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13 Commits
10.7.0-rc1
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10.1.0
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3
.lastclean
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3
.lastclean
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@@ -0,0 +1,3 @@
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39
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15
CHANGES
15
CHANGES
@@ -8,6 +8,21 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes since Asterisk 10.0.0 ------------------------------
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------------------------------------------------------------------------------
|
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|
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RTP changes
|
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-------------
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* A new option, 'probation' has been added to rtp.conf
|
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RTP in strictrtp mode can now require more than 1 packet to exit learning
|
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mode with a new source (and by default requires 4). The probation option
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allows the user to change the required number of packets in sequence to any
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desired value. Use a value of 1 to essentially restore the old behavior.
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Also, with strictrtp on, Asterisk will now drop all packets until learning
|
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mode has successfully exited. These changes are based on how pjmedia handles
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media sources and source changes.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
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------------------------------------------------------------------------------
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@@ -2397,9 +2397,8 @@ static struct call_queue *find_load_queue_rt_friendly(const char *queuename)
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if (queue_vars) {
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member_config = ast_load_realtime_multientry("queue_members", "interface LIKE", "%", "queue_name", queuename, SENTINEL);
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if (!member_config) {
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ast_log(LOG_ERROR, "no queue_members defined in your config (extconfig.conf).\n");
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ast_variables_destroy(queue_vars);
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return NULL;
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ast_debug(1, "No queue_members defined in config extconfig.conf\n");
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member_config = ast_config_new();
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}
|
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}
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if (q) {
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287
asterisk-10.1.0-summary.html
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287
asterisk-10.1.0-summary.html
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@@ -0,0 +1,287 @@
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
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<html xmlns="http://www.w3.org/1999/xhtml">
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<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.1.0</title></head>
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<body>
|
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<h1 align="center"><a name="top">Release Summary</a></h1>
|
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<h3 align="center">asterisk-10.1.0</h3>
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<h3 align="center">Date: 2012-01-27</h3>
|
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<h3 align="center"><asteriskteam@digium.com></h3>
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<hr/>
|
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<h2 align="center">Table of Contents</h2>
|
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<ol>
|
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<li><a href="#summary">Summary</a></li>
|
||||
<li><a href="#contributors">Contributors</a></li>
|
||||
<li><a href="#commits">Other Changes</a></li>
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||||
<li><a href="#diffstat">Diffstat</a></li>
|
||||
</ol>
|
||||
<hr/>
|
||||
<a name="summary"><h2 align="center">Summary</h2></a>
|
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<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>
|
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<p>The data in this summary reflects changes that have been made since the previous release, asterisk-10.0.0.</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>
|
||||
33 rmudgett<br/>
|
||||
12 mjordan<br/>
|
||||
12 wdoekes<br/>
|
||||
11 jrose<br/>
|
||||
10 twilson<br/>
|
||||
8 kmoore<br/>
|
||||
3 kpfleming<br/>
|
||||
3 may<br/>
|
||||
3 mnicholson<br/>
|
||||
3 seanbright<br/>
|
||||
2 bebuild<br/>
|
||||
2 dvossel<br/>
|
||||
2 lmadsen<br/>
|
||||
2 pabelanger<br/>
|
||||
2 schmidts<br/>
|
||||
2 tilghman<br/>
|
||||
1 irroot<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/branches/10?view=revision&revision=344004">344004</a></td><td>rmudgett</td><td>Residual changes for Asterisk v10 branch from ASTERISK-18747.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18747">ASTERISK-18747</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344049">344049</a></td><td>mnicholson</td><td>don't call ltohl() twice on the same value</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18739">ASTERISK-18739</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344103">344103</a></td><td>kmoore</td><td>Fix pin parameter behavior regression in MeetMe</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18488">ASTERISK-18488</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344159">344159</a></td><td>may</td><td>Generate response to Status Enquiry message with Status q.931 message.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18748">ASTERISK-18748</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344160">344160</a></td><td>may</td><td>delete svn:mergeinfo</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344175">344175</a></td><td>twilson</td><td>Add a unit test for ast_sockaddr_split_hostport</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344216">344216</a></td><td>twilson</td><td>Don't treat a host:port string as a domain</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344271">344271</a></td><td>rmudgett</td><td>Fix deadlock during dialplan reload.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18740">ASTERISK-18740</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344334">344334</a></td><td>mnicholson</td><td>only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18490">ASTERISK-18490</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344386">344386</a></td><td>kmoore</td><td>Fix several bugs with SDP parsing and well-formedness of responses</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16558">ASTERISK-16558</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344440">344440</a></td><td>kmoore</td><td>Fix another incorrect case with meetme's PIN logic and add documentation</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344493">344493</a></td><td>dvossel</td><td>Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18829">ASTERISK-18829</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344537">344537</a></td><td>rmudgett</td><td>Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18152">ASTERISK-18152</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344540">344540</a></td><td>rmudgett</td><td>Fix potential deadlock calling ast_call() with channel locks held.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344557">344557</a></td><td>rmudgett</td><td>Fix app_macro.c MODULEINFO section termination.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18848">ASTERISK-18848</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344609">344609</a></td><td>jrose</td><td>Fix a segmentation fault when using an extension with CID matching and no CID.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18392">ASTERISK-18392</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344662">344662</a></td><td>rmudgett</td><td>Make CLI "core show channel" not hold the channel lock during console output.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18571">ASTERISK-18571</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344716">344716</a></td><td>rmudgett</td><td>Check sip.conf maxforwards parameter for range 1 <= x <= 255.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344770">344770</a></td><td>kmoore</td><td>Fix regression introduced by SDP fixups</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344836">344836</a></td><td>wdoekes</td><td>Fix bad quoting of multiline mxml opaque_data that caused invalid xml.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18852">ASTERISK-18852</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344839">344839</a></td><td>wdoekes</td><td>Remove unneeded if(params) checks in reqresp_parser.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344842">344842</a></td><td>mjordan</td><td>Video format was treated as audio when removed from the file playback scheduler</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18682">ASTERISK-18682</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344845">344845</a></td><td>wdoekes</td><td>Use __alignof__ instead of sizeof for stringfield length storage.