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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-07-20 15:01:23 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-07-20 15:01:23 +0000
commit8db98ddeb970b69cc2b1687a2a1bd5d6299d2c82 (patch)
tree05a442e8acf50be47c36277cb5237d92cee84176
parentcabfea348d67d60507a37ebbfa7d928b033f4c2b (diff)
Importing release summary for 1.4.26 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.26@207358 f38db490-d61c-443f-a65b-d21fe96a405b
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+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-1.4.26</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-1.4.26</h3>
+<h3 align="center">Date: 2009-07-20</h3>
+<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+ <li><a href="#summary">Summary</a></li>
+ <li><a href="#contributors">Contributors</a></li>
+ <li><a href="#issues">Closed Issues</a></li>
+ <li><a href="#commits">Other Changes</a></li>
+ <li><a href="#diffstat">Diffstat</a></li>
+</ol>
+<hr/>
+<a name="summary"><h2 align="center">Summary</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
+<p>The data in this summary reflects changes that have been made since the previous release, asterisk-1.4.25.</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>
+20 mmichelson<br/>
+19 dvossel<br/>
+13 seanbright<br/>
+8 russell<br/>
+7 rmudgett<br/>
+7 tilghman<br/>
+5 mnicholson<br/>
+4 jpeeler<br/>
+3 alecdavis<br/>
+2 dbrooks<br/>
+2 eliel<br/>
+2 file<br/>
+2 guillecabeza<br/>
+2 klaus3000<br/>
+2 kpfleming<br/>
+2 lmadsen<br/>
+2 p<br/>
+2 qwell<br/>
+2 twilson<br/>
+1 adomjan<br/>
+1 garlew<br/>
+1 Jamuel<br/>
+1 lacoursj<br/>
+1 latinsud<br/>
+1 murf<br/>
+1 Nick<br/>
+1 oej<br/>
+1 phsultan<br/>
+1 srt<br/>
+1 tzafrir<br/>
+</td>
+<td>
+6 dvossel<br/>
+5 aragon<br/>
+4 dbrooks<br/>
+4 p_lindheimer<br/>
+3 alecdavis<br/>
+3 scottbmilne<br/>
+3 seanbright<br/>
+2 guillecabeza<br/>
+2 klaus3000<br/>
+2 loloski<br/>
+2 mnicholson<br/>
+2 rmudgett<br/>
+2 sh0t<br/>
+2 sum<br/>
+1 amilcar<br/>
+1 andrebarbosa<br/>
+1 Jamuel<br/>
+1 lacoursj<br/>
+1 legart<br/>
+1 lmadsen<br/>
+1 madkins<br/>
+1 manwe<br/>
+1 murf<br/>
+1 Nick_Lewis<br/>
+1 snblitz<br/>
+1 tootai<br/>
+1 volivier<br/>
+</td>
+<td>
+4 p_lindheimer<br/>
+3 aragon<br/>
+3 klaus3000<br/>
+2 ibc<br/>
+2 loloski<br/>
+2 sodom<br/>
+1 adomjan<br/>
+1 alecdavis<br/>
+1 alexh<br/>
+1 andrebarbosa<br/>
+1 Andrey Sofronov<br/>
+1 avinoash<br/>
+1 cmoss28<br/>
+1 davidw<br/>
+1 deepesh<br/>
+1 dimas<br/>
+1 eboscani<br/>
+1 festr<br/>
+1 ffloimair<br/>
+1 flefoll<br/>
+1 fvdb<br/>
+1 garlew<br/>
+1 geoffs<br/>
+1 greenfieldtech<br/>
+1 guillecabeza<br/>
+1 hotsblanc<br/>
+1 Jamuel<br/>
+1 jkroon<br/>
+1 jpiszcz<br/>
+1 lacoursj<br/>
+1 latinsud<br/>
+1 legart<br/>
+1 lidocaineus<br/>
+1 lmadsen<br/>
+1 lmamane<br/>
+1 madkins<br/>
+1 markd<br/>
+1 mbrancaleoni<br/>
+1 meral<br/>
+1 Nick_Lewis<br/>
+1 okrief<br/>
+1 paravoid<br/>
+1 pdavis<br/>
+1 pj<br/>
+1 pkempgen<br/>
+1 ramonpeek<br/>
+1 samy<br/>
+1 schmidts<br/>
+1 scottbmilne<br/>
+1 seandarcy<br/>
+1 sh0t<br/>
+1 srt<br/>
+1 sum<br/>
+1 timking<br/>
+1 travisghansen<br/>
+1 volivier<br/>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="issues"><h2 align="center">Closed Issues</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
+<h3>Category: Applications/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15022">#15022</a>: [patch] Language handling for numbers, dates, etc is misbehaving when utilizing sub-regional languages<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204556">204556</a><br/>
+Reporter: greenfieldtech<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Applications/app_dial</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15056">#15056</a>: [patch] Blank FORWARD_CONTEXT is not ignored<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198251">198251</a><br/>
+Reporter: p_lindheimer<br/>
+Testers: p_lindheimer<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Applications/app_forkcdr</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=13797">#13797</a>: [patch] forkcdr() doesn't fork when call disposition is ANSWERED<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=195881">195881</a><br/>
+Reporter: sh0t<br/>
+Testers: sh0t<br/>
+Coders: mnicholson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14744">#14744</a>: ForkCDR creates 3 CDR's<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=195881">195881</a><br/>
+Reporter: deepesh<br/>
+Testers: sh0t<br/>
+Coders: mnicholson<br/>
+<br/>
+<h3>Category: Applications/app_mixmonitor</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=13745">#13745</a>: Recordings out of sync when using chanspy<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197537">197537</a><br/>
+Reporter: geoffs<br/>
+Testers: snblitz<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15259">#15259</a>: MixMonitor is not releasing the file handle on the recorded file<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201423">201423</a><br/>
+Reporter: travisghansen<br/>
+Testers: dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15400">#15400</a>: /include/asterisk/lock.h:531 __ast_pthread_mutex_unlock: app_mixmonitor.c line 277 (mixmonitor_thread): Error releasing mutex: O<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204012">204012</a><br/>
+Reporter: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Applications/app_queue</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14631">#14631</a>: [patch] Ghost calls with queues and spa942 and 922<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205349">205349</a><br/>
+Reporter: latinsud<br/>
+Coders: latinsud<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15122">#15122</a>: Abandoned queue calls won't show on CDR<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198068">198068</a><br/>
+Reporter: sum<br/>
+Testers: mnicholson, dbrooks, sum<br/>
+Coders: mnicholson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15198">#15198</a>: [patch] adding queue member from asterisk console with state_interface parameter not working<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197024">197024</a><br/>
+Reporter: loloski<br/>
+Testers: loloski<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Applications/app_voicemail</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14554">#14554</a>: [patch] # for fastforward goes beyond end of message<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203785">203785</a><br/>
+Reporter: lacoursj<br/>
+Testers: lacoursj<br/>
+Coders: lacoursj<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14932">#14932</a>: [patch] asterisk-1.6.0.9-x86_64 segfaults when leaving a voicemail internally to another extension<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199626">199626</a><br/>
+Reporter: jpiszcz<br/>
+Testers: seanbright<br/>
+Coders: seanbright<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15331">#15331</a>: [patch] Log message does not match conditional check<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203719">203719</a><br/>
+Reporter: markd<br/>
+Coders: dbrooks<br/>
+<br/>
+<h3>Category: CDR/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=12946">#12946</a>: [patch] ResetCDR does not work on non-answered channel<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198068">198068</a><br/>
+Reporter: meral<br/>
+Testers: mnicholson, dbrooks, sum<br/>
+Coders: mnicholson<br/>
+<br/>
+<h3>Category: Channels/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15330">#15330</a>: [patch] Using CHANNEL function from ZOMBIE channel stops Asterisk<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201380">201380</a><br/>
+Reporter: okrief<br/>
+Testers: dbrooks<br/>
+Coders: dbrooks<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15416">#15416</a>: No voice on PRI calls with asterisk 1.4.25 & 26<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205728">205728</a><br/>
