Importing release summary for 10.1.3 release.

git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.3@356571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Merge 355733, 356476 for 10.1.3
2025-12-19 21:19:03 +00:00 · 2012-02-23 23:37:04 +00:00 · 2012-02-23 23:31:02 +00:00 · 2012-02-23 22:45:01 +00:00 · 2012-02-09 19:30:00 +00:00 · 2012-02-09 19:13:16 +00:00
9 changed files with 701 additions and 986 deletions
--- a/.version
+++ b/.version
@@ -1 +1 @@
-10.1.0
+10.1.3
--- a/98
+++ b/98
@@ -1,3 +1,101 @@
+2012-02-23  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Asterisk 10.1.3 Released.
+
+	* channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+
+	  When we send an ACK for a 2xx response to an INVITE, we are supposed
+	  to use the learned route set. However, when we receive a non-2xx
+	  final response to an INVITE, we are supposed to send the ACK to the
+	  same place we initially sent the INVITE.
+
+	  We had been doing this up until the changes went in that would build
+	  a route set from provisional responses. That introduced a regression
+	  where we would use the learned route set under all circumstances.
+
+	  With this change, we now will set the destination of our ACK based on
+	  the invitestate. If it is INV_COMPLETED then that means that we have
+	  received a non-2xx final response (INV_TERMINATED indicates a 2xx
+	  response was received).  If it is INV_CANCELLED, then that means the
+	  call is being canceled, which means that we should be ACKing a 487
+	  response.
+
+	  The other change introduced here is setting the invitestate to
+	  INV_CONFIRMED when we send an ACK *after* the reqprep instead of
+	  before. This way, we can tell in reqprep more easily what the
+	  invitestate is prior to sending the ACK.
+
+	  (closes issue ASTERISK-19389)
+ 	  reported by Karsten Wemheuer
+	  patches:
+	    ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
+
+	* channels/chan_sip.c: Fix regressions with regards to route-set
+	  creation on early dialogs.
+
+	  This fixes two main issues:
+	  1. Asterisk would send a CANCEL to the route created by the provisional
+	     response instead of using the same destination it did in the initial
+	     INVITE.
+	  2. If a new route set arrives in a 200 OK than was in the 1XX response
+	     (perfectly possible if our outbound INVITE gets forked), then the
+	     route set in the 200 OK needs to overwrite the route set in the 1XX
+	     response.
+	  (closes issue ASTERISK-19358)
+	  Reported by: Karsten Wemheuer
+	  Tested by: Karsten Wemheuer
+	  patches:
+	   ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
+	   ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
+
+ 	  Review: https://reviewboard.asterisk.org/r/1749
+
+2012-02-09  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Asterisk 10.1.2 Released.
+
+	* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+	  codes.  In ASTERISK-18924, SIP INFO DTMF handling was changed to
+	  account for both lowercase alphabetic DTMF events, as well as
+	  uppercase alphabetic DTMF events.  When this occurred, the comparison
+	  of the character buffer containing the event code was changed such
+	  that the buffer was first compared against '0' and '9' to determine if
+	  it was numeric.  Unfortunately, since the first character in the
+	  buffer will typically be '1' in the case of non-numeric event codes
+	  (10-16), this caused those codes to be converted to a DTMF event of
+	  '1'.  This patch fixes that, and cleans up handling of both
+	  application/dtmf-relay and application/dtmf content types.
+	  Review: https://reviewboard.asterisk.org/r/1722/
+	  (closes issue ASTERISK-19290) Reported by: Ira Emus
+	  Tested by: mjordan
+
+	* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
+	  uninitiliazed caller_id storage (closes issue ASTERISK-19311)
+	  Reported by: tootai
+	  Tested by: rmudgett
+
+2012-02-06  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Asterisk 10.1.1 Released.
+
+	* channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to
+	  r335976. Bad locking order was added to chan_agent to prevent
+	  segfaults from having no locking in a patch by irroot. This patch
+	  addresses the bad locking order by releasing locks before getting the
+	  right locking order to stop deadlocks from occuring when doing
+	  multiple interactions with agents. (closes issue ASTERISK-19285)
+	  Reported by: Alex Villacis Lasso
+	  Review: https://reviewboard.asterisk.org/r/1708/
+
+	* channels/chan_sip.c: Ensure entering T.38 passthrough does not cause
+	  an infinite loop. After R340970 Asterisk was still polling the RTCP
+	  file descriptor after RTCP is shut down and removed. If the
+	  descriptor happened to have data ready when the removal occured then
+	  Asterisk would go into an infinite loop trying to read data that it
+	  can never actually access. This change disables the audio RTCP file
+	  descriptor for the duration of the T.38 transaction. (closes issue
+	  ASTERISK-18951) Reported-by: Kristijan Vrban
+
 2012-01-27  Asterisk Development Team <asteriskteam@digium.com>

 	* Asterisk 10.1.0 Released.
--- a/apps/app_parkandannounce.c
+++ b/apps/app_parkandannounce.c
@@ -150,6 +150,7 @@ static int parkandannounce_exec(struct ast_channel *chan, const char *data)
 	}