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344900">344900</a></td><td>twilson</td><td>Don't forget to rescan MOH files for cached realtime classes</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18039">ASTERISK-18039</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344966">344966</a></td><td>irroot</td><td>mISDN Round Robin break when no channel is available</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345064">345064</a></td><td>kmoore</td><td>Ensure that a null vmexten does not cause a segfault</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345117">345117</a></td><td>jrose</td><td>Moves voicemail setup password entry to the end of the setup process.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18282">ASTERISK-18282</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345161">345161</a></td><td>wdoekes</td><td>Update reqresp_parser parse_uri doxygen comments.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18572">ASTERISK-18572</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345164">345164</a></td><td>twilson</td><td>Don't read past end of input when calling write()</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345220">345220</a></td><td>rmudgett</td><td>Fix Progress spelling error in main/pbx.c.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18857">ASTERISK-18857</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345275">345275</a></td><td>rmudgett</td><td>Restore SIP DTMF overlap dialing method.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17288">ASTERISK-17288</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18702">ASTERISK-18702</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345290">345290</a></td><td>rmudgett</td><td>Make queue log indicate if ADDMEMBER is paused for AMI and realtime.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18645">ASTERISK-18645</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345371">345371</a></td><td>rmudgett</td><td>Fix typo in sig_pri using wrong structure name.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18868">ASTERISK-18868</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345432">345432</a></td><td>rmudgett</td><td>Make FastAGI HANGUP show up in AGI debug output.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18723">ASTERISK-18723</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345488">345488</a></td><td>jrose</td><td>Guarantee messages go into the right folders with multiple recipients</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18245">ASTERISK-18245</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18246">ASTERISK-18246</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345558">345558</a></td><td>rmudgett</td><td>Remove dead code since pri_grab() can never fail.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345640">345640</a></td><td>tilghman</td><td>Fix a change in behavior in 'database show' from 1.8.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18886">ASTERISK-18886</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345683">345683</a></td><td>tilghman</td><td>Update the documentation to better clarify how the existing commands work.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345830">345830</a></td><td>twilson</td><td>Default to nat=yes; warn when nat in general and peer differ</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18862">ASTERISK-18862</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345882">345882</a></td><td>pabelanger</td><td>Add missing sound_only_one config variable</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18895">ASTERISK-18895</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345924">345924</a></td><td>wdoekes</td><td>Clarify why the AST_LOG_* macros exist next to the LOG_* macros.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17973">ASTERISK-17973</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345977">345977</a></td><td>rmudgett</td><td>Fix dnsmgr entries to ask for the same address family each time.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346029">346029</a></td><td>pabelanger</td><td>Added support level for new modules</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346031">346031</a></td><td>twilson</td><td>Resume playing existing hold music for cached realtime MOH</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18039">ASTERISK-18039</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18912">ASTERISK-18912</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346040">346040</a></td><td>mjordan</td><td>Fixed SendMessage stripping extension from To: header in SIP MESSAGE</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18903">ASTERISK-18903</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346087">346087</a></td><td>kmoore</td><td>Fix res_jabber resource leaks</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346145">346145</a></td><td>wdoekes</td><td>Fix ast_str_truncate signedness warning and documentation.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346198">346198</a></td><td>wdoekes</td><td>Minor cleanup in chan_sip get_msg_text() function.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346240">346240</a></td><td>rmudgett</td><td>Fix calls to ast_get_ip() not initializing the address family.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346293">346293</a></td><td>schmidts</td><td>Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18693">ASTERISK-18693</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346349">346349</a></td><td>dvossel</td><td>Fixes memory leak in message API.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346473">346473</a></td><td>lmadsen</td><td>Update queues.conf.sample documentation.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17413">ASTERISK-17413</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346565">346565</a></td><td>jrose</td><td>r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18700">ASTERISK-18700</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18345">ASTERISK-18345</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18342">ASTERISK-18342</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346698">346698</a></td><td>jrose</td><td>Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18925">ASTERISK-18925</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346701">346701</a></td><td>rmudgett</td><td>Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18327">ASTERISK-18327</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346763">346763</a></td><td>may</td><td>process null frame pointer returned by ast_rtp_instance_read correctly</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16697">ASTERISK-16697</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346856">346856</a></td><td>mjordan</td><td>Update SIP MESSAGE To parsing to correctly handle URI</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18903">ASTERISK-18903</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346900">346900</a></td><td>wdoekes</td><td>For SIP REGISTER fix domain-only URIs and domain ACL bypass.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18389">ASTERISK-18389</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18741">ASTERISK-18741</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346952">346952</a></td><td>kmoore</td><td>Fix chan_jingle/gtalk load regression introduced in r346087</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346955">346955</a></td><td>jrose</td><td>Resolve duplicate label used in multiple priorities for the same extension.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18807">ASTERISK-18807</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347007">347007</a></td><td>rmudgett</td><td>Restore call progress code for analog ports.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18841">ASTERISK-18841</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347068">347068</a></td><td>mjordan</td><td>Fixed crash from orphaned MWI subscriptions in chan_sip</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18663">ASTERISK-18663</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347124">347124</a></td><td>wdoekes</td><td>Move setting of voicemail zonetag and locale up a bit.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18838">ASTERISK-18838</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347146">347146</a></td><td>wdoekes</td><td>Add regression tests for issue ASTERISK-18838.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347167">347167</a></td><td>wdoekes</td><td>Don't allow transport=tcp when tcpenable=no.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18837">ASTERISK-18837</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347240">347240</a></td><td>jrose</td><td>Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18804">ASTERISK-18804</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347293">347293</a></td><td>rmudgett</td><td>Make SIP INFO messages for dtmf-relay signals case insensitive.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18924">ASTERISK-18924</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347344">347344</a></td><td>twilson</td><td>Add ASTSBINDIR to the list of configurable paths</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18959">ASTERISK-18959</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347383">347383</a></td><td>jrose</td><td>Fix: Meetme recording variables from realtime DB use null entries over channel variables</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18873">ASTERISK-18873</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347439">347439</a></td><td>rmudgett</td><td>Update AMI Getvar and Setvar documentation about supplying a channel name.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18958">ASTERISK-18958</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347532">347532</a></td><td>twilson</td><td>Don't crash on INFO automon request with no channel</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18805">ASTERISK-18805</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347600">347600</a></td><td>rmudgett</td><td>Mark channel running the h exten with the soft-hangup flag.