+Reporter: avinoash<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Channels/chan_dahdi</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14383">#14383</a>: priexclusive parameter ignored if pri = pri_cpe ?<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203908">203908</a><br/>
+Reporter: mbrancaleoni<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14471">#14471</a>: [patch] Making outgoing calls DTMF can't be detected<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=207092">207092</a><br/>
+Reporter: eboscani<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14477">#14477</a>: pseudo channel disappears after dahdi restart<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203848">203848</a><br/>
+Reporter: timking<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14726">#14726</a>: Conditional compilation of a diagnostic message needs an L modifier to %d for a 64 bit integer<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=207155">207155</a><br/>
+Reporter: lmadsen<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15389">#15389</a>: [patch] no audio with SIP call to ISDN PRI, if neither Progress or Proceeding are received.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205728">205728</a><br/>
+Reporter: alecdavis<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Channels/chan_iax2</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14216">#14216</a>: Random audio dropouts when jitterbuffer = yes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=195991">195991</a><br/>
+Reporter: Andrey Sofronov<br/>
+Testers: guillecabeza<br/>
+Coders: guillecabeza<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15032">#15032</a>: [patch] Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=195991">195991</a><br/>
+Reporter: guillecabeza<br/>
+Testers: guillecabeza<br/>
+Coders: guillecabeza<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15361">#15361</a>: [patch] AST-2009-001 breaks IAX2 RFC5456 compliance - Timestamps in POKE/PONG zero in 2 of 4 Bytes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201993">201993</a><br/>
+Reporter: ffloimair<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15377">#15377</a>: [patch] segfault in iax2_hangup Asterisk revision 201600<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204067">204067</a><br/>
+Reporter: aragon<br/>
+Testers: aragon, dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Channels/chan_misdn</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=11974">#11974</a>: external lines connected with message !! Got Busy in Connected State !?!<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204834">204834</a><br/>
+Reporter: fvdb<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=12286">#12286</a>: [patch] immediate=yes treats s extension differently than others: doesn't return cause code 1 if doesn't exist<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=196116">196116</a><br/>
+Reporter: lmamane<br/>
+Coders: file<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14355">#14355</a>: [patch] Segfault if you transfer a call into a meetme room<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206487">206487</a><br/>
+Reporter: sodom<br/>
+Testers: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14692">#14692</a>: [patch] ISDN-Transfer causes backcall attempt of attendent phone<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206487">206487</a><br/>
+Reporter: sodom<br/>
+Testers: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=11231">#11231</a>: [patch] Many retransmits when chan_sip generates multiple outstanding requests<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204243">204243</a><br/>
+Reporter: flefoll<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=13432">#13432</a>: [patch] outboundproxy=proxy.mmmydomain.net where domain can not be resolved silently removes the sip section<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204300">204300</a><br/>
+Reporter: p_lindheimer<br/>
+Coders: p<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=13823">#13823</a>: Response to REGISTER is sent to wrong address when using rport<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197466">197466</a><br/>
+Reporter: dimas<br/>
+Coders: file<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14464">#14464</a>: [patch] lock during simple call processing<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202336">202336</a><br/>
+Reporter: pj<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14659">#14659</a>: [patch] MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port (5060)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202671">202671</a><br/>
+Reporter: klaus3000<br/>
+Testers: dvossel, klaus3000<br/>
+Coders: klaus3000, dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15213">#15213</a>: [patch] asterisk lock in sipsock_read for several seconds and drop sip packets<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202336">202336</a><br/>
+Reporter: schmidts<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15345">#15345</a>: [patch] SIP deadlock in 1.4 revision 199472<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202336">202336</a><br/>
+Reporter: aragon<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15349">#15349</a>: Deadlock in do_monitor() of chan_sip<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202336">202336</a><br/>
+Reporter: samy<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15366">#15366</a>: recent commit on svn 1.4 chan_sip failed to load<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202153">202153</a><br/>
+Reporter: loloski<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=13630">#13630</a>: [patch] Potential spiral detected problem<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198891">198891</a><br/>
+Reporter: festr<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14465">#14465</a>: [patch] Incorrect From: header information when CALLERPRES=PRES_PROHIB<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206938">206938</a><br/>
+Reporter: Nick_Lewis<br/>
+Testers: Nick_Lewis, dvossel<br/>
+Coders: Nick, dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14584">#14584</a>: [patch] Asterisk does not stop retransmission<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202341">202341</a><br/>
+Reporter: klaus3000<br/>
+Testers: klaus3000<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=14725">#14725</a>: Asterisk doesn't add Route headers in NOTIFY when the SUBSCRIBE came from a proxy<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205775">205775</a><br/>
+Reporter: ibc<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15158">#15158</a>: [patch] Message: "Unable to handle indication 3"<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200360">200360</a><br/>
+Reporter: madkins<br/>
+Testers: madkins<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15102">#15102</a>: [patch] Registration Deadlock between Asterisk and Polycom Soundpoint IP 450<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205804">205804</a><br/>
+Reporter: Jamuel<br/>
+Testers: Jamuel<br/>
+Coders: Jamuel, dvossel<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15194">#15194</a>: [patch] [SIP realtime] "sip reload" makes UNREACHABLE users behind NAT<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197562">197562</a><br/>
+Reporter: ibc<br/>
+Testers: manwe<br/>
+Coders: eliel<br/>
+<br/>
+<h3>Category: Core/BuildSystem</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15209">#15209</a>: make install: build_tools/mkpkgconfig: 21: [[: not found<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197264">197264</a><br/>