 	/* Save the CallerID because the masquerade turns chan into a ZOMBIE. */
+	ast_party_id_init(&caller_id);
 	ast_channel_lock(chan);
 	ast_party_id_copy(&caller_id, &chan->caller.id);
 	ast_channel_unlock(chan);
--- a/asterisk-10.1.0-summary.html
+++ b/asterisk-10.1.0-summary.html
@@ -1,287 +0,0 @@
-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
-<html xmlns="http://www.w3.org/1999/xhtml">
-<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.1.0</title></head>
-<body>
-<h1 align="center"><a name="top">Release Summary</a></h1>
-<h3 align="center">asterisk-10.1.0</h3>
-<h3 align="center">Date: 2012-01-27</h3>
-<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
-<hr/>
-<h2 align="center">Table of Contents</h2>
-<ol>
-   <li><a href="#summary">Summary</a></li>
-   <li><a href="#contributors">Contributors</a></li>
-   <li><a href="#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-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>
--- a/asterisk-10.1.0-summary.txt
+++ b/asterisk-10.1.0-summary.txt
@@ -1,571 +0,0 @@
-                                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(-)
-
-     ----------------------------------------------------------------------
--- a/asterisk-10.1.3-summary.html
+++ b/asterisk-10.1.3-summary.html
@@ -0,0 +1,137 @@
+<!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-10.1.3</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-10.1.3</h3>
+<h3 align="center">Date: 2012-02-23</h3>
+<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+   <li><a href="#summary">Summary</a></li>
+   <li><a href="#contributors">Contributors</a></li>
+   <li><a href="#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-10.1.2.</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>
+13 seanbright<br/>
+8 rmudgett<br/>
+4 pabelanger<br/>
+3 kmoore<br/>
+3 mmichelson<br/>
+2 file<br/>
+2 qwell<br/>
+2 twilson<br/>
+1 alecdavis<br/>
+1 may<br/>
+1 mjordan<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=354656">354656</a></td><td>kmoore</td><td>Make the config parser remove escaping backslashes</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17121">ASTERISK-17121</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354703">354703</a></td><td>kmoore</td><td>Fix parsing of SIP headers where compact and non-compact headers are mixed</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17192">ASTERISK-17192</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354750">354750</a></td><td>twilson</td><td>Note that CDRs are immutable once a bridge is torn down</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16923">ASTERISK-16923</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354836">354836</a></td><td>rmudgett</td><td>Fix AMI Redirect ExtraChannel not redirecting to the same exten and context.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16974">ASTERISK-16974</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354890">354890</a></td><td>qwell</td><td>Fix a voicemail memory leak with heard/deleted messages.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354938">354938</a></td><td>file</td><td>Don't try to play sound files that do not exist.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19188">ASTERISK-19188</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354959">354959</a></td><td>rmudgett</td><td>Fix reconnecting to pgsql database after connection loss.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16982">ASTERISK-16982</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355010">355010</a></td><td>file</td><td>Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355057">355057</a></td><td>rmudgett</td><td>Fix occasional incorrectly delayed call-file execution.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19081">ASTERISK-19081</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355137">355137</a></td><td>may</td><td>call manager_event only if there is not null channel structure</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19298">ASTERISK-19298</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355183">355183</a></td><td>seanbright</td><td>Clear the high order bit from the destination call number before sending.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355229">355229</a></td><td>qwell</td><td>Don't enable sqlite3 CDRs by default in sample configs.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355271">355271</a></td><td>mmichelson</td><td>Properly invert the return of a strncmp call.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355320">355320</a></td><td>rmudgett</td><td>Fix lock typo that should be unlock in cel_sqlite_custom reload.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19356">ASTERISK-19356</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355375">355375</a></td><td>rmudgett</td><td>Fix voicemail problems when using ogg/vorbis.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16926">ASTERISK-16926</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355449">355449</a></td><td>seanbright</td><td>Use TRUNK_CALL_START as originally intended.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355467">355467</a></td><td>seanbright</td><td>Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355530">355530</a></td><td>seanbright</td><td>When IAX2 debugging is enabled, make sure to log 'apathetic' messages too.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355575">355575</a></td><td>rmudgett</td><td>Fix AMI Monitor action without File header converting channel name into filename.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355620">355620</a></td><td>rmudgett</td><td>Fix compile problem when old version of libvorbisfile v1.1.2 is used.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19370">ASTERISK-19370</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355623">355623</a></td><td>seanbright</td><td>Revert a change to audio_audiohook_write_list that had no affect.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355733">355733</a></td><td>mmichelson</td><td>Fix regressions with regards to route-set creation on early dialogs.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19358">ASTERISK-19358</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355747">355747</a></td><td>seanbright</td><td>Pass the correct value to ast_timer_set_rate() for IAX2 trunking.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355794">355794</a></td><td>seanbright</td><td>Don't allow trunkfreq to be greater than 1000ms.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355851">355851</a></td><td>alecdavis</td><td>push 'outgoing' flag from sig_XXX up to chan_dahdi</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19316">ASTERISK-19316</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355895">355895</a></td><td>pabelanger</td><td>Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355896">355896</a></td><td>pabelanger</td><td>Revert commit</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355902">355902</a></td><td>seanbright</td><td>Set the length of the ast_sockaddr, so that we can set it's port later.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355905">355905</a></td><td>seanbright</td><td>Add some boilerplate documentation for IAXVAR and IAXPEER.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355950">355950</a></td><td>seanbright</td><td>Change some debug messages from LOG_DEBUG to ast_debug.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355953">355953</a></td><td>seanbright</td><td>This was a LOG_NOTICE, so roll it back.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355998">355998</a></td><td>seanbright</td><td>Remove spurious warning when 'qualifyfreqnotok' is set successfully.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17176">ASTERISK-17176</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356074">356074</a></td><td>kmoore</td><td>Add missing newline to ccss state change notification</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356108">356108</a></td><td>seanbright</td><td>Make 'iax2 show callnumber usage' output make sense when an IP is passed in.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356215">356215</a></td><td>mjordan</td><td>Fix potential buffer overrun and memory leak when executing "sip show peers"</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19231">ASTERISK-19231</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19361">ASTERISK-19361</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356297">356297</a></td><td>twilson</td><td>Track module use count for res_calendar</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356428">356428</a></td><td>pabelanger</td><td>Multiple revisions 356290,356335,356337</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356431">356431</a></td><td>pabelanger</td><td>Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356476">356476</a></td><td>mmichelson</td><td>Fix ACK routing for non-2xx responses.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19389">ASTERISK-19389</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356522">356522</a></td><td>rmudgett</td><td>Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19322">ASTERISK-19322</a></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+addons/chan_ooh323.c                   |    7
+apps/app_confbridge.c                  |    8
+apps/app_rpt.c                         |   69 +----
+apps/app_voicemail.c                   |   23 -
+autoconf/ast_c_declare_check.m4        |   31 ++
+cel/cel_sqlite3_custom.c               |    2
+channels/chan_dahdi.c                  |   10
+channels/chan_iax2.c                   |   62 +++--
+channels/chan_sip.c                    |  115 +++++----
+channels/sig_analog.c                  |   16 -
+channels/sig_analog.h                  |    1
+channels/sig_pri.c                     |   16 -
+channels/sig_pri.h                     |    1
+channels/sig_ss7.c                     |   14 -
+channels/sig_ss7.h                     |    1
+configs/cdr_sqlite3_custom.conf.sample |    8
+configs/extconfig.conf.sample          |    4
+configs/iax.conf.sample                |    3
+configure.ac                           |    5
+formats/format_ogg_vorbis.c            |  399 +++++++++++----------------------
+funcs/func_cdr.c                       |    4
+include/asterisk/autoconfig.h.in       |   16 -
+include/asterisk/calendar.h            |    2
+main/audiohook.c                       |    4
+main/ccss.c                            |    2
+main/config.c                          |    4
+main/features.c                        |   81 +++---
+main/loader.c                          |    4
+main/manager.c                         |    7
+pbx/pbx_config.c                       |   10
+pbx/pbx_spool.c                        |   41 ++-
+res/res_calendar.c                     |    2
+res/res_config_pgsql.c                 |   66 +++++
+res/res_monitor.c                      |   22 +
+34 files changed, 567 insertions(+), 493 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
--- a/asterisk-10.1.3-summary.txt
+++ b/asterisk-10.1.3-summary.txt
@@ -0,0 +1,257 @@
+                                Release Summary
+
+                                asterisk-10.1.3
+
+                                Date: 2012-02-23
+
+                           <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-10.1.2.
+
+     ----------------------------------------------------------------------
+
+                                  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              
+   13 seanbright            
+   8 rmudgett               
+   4 pabelanger             
+   3 kmoore                 
+   3 mmichelson             
+   2 file                   
+   2 qwell                  
+   2 twilson                
+   1 alecdavis              
+   1 may                    
+   1 mjordan                
+