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18811">ASTERISK-18811</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347656">347656</a></td><td>jrose</td><td>Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347727">347727</a></td><td>wdoekes</td><td>Fix regression when using tcpenable=no and tlsenable=yes.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347812">347812</a></td><td>rmudgett</td><td>Fix some parsing issues in add_exten_to_pattern_tree().</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18909">ASTERISK-18909</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347953">347953</a></td><td>rmudgett</td><td>Update sample configs to put incoming calls into context public.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-14122">ASTERISK-14122</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347955">347955</a></td><td>rmudgett</td><td>Reverting -r347953 for ASTERISK-14122</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347996">347996</a></td><td>twilson</td><td>Add a separate buffer for SRTCP packets</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18889">ASTERISK-18889</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348056">348056</a></td><td>schmidts</td><td>Fix possible misshandling of an incoming SIP response as a peer poke response.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18940">ASTERISK-18940</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348102">348102</a></td><td>rmudgett</td><td>Fix FollowMe CallerID on outgoing calls.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17557">ASTERISK-17557</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348155">348155</a></td><td>jrose</td><td>Document PARKINGSLOT variable in features.conf.sample</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16239">ASTERISK-16239</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348158">348158</a></td><td>jrose</td><td>Fix accidental use of tabs instead of spaces from previous features.conf.sample change</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348211">348211</a></td><td>mjordan</td><td>Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348213">348213</a></td><td>mnicholson</td><td>Don't clear LOCALSTATIONID before sending or receiving. The user may set that</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18921">ASTERISK-18921</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348265">348265</a></td><td>mjordan</td><td>Added support for all slin formats to app_originate</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348311">348311</a></td><td>rmudgett</td><td>Fix ParkAndAnnounce to pass the CallerID to the announcing channel.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348363">348363</a></td><td>rmudgett</td><td>Fix crash during CDR update.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18836">ASTERISK-18836</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348405">348405</a></td><td>rmudgett</td><td>Fix cut and past error in ast_call_forward().</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18836">ASTERISK-18836</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348465">348465</a></td><td>rmudgett</td><td>Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348517">348517</a></td><td>kpfleming</td><td>Correct two flaws in sip.conf.sample related to AST-2011-013.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348605">348605</a></td><td>lmadsen</td><td>Update documentation for MESSAGE_SEND_STATUS variable.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19056">ASTERISK-19056</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348648">348648</a></td><td>rmudgett</td><td>Fix crashes on other platforms caused by interference from Darwin weak symbol support.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18728">ASTERISK-18728</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348736">348736</a></td><td>rmudgett</td><td>Fix chan_iax2 to not report an RDNIS number if it is blank.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17152">ASTERISK-17152</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348790">348790</a></td><td>rmudgett</td><td>Make apps/confbridge ignore *.i files also.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348793">348793</a></td><td>rmudgett</td><td>Make codecs/speex ignore *.i files also.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348845">348845</a></td><td>twilson</td><td>Allow packetization vaules > 127</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18876">ASTERISK-18876</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348846">348846</a></td><td>mjordan</td><td>Add Asterisk TestSuite event hooks to support ConfBridge testing</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19059">ASTERISK-19059</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348889">348889</a></td><td>mjordan</td><td>Fix for memory leaks / cleanup in cel_pgsql</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18879">ASTERISK-18879</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348952">348952</a></td><td>rmudgett</td><td>Fix extension state callback references in chan_sip.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17760">ASTERISK-17760</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18844">ASTERISK-18844</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348993">348993</a></td><td>kmoore</td><td>Fix missing doc tags found while fixing ASTERISK-18689</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18689">ASTERISK-18689</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349045">349045</a></td><td>seanbright</td><td>In ChanSpy, don't create audiohooks that will never be used.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349145">349145</a></td><td>seanbright</td><td>Once an audiohook is attached to a channel, we continue to transcode all of the</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349195">349195</a></td><td>mjordan</td><td>Fix timing source dependency issues with MOH</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17474">ASTERISK-17474</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349248">349248</a></td><td>kpfleming</td><td>Improve T.38 gateway V.21 preamble detection.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349250">349250</a></td><td>kpfleming</td><td>Tell Subversion to gnore the 'astdb2bdb' binary file if it exists.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349290">349290</a></td><td>seanbright</td><td>Use ast_audiohook_write_list_empty to determine if our lists are empty instead</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349340">349340</a></td><td>mjordan</td><td>Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19040">ASTERISK-19040</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19128">ASTERISK-19128</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-17725">ASTERISK-17725</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18340">ASTERISK-18340</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19095">ASTERISK-19095</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352285">352285</a></td><td>mjordan</td><td>Create 10.1.0-rc2</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352290">352290</a></td><td>mjordan</td><td>Merged 349732, 350553, 352228, 352015, 351505, 351289, 351308</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352346">352346</a></td><td>bebuild</td><td>Updated with test results</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352347">352347</a></td><td>bebuild</td><td>Importing release summary for 10.1.0-rc2 release.</td>
|
||||
<td></td></tr></table>
|
||||
<hr/>
|
||||
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
|
||||
<pre>
|
||||
.version | 2
|
||||
CHANGES | 20
|
||||
ChangeLog | 48 +
|
||||
Makefile | 4
|
||||
UPGRADE-1.8.txt | 23
|
||||
addons/chan_ooh323.c | 2
|
||||
addons/ooh323c/src/oochannels.c | 4
|
||||
addons/ooh323c/src/ooh245.c | 17
|
||||
addons/ooh323c/src/ooh323.c | 1
|
||||
addons/ooh323c/src/ooq931.c | 179 ++++++
|
||||
addons/ooh323c/src/ooq931.h | 8
|
||||
addons/ooh323c/src/ootypes.h | 3
|
||||
apps/app_authenticate.c | 15
|
||||
apps/app_chanspy.c | 56 +
|
||||
apps/app_confbridge.c | 6
|
||||
apps/app_dial.c | 2
|
||||
apps/app_followme.c | 201 +++----
|
||||
apps/app_macro.c | 2
|
||||
apps/app_meetme.c | 34 -
|
||||
apps/app_originate.c | 8
|
||||
apps/app_parkandannounce.c | 19
|
||||
apps/app_queue.c | 192 ++++--
|
||||
apps/app_voicemail.c | 329 +++++++----
|
||||
apps/confbridge/conf_config_parser.c | 2
|
||||
asterisk-10.1.0-rc1-summary.html | 275 ---------
|
||||
asterisk-10.1.0-rc1-summary.txt | 553 -------------------
|
||||
asterisk-10.1.0-rc2-summary.html | 68 ++
|
||||
asterisk-10.1.0-rc2-summary.txt | 99 +++
|
||||
bridges/bridge_builtin_features.c | 13
|
||||
build_tools/make_defaults_h | 1
|
||||
cel/cel_pgsql.c | 37 -
|
||||
channels/chan_dahdi.c | 12
|
||||
channels/chan_gtalk.c | 25
|
||||
channels/chan_h323.c | 3
|
||||
channels/chan_iax2.c | 10
|
||||
channels/chan_jingle.c | 46 +
|
||||
channels/chan_misdn.c | 16
|
||||
channels/chan_sip.c | 965 +++++++++++++++++++++-------------
|
||||
channels/chan_skinny.c | 1
|
||||
channels/sig_analog.c | 13
|
||||
channels/sig_analog.h | 1
|
||||
channels/sig_pri.c | 175 ++----
|
||||
channels/sip/include/reqresp_parser.h | 14
|
||||
channels/sip/include/sip.h | 82 +-
|
||||
channels/sip/reqresp_parser.c | 198 +++---
|
||||
configs/asterisk.conf.sample | 1
|
||||
configs/features.conf.sample | 2
|
||||
configs/queues.conf.sample | 9
|
||||
configs/res_stun_monitor.conf.sample | 17
|
||||
configs/rtp.conf.sample | 7
|
||||
configs/sip.conf.sample | 26
|
||||
configure.ac | 34 +
|
||||
formats/format_wav.c | 6
|
||||
funcs/func_cdr.c | 20
|
||||
include/asterisk/acl.h | 25
|
||||
include/asterisk/cdr.h | 32 -
|
||||
include/asterisk/dnsmgr.h | 19
|
||||
include/asterisk/dsp.h | 5
|
||||
include/asterisk/format_pref.h | 2
|
||||
include/asterisk/jabber.h | 5
|
||||
include/asterisk/logger.h | 4
|
||||
include/asterisk/message.h | 3
|
||||
include/asterisk/module.h | 1
|
||||
include/asterisk/paths.h | 1
|
||||
include/asterisk/pbx.h | 40 +