+Reporter: seandarcy<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=13778">#13778</a>: asterisk blocked at startup between main/asterisk.c/loader.c/load_modules and manager.c/loader.c/ast_module_reload<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199022">199022</a><br/>
+Reporter: hotsblanc<br/>
+Testers: p_lindheimer, seanbright<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Core/Internationalization</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15346">#15346</a>: [patch] TW is not an ISO Language Code<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204469">204469</a><br/>
+Reporter: volivier<br/>
+Testers: volivier<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Core/PBX</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14702">#14702</a>: [patch] debugging CID matching<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206126">206126</a><br/>
+Reporter: klaus3000<br/>
+Coders: klaus3000<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15057">#15057</a>: [patch] hints with 2+ devices that include ONHOLD are often set wrong<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199297">199297</a><br/>
+Reporter: p_lindheimer<br/>
+Testers: p_lindheimer, dvossel<br/>
+Coders: dvossel, p<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15189">#15189</a>: [patch] #exec script can't access manager on first asterisk load<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199022">199022</a><br/>
+Reporter: p_lindheimer<br/>
+Testers: p_lindheimer, seanbright<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Documentation</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15518">#15518</a>: iax.conf, IP-based access control<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206872">206872</a><br/>
+Reporter: pkempgen<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Functions/func_cut</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15208">#15208</a>: [patch] The CUT function does not show the leading field-separators until it finds a value.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197194">197194</a><br/>
+Reporter: ramonpeek<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Functions/func_devstate</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15413">#15413</a>: [patch] Mapping of extension state to device state is incorrect<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204681">204681</a><br/>
+Reporter: legart<br/>
+Testers: dvossel, legart, amilcar<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Functions/func_odbc</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15317">#15317</a>: func_odbc is broken in 1.4.25<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204170">204170</a><br/>
+Reporter: alexh<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Functions/func_realtime</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15517">#15517</a>: [patch] memory leak in func_realtime<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206807">206807</a><br/>
+Reporter: adomjan<br/>
+Coders: adomjan<br/>
+<br/>
+<h3>Category: General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15225">#15225</a>: make config on slackware<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197998">197998</a><br/>
+Reporter: pdavis<br/>
+Coders: seanbright<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15420">#15420</a>: [patch] No audio on calls from asterisk sip phones to nortel set until dtmf from sip phone<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205728">205728</a><br/>
+Reporter: scottbmilne<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: PBX/pbx_spool</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15072">#15072</a>: [patch] Possible crash in pbx_spool.c - operation on previously freed structure<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198957">198957</a><br/>
+Reporter: garlew<br/>
+Coders: garlew<br/>
+<br/>
+<h3>Category: Resources/res_convert</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=15181">#15181</a>: [patch] file convert leaks input file descriptor<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=196826">196826</a><br/>
+Reporter: jkroon<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: Resources/res_features</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=13794">#13794</a>: [patch] CDR for picked up parked call gives answer time < start time and no record for parking<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203375">203375</a><br/>
+Reporter: davidw<br/>
+Testers: murf, dbrooks<br/>
+Coders: murf<br/>
+<br/>
+<a href="http://bugs.digium.com/view.php?id=15183">#15183</a>: [patch] Attended Transfers are not working<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197124">197124</a><br/>
+Reporter: andrebarbosa<br/>
+Testers: andrebarbosa, tootai, loloski<br/>
+Coders: jpeeler<br/>
+<br/>
+<h3>Category: Resources/res_jabber</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14876">#14876</a>: [patch] Missing \r\n in response to JabberSend manager action<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198370">198370</a><br/>
+Reporter: srt<br/>
+Coders: seanbright, srt, phsultan<br/>
+<br/>
+<h3>Category: Resources/res_musiconhold</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14759">#14759</a>: [patch] Asterisk commands "moh reload" or "reload res_musiconhold.so" causes MOH not to work properly<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198665">198665</a><br/>
+Reporter: lidocaineus<br/>
+Testers: lmadsen<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Resources/res_smdi</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=14561">#14561</a>: When using SMDI Asterisk crashes after message is left<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=198311">198311</a><br/>
+Reporter: cmoss28<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: Utilities/General</h3><br/>
+<a href="http://bugs.digium.com/view.php?id=10812">#10812</a>: [patch] safe_asterisk includes bashisms<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=196657">196657</a><br/>
+Reporter: paravoid<br/>
+Coders: tzafrir<br/>
+<br/>
+<hr/>
+<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
+<table width="100%" border="1">
+<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197259">197259</a></td><td>oej</td><td>Typo fix</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197588">197588</a></td><td>mmichelson</td><td>Allow for media to arrive from an alternate source when responding to a reinvite with 491.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197620">197620</a></td><td>dvossel</td><td>'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=197895">197895</a></td><td>lmadsen</td><td>Update MixMonitor documentation.</td>
+<td><a href="http://bugs.digium.com/view.php?id=14829">#14829</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199138">199138</a></td><td>dvossel</td><td>Additional updates to AST-2009-001</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199628">199628</a></td><td>seanbright</td><td>Fix a typo in the stack size calculation just introduced.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=199856">199856</a></td><td>seanbright</td><td>__WORDSIZE is not available on all platforms, so use sizeof(void *) instead.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200037">200037</a></td><td>lmadsen</td><td>Fix path for .flavor and .version.</td>