+     ----------------------------------------------------------------------
+
+                      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     |
+   |--------+----------+------------------------------------+---------------|
+   |354656  |kmoore    |Make the config parser remove       |ASTERISK-17121 |
+   |        |          |escaping backslashes                |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Fix parsing of SIP headers where    |               |
+   |354703  |kmoore    |compact and non-compact headers are |ASTERISK-17192 |
+   |        |          |mixed                               |               |
+   |--------+----------+------------------------------------+---------------|
+   |354750  |twilson   |Note that CDRs are immutable once a |ASTERISK-16923 |
+   |        |          |bridge is torn down                 |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Fix AMI Redirect ExtraChannel not   |               |
+   |354836  |rmudgett  |redirecting to the same exten and   |ASTERISK-16974 |
+   |        |          |context.                            |               |
+   |--------+----------+------------------------------------+---------------|
+   |354890  |qwell     |Fix a voicemail memory leak with    |               |
+   |        |          |heard/deleted messages.             |               |
+   |--------+----------+------------------------------------+---------------|
+   |354938  |file      |Don't try to play sound files that  |ASTERISK-19188 |
+   |        |          |do not exist.                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |354959  |rmudgett  |Fix reconnecting to pgsql database  |ASTERISK-16982 |
+   |        |          |after connection loss.              |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Only allow one 'dialplan reload' to |               |
+   |355010  |file      |execute at a time as otherwise they |               |
+   |        |          |would share the same common local   |               |
+   |        |          |context list.                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |355057  |rmudgett  |Fix occasional incorrectly delayed  |ASTERISK-19081 |
+   |        |          |call-file execution.                |               |
+   |--------+----------+------------------------------------+---------------|
+   |355137  |may       |call manager_event only if there is |ASTERISK-19298 |
+   |        |          |not null channel structure          |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Clear the high order bit from the   |               |
+   |355183  |seanbright|destination call number before      |               |
+   |        |          |sending.                            |               |
+   |--------+----------+------------------------------------+---------------|
+   |355229  |qwell     |Don't enable sqlite3 CDRs by default|               |
+   |        |          |in sample configs.                  |               |
+   |--------+----------+------------------------------------+---------------|
+   |355271  |mmichelson|Properly invert the return of a     |               |
+   |        |          |strncmp call.                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |355320  |rmudgett  |Fix lock typo that should be unlock |ASTERISK-19356 |
+   |        |          |in cel_sqlite_custom reload.        |               |
+   |--------+----------+------------------------------------+---------------|
+   |355375  |rmudgett  |Fix voicemail problems when using   |ASTERISK-16926 |
+   |        |          |ogg/vorbis.                         |               |
+   |--------+----------+------------------------------------+---------------|
+   |355449  |seanbright|Use TRUNK_CALL_START as originally  |               |
+   |        |          |intended.                           |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Only use maxtrunkcall and           |               |
+   |355467  |seanbright|maxnontrunkcall in chan_iax2 if     |               |
+   |        |          |IAX_OLD_FIND is specified.          |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |When IAX2 debugging is enabled, make|               |
+   |355530  |seanbright|sure to log 'apathetic' messages    |               |
+   |        |          |too.                                |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Fix AMI Monitor action without File |               |
+   |355575  |rmudgett  |header converting channel name into |               |
+   |        |          |filename.                           |               |
+   |--------+----------+------------------------------------+---------------|
+   |355620  |rmudgett  |Fix compile problem when old version|ASTERISK-19370 |
+   |        |          |of libvorbisfile v1.1.2 is used.    |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Revert a change to                  |               |
+   |355623  |seanbright|audio_audiohook_write_list that had |               |
+   |        |          |no affect.                          |               |
+   |--------+----------+------------------------------------+---------------|
+   |355733  |mmichelson|Fix regressions with regards to     |ASTERISK-19358 |
+   |        |          |route-set creation on early dialogs.|               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Pass the correct value to           |               |
+   |355747  |seanbright|ast_timer_set_rate() for IAX2       |               |
+   |        |          |trunking.                           |               |
+   |--------+----------+------------------------------------+---------------|
+   |355794  |seanbright|Don't allow trunkfreq to be greater |               |
+   |        |          |than 1000ms.                        |               |
+   |--------+----------+------------------------------------+---------------|
+   |355851  |alecdavis |push 'outgoing' flag from sig_XXX up|ASTERISK-19316 |
+   |        |          |to chan_dahdi                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |355895  |pabelanger|Fix -Werror=unused-but-set-variable |               |
+   |        |          |compiler error (gcc 4.6.2)          |               |
+   |--------+----------+------------------------------------+---------------|
+   |355896  |pabelanger|Revert commit                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |355902  |seanbright|Set the length of the ast_sockaddr, |               |
+   |        |          |so that we can set it's port later. |               |
+   |--------+----------+------------------------------------+---------------|
+   |355905  |seanbright|Add some boilerplate documentation  |               |
+   |        |          |for IAXVAR and IAXPEER.             |               |
+   |--------+----------+------------------------------------+---------------|
+   |355950  |seanbright|Change some debug messages from     |               |
+   |        |          |LOG_DEBUG to ast_debug.             |               |
+   |--------+----------+------------------------------------+---------------|
+   |355953  |seanbright|This was a LOG_NOTICE, so roll it   |               |
+   |        |          |back.                               |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Remove spurious warning when        |               |
+   |355998  |seanbright|'qualifyfreqnotok' is set           |ASTERISK-17176 |
+   |        |          |successfully.                       |               |
+   |--------+----------+------------------------------------+---------------|
+   |356074  |kmoore    |Add missing newline to ccss state   |               |
+   |        |          |change notification                 |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Make 'iax2 show callnumber usage'   |               |
+   |356108  |seanbright|output make sense when an IP is     |               |
+   |        |          |passed in.                          |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Fix potential buffer overrun and    |ASTERISK-19231,|
+   |356215  |mjordan   |memory leak when executing "sip show|ASTERISK-19361 |
+   |        |          |peers"                              |               |
+   |--------+----------+------------------------------------+---------------|
+   |356297  |twilson   |Track module use count for          |               |
+   |        |          |res_calendar                        |               |
+   |--------+----------+------------------------------------+---------------|
+   |356428  |pabelanger|Multiple revisions                  |               |
+   |        |          |356290,356335,356337                |               |
+   |--------+----------+------------------------------------+---------------|
+   |356431  |pabelanger|Fix -Werror=unused-but-set-variable |               |
+   |        |          |compiler error (gcc 4.6.2)          |               |
+   |--------+----------+------------------------------------+---------------|
+   |356476  |mmichelson|Fix ACK routing for non-2xx         |ASTERISK-19389 |
+   |        |          |responses.                          |               |
+   |--------+----------+------------------------------------+---------------|
+   |        |          |Fix blind transfer parking issues if|               |
+   |356522  |rmudgett  |the dialed extension is not         |ASTERISK-19322 |
+   |        |          |recognized as a parking extension.  |               |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                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.
+
+ addons/chan_ooh323.c                   |    7
+ apps/app_confbridge.c                  |    8
+ apps/app_rpt.c                         |   69 +----
+ apps/app_voicemail.c                   |   23 -
+ autoconf/ast_c_declare_check.m4        |   31 ++
+ cel/cel_sqlite3_custom.c               |    2
+ channels/chan_dahdi.c                  |   10
+ channels/chan_iax2.c                   |   62 +++--
+ channels/chan_sip.c                    |  115 +++++----
+ channels/sig_analog.c                  |   16 -
+ channels/sig_analog.h                  |    1
+ channels/sig_pri.c                     |   16 -
+ channels/sig_pri.h                     |    1
+ channels/sig_ss7.c                     |   14 -
+ channels/sig_ss7.h                     |    1
+ configs/cdr_sqlite3_custom.conf.sample |    8
+ configs/extconfig.conf.sample          |    4
+ configs/iax.conf.sample                |    3
+ configure.ac                           |    5
+ formats/format_ogg_vorbis.c            |  399 +++++++++++----------------------
+ funcs/func_cdr.c                       |    4
+ include/asterisk/autoconfig.h.in       |   16 -
+ include/asterisk/calendar.h            |    2
+ main/audiohook.c                       |    4
+ main/ccss.c                            |    2
+ main/config.c                          |    4
+ main/features.c                        |   81 +++---
+ main/loader.c                          |    4
+ main/manager.c                         |    7
+ pbx/pbx_config.c                       |   10
+ pbx/pbx_spool.c                        |   41 ++-
+ res/res_calendar.c                     |    2
+ res/res_config_pgsql.c                 |   66 +++++
+ res/res_monitor.c                      |   22 +
+ 34 files changed, 567 insertions(+), 493 deletions(-)
+
+     ----------------------------------------------------------------------
--- a/channels/chan_agent.c
+++ b/channels/chan_agent.c
@@ -1,7 +1,7 @@
 /*
 * Asterisk -- An open source telephony toolkit.
 *
- * Copyright (C) 1999 - 2006, Digium, Inc.
+ * Copyright (C) 1999 - 2012, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
@@ -373,6 +373,43 @@ static struct ast_channel_tech agent_tech = {
 	.set_base_channel = agent_set_base_channel,
 };