|
||||
include/asterisk/res_fax.h | 4
|
||||
include/asterisk/stringfields.h | 7
|
||||
include/asterisk/strings.h | 10
|
||||
include/asterisk/stun.h | 43 +
|
||||
include/asterisk/tcptls.h | 7
|
||||
include/asterisk/utils.h | 63 +-
|
||||
main/acl.c | 12
|
||||
main/app.c | 3
|
||||
main/asterisk.c | 18
|
||||
main/audiohook.c | 4
|
||||
main/bridging.c | 25
|
||||
main/channel.c | 128 +++-
|
||||
main/cli.c | 32 -
|
||||
main/db.c | 36 -
|
||||
main/dnsmgr.c | 18
|
||||
main/dsp.c | 147 -----
|
||||
main/features.c | 39 +
|
||||
main/file.c | 73 +-
|
||||
main/manager.c | 15
|
||||
main/message.c | 12
|
||||
main/pbx.c | 515 ++++++++++++------
|
||||
main/rtp_engine.c | 8
|
||||
main/say.c | 2
|
||||
main/stun.c | 126 ++--
|
||||
main/tcptls.c | 55 +
|
||||
main/utils.c | 18
|
||||
res/res_agi.c | 4
|
||||
res/res_fax.c | 195 ++++--
|
||||
res/res_fax_spandsp.c | 85 ++
|
||||
res/res_format_attr_celt.c | 4
|
||||
res/res_format_attr_silk.c | 4
|
||||
res/res_jabber.c | 198 +++---
|
||||
res/res_jabber.exports.in | 2
|
||||
res/res_monitor.c | 6
|
||||
res/res_musiconhold.c | 38 -
|
||||
res/res_rtp_asterisk.c | 120 ++++
|
||||
res/res_srtp.c | 10
|
||||
res/res_stun_monitor.c | 302 ++++++----
|
||||
res/res_timing_dahdi.c | 2
|
||||
res/res_timing_pthread.c | 2
|
||||
res/res_timing_timerfd.c | 2
|
||||
tests/test_netsock2.c | 71 ++
|
||||
107 files changed, 3809 insertions(+), 2699 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
571
asterisk-10.1.0-summary.txt
Normal file
571
asterisk-10.1.0-summary.txt
Normal file
@@ -0,0 +1,571 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-10.1.0
|
||||
|
||||
Date: 2012-01-27
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Other Changes
|
||||
4. 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-10.0.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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
|
||||
33 rmudgett
|
||||
12 mjordan
|
||||
12 wdoekes
|
||||
11 jrose
|
||||
10 twilson
|
||||
8 kmoore
|
||||
3 kpfleming
|
||||
3 may
|
||||
3 mnicholson
|
||||
3 seanbright
|
||||
2 bebuild
|
||||
2 dvossel
|
||||
2 lmadsen
|
||||
2 pabelanger
|
||||
2 schmidts
|
||||
2 tilghman
|
||||
1 irroot
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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|
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344004 |rmudgett |Residual changes for Asterisk v10 |ASTERISK-18747 |
|
||||
| | |branch from ASTERISK-18747. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344049 |mnicholson|don't call ltohl() twice on the |ASTERISK-18739 |
|
||||
| | |same value | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344103 |kmoore |Fix pin parameter behavior |ASTERISK-18488 |
|
||||
| | |regression in MeetMe | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Generate response to Status | |
|
||||
|344159 |may |Enquiry message with Status q.931 |ASTERISK-18748 |
|
||||
| | |message. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344160 |may |delete svn:mergeinfo | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344175 |twilson |Add a unit test for | |
|
||||
| | |ast_sockaddr_split_hostport | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344216 |twilson |Don't treat a host:port string as | |
|
||||
| | |a domain | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344271 |rmudgett |Fix deadlock during dialplan |ASTERISK-18740 |
|
||||
| | |reload. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |only attempt to do stun handling | |
|
||||
|344334 |mnicholson|on ipv4 or ipv4 mapped to ipv6 |ASTERISK-18490 |
|
||||
| | |addresses | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344386 |kmoore |Fix several bugs with SDP parsing |ASTERISK-16558 |
|
||||
| | |and well-formedness of responses | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix another incorrect case with | |
|
||||
|344440 |kmoore |meetme's PIN logic and add | |
|
||||
| | |documentation | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fixes issue with ConfBridge | |
|
||||
|344493 |dvossel |participants hanging up during |ASTERISK-18829 |
|
||||
| | |DTMF feature menu usage getting | |
|
||||
| | |stuck in conference forever. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Make AMI event AgentCalled get | |
|
||||
|344537 |rmudgett |CallerID/ConnectedLine info from |ASTERISK-18152 |
|
||||
| | |the incoming channel. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix potential deadlock calling | |
|
||||
|344540 |rmudgett |ast_call() with channel locks | |
|
||||
| | |held. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344557 |rmudgett |Fix app_macro.c MODULEINFO section|ASTERISK-18848 |
|
||||
| | |termination. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix a segmentation fault when | |
|
||||
|344609 |jrose |using an extension with CID |ASTERISK-18392 |
|
||||
| | |matching and no CID. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Make CLI "core show channel" not | |
|
||||
|344662 |rmudgett |hold the channel lock during |ASTERISK-18571 |
|
||||
| | |console output. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344716 |rmudgett |Check sip.conf maxforwards | |
|
||||
| | |parameter for range 1 <= x <= 255.| |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344770 |kmoore |Fix regression introduced by SDP | |
|
||||
| | |fixups | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix bad quoting of multiline mxml | |
|
||||
|344836 |wdoekes |opaque_data that caused invalid |ASTERISK-18852 |
|
||||
| | |xml. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344839 |wdoekes |Remove unneeded if(params) checks | |
|
||||
| | |in reqresp_parser. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Video format was treated as audio | |
|
||||
|344842 |mjordan |when removed from the file |ASTERISK-18682 |
|
||||
| | |playback scheduler | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344845 |wdoekes |Use __alignof__ instead of sizeof | |
|
||||
| | |for stringfield length storage. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344900 |twilson |Don't forget to rescan MOH files |ASTERISK-18039 |
|
||||
| | |for cached realtime classes | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|344966 |irroot |mISDN Round Robin break when no | |
|
||||
| | |channel is available | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345064 |kmoore |Ensure that a null vmexten does | |
|
||||
| | |not cause a segfault | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Moves voicemail setup password | |
|
||||
|345117 |jrose |entry to the end of the setup |ASTERISK-18282 |
|
||||
| | |process. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345161 |wdoekes |Update reqresp_parser parse_uri |ASTERISK-18572 |
|
||||
| | |doxygen comments. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345164 |twilson |Don't read past end of input when | |
|
||||
| | |calling write() | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345220 |rmudgett |Fix Progress spelling error in |ASTERISK-18857 |
|
||||
| | |main/pbx.c. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345275 |rmudgett |Restore SIP DTMF overlap dialing |ASTERISK-17288, |
|
||||
| | |method. |ASTERISK-18702 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Make queue log indicate if | |
|
||||
|345290 |rmudgett |ADDMEMBER is paused for AMI and |ASTERISK-18645 |
|
||||
| | |realtime. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345371 |rmudgett |Fix typo in sig_pri using wrong |ASTERISK-18868 |
|
||||
| | |structure name. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345432 |rmudgett |Make FastAGI HANGUP show up in AGI|ASTERISK-18723 |
|
||||
| | |debug output. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Guarantee messages go into the |ASTERISK-18245, |
|
||||
|345488 |jrose |right folders with multiple |ASTERISK-18246 |
|
||||
| | |recipients | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345558 |rmudgett |Remove dead code since pri_grab() | |
|
||||
| | |can never fail. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345640 |tilghman |Fix a change in behavior in |ASTERISK-18886 |
|
||||
| | |'database show' from 1.8. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Update the documentation to better| |
|
||||
|345683 |tilghman |clarify how the existing commands | |
|
||||
| | |work. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345830 |twilson |Default to nat=yes; warn when nat |ASTERISK-18862 |
|
||||
| | |in general and peer differ | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345882 |pabelanger|Add missing sound_only_one config |ASTERISK-18895 |
|
||||
| | |variable | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345924 |wdoekes |Clarify why the AST_LOG_* macros |ASTERISK-17973 |
|
||||
| | |exist next to the LOG_* macros. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|345977 |rmudgett |Fix dnsmgr entries to ask for the | |
|
||||
| | |same address family each time. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346029 |pabelanger|Added support level for new | |
|
||||
| | |modules | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346031 |twilson |Resume playing existing hold music|ASTERISK-18039, |
|
||||
| | |for cached realtime MOH |ASTERISK-18912 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fixed SendMessage stripping | |
|
||||
|346040 |mjordan |extension from To: header in SIP |ASTERISK-18903 |
|
||||
| | |MESSAGE | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346087 |kmoore |Fix res_jabber resource leaks | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346145 |wdoekes |Fix ast_str_truncate signedness | |
|
||||
| | |warning and documentation. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346198 |wdoekes |Minor cleanup in chan_sip | |
|
||||
| | |get_msg_text() function. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346240 |rmudgett |Fix calls to ast_get_ip() not | |
|
||||
| | |initializing the address family. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix regression that 'rtp/rtcp set | |