+<td><a href="http://bugs.digium.com/view.php?id=14737">#14737</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200185">200185</a></td><td>seanbright</td><td>Backport fix for parallel build warnings from trunk r199781.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200513">200513</a></td><td>mmichelson</td><td>Add INFO to our allowed methods so that endpoints know they may send it to us.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200875">200875</a></td><td>eliel</td><td>Show the interface name on error, if it is not found.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=200991">200991</a></td><td>kpfleming</td><td>Improve support for media paths that can generate multiple frames at once.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201261">201261</a></td><td>kpfleming</td><td>Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201450">201450</a></td><td>mmichelson</td><td>Change the datastore traversal in ast_do_masquerade to use a safe list traversal.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201600">201600</a></td><td>russell</td><td>Fix memory corruption and leakage related reloads of non files mode MoH classes.</td>
+<td><a href="http://bugs.digium.com/view.php?id=15109">#15109</a>, <a href="http://bugs.digium.com/view.php?id=15123">#15123</a>, <a href="http://bugs.digium.com/view.php?id=15195">#15195</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=201828">201828</a></td><td>tilghman</td><td>If the "h" extension fails, give it another chance in main/pbx.c.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202022">202022</a></td><td>mnicholson</td><td>Added deadlock protection to try_suggested_sip_codec in chan_sip.c.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202342">202342</a></td><td>mmichelson</td><td>Remove an extra debug line left from previous commit.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202414">202414</a></td><td>russell</td><td>Make Polycom subscription type override check more explicit.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202496">202496</a></td><td>russell</td><td>Report CallerID change during a masquerade.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202572">202572</a></td><td>mmichelson</td><td>Fix potential memory leak in chan_sip when video rtp is not allocated properly.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202601">202601</a></td><td>mmichelson</td><td>Fix more memory leaks that may result if rtp is not successfully allocated.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=202966">202966</a></td><td>mmichelson</td><td>Use the handy UNLINK macro instead of hand-coding the same thing in-line.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203036">203036</a></td><td>rmudgett</td><td>Improved chan_dahdi.conf pritimer error checking.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203115">203115</a></td><td>russell</td><td>Resolve a crash related to a T.38 reinvite race condition.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203230">203230</a></td><td>mmichelson</td><td>Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203311">203311</a></td><td>twilson</td><td>Don't try to free NULL</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=203380">203380</a></td><td>twilson</td><td>I didn't see that Mark already fixed the underlying issue!</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204246">204246</a></td><td>mmichelson</td><td>Fix build oops.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204474">204474</a></td><td>qwell</td><td>Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=204755">204755</a></td><td>dvossel</td><td>moving device state functions from pbx.h to devicestate.h to sync with other branches</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205149">205149</a></td><td>russell</td><td>Make OpenSSL usage thread-safe.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205188">205188</a></td><td>tilghman</td><td>Add redirection warnings for the invalid language codes previously removed.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205215">205215</a></td><td>dvossel</td><td>ast_samp2tv needs floating point for 16khz audio</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205288">205288</a></td><td>qwell</td><td>Update config.guess and config.sub from the savannah.gnu.org git repo.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205409">205409</a></td><td>dvossel</td><td>moving ast_devstate_to_extenstate to pbx.c from devicestate.c</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205471">205471</a></td><td>dvossel</td><td>Fixes 8khz assumptions</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205599">205599</a></td><td>dvossel</td><td>Changing ast_samp2tv to not use floating point.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=205877">205877</a></td><td>mmichelson</td><td>Properly ACK 487 responses to canceled INVITEs.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206284">206284</a></td><td>rmudgett</td><td>Fix some memory leaks in chan_misdn.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206385">206385</a></td><td>russell</td><td>Ensure apathetic replies are sent out on the proper socket.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206635">206635</a></td><td>seanbright</td><td>Only print debug info in codec_dahdi if we are asking for it.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206706">206706</a></td><td>rmudgett</td><td>Merged revision 206700 from</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=206867">206867</a></td><td>dvossel</td><td>avoid segfault caused by user error</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=207033">207033</a></td><td>dvossel</td><td>sip option flags handled incorrectly</td>
+<td><a href="http://bugs.digium.com/view.php?id=15376">#15376</a></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+Makefile | 14
+UPGRADE.txt | 10
+apps/app_chanspy.c | 20
+apps/app_dial.c | 3
+apps/app_meetme.c | 16
+apps/app_mixmonitor.c | 98 +++-
+apps/app_queue.c | 9
+apps/app_voicemail.c | 2
+build_tools/cflags-devmode.xml | 2
+build_tools/make_version_h | 6
+channels/chan_dahdi.c | 99 +++-
+channels/chan_iax2.c | 63 ++-
+channels/chan_misdn.c | 785 +++++++++++++++++++++++----------------
+channels/chan_sip.c | 362 +++++++++++++----
+channels/misdn/isdn_lib.c | 155 ++++---
+channels/misdn/isdn_lib.h | 3
+channels/misdn/isdn_lib_intern.h | 1
+codecs/codec_dahdi.c | 12
+config.guess | 168 +++++---
+config.sub | 160 ++++++-
+configs/chan_dahdi.conf.sample | 5
+configs/iax.conf.sample | 2
+contrib/scripts/safe_asterisk | 8
+doc/asterisk-conf.txt | 4
+funcs/func_cut.c | 11
+funcs/func_odbc.c | 11
+funcs/func_realtime.c | 2
+funcs/func_strings.c | 26 -
+include/asterisk.h | 12
+include/asterisk/audiohook.h | 4
+include/asterisk/cdr.h | 19
+include/asterisk/channel.h | 36 +
+include/asterisk/devicestate.h | 81 +++-
+include/asterisk/frame.h | 11
+include/asterisk/linkedlists.h | 39 +
+include/asterisk/pbx.h | 9
+include/asterisk/rtp.h | 17
+include/asterisk/time.h | 2
+include/asterisk/utils.h | 6
+main/asterisk.c | 17
+main/astmm.c | 8
+main/audiohook.c | 12
+main/autoservice.c | 17
+main/callerid.c | 4
+main/cdr.c | 36 +
+main/channel.c | 394 ++++++++++++++-----
+main/cli.c | 1
+main/devicestate.c | 108 ++++-
+main/file.c | 25 -
+main/frame.c | 125 ++----
+main/loader.c | 81 ++++
+main/pbx.c | 138 +++---
+main/rtp.c | 71 ++-
+main/say.c | 505 ++++++++++++++-----------
+main/slinfactory.c | 30 -
+pbx/pbx_spool.c | 6
+res/res_convert.c | 4
+res/res_crypto.c | 46 ++
+res/res_features.c | 45 +-
+res/res_jabber.c | 16
+res/res_musiconhold.c | 20
+res/res_smdi.c | 4
+62 files changed, 2676 insertions(+), 1330 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
diff --git a/asterisk-1.4.26-summary.txt b/asterisk-1.4.26-summary.txt
new file mode 100644
index 000000000..f8b62b5bc
--- /dev/null
+++ b/asterisk-1.4.26-summary.txt
@@ -0,0 +1,808 @@
+ Release Summary
+
+ asterisk-1.4.26
+
+ Date: 2009-07-20
+
+ <asteriskteam@digium.com>
+
+ ----------------------------------------------------------------------
+
+ Table of Contents
+
+ 1. Summary
+ 2. Contributors
+ 3. Closed Issues
+ 4. Other Changes
+ 5. Diffstat
+
+ ----------------------------------------------------------------------
+
+ Summary
+
+ [Back to Top]
+
+ This release includes only bug fixes. The changes included were made only
+ to address problems that have been identified in this release series.