+/*!
+ * \brief Locks the owning channel for a LOCKED pvt while obeying locking order. The pvt
+ * must enter this function locked and will be returned locked, but this function will
+ * unlock the pvt for a short time, so it can't be used while expecting the pvt to remain
+ * static. If function returns a non NULL channel, it will need to be unlocked and
+ * unrefed once it is no longer needed.
+ *
+ * \param pvt Pointer to the LOCKED agent_pvt for which the owner is needed
+ * \ret locked channel which owns the pvt at the time of completion. NULL if not available.
+ */
+static struct ast_channel *agent_lock_owner(struct agent_pvt *pvt)
+{
+	struct ast_channel *owner;
+
+	for (;;) {
+		if (!pvt->owner) { /* No owner. Nothing to do. */
+			return NULL;
+		}
+
+		/* If we don't ref the owner, it could be killed when we unlock the pvt. */
+		owner = ast_channel_ref(pvt->owner);
+
+		/* Locking order requires us to lock channel, then pvt. */
+		ast_mutex_unlock(&pvt->lock);
+		ast_channel_lock(owner);
+		ast_mutex_lock(&pvt->lock);
+
+		/* Check if owner changed during pvt unlock period */
+		if (owner != pvt->owner) { /* Channel changed. Unref and do another pass. */
+			ast_channel_unlock(owner);
+			owner = ast_channel_unref(owner);
+		} else { /* Channel stayed the same. Return it. */
+			return owner;
+		}
+	}
+}
+
 /*!
 * Adds an agent to the global list of agents.
 *
@@ -553,7 +590,11 @@ static struct ast_frame *agent_read(struct ast_channel *ast)
 	struct ast_frame *f = NULL;
 	static struct ast_frame answer_frame = { AST_FRAME_CONTROL, { AST_CONTROL_ANSWER } };
 	int cur_time = time(NULL);
+	struct ast_channel *owner;
+
 	ast_mutex_lock(&p->lock);
+	owner = agent_lock_owner(p);
+
 	CHECK_FORMATS(ast, p);
 	if (!p->start) {
 		p->start = cur_time;
@@ -583,13 +624,11 @@ static struct ast_frame *agent_read(struct ast_channel *ast)
 			int howlong = cur_time - p->start;
 			if (p->autologoff && (howlong >= p->autologoff)) {
 				ast_log(LOG_NOTICE, "Agent '%s' didn't answer/confirm within %d seconds (waited %d)\n", p->name, p->autologoff, howlong);
-				if (p->owner || p->chan) {
-					while (p->owner && ast_channel_trylock(p->owner)) {
-						DEADLOCK_AVOIDANCE(&p->lock);
-					}
-					if (p->owner) {
-						ast_softhangup(p->owner, AST_SOFTHANGUP_EXPLICIT);
-						ast_channel_unlock(p->owner);
+				if (owner || p->chan) {
+					if (owner) {
+						ast_softhangup(owner, AST_SOFTHANGUP_EXPLICIT);
+						ast_channel_unlock(owner);
+						owner = ast_channel_unref(owner);
 					}