|
||||
|346293 |schmidts |debup ip' only works when also a |ASTERISK-18693 |
|
||||
| | |port was specified. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346349 |dvossel |Fixes memory leak in message API. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346473 |lmadsen |Update queues.conf.sample |ASTERISK-17413 |
|
||||
| | |documentation. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |r346525 | jrose | 2011-11-30 |ASTERISK-18700, |
|
||||
|346565 |jrose |15:10:38 -0600 (Wed, 30 Nov 2011) |ASTERISK-18345, |
|
||||
| | || 18 lines |ASTERISK-18342 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Change 183 Ringing in sipfrag body| |
|
||||
|346698 |jrose |to 180 ringing. 183 Ringing isn't |ASTERISK-18925 |
|
||||
| | |even a thing. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Re-resolve the STUN address if a | |
|
||||
|346701 |rmudgett |STUN poll fails for |ASTERISK-18327 |
|
||||
| | |res_stun_monitor. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |process null frame pointer | |
|
||||
|346763 |may |returned by ast_rtp_instance_read |ASTERISK-16697 |
|
||||
| | |correctly | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346856 |mjordan |Update SIP MESSAGE To parsing to |ASTERISK-18903 |
|
||||
| | |correctly handle URI | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346900 |wdoekes |For SIP REGISTER fix domain-only |ASTERISK-18389, |
|
||||
| | |URIs and domain ACL bypass. |ASTERISK-18741 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|346952 |kmoore |Fix chan_jingle/gtalk load | |
|
||||
| | |regression introduced in r346087 | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Resolve duplicate label used in | |
|
||||
|346955 |jrose |multiple priorities for the same |ASTERISK-18807 |
|
||||
| | |extension. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347007 |rmudgett |Restore call progress code for |ASTERISK-18841 |
|
||||
| | |analog ports. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347068 |mjordan |Fixed crash from orphaned MWI |ASTERISK-18663 |
|
||||
| | |subscriptions in chan_sip | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347124 |wdoekes |Move setting of voicemail zonetag |ASTERISK-18838 |
|
||||
| | |and locale up a bit. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347146 |wdoekes |Add regression tests for issue | |
|
||||
| | |ASTERISK-18838. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347167 |wdoekes |Don't allow transport=tcp when |ASTERISK-18837 |
|
||||
| | |tcpenable=no. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Documents CHANNEL(musicclass) | |
|
||||
|347240 |jrose |taking priority over m([x]) in |ASTERISK-18804 |
|
||||
| | |waitExten | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Make SIP INFO messages for | |
|
||||
|347293 |rmudgett |dtmf-relay signals case |ASTERISK-18924 |
|
||||
| | |insensitive. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347344 |twilson |Add ASTSBINDIR to the list of |ASTERISK-18959 |
|
||||
| | |configurable paths | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix: Meetme recording variables | |
|
||||
|347383 |jrose |from realtime DB use null entries |ASTERISK-18873 |
|
||||
| | |over channel variables | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Update AMI Getvar and Setvar | |
|
||||
|347439 |rmudgett |documentation about supplying a |ASTERISK-18958 |
|
||||
| | |channel name. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347532 |twilson |Don't crash on INFO automon |ASTERISK-18805 |
|
||||
| | |request with no channel | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347600 |rmudgett |Mark channel running the h exten |ASTERISK-18811 |
|
||||
| | |with the soft-hangup flag. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix regressed behavior of queue | |
|
||||
|347656 |jrose |set penalty to work without | |
|
||||
| | |specifying 'in ' | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347727 |wdoekes |Fix regression when using | |
|
||||
| | |tcpenable=no and tlsenable=yes. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347812 |rmudgett |Fix some parsing issues in |ASTERISK-18909 |
|
||||
| | |add_exten_to_pattern_tree(). | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Update sample configs to put | |
|
||||
|347953 |rmudgett |incoming calls into context |ASTERISK-14122 |
|
||||
| | |public. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347955 |rmudgett |Reverting -r347953 for | |
|
||||
| | |ASTERISK-14122 | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|347996 |twilson |Add a separate buffer for SRTCP |ASTERISK-18889 |
|
||||
| | |packets | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix possible misshandling of an | |
|
||||
|348056 |schmidts |incoming SIP response as a peer |ASTERISK-18940 |
|
||||
| | |poke response. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348102 |rmudgett |Fix FollowMe CallerID on outgoing |ASTERISK-17557 |
|
||||
| | |calls. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348155 |jrose |Document PARKINGSLOT variable in |ASTERISK-16239 |
|
||||
| | |features.conf.sample | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix accidental use of tabs instead| |
|
||||
|348158 |jrose |of spaces from previous | |
|
||||
| | |features.conf.sample change | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fixed Asterisk crash when function| |
|
||||
|348211 |mjordan |QUEUE_MEMBER receives invalid | |
|
||||
| | |input | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Don't clear LOCALSTATIONID before | |
|
||||
|348213 |mnicholson|sending or receiving. The user may|ASTERISK-18921 |
|
||||
| | |set that | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348265 |mjordan |Added support for all slin formats| |
|
||||
| | |to app_originate | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix ParkAndAnnounce to pass the | |
|
||||
|348311 |rmudgett |CallerID to the announcing | |
|
||||
| | |channel. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348363 |rmudgett |Fix crash during CDR update. |ASTERISK-18836 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348405 |rmudgett |Fix cut and past error in |ASTERISK-18836 |
|
||||
| | |ast_call_forward(). | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Clean-up on isle five for | |
|
||||
|348465 |rmudgett |__ast_request_and_dial() and | |
|
||||
| | |ast_call_forward(). | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Correct two flaws in | |
|
||||
|348517 |kpfleming |sip.conf.sample related to | |
|
||||
| | |AST-2011-013. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348605 |lmadsen |Update documentation for |ASTERISK-19056 |
|
||||
| | |MESSAGE_SEND_STATUS variable. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Fix crashes on other platforms | |
|
||||
|348648 |rmudgett |caused by interference from Darwin|ASTERISK-18728 |
|
||||
| | |weak symbol support. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348736 |rmudgett |Fix chan_iax2 to not report an |ASTERISK-17152 |
|
||||
| | |RDNIS number if it is blank. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348790 |rmudgett |Make apps/confbridge ignore *.i | |
|
||||
| | |files also. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348793 |rmudgett |Make codecs/speex ignore *.i files| |
|
||||
| | |also. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348845 |twilson |Allow packetization vaules > 127 |ASTERISK-18876 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348846 |mjordan |Add Asterisk TestSuite event hooks|ASTERISK-19059 |
|
||||
| | |to support ConfBridge testing | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348889 |mjordan |Fix for memory leaks / cleanup in |ASTERISK-18879 |
|
||||
| | |cel_pgsql | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348952 |rmudgett |Fix extension state callback |ASTERISK-17760, |
|
||||
| | |references in chan_sip. |ASTERISK-18844 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|348993 |kmoore |Fix missing doc tags found while |ASTERISK-18689 |
|
||||
| | |fixing ASTERISK-18689 | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |In ChanSpy, don't create | |
|
||||
|349045 |seanbright|audiohooks that will never be | |
|
||||
| | |used. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Once an audiohook is attached to a| |
|
||||
|349145 |seanbright|channel, we continue to transcode | |
|
||||
| | |all of the | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|349195 |mjordan |Fix timing source dependency |ASTERISK-17474 |
|
||||
| | |issues with MOH | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|349248 |kpfleming |Improve T.38 gateway V.21 preamble| |
|
||||
| | |detection. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Tell Subversion to gnore the | |
|
||||
|349250 |kpfleming |'astdb2bdb' binary file if it | |
|
||||
| | |exists. | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | |Use ast_audiohook_write_list_empty| |
|
||||
|349290 |seanbright|to determine if our lists are | |
|
||||
| | |empty instead | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
| | | |ASTERISK-19040, |
|
||||
| | |Handle AST_CONTROL_UPDATE_RTP_PEER|ASTERISK-19128, |
|
||||
|349340 |mjordan |frames in local bridge loop |ASTERISK-17725, |
|
||||
| | | |ASTERISK-18340, |
|
||||
| | | |ASTERISK-19095 |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|352285 |mjordan |Create 10.1.0-rc2 | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|352290 |mjordan |Merged 349732, 350553, 352228, | |
|
||||
| | |352015, 351505, 351289, 351308 | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|352346 |bebuild |Updated with test results | |
|
||||
|--------+----------+----------------------------------+-----------------|
|
||||
|352347 |bebuild |Importing release summary for | |
|
||||
| | |10.1.0-rc2 release. | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Diffstat Results
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a summary of the changes to the source code that went into this
|
||||
release that was generated using the diffstat utility.