+ Users should be able to safely upgrade to this version if this release
+ series is already in use. Users considering upgrading from a previous
+ release series are strongly encouraged to review the UPGRADE.txt document
+ as well as the CHANGES document for information about upgrading to this
+ release series.
+
+ The data in this summary reflects changes that have been made since the
+ previous release, asterisk-1.4.25.
+
+ ----------------------------------------------------------------------
+
+ 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
+ 20 mmichelson 6 dvossel 4 p_lindheimer
+ 19 dvossel 5 aragon 3 aragon
+ 13 seanbright 4 dbrooks 3 klaus3000
+ 8 russell 4 p_lindheimer 2 ibc
+ 7 rmudgett 3 alecdavis 2 loloski
+ 7 tilghman 3 scottbmilne 2 sodom
+ 5 mnicholson 3 seanbright 1 adomjan
+ 4 jpeeler 2 guillecabeza 1 alecdavis
+ 3 alecdavis 2 klaus3000 1 alexh
+ 2 dbrooks 2 loloski 1 andrebarbosa
+ 2 eliel 2 mnicholson 1 Andrey Sofronov
+ 2 file 2 rmudgett 1 avinoash
+ 2 guillecabeza 2 sh0t 1 cmoss28
+ 2 klaus3000 2 sum 1 davidw
+ 2 kpfleming 1 amilcar 1 deepesh
+ 2 lmadsen 1 andrebarbosa 1 dimas
+ 2 p 1 Jamuel 1 eboscani
+ 2 qwell 1 lacoursj 1 festr
+ 2 twilson 1 legart 1 ffloimair
+ 1 adomjan 1 lmadsen 1 flefoll
+ 1 garlew 1 madkins 1 fvdb
+ 1 Jamuel 1 manwe 1 garlew
+ 1 lacoursj 1 murf 1 geoffs
+ 1 latinsud 1 Nick_Lewis 1 greenfieldtech
+ 1 murf 1 snblitz 1 guillecabeza
+ 1 Nick 1 tootai 1 hotsblanc
+ 1 oej 1 volivier 1 Jamuel
+ 1 phsultan 1 jkroon
+ 1 srt 1 jpiszcz
+ 1 tzafrir 1 lacoursj
+ 1 latinsud
+ 1 legart
+ 1 lidocaineus
+ 1 lmadsen
+ 1 lmamane
+ 1 madkins
+ 1 markd
+ 1 mbrancaleoni
+ 1 meral
+ 1 Nick_Lewis
+ 1 okrief
+ 1 paravoid
+ 1 pdavis
+ 1 pj
+ 1 pkempgen
+ 1 ramonpeek
+ 1 samy
+ 1 schmidts
+ 1 scottbmilne
+ 1 seandarcy
+ 1 sh0t
+ 1 srt
+ 1 sum
+ 1 timking
+ 1 travisghansen
+ 1 volivier
+
+ ----------------------------------------------------------------------
+
+ Closed Issues
+
+ [Back to Top]
+
+ This is a list of all issues from the issue tracker that were closed by
+ changes that went into this release.