 					while (p->chan && ast_channel_trylock(p->chan)) {
@@ -651,6 +690,11 @@ static struct ast_frame *agent_read(struct ast_channel *ast)
 		}
 	}

+	if (owner) {
+		ast_channel_unlock(owner);
+		owner = ast_channel_unref(owner);
+	}
+
 	CLEANUP(ast,p);
 	if (p->chan && !p->chan->_bridge) {
 		if (strcasecmp(p->chan->tech->type, "Local")) {
@@ -888,6 +932,14 @@ int agent_set_base_channel(struct ast_channel *chan, struct ast_channel *base)
 static int agent_hangup(struct ast_channel *ast)
 {
 	struct agent_pvt *p = ast->tech_pvt;
+	struct ast_channel *indicate_chan = NULL;
+	char *tmp_moh; /* moh buffer for indicating after unlocking p */
+
+	if (p->pending) {
+		AST_LIST_LOCK(&agents);
+		AST_LIST_REMOVE(&agents, p, list);
+		AST_LIST_UNLOCK(&agents);
+	}

 	ast_mutex_lock(&p->lock);
 	p->owner = NULL;
@@ -910,7 +962,7 @@ static int agent_hangup(struct ast_channel *ast)
 	if (p->start && (ast->_state != AST_STATE_UP)) {
 		p->start = 0;
 	} else
-		p->start = 0; 
+		p->start = 0;
 	if (p->chan) {
 		p->chan->_bridge = NULL;
 		/* If they're dead, go ahead and hang up on the agent now */
@@ -919,15 +971,21 @@ static int agent_hangup(struct ast_channel *ast)
 			ast_softhangup(p->chan, AST_SOFTHANGUP_EXPLICIT);
 			ast_channel_unlock(p->chan);
 		} else if (p->loginstart) {
-			ast_channel_lock(p->chan);
-			ast_indicate_data(p->chan, AST_CONTROL_HOLD, 
-				S_OR(p->moh, NULL),
-				!ast_strlen_zero(p->moh) ? strlen(p->moh) + 1 : 0);
-			ast_channel_unlock(p->chan);
+			indicate_chan = ast_channel_ref(p->chan);
+			tmp_moh = ast_strdupa(p->moh);
 		}
 	}
 	ast_mutex_unlock(&p->lock);

+	if (indicate_chan) {
+		ast_channel_lock(indicate_chan);
+		ast_indicate_data(indicate_chan, AST_CONTROL_HOLD,
+			S_OR(tmp_moh, NULL),
+			!ast_strlen_zero(tmp_moh) ? strlen(tmp_moh) + 1 : 0);
+		ast_channel_unlock(indicate_chan);
+		indicate_chan = ast_channel_unref(indicate_chan);
+	}
+
 	/* Only register a device state change if the agent is still logged in */
 	if (!p->loginstart) {
 		p->logincallerid[0] = '\0';
@@ -935,11 +993,6 @@ static int agent_hangup(struct ast_channel *ast)
 		ast_devstate_changed(AST_DEVICE_NOT_INUSE, "Agent/%s", p->agent);
 	}

-	if (p->pending) {
-		AST_LIST_LOCK(&agents);
-		AST_LIST_REMOVE(&agents, p, list);
-		AST_LIST_UNLOCK(&agents);
-	}
 	if (p->abouttograb) {
 		/* Let the "about to grab" thread know this isn't valid anymore, and let it
 		   kill it later */
@@ -1492,6 +1545,8 @@ static force_inline int powerof(unsigned int d)
 /*!
 * Lists agents and their status to the Manager API.
 * It is registered on load_module() and it gets called by the manager backend.
+ * This function locks both the pvt and the channel that owns it for a while, but
+ * does not keep these locks.
 * \param s
 * \param m
 * \returns 
@@ -1514,7 +1569,9 @@ static int action_agents(struct mansession *s, const struct message *m)
 	astman_send_ack(s, m, "Agents will follow");
 	AST_LIST_LOCK(&agents);
 	AST_LIST_TRAVERSE(&agents, p, list) {
-        	ast_mutex_lock(&p->lock);
+		struct ast_channel *owner;
+		ast_mutex_lock(&p->lock);
+		owner = agent_lock_owner(p);

 		/* Status Values:
 		   AGENT_LOGGEDOFF - Agent isn't logged in
@@ -1529,16 +1586,14 @@ static int action_agents(struct mansession *s, const struct message *m)

 		if (p->chan) {
 			loginChan = ast_strdupa(p->chan->name);
-			if (p->owner && p->owner->_bridge) {
+			if (owner && owner->_bridge) {
 				talkingto = S_COR(p->chan->caller.id.number.valid,
 					p->chan->caller.id.number.str, "n/a");
-				ast_channel_lock(p->owner);
-				if ((bridge = ast_bridged_channel(p->owner))) {
+				if ((bridge = ast_bridged_channel(owner))) {
 					talkingtoChan = ast_strdupa(bridge->name);
 				} else {
 					talkingtoChan = "n/a";
 				}
-				ast_channel_unlock(p->owner);
 				status = "AGENT_ONCALL";
 			} else {
 				talkingto = "n/a";
@@ -1552,6 +1607,11 @@ static int action_agents(struct mansession *s, const struct message *m)
 			status = "AGENT_LOGGEDOFF";
 		}

