-]
[+]
|
Changed |
asterisk-1.4.43.tar.bz2/channels/chan_sip.c
^
|
| @@ -94,7 +94,7 @@
#include "asterisk.h"
-ASTERISK_FILE_VERSION(__FILE__, "$Revision: 325275 $")
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 345776 $")
#include <stdio.h>
#include <ctype.h>
@@ -18206,15 +18206,14 @@
}
} else if (!strcasecmp(v->name, "nat")) {
ast_set_flag(&mask[0], SIP_NAT);
- ast_clear_flag(&flags[0], SIP_NAT);
- if (!strcasecmp(v->value, "never"))
- ast_set_flag(&flags[0], SIP_NAT_NEVER);
- else if (!strcasecmp(v->value, "route"))
- ast_set_flag(&flags[0], SIP_NAT_ROUTE);
- else if (ast_true(v->value))
- ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
- else
- ast_set_flag(&flags[0], SIP_NAT_RFC3581);
+ ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
+ if (!strcasecmp(v->value, "never")) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_NEVER);
+ } else if (!strcasecmp(v->value, "route")) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_ROUTE);
+ } else if (ast_false(v->value)) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_RFC3581);
+ }
} else if (!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
@@ -18956,6 +18955,18 @@
return peer;
}
+static void display_nat_warning(const char *cat, int reason, struct ast_flags *flags) {
+ int global_nat, specific_nat;
+
+ if (reason == CHANNEL_MODULE_LOAD && (specific_nat = ast_test_flag(&flags[0], SIP_NAT)) != (global_nat = ast_test_flag(&global_flags[0], SIP_NAT))) {
+ ast_log(LOG_WARNING, "!!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make\n");
+ ast_log(LOG_WARNING, "!!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users\n");
+ ast_log(LOG_WARNING, "!!! will be sent to a different port than replies for an existing peer/user. If at all possible,\n");
+ ast_log(LOG_WARNING, "!!! use the global 'nat' setting and do not set 'nat' per peer/user.\n");
+ ast_log(LOG_WARNING, "!!! (config category='%s' global='%s' peer/user='%s')\n", cat, nat2str(global_nat), nat2str(specific_nat));
+ }
+}
+
/*! \brief Re-read SIP.conf config file
\note This function reloads all config data, except for
active peers (with registrations). They will only
@@ -19095,9 +19106,10 @@
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
- ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
- ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
- ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
+ ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
+ ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_NAT_ALWAYS); /*!< Default to nat=yes */
ast_set_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED); /*!< Set up call forward on 482 Loop Detected */
/* Debugging settings, always default to off */
@@ -19477,6 +19489,7 @@
if (is_user) {
user = build_user(cat, ast_variable_browse(cfg, cat), NULL, 0);
if (user) {
+ display_nat_warning(cat, reason, &user->flags[0]);
ASTOBJ_CONTAINER_LINK(&userl,user);
ASTOBJ_UNREF(user, sip_destroy_user);
user_count++;
@@ -19485,6 +19498,9 @@
if (is_peer) {
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
if (peer) {
+ if (!is_user) {
+ display_nat_warning(cat, reason, &peer->flags[0]);
+ }
ASTOBJ_CONTAINER_LINK(&peerl,peer);
ASTOBJ_UNREF(peer, sip_destroy_peer);
peer_count++;
@@ -19492,6 +19508,7 @@
}
}
}
+
if (ast_find_ourip(&__ourip, bindaddr)) {
ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n");
ast_config_destroy(cfg);
|
|
[-]
[+]
|
Changed |
asterisk-1.4.43.tar.bz2/configs/sip.conf.sample
^
|
| @@ -354,12 +354,20 @@
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
-;nat=no ; Global NAT settings (Affects all peers and users)
- ; yes = Always ignore info and assume NAT
+;nat=yes ; Global NAT settings (Affects all peers and users)
+ ; yes = Always ignore info and assume NAT (default)
; no = Use NAT mode only according to RFC3581 (;rport)
; never = Never attempt NAT mode or RFC3581 support
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)
+;
+; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
+; the nat setting in a peer definition, then the peer username will be discoverable
+; by outside parties as Asterisk will respond to different ports for defined and
+; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
+; GENERAL SECTION. Specifically, if nat=route or nat=yes in one section and nat=no or
+; nat=never in the other, then valid users with settings differing from those in the
+; general section will be discoverable.
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
@@ -627,7 +635,6 @@
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
-;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
@@ -659,7 +666,6 @@
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
-;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
@@ -725,9 +731,6 @@
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
-;nat=yes ; This phone may be natted
- ; Send SIP and RTP to the IP address that packet is
- ; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
|