+
+ Category: Applications/General
+
+ #15022: [patch] Language handling for numbers, dates, etc is misbehaving
+ when utilizing sub-regional languages
+ Revision: 204556
+ Reporter: greenfieldtech
+ Coders: tilghman
+
+ Category: Applications/app_dial
+
+ #15056: [patch] Blank FORWARD_CONTEXT is not ignored
+ Revision: 198251
+ Reporter: p_lindheimer
+ Testers: p_lindheimer
+ Coders: seanbright
+
+ Category: Applications/app_forkcdr
+
+ #13797: [patch] forkcdr() doesn't fork when call disposition is ANSWERED
+ Revision: 195881
+ Reporter: sh0t
+ Testers: sh0t
+ Coders: mnicholson
+
+ #14744: ForkCDR creates 3 CDR's
+ Revision: 195881
+ Reporter: deepesh
+ Testers: sh0t
+ Coders: mnicholson
+
+ Category: Applications/app_mixmonitor
+
+ #13745: Recordings out of sync when using chanspy
+ Revision: 197537
+ Reporter: geoffs
+ Testers: snblitz
+ Coders: mmichelson
+
+ #15259: MixMonitor is not releasing the file handle on the recorded file
+ Revision: 201423
+ Reporter: travisghansen
+ Testers: dvossel
+ Coders: dvossel
+
+ #15400: /include/asterisk/lock.h:531 __ast_pthread_mutex_unlock:
+ app_mixmonitor.c line 277 (mixmonitor_thread): Error releasing mutex: O
+ Revision: 204012
+ Reporter: aragon
+ Coders: mmichelson
+
+ Category: Applications/app_queue
+
+ #14631: [patch] Ghost calls with queues and spa942 and 922
+ Revision: 205349
+ Reporter: latinsud
+ Coders: latinsud
+
+ #15122: Abandoned queue calls won't show on CDR
+ Revision: 198068
+ Reporter: sum
+ Testers: mnicholson, dbrooks, sum
+ Coders: mnicholson
+
+ #15198: [patch] adding queue member from asterisk console with
+ state_interface parameter not working
+ Revision: 197024
+ Reporter: loloski
+ Testers: loloski
+ Coders: seanbright
+
+ Category: Applications/app_voicemail
+
+ #14554: [patch] # for fastforward goes beyond end of message
+ Revision: 203785
+ Reporter: lacoursj
+ Testers: lacoursj
+ Coders: lacoursj
+
+ #14932: [patch] asterisk-1.6.0.9-x86_64 segfaults when leaving a voicemail
+ internally to another extension
+ Revision: 199626
+ Reporter: jpiszcz
+ Testers: seanbright
+ Coders: seanbright
+
+ #15331: [patch] Log message does not match conditional check
+ Revision: 203719
+ Reporter: markd
+ Coders: dbrooks
+
+ Category: CDR/General
+
+ #12946: [patch] ResetCDR does not work on non-answered channel
+ Revision: 198068
+ Reporter: meral
+ Testers: mnicholson, dbrooks, sum
+ Coders: mnicholson
+
+ Category: Channels/General
+
+ #15330: [patch] Using CHANNEL function from ZOMBIE channel stops Asterisk
+ Revision: 201380
+ Reporter: okrief
+ Testers: dbrooks
+ Coders: dbrooks
+
+ #15416: No voice on PRI calls with asterisk 1.4.25 & 26
+ Revision: 205728
+ Reporter: avinoash
+ Testers: scottbmilne, alecdavis
+ Coders: alecdavis
+
+ Category: Channels/chan_dahdi
+
+ #14383: priexclusive parameter ignored if pri = pri_cpe ?
+ Revision: 203908
+ Reporter: mbrancaleoni
+ Coders: rmudgett
+
+ #14471: [patch] Making outgoing calls DTMF can't be detected
+ Revision: 207092
+ Reporter: eboscani
+ Coders: jpeeler
+
+ #14477: pseudo channel disappears after dahdi restart
+ Revision: 203848
+ Reporter: timking
+ Coders: jpeeler
+
+ #14726: Conditional compilation of a diagnostic message needs an L
+ modifier to %d for a 64 bit integer
+ Revision: 207155
+ Reporter: lmadsen
+ Coders: jpeeler
+
+ #15389: [patch] no audio with SIP call to ISDN PRI, if neither Progress or
+ Proceeding are received.
+ Revision: 205728
+ Reporter: alecdavis
+ Testers: scottbmilne, alecdavis
+ Coders: alecdavis
+
+ Category: Channels/chan_iax2
+
+ #14216: Random audio dropouts when jitterbuffer = yes
+ Revision: 195991
+ Reporter: Andrey Sofronov
+ Testers: guillecabeza
+ Coders: guillecabeza
+
+ #15032: [patch] Sign problem calculating timestamp for iax frame leads to
+ no audio on the receiving peer.
+ Revision: 195991
+ Reporter: guillecabeza
+ Testers: guillecabeza
+ Coders: guillecabeza
+
+ #15361: [patch] AST-2009-001 breaks IAX2 RFC5456 compliance - Timestamps
+ in POKE/PONG zero in 2 of 4 Bytes
+ Revision: 201993
+ Reporter: ffloimair
+ Coders: dvossel
+
+ #15377: [patch] segfault in iax2_hangup Asterisk revision 201600
+ Revision: 204067
+ Reporter: aragon
+ Testers: aragon, dvossel
+ Coders: dvossel
+
+ Category: Channels/chan_misdn
+
+ #11974: external lines connected with message !! Got Busy in Connected
+ State !?!
+ Revision: 204834
+ Reporter: fvdb
+ Coders: rmudgett
+
+ #12286: [patch] immediate=yes treats s extension differently than others:
+ doesn't return cause code 1 if doesn't exist
+ Revision: 196116
+ Reporter: lmamane
+ Coders: file
+
+ #14355: [patch] Segfault if you transfer a call into a meetme room
+ Revision: 206487
+ Reporter: sodom
+ Testers: rmudgett
+ Coders: rmudgett
+
+ #14692: [patch] ISDN-Transfer causes backcall attempt of attendent phone
+ Revision: 206487
+ Reporter: sodom
+ Testers: rmudgett
+ Coders: rmudgett
+
+ Category: Channels/chan_sip/General
+
+ #11231: [patch] Many retransmits when chan_sip generates multiple
+ outstanding requests
+ Revision: 204243
+ Reporter: flefoll
+ Coders: mmichelson
+
+ #13432: [patch] outboundproxy=proxy.mmmydomain.net where domain can not be
+ resolved silently removes the sip section
+ Revision: 204300
+ Reporter: p_lindheimer
+ Coders: p
+
+ #13823: Response to REGISTER is sent to wrong address when using rport
+ Revision: 197466
+ Reporter: dimas
+ Coders: file
+
+ #14464: [patch] lock during simple call processing
+ Revision: 202336
+ Reporter: pj
+ Testers: aragon
+ Coders: mmichelson
+
+ #14659: [patch] MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port (5060)
+ Revision: 202671
+ Reporter: klaus3000
+ Testers: dvossel, klaus3000
+ Coders: klaus3000, dvossel
+
+ #15213: [patch] asterisk lock in sipsock_read for several seconds and drop
+ sip packets
+ Revision: 202336
+ Reporter: schmidts
+ Testers: aragon
+ Coders: mmichelson
+
+ #15345: [patch] SIP deadlock in 1.4 revision 199472
+ Revision: 202336
+ Reporter: aragon
+ Testers: aragon
+ Coders: mmichelson
+
+ #15349: Deadlock in do_monitor() of chan_sip
+ Revision: 202336
+ Reporter: samy