+		if (owner) {
+			ast_channel_unlock(owner);
+			owner = ast_channel_unref(owner);
+		}
+
 		astman_append(s, "Event: Agents\r\n"
 			"Agent: %s\r\n"
 			"Name: %s\r\n"
@@ -1583,14 +1643,14 @@ static int agent_logoff(const char *agent, int soft)
 			ret = 0;
 			if (p->owner || p->chan) {
 				if (!soft) {
+					struct ast_channel *owner;
 					ast_mutex_lock(&p->lock);
+					owner = agent_lock_owner(p);

-					while (p->owner && ast_channel_trylock(p->owner)) {
-						DEADLOCK_AVOIDANCE(&p->lock);
-					}
-					if (p->owner) {
-						ast_softhangup(p->owner, AST_SOFTHANGUP_EXPLICIT);
-						ast_channel_unlock(p->owner);
+					if (owner) {
+						ast_softhangup(owner, AST_SOFTHANGUP_EXPLICIT);
+						ast_channel_unlock(owner);
+						owner = ast_channel_unref(owner);
 					}

 					while (p->chan && ast_channel_trylock(p->chan)) {
@@ -1727,7 +1787,9 @@ static char *agents_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *

 	AST_LIST_LOCK(&agents);
 	AST_LIST_TRAVERSE(&agents, p, list) {
+		struct ast_channel *owner;
 		ast_mutex_lock(&p->lock);
+		owner = agent_lock_owner(p);
 		if (p->pending) {
 			if (p->group)
 				ast_cli(a->fd, "-- Pending call to group %d\n", powerof(p->group));
@@ -1740,10 +1802,11 @@ static char *agents_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *
 				username[0] = '\0';
 			if (p->chan) {
 				snprintf(location, sizeof(location), "logged in on %s", p->chan->name);
-				if (p->owner && ast_bridged_channel(p->owner))
+				if (owner && ast_bridged_channel(owner)) {
 					snprintf(talkingto, sizeof(talkingto), " talking to %s", ast_bridged_channel(p->owner)->name);
-				 else 
+				} else {
 					strcpy(talkingto, " is idle");
+				}
 				online_agents++;
 			} else {
 				strcpy(location, "not logged in");
@@ -1756,6 +1819,11 @@ static char *agents_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *
 				username, location, talkingto, music);
 			count_agents++;
 		}
+
+		if (owner) {
+			ast_channel_unlock(owner);
+			owner = ast_channel_unref(owner);
+		}
 		ast_mutex_unlock(&p->lock);
 	}
 	AST_LIST_UNLOCK(&agents);
@@ -1796,21 +1864,32 @@ static char *agents_show_online(struct ast_cli_entry *e, int cmd, struct ast_cli

 	AST_LIST_LOCK(&agents);
 	AST_LIST_TRAVERSE(&agents, p, list) {
+		struct ast_channel *owner;
+
 		agent_status = 0;       /* reset it to offline */
 		ast_mutex_lock(&p->lock);
+		owner = agent_lock_owner(p);
+
 		if (!ast_strlen_zero(p->name))
 			snprintf(username, sizeof(username), "(%s) ", p->name);
 		else
 			username[0] = '\0';
 		if (p->chan) {
 			snprintf(location, sizeof(location), "logged in on %s", p->chan->name);
-			if (p->owner && ast_bridged_channel(p->owner)) 
-				snprintf(talkingto, sizeof(talkingto), " talking to %s", ast_bridged_channel(p->owner)->name);
-			else 
+			if (owner && ast_bridged_channel(owner)) {
+				snprintf(talkingto, sizeof(talkingto), " talking to %s", ast_bridged_channel(owner)->name);
+			} else {
 				strcpy(talkingto, " is idle");
+			}
 			agent_status = 1;
 			online_agents++;
 		}
+
+		if (owner) {
+			ast_channel_unlock(owner);
+			owner = ast_channel_unref(owner);
+		}
+
 		if (!ast_strlen_zero(p->moh))
 			snprintf(music, sizeof(music), " (musiconhold is '%s')", p->moh);
 		if (agent_status)
@@ -2386,12 +2465,16 @@ static int agents_data_provider_get(const struct ast_data_search *search,

 	AST_LIST_LOCK(&agents);
 	AST_LIST_TRAVERSE(&agents, p, list) {
+		struct ast_channel *owner;
+
 		data_agent = ast_data_add_node(data_root, "agent");
 		if (!data_agent) {
 			continue;
 		}

 		ast_mutex_lock(&p->lock);
+		owner = agent_lock_owner(p);
+
 		if (!(p->pending)) {
 			ast_data_add_str(data_agent, "id", p->agent);
 			ast_data_add_structure(agent_pvt, data_agent, p);
@@ -2402,17 +2485,25 @@ static int agents_data_provider_get(const struct ast_data_search *search,
 				if (!data_channel) {
 					ast_mutex_unlock(&p->lock);
 					ast_data_remove_node(data_root, data_agent);
+					if (owner) {
+						ast_channel_unlock(owner);
+						owner = ast_channel_unref(owner);
+					}
 					continue;
 				}
 				ast_channel_data_add_structure(data_channel, p->chan, 0);
-				if (p->owner && ast_bridged_channel(p->owner)) {
+				if (owner && ast_bridged_channel(owner)) {
 					data_talkingto = ast_data_add_node(data_agent, "talkingto");
 					if (!data_talkingto) {
 						ast_mutex_unlock(&p->lock);
 						ast_data_remove_node(data_root, data_agent);
+						if (owner) {
+							ast_channel_unlock(owner);
+							owner = ast_channel_unref(owner);
+						}
 						continue;
 					}
-					ast_channel_data_add_structure(data_talkingto, ast_bridged_channel(p->owner), 0);
+					ast_channel_data_add_structure(data_talkingto, ast_bridged_channel(owner), 0);
 				}
 			} else {
 				ast_data_add_node(data_agent, "talkingto");
@@ -2420,6 +2511,12 @@ static int agents_data_provider_get(const struct ast_data_search *search,
 			}
 			ast_data_add_str(data_agent, "musiconhold", p->moh);
 		}
+
+		if (owner) {
+			ast_channel_unlock(owner);
+			owner = ast_channel_unref(owner);
+		}
+
 		ast_mutex_unlock(&p->lock);