|
||||
|
||||
.version | 2
|
||||
CHANGES | 20
|
||||
ChangeLog | 48 +
|
||||
Makefile | 4
|
||||
UPGRADE-1.8.txt | 23
|
||||
addons/chan_ooh323.c | 2
|
||||
addons/ooh323c/src/oochannels.c | 4
|
||||
addons/ooh323c/src/ooh245.c | 17
|
||||
addons/ooh323c/src/ooh323.c | 1
|
||||
addons/ooh323c/src/ooq931.c | 179 ++++++
|
||||
addons/ooh323c/src/ooq931.h | 8
|
||||
addons/ooh323c/src/ootypes.h | 3
|
||||
apps/app_authenticate.c | 15
|
||||
apps/app_chanspy.c | 56 +
|
||||
apps/app_confbridge.c | 6
|
||||
apps/app_dial.c | 2
|
||||
apps/app_followme.c | 201 +++----
|
||||
apps/app_macro.c | 2
|
||||
apps/app_meetme.c | 34 -
|
||||
apps/app_originate.c | 8
|
||||
apps/app_parkandannounce.c | 19
|
||||
apps/app_queue.c | 192 ++++--
|
||||
apps/app_voicemail.c | 329 +++++++----
|
||||
apps/confbridge/conf_config_parser.c | 2
|
||||
asterisk-10.1.0-rc1-summary.html | 275 ---------
|
||||
asterisk-10.1.0-rc1-summary.txt | 553 -------------------
|
||||
asterisk-10.1.0-rc2-summary.html | 68 ++
|
||||
asterisk-10.1.0-rc2-summary.txt | 99 +++
|
||||
bridges/bridge_builtin_features.c | 13
|
||||
build_tools/make_defaults_h | 1
|
||||
cel/cel_pgsql.c | 37 -
|
||||
channels/chan_dahdi.c | 12
|
||||
channels/chan_gtalk.c | 25
|
||||
channels/chan_h323.c | 3
|
||||
channels/chan_iax2.c | 10
|
||||
channels/chan_jingle.c | 46 +
|
||||
channels/chan_misdn.c | 16
|
||||
channels/chan_sip.c | 965 +++++++++++++++++++++-------------
|
||||
channels/chan_skinny.c | 1
|
||||
channels/sig_analog.c | 13
|
||||
channels/sig_analog.h | 1
|
||||
channels/sig_pri.c | 175 ++----
|
||||
channels/sip/include/reqresp_parser.h | 14
|
||||
channels/sip/include/sip.h | 82 +-
|
||||
channels/sip/reqresp_parser.c | 198 +++---
|
||||
configs/asterisk.conf.sample | 1
|
||||
configs/features.conf.sample | 2
|
||||
configs/queues.conf.sample | 9
|
||||
configs/res_stun_monitor.conf.sample | 17
|
||||
configs/rtp.conf.sample | 7
|
||||
configs/sip.conf.sample | 26
|
||||
configure.ac | 34 +
|
||||
formats/format_wav.c | 6
|
||||
funcs/func_cdr.c | 20
|
||||
include/asterisk/acl.h | 25
|
||||
include/asterisk/cdr.h | 32 -
|
||||
include/asterisk/dnsmgr.h | 19
|
||||
include/asterisk/dsp.h | 5
|
||||
include/asterisk/format_pref.h | 2
|
||||
include/asterisk/jabber.h | 5
|
||||
include/asterisk/logger.h | 4
|
||||
include/asterisk/message.h | 3
|
||||
include/asterisk/module.h | 1
|
||||
include/asterisk/paths.h | 1
|
||||
include/asterisk/pbx.h | 40 +
|
||||
include/asterisk/res_fax.h | 4
|
||||
include/asterisk/stringfields.h | 7
|
||||
include/asterisk/strings.h | 10
|
||||
include/asterisk/stun.h | 43 +
|
||||
include/asterisk/tcptls.h | 7
|
||||
include/asterisk/utils.h | 63 +-
|
||||
main/acl.c | 12
|
||||
main/app.c | 3
|
||||
main/asterisk.c | 18
|
||||
main/audiohook.c | 4
|
||||
main/bridging.c | 25
|
||||
main/channel.c | 128 +++-
|
||||
main/cli.c | 32 -
|
||||
main/db.c | 36 -
|
||||
main/dnsmgr.c | 18
|
||||
main/dsp.c | 147 -----
|
||||
main/features.c | 39 +
|
||||
main/file.c | 73 +-
|
||||
main/manager.c | 15
|
||||
main/message.c | 12
|
||||
main/pbx.c | 515 ++++++++++++------
|
||||
main/rtp_engine.c | 8
|
||||
main/say.c | 2
|
||||
main/stun.c | 126 ++--
|
||||
main/tcptls.c | 55 +
|
||||
main/utils.c | 18
|
||||
res/res_agi.c | 4
|
||||
res/res_fax.c | 195 ++++--
|
||||
res/res_fax_spandsp.c | 85 ++
|
||||
res/res_format_attr_celt.c | 4
|
||||
res/res_format_attr_silk.c | 4
|
||||
res/res_jabber.c | 198 +++---
|
||||
res/res_jabber.exports.in | 2
|
||||
res/res_monitor.c | 6
|
||||
res/res_musiconhold.c | 38 -
|
||||
res/res_rtp_asterisk.c | 120 ++++
|
||||
res/res_srtp.c | 10
|
||||
res/res_stun_monitor.c | 302 ++++++----
|
||||
res/res_timing_dahdi.c | 2
|
||||
res/res_timing_pthread.c | 2
|
||||
res/res_timing_timerfd.c | 2
|
||||
tests/test_netsock2.c | 71 ++
|
||||
107 files changed, 3809 insertions(+), 2699 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
@@ -3882,6 +3882,7 @@ static int __sip_autodestruct(const void *data)
|
||||
ast_channel_unref(owner);
|
||||
} else if (p->refer && !p->alreadygone) {
|
||||
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
|
||||
stop_media_flows(p);
|
||||
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
|
||||
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
|
||||
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
|
||||
@@ -20714,15 +20715,22 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
|
||||
case 200: /* Notify accepted */
|
||||
/* They got the notify, this is the end */
|
||||
if (p->owner) {
|
||||
if (!p->refer) {
|
||||
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
|
||||
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
|
||||
if (p->refer) {
|
||||
ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
|
||||
} else {
|
||||
ast_debug(4, "Got OK on REFER Notify message\n");
|
||||
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
|
||||
/*
|
||||
* XXX There is discrepancy on whether a hangup should be queued
|
||||
* or not. This code used to be duplicated in two places, and the more
|
||||
* frequently hit area had this disabled, making it the de facto
|
||||
* "correct" way to go.