+ Testers: aragon
+ Coders: mmichelson
+
+ #15366: recent commit on svn 1.4 chan_sip failed to load
+ Revision: 202153
+ Reporter: loloski
+ Coders: seanbright
+
+ Category: Channels/chan_sip/Interoperability
+
+ #13630: [patch] Potential spiral detected problem
+ Revision: 198891
+ Reporter: festr
+ Coders: dvossel
+
+ #14465: [patch] Incorrect From: header information when
+ CALLERPRES=PRES_PROHIB
+ Revision: 206938
+ Reporter: Nick_Lewis
+ Testers: Nick_Lewis, dvossel
+ Coders: Nick, dvossel
+
+ #14584: [patch] Asterisk does not stop retransmission
+ Revision: 202341
+ Reporter: klaus3000
+ Testers: klaus3000
+ Coders: mmichelson
+
+ #14725: Asterisk doesn't add Route headers in NOTIFY when the SUBSCRIBE
+ came from a proxy
+ Revision: 205775
+ Reporter: ibc
+ Coders: mmichelson
+
+ #15158: [patch] Message: "Unable to handle indication 3"
+ Revision: 200360
+ Reporter: madkins
+ Testers: madkins
+ Coders: mmichelson
+
+ Category: Channels/chan_sip/Registration
+
+ #15102: [patch] Registration Deadlock between Asterisk and Polycom
+ Soundpoint IP 450
+ Revision: 205804
+ Reporter: Jamuel
+ Testers: Jamuel
+ Coders: Jamuel, dvossel
+
+ #15194: [patch] [SIP realtime] "sip reload" makes UNREACHABLE users behind
+ NAT
+ Revision: 197562
+ Reporter: ibc
+ Testers: manwe
+ Coders: eliel
+
+ Category: Core/BuildSystem
+
+ #15209: make install: build_tools/mkpkgconfig: 21: [[: not found
+ Revision: 197264
+ Reporter: seandarcy
+ Coders: seanbright
+
+ Category: Core/General
+
+ #13778: asterisk blocked at startup between
+ main/asterisk.c/loader.c/load_modules and
+ manager.c/loader.c/ast_module_reload
+ Revision: 199022
+ Reporter: hotsblanc
+ Testers: p_lindheimer, seanbright
+ Coders: seanbright
+
+ Category: Core/Internationalization
+
+ #15346: [patch] TW is not an ISO Language Code
+ Revision: 204469
+ Reporter: volivier
+ Testers: volivier
+ Coders: tilghman
+
+ Category: Core/PBX
+
+ #14702: [patch] debugging CID matching
+ Revision: 206126
+ Reporter: klaus3000
+ Coders: klaus3000
+
+ #15057: [patch] hints with 2+ devices that include ONHOLD are often set
+ wrong
+ Revision: 199297
+ Reporter: p_lindheimer
+ Testers: p_lindheimer, dvossel
+ Coders: dvossel, p
+
+ #15189: [patch] #exec script can't access manager on first asterisk load
+ Revision: 199022
+ Reporter: p_lindheimer
+ Testers: p_lindheimer, seanbright
+ Coders: seanbright
+
+ Category: Documentation
+
+ #15518: iax.conf, IP-based access control
+ Revision: 206872
+ Reporter: pkempgen
+ Coders: dvossel
+
+ Category: Functions/func_cut
+
+ #15208: [patch] The CUT function does not show the leading
+ field-separators until it finds a value.
+ Revision: 197194
+ Reporter: ramonpeek
+ Coders: tilghman
+
+ Category: Functions/func_devstate
+
+ #15413: [patch] Mapping of extension state to device state is incorrect
+ Revision: 204681
+ Reporter: legart
+ Testers: dvossel, legart, amilcar
+ Coders: dvossel
+
+ Category: Functions/func_odbc
+
+ #15317: func_odbc is broken in 1.4.25
+ Revision: 204170
+ Reporter: alexh
+ Coders: tilghman
+
+ Category: Functions/func_realtime
+
+ #15517: [patch] memory leak in func_realtime
+ Revision: 206807
+ Reporter: adomjan
+ Coders: adomjan
+
+ Category: General
+
+ #15225: make config on slackware
+ Revision: 197998
+ Reporter: pdavis
+ Coders: seanbright
+
+ #15420: [patch] No audio on calls from asterisk sip phones to nortel set
+ until dtmf from sip phone
+ Revision: 205728
+ Reporter: scottbmilne
+ Testers: scottbmilne, alecdavis
+ Coders: alecdavis
+
+ Category: PBX/pbx_spool
+
+ #15072: [patch] Possible crash in pbx_spool.c - operation on previously
+ freed structure
+ Revision: 198957
+ Reporter: garlew
+ Coders: garlew
+
+ Category: Resources/res_convert
+
+ #15181: [patch] file convert leaks input file descriptor
+ Revision: 196826
+ Reporter: jkroon
+ Coders: russell
+
+ Category: Resources/res_features
+
+ #13794: [patch] CDR for picked up parked call gives answer time < start
+ time and no record for parking
+ Revision: 203375
+ Reporter: davidw
+ Testers: murf, dbrooks
+ Coders: murf
+
+ #15183: [patch] Attended Transfers are not working
+ Revision: 197124
+ Reporter: andrebarbosa
+ Testers: andrebarbosa, tootai, loloski
+ Coders: jpeeler
+
+ Category: Resources/res_jabber
+
+ #14876: [patch] Missing \r\n in response to JabberSend manager action
+ Revision: 198370
+ Reporter: srt
+ Coders: seanbright, srt, phsultan
+
+ Category: Resources/res_musiconhold
+
+ #14759: [patch] Asterisk commands "moh reload" or "reload
+ res_musiconhold.so" causes MOH not to work properly
+ Revision: 198665
+ Reporter: lidocaineus
+ Testers: lmadsen
+ Coders: tilghman
+
+ Category: Resources/res_smdi
+
+ #14561: When using SMDI Asterisk crashes after message is left
+ Revision: 198311
+ Reporter: cmoss28
+ Coders: russell
+
+ Category: Utilities/General
+
+ #10812: [patch] safe_asterisk includes bashisms
+ Revision: 196657
+ Reporter: paravoid
+ Coders: tzafrir
+
+ ----------------------------------------------------------------------
+
+ 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 |
+ |----------+------------+-----------------------------------+------------|
+ | 197259 | oej | Typo fix | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Allow for media to arrive from an | |
+ | 197588 | mmichelson | alternate source when responding | |
+ | | | to a reinvite with 491. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | 'iax show peer blah' now outputs | |
+ | 197620 | dvossel | whether or not peer 'blah' is in | |
+ | | | trunk mode or not. | |
+ |----------+------------+-----------------------------------+------------|
+ | 197895 | lmadsen | Update MixMonitor documentation. | #14829 |
+ |----------+------------+-----------------------------------+------------|
+ | 199138 | dvossel | Additional updates to | |
+ | | | AST-2009-001 | |
+ |----------+------------+-----------------------------------+------------|
+ | 199628 | seanbright | Fix a typo in the stack size | |
+ | | | calculation just introduced. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | __WORDSIZE is not available on | |