 		/* if this agent doesn't match remove the added agent. */
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1283,7 +1283,7 @@ static int auto_congest(const void *arg);
 static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method);
 static void free_old_route(struct sip_route *route);
 static void list_route(struct sip_route *route);
-static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
+static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp);
 static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
 					      struct sip_request *req, const char *uri);
 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
@@ -8023,7 +8023,7 @@ static void forked_invite_init(struct sip_request *req, const char *new_theirtag
 	ast_string_field_set(p, our_contact, original->our_contact);
 	ast_string_field_set(p, fullcontact, original->fullcontact);
 	parse_ok_contact(p, req);
-	build_route(p, req, 1);
+	build_route(p, req, 1, 0);

 	transmit_request(p, SIP_ACK, p->ocseq, XMIT_UNRELIABLE, TRUE);
 	transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
@@ -9311,6 +9311,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			/* Ensure RTCP is enabled since it may be inactive
 			   if we're coming back from a T.38 session */
 			ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+			/* Ensure audio RTCP reads are enabled */
+			if (p->owner) {
+				ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+			}

 			if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
 				ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -9327,6 +9331,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		} else if (udptlportno > 0) {
 			if (debug)
 				ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
+			/* Prevent audio RTCP reads */
+			if (p->owner) {
+				ast_channel_set_fd(p->owner, 1, -1);
+			}
 			/* Silence RTCP while audio RTP is inactive */
 			ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
 		} else {
@@ -10465,7 +10473,15 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
 	snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);

 	add_header(req, "Via", p->via);
-	if (p->route) {
+	/*
+	 * Use the learned route set unless this is a CANCEL on an ACK for a non-2xx
+	 * final response. For a CANCEL or ACK, we have to send to the same destination
+	 * as the original INVITE.
+	 */
+	if (sipmethod == SIP_CANCEL ||
+			(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED))) {
+		set_destination(p, ast_strdupa(p->uri));
+	} else if (p->route) {
 		set_destination(p, p->route->hop);
 		add_route(req, is_strict ? p->route->next : p->route);
 	}
@@ -13700,15 +13716,15 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
 {
 	struct sip_request resp;
 	
-	if (sipmethod == SIP_ACK) {
-		p->invitestate = INV_CONFIRMED;
-	}
-
 	reqprep(&resp, p, sipmethod, seqno, newbranch);
 	if (sipmethod == SIP_CANCEL && p->answered_elsewhere) {
 		add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
 	}

+	if (sipmethod == SIP_ACK) {
+		p->invitestate = INV_CONFIRMED;
+	}
+
 	return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
 }

@@ -14277,8 +14293,9 @@ static void list_route(struct sip_route *route)
 	}
 }

-/*! \brief Build route list from Record-Route header */
-static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards)
+/*! \brief Build route list from Record-Route header 
+    \param resp the SIP response code or 0 for a request */
+static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp)
 {
 	struct sip_route *thishop, *head, *tail;
 	int start = 0;
@@ -14296,8 +14313,11 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward
 		p->route = NULL;
 	}

-	/* We only want to create the route set the first time this is called */
-	p->route_persistent = 1;
+	/* We only want to create the route set the first time this is called except
+	   it is called from a provisional response.*/
+	if ((resp < 100) || (resp > 199)) {
+		p->route_persistent = 1;
+	}

 	/* Build a tailq, then assign it to p->route when done.
 	 * If backwards, we add entries from the head so they end up
@@ -19016,7 +19036,8 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)

 	/* Need to check the media/type */
 	if (!strcasecmp(c, "application/dtmf-relay") ||
-	    !strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
+	    !strcasecmp(c, "application/vnd.nortelnetworks.digits") ||
+	    !strcasecmp(c, "application/dtmf")) {
 		unsigned int duration = 0;

 		if (!p->owner) {	/* not a PBX call */
@@ -19025,44 +19046,55 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
 			return;
 		}

-		/* Try getting the "signal=" part */
-		if (ast_strlen_zero(c = get_body(req, "Signal", '=')) && ast_strlen_zero(c = get_body(req, "d", '='))) {
-			ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid);
-			transmit_response(p, "200 OK", req); /* Should return error */
-			return;
+		/* If dtmf-relay or vnd.nortelnetworks.digits, parse the signal and duration;
+		 * otherwise use the body as the signal */
+		if (strcasecmp(c, "application/dtmf")) {
+			const char *msg_body;
+
+			if (   ast_strlen_zero(msg_body = get_body(req, "Signal", '='))
+				&& ast_strlen_zero(msg_body = get_body(req, "d", '='))) {
+				ast_log(LOG_WARNING, "Unable to retrieve DTMF signal for INFO message on "
+						"call %s\n", p->callid);
+				transmit_response(p, "200 OK", req);
+				return;
+			}
+			ast_copy_string(buf, msg_body, sizeof(buf));
+
+			if (!ast_strlen_zero((msg_body = get_body(req, "Duration", '=')))) {
+				sscanf(msg_body, "%30u", &duration);
+			}
 		} else {
-			ast_copy_string(buf, c, sizeof(buf));
+			/* Type is application/dtmf, simply use what's in the message body */
+			get_msg_text(buf, sizeof(buf), req);
 		}

-		if (!ast_strlen_zero((c = get_body(req, "Duration", '=')))) {
-			duration = atoi(c);
-		}
-		if (!duration) {
-			duration = 100; /* 100 ms */
-		}
-
-
+		/* An empty message body requires us to send a 200 OK */
 		if (ast_strlen_zero(buf)) {
 			transmit_response(p, "200 OK", req);
 			return;
 		}