|
||||
*
|
||||
* ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
|
||||
*/
|
||||
}
|
||||
} else {
|
||||
if (p->subscribed == NONE) {
|
||||
ast_debug(4, "Got 200 accepted on NOTIFY\n");
|
||||
if (p->subscribed == NONE && !p->refer) {
|
||||
ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
|
||||
pvt_set_needdestroy(p, "received 200 response");
|
||||
}
|
||||
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
|
||||
@@ -20747,6 +20755,9 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
|
||||
pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
|
||||
}
|
||||
break;
|
||||
case 481: /* Call leg does not exist */
|
||||
pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
@@ -21389,6 +21400,9 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
} else if (sipmethod == SIP_MESSAGE) {
|
||||
/* More good gravy! */
|
||||
handle_response_message(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_NOTIFY) {
|
||||
/* The gravy train continues to roll */
|
||||
handle_response_notify(p, resp, rest, req, seqno);
|
||||
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
|
||||
switch(resp) {
|
||||
case 100: /* 100 Trying */
|
||||
@@ -21404,8 +21418,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
p->authtries = 0; /* Reset authentication counter */
|
||||
if (sipmethod == SIP_INVITE) {
|
||||
handle_response_invite(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_NOTIFY) {
|
||||
handle_response_notify(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_REGISTER) {
|
||||
handle_response_register(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_SUBSCRIBE) {
|
||||
@@ -21420,8 +21432,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
case 407: /* Proxy auth required */
|
||||
if (sipmethod == SIP_INVITE)
|
||||
handle_response_invite(p, resp, rest, req, seqno);
|
||||
else if (sipmethod == SIP_NOTIFY)
|
||||
handle_response_notify(p, resp, rest, req, seqno);
|
||||
else if (sipmethod == SIP_SUBSCRIBE)
|
||||
handle_response_subscribe(p, resp, rest, req, seqno);
|
||||
else if (p->registry && sipmethod == SIP_REGISTER)
|
||||
@@ -21496,8 +21506,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
handle_response_invite(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_SUBSCRIBE) {
|
||||
handle_response_subscribe(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_NOTIFY) {
|
||||
pvt_set_needdestroy(p, "received 481 response");
|
||||
} else if (sipmethod == SIP_BYE) {
|
||||
/* The other side has no transaction to bye,
|
||||
just assume it's all right then */
|
||||
@@ -21658,24 +21666,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
ast_debug(1, "Got 200 OK on CANCEL\n");
|
||||
|
||||
/* Wait for 487, then destroy */
|
||||
} else if (sipmethod == SIP_NOTIFY) {
|
||||
/* They got the notify, this is the end */
|
||||
if (p->owner) {
|
||||
if (p->refer) {
|
||||
ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
|
||||
} else
|
||||
ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
|
||||
/* ast_queue_hangup(p->owner); Disabled */
|
||||
} else {
|
||||
if (!p->subscribed && !p->refer) {
|
||||
pvt_set_needdestroy(p, "transaction completed");
|
||||
}
|
||||
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
|
||||
/* Ready to send the next state we have on queue */
|
||||
ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
|
||||
cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
|
||||
}
|
||||
}
|
||||
} else if (sipmethod == SIP_BYE) {
|
||||
pvt_set_needdestroy(p, "transaction completed");
|
||||
}
|
||||
@@ -21697,8 +21687,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
|
||||
handle_response_invite(p, resp, rest, req, seqno);
|
||||
} else if (sipmethod == SIP_BYE) {
|
||||
pvt_set_needdestroy(p, "received 481 response");
|
||||
} else if (sipmethod == SIP_NOTIFY) {
|
||||
pvt_set_needdestroy(p, "received 481 response");
|
||||
} else if (sipdebug) {
|
||||
ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
|
||||
}
|
||||
@@ -30103,6 +30091,12 @@ static int setup_srtp(struct sip_srtp **srtp)
|
||||
|
||||
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
|
||||
{
|
||||
/* If no RTP instance exists for this media stream don't bother processing the crypto line */
|
||||
if (!rtp) {
|
||||
ast_debug(3, "Received offer with crypto line for media stream that is not enabled\n");
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
if (strncasecmp(a, "crypto:", 7)) {
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
@@ -25,3 +25,10 @@ rtpend=20000
|
||||
; do not come from the source of the RTP stream. This option is
|
||||
; enabled by default.
|
||||
; strictrtp=yes
|
||||
;
|
||||
; Number of packets containing consecutive sequence values needed
|
||||
; to change the RTP source socket address. This option only comes
|
||||
; into play while using strictrtp=yes. Consider changing this value
|
||||
; if rtp packets are dropped from one or both ends after a call is
|
||||
; connected. This option is set to 4 by default.
|
||||
; probation=8
|
||||
|
||||
@@ -4110,6 +4110,17 @@ int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct a
|
||||
if (!f || (f->frametype == AST_FRAME_CONTROL &&
|
||||
(f->subclass.integer == AST_CONTROL_HANGUP || f->subclass.integer == AST_CONTROL_BUSY ||
|
||||
f->subclass.integer == AST_CONTROL_CONGESTION))) {
|
||||
/*
|
||||
* If the bridge was broken for a hangup that isn't real, then
|
||||
* then don't run the h extension, because the channel isn't
|
||||
* really hung up. This should really only happen with AST_SOFTHANGUP_ASYNCGOTO,
|
||||
* but it doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either.