-		if ('0' <= buf[0] && buf[0] <= '9') {
-			event = buf[0] - '0';
-		} else if (buf[0] == '*') {
+		if (!duration) {
+			duration = 100; /* 100 ms */
+		}
+
+		if (buf[0] == '*') {
 			event = 10;
 		} else if (buf[0] == '#') {
 			event = 11;
+		} else if (buf[0] == '!') {
+			event = 16;
 		} else if ('A' <= buf[0] && buf[0] <= 'D') {
 			event = 12 + buf[0] - 'A';
 		} else if ('a' <= buf[0] && buf[0] <= 'd') {
 			event = 12 + buf[0] - 'a';
-		} else if (buf[0] == '!') {
-			event = 16;
-		} else {
-			/* Unknown digit */
-			event = 0;
+		} else if ((sscanf(buf, "%30u", &event) != 1) || event > 16) {
+			ast_log(AST_LOG_WARNING, "Unable to convert DTMF event signal code to a valid "
+					"value for INFO message on call %s\n", p->callid);
+			transmit_response(p, "200 OK", req);
+			return;
 		}
+
 		if (event == 16) {
 			/* send a FLASH event */
 			struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
@@ -19079,56 +19111,8 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
 				f.subclass.integer = '*';
 			} else if (event == 11) {
 				f.subclass.integer = '#';
-			} else if (event < 16) {
-				f.subclass.integer = 'A' + (event - 12);
-			}
-			f.len = duration;
-			ast_queue_frame(p->owner, &f);
-			if (sipdebug) {
-				ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
-			}
-		}
-		transmit_response(p, "200 OK", req);
-		return;
-	} else if (!strcasecmp(c, "application/dtmf")) {
-		/*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
-		unsigned int duration = 0;
-
-		if (!p->owner) {	/* not a PBX call */
-			transmit_response(p, "481 Call leg/transaction does not exist", req);
-			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-			return;
-		}
-
-		get_msg_text(buf, sizeof(buf), req);
-		duration = 100; /* 100 ms */
-
-		if (ast_strlen_zero(buf)) {
-			transmit_response(p, "200 OK", req);
-			return;
-		}
-		event = atoi(buf);
-		if (event == 16) {
-			/* send a FLASH event */
-			struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH }, };
-			ast_queue_frame(p->owner, &f);
-			if (sipdebug) {
-				ast_verbose("* DTMF-relay event received: FLASH\n");
-			}
-		} else {
-			/* send a DTMF event */
-			struct ast_frame f = { AST_FRAME_DTMF, };
-			if (event < 10) {
-				f.subclass.integer = '0' + event;
-			} else if (event == 10) {
-				f.subclass.integer = '*';
-			} else if (event == 11) {
-				f.subclass.integer = '#';
-			} else if (event < 16) {
-				f.subclass.integer = 'A' + (event - 12);
 			} else {
-				/* Unknown digit. */
-				f.subclass.integer = '0';
+				f.subclass.integer = 'A' + (event - 12);
 			}
 			f.len = duration;
 			ast_queue_frame(p->owner, &f);
@@ -19138,7 +19122,6 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
 		}
 		transmit_response(p, "200 OK", req);
 		return;
-
 	} else if (!strcasecmp(c, "application/media_control+xml")) {
 		/* Eh, we'll just assume it's a fast picture update for now */
 		if (p->owner) {
@@ -20317,7 +20300,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		 * */
 		parse_ok_contact(p, req);
 		if (!reinvite) {
-			build_route(p, req, 1);
+			build_route(p, req, 1, resp);
 		}
 		if (!req->ignore && p->owner) {
 			if (get_rpid(p, req)) {
@@ -20367,7 +20350,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		 * */
 		parse_ok_contact(p, req);
 		if (!reinvite) {
-			build_route(p, req, 1);
+			build_route(p, req, 1, resp);
 		}
 		if (!req->ignore && p->owner) {
 			struct ast_party_redirecting redirecting;
@@ -20393,7 +20376,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		 * */
 		parse_ok_contact(p, req);
 		if (!reinvite) {
-			build_route(p, req, 1);
+			build_route(p, req, 1, resp);
 		}
 		if (!req->ignore && p->owner) {
 			if (get_rpid(p, req)) {
@@ -20493,7 +20476,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 			parse_ok_contact(p, req);
 			/* Save Record-Route for any later requests we make on this dialogue */
 			if (!reinvite)
-				build_route(p, req, 1);
+				build_route(p, req, 1, resp);

 			if(set_address_from_contact(p)) {
 				/* Bad contact - we don't know how to reach this device */
@@ -23078,7 +23061,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			*recount = 1;

 			/* Save Record-Route for any later requests we make on this dialogue */
-			build_route(p, req, 0);
+			build_route(p, req, 0, 0);

 			if (c) {
 				ast_party_redirecting_init(&redirecting);
@@ -25043,7 +25026,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
 		if (sipdebug)
 			ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
 		check_via(p, req);
-		build_route(p, req, 0);
+		build_route(p, req, 0, 0);
 	} else if (req->debug && req->ignore)
 		ast_verbose("Ignoring this SUBSCRIBE request\n");
Author	SHA1	Message	Date
Asterisk Autobuilder	df35f748ef	Importing release summary for 10.1.3 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.3@356571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-23 23:37:04 +00:00
Matthew Jordan	9d63a27b7a	Merge 355733, 356476 for 10.1.3 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.3@356569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-23 23:31:02 +00:00
Matthew Jordan	b0022437ae	Create tag for 10.1.3 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.3@356567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-23 22:45:01 +00:00
Asterisk Autobuilder	35a612c868	Importing release summary for 10.1.2 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.2@354654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-09 19:30:00 +00:00
Matthew Jordan	638257e6b3	Committing 354496, 354543, 354548 for 10.1.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.2@354642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-09 19:13:16 +00:00
Matthew Jordan	0ac98cb7c0	Create tag for 10.1.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.2@354578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-09 17:26:55 +00:00
Asterisk Autobuilder	2baf74df58	Importing release summary for 10.1.1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.1@354215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-06 21:50:38 +00:00
Jason Parker	aa36e5f198	Remove old summary files. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.1@354214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-06 21:47:04 +00:00
Jason Parker	3c2a9bc967	Update .version and ChangeLog. Merge fixes. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.1@354211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-06 21:40:48 +00:00
Jason Parker	35c6193ce8	Create tag for Asterisk 10.1.1 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.1.1@354206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-02-06 21:25:59 +00:00
@@ -1 +1 @@
 .1.0
 .1.3