|
||||
*/
|
||||
ast_channel_lock(chan);
|
||||
if (chan->_softhangup & (AST_SOFTHANGUP_ASYNCGOTO | AST_SOFTHANGUP_UNBRIDGE)) {
|
||||
ast_set_flag(chan, AST_FLAG_BRIDGE_HANGUP_DONT);
|
||||
}
|
||||
ast_channel_unlock(chan);
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
|
||||
16
main/file.c
16
main/file.c
@@ -1012,6 +1012,7 @@ int ast_streamfile(struct ast_channel *chan, const char *filename, const char *p
|
||||
struct ast_filestream *fs;
|
||||
struct ast_filestream *vfs=NULL;
|
||||
char fmt[256];
|
||||
off_t pos;
|
||||
int seekattempt;
|
||||
int res;
|
||||
|
||||
@@ -1024,12 +1025,17 @@ int ast_streamfile(struct ast_channel *chan, const char *filename, const char *p
|
||||
/* check to see if there is any data present (not a zero length file),
|
||||
* done this way because there is no where for ast_openstream_full to
|
||||
* return the file had no data. */
|
||||
seekattempt = fseek(fs->f, -1, SEEK_END);
|
||||
if (seekattempt && errno == EINVAL) {
|
||||
/* Zero-length file, as opposed to a pipe */
|
||||
return 0;
|
||||
pos = ftello(fs->f);
|
||||
seekattempt = fseeko(fs->f, -1, SEEK_END);
|
||||
if (seekattempt) {
|
||||
if (errno == EINVAL) {
|
||||
/* Zero-length file, as opposed to a pipe */
|
||||
return 0;
|
||||
} else {
|
||||
ast_seekstream(fs, 0, SEEK_SET);
|
||||
}
|
||||
} else {
|
||||
ast_seekstream(fs, 0, SEEK_SET);
|
||||
fseeko(fs->f, pos, SEEK_SET);
|
||||
}
|
||||
|
||||
vfs = ast_openvstream(chan, filename, preflang);
|
||||
|
||||
@@ -80,6 +80,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#define ZFONE_PROFILE_ID 0x505a
|
||||
|
||||
#define DEFAULT_LEARNING_MIN_SEQUENTIAL 4
|
||||
|
||||
extern struct ast_srtp_res *res_srtp;
|
||||
static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
||||
|
||||
@@ -96,7 +98,8 @@ static int rtcpdebugport; /*< Debug only RTCP packets from IP or IP+Port if por
|
||||
#ifdef SO_NO_CHECK
|
||||
static int nochecksums;
|
||||
#endif
|
||||
static int strictrtp;
|
||||
static int strictrtp; /*< Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode. */
|
||||
static int learning_min_sequential; /*< Number of sequential RTP frames needed from a single source during learning mode to accept new source. */
|
||||
|
||||
enum strict_rtp_state {
|
||||
STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
|
||||
@@ -176,6 +179,13 @@ struct ast_rtp {
|
||||
struct ast_sockaddr strict_rtp_address; /*!< Remote address information for strict RTP purposes */
|
||||
struct ast_sockaddr alt_rtp_address; /*!<Alternate remote address information */
|
||||
|
||||
/*
|
||||
* Learning mode values based on pjmedia's probation mode. Many of these values are redundant to the above,
|
||||
* but these are in place to keep learning mode sequence values sealed from their normal counterparts.
|
||||
*/
|
||||
uint16_t learning_max_seq; /*!< Highest sequence number heard */
|
||||
int learning_probation; /*!< Sequential packets untill source is valid */
|
||||
|
||||
struct rtp_red *red;
|
||||
};
|
||||
|
||||
@@ -460,6 +470,50 @@ static int create_new_socket(const char *type, int af)
|
||||
return sock;
|
||||
}
|
||||
|
||||
/*!
|
||||
* \internal
|
||||
* \brief Initializes sequence values and probation for learning mode.
|
||||
* \note This is an adaptation of pjmedia's pjmedia_rtp_seq_init function.
|
||||
*
|
||||
* \param rtp pointer to rtp struct used with the received rtp packet.
|
||||
* \param seq sequence number read from the rtp header
|
||||
*/
|
||||
static void rtp_learning_seq_init(struct ast_rtp *rtp, uint16_t seq)
|
||||
{
|
||||
rtp->learning_max_seq = seq - 1;
|
||||
rtp->learning_probation = learning_min_sequential;
|
||||
}
|
||||
|
||||
/*!
|
||||
* \internal
|
||||
* \brief Updates sequence information for learning mode and determines if probation/learning mode should remain in effect.
|
||||
* \note This function was adapted from pjmedia's pjmedia_rtp_seq_update function.
|
||||
*
|
||||
* \param rtp pointer to rtp struct used with the received rtp packet.
|
||||
* \param seq sequence number read from the rtp header
|
||||
* \return boolean value indicating if probation mode is active at the end of the function
|
||||
*/
|
||||
static int rtp_learning_rtp_seq_update(struct ast_rtp *rtp, uint16_t seq)
|
||||
{
|
||||
int probation = 1;
|
||||
|
||||
ast_debug(1, "%p -- probation = %d, seq = %d\n", rtp, rtp->learning_probation, seq);
|
||||
|
||||
if (seq == rtp->learning_max_seq + 1) {
|
||||
/* packet is in sequence */
|
||||
rtp->learning_probation--;
|
||||
rtp->learning_max_seq = seq;
|
||||
if (rtp->learning_probation == 0) {
|
||||
probation = 0;
|
||||
}
|
||||
} else {
|
||||
rtp->learning_probation = learning_min_sequential - 1;
|
||||
rtp->learning_max_seq = seq;
|
||||
}
|
||||
|
||||
return probation;
|
||||
}
|
||||
|
||||
static int ast_rtp_new(struct ast_rtp_instance *instance,
|
||||
struct ast_sched_context *sched, struct ast_sockaddr *addr,
|
||||
void *data)
|
||||
@@ -476,6 +530,9 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
|
||||
rtp->ssrc = ast_random();
|
||||
rtp->seqno = ast_random() & 0xffff;
|
||||
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
|
||||
if (strictrtp) {
|
||||
rtp_learning_seq_init(rtp, (uint16_t)rtp->seqno);
|
||||
}
|
||||
|
||||
/* Create a new socket for us to listen on and use */
|
||||
if ((rtp->s =
|
||||
@@ -2082,7 +2139,17 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
|
||||
|
||||
/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
|
||||
if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
|
||||
ast_debug(1, "%p -- start learning mode pass with addr = %s\n", rtp, ast_sockaddr_stringify(&addr));
|
||||
/* For now, we always copy the address. */
|
||||
ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
|
||||
|
||||
/* Send the rtp and the seqno from header to rtp_learning_rtp_seq_update to see whether we can exit or not*/
|
||||
if (rtp_learning_rtp_seq_update(rtp, ntohl(rtpheader[0]))) {
|
||||
ast_debug(1, "%p -- Condition for learning hasn't exited, so reject the frame.\n", rtp);
|
||||
return &ast_null_frame;
|
||||
}
|
||||
|
||||
ast_debug(1, "%p -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address %s\n", rtp, ast_sockaddr_stringify(&addr));
|
||||
rtp->strict_rtp_state = STRICT_RTP_CLOSED;
|
||||
} else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
|
||||
if (ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
|
||||
@@ -2497,6 +2564,7 @@ static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct
|
||||
|
||||
if (strictrtp) {
|
||||
rtp->strict_rtp_state = STRICT_RTP_LEARN;
|
||||
rtp_learning_seq_init(rtp, rtp->seqno);
|
||||
}
|
||||
|
||||
return;
|
||||
@@ -2884,6 +2952,7 @@ static int rtp_reload(int reload)
|
||||
rtpend = DEFAULT_RTP_END;
|
||||
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
||||
strictrtp = STRICT_RTP_CLOSED;
|
||||
learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
|
||||
if (cfg) {
|
||||
if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
|
||||
rtpstart = atoi(s);
|
||||
@@ -2927,6 +2996,12 @@ static int rtp_reload(int reload)
|
||||
if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
|
||||
strictrtp = ast_true(s);
|
||||
}
|
||||
if ((s = ast_variable_retrieve(cfg, "general", "probation"))) {
|
||||
if ((sscanf(s, "%d", &learning_min_sequential) <= 0) || learning_min_sequential <= 0) {
|
||||
ast_log(LOG_WARNING, "Value for 'probation' could not be read, using default of '%d' instead\n",
|
||||
DEFAULT_LEARNING_MIN_SEQUENTIAL);
|
||||
}
|
||||
}
|
||||
ast_config_destroy(cfg);
|
||||
}
|
||||
if (rtpstart >= rtpend) {
|
||||
|
||||
Reference in New Issue
Block a user