+ | 199856 | seanbright | all platforms, so use sizeof(void | |
+ | | | *) instead. | |
+ |----------+------------+-----------------------------------+------------|
+ | 200037 | lmadsen | Fix path for .flavor and | #14737 |
+ | | | .version. | |
+ |----------+------------+-----------------------------------+------------|
+ | 200185 | seanbright | Backport fix for parallel build | |
+ | | | warnings from trunk r199781. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Add INFO to our allowed methods | |
+ | 200513 | mmichelson | so that endpoints know they may | |
+ | | | send it to us. | |
+ |----------+------------+-----------------------------------+------------|
+ | 200875 | eliel | Show the interface name on error, | |
+ | | | if it is not found. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Improve support for media paths | |
+ | 200991 | kpfleming | that can generate multiple frames | |
+ | | | at once. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Correct AST_LIST_APPEND_LIST | |
+ | 201261 | kpfleming | behavior when list to be appended | |
+ | | | is empty. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Change the datastore traversal in | |
+ | 201450 | mmichelson | ast_do_masquerade to use a safe | |
+ | | | list traversal. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Fix memory corruption and leakage | #15109, |
+ | 201600 | russell | related reloads of non files mode | #15123, |
+ | | | MoH classes. | #15195 |
+ |----------+------------+-----------------------------------+------------|
+ | 201828 | tilghman | If the "h" extension fails, give | |
+ | | | it another chance in main/pbx.c. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Added deadlock protection to | |
+ | 202022 | mnicholson | try_suggested_sip_codec in | |
+ | | | chan_sip.c. | |
+ |----------+------------+-----------------------------------+------------|
+ | 202342 | mmichelson | Remove an extra debug line left | |
+ | | | from previous commit. | |
+ |----------+------------+-----------------------------------+------------|
+ | 202414 | russell | Make Polycom subscription type | |
+ | | | override check more explicit. | |
+ |----------+------------+-----------------------------------+------------|
+ | 202496 | russell | Report CallerID change during a | |
+ | | | masquerade. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Fix potential memory leak in | |
+ | 202572 | mmichelson | chan_sip when video rtp is not | |
+ | | | allocated properly. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Fix more memory leaks that may | |
+ | 202601 | mmichelson | result if rtp is not successfully | |
+ | | | allocated. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Use the handy UNLINK macro | |
+ | 202966 | mmichelson | instead of hand-coding the same | |
+ | | | thing in-line. | |
+ |----------+------------+-----------------------------------+------------|
+ | 203036 | rmudgett | Improved chan_dahdi.conf pritimer | |
+ | | | error checking. | |
+ |----------+------------+-----------------------------------+------------|
+ | 203115 | russell | Resolve a crash related to a T.38 | |
+ | | | reinvite race condition. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Prevent false positives when | |
+ | 203230 | mmichelson | freeing a NULL pointer with | |
+ | | | MALLOC_DEBUG enabled. | |
+ |----------+------------+-----------------------------------+------------|
+ | 203311 | twilson | Don't try to free NULL | |
+ |----------+------------+-----------------------------------+------------|
+ | 203380 | twilson | I didn't see that Mark already | |
+ | | | fixed the underlying issue! | |
+ |----------+------------+-----------------------------------+------------|
+ | 204246 | mmichelson | Fix build oops. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Fix ast_say_counted_noun to | |
+ | 204474 | qwell | correctly handle Polish. Fix a | |
+ | | | comment typo in passing. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | moving device state functions | |
+ | 204755 | dvossel | from pbx.h to devicestate.h to | |
+ | | | sync with other branches | |
+ |----------+------------+-----------------------------------+------------|
+ | 205149 | russell | Make OpenSSL usage thread-safe. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Add redirection warnings for the | |
+ | 205188 | tilghman | invalid language codes previously | |
+ | | | removed. | |
+ |----------+------------+-----------------------------------+------------|
+ | 205215 | dvossel | ast_samp2tv needs floating point | |
+ | | | for 16khz audio | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Update config.guess and | |
+ | 205288 | qwell | config.sub from the | |
+ | | | savannah.gnu.org git repo. | |
+ |----------+------------+-----------------------------------+------------|
+ | 205409 | dvossel | moving ast_devstate_to_extenstate | |
+ | | | to pbx.c from devicestate.c | |
+ |----------+------------+-----------------------------------+------------|
+ | 205471 | dvossel | Fixes 8khz assumptions | |
+ |----------+------------+-----------------------------------+------------|
+ | 205599 | dvossel | Changing ast_samp2tv to not use | |
+ | | | floating point. | |
+ |----------+------------+-----------------------------------+------------|
+ | 205877 | mmichelson | Properly ACK 487 responses to | |
+ | | | canceled INVITEs. | |
+ |----------+------------+-----------------------------------+------------|
+ | 206284 | rmudgett | Fix some memory leaks in | |
+ | | | chan_misdn. | |
+ |----------+------------+-----------------------------------+------------|
+ | 206385 | russell | Ensure apathetic replies are sent | |
+ | | | out on the proper socket. | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Only print debug info in | |
+ | 206635 | seanbright | codec_dahdi if we are asking for | |
+ | | | it. | |
+ |----------+------------+-----------------------------------+------------|
+ | 206706 | rmudgett | Merged revision 206700 from | |
+ |----------+------------+-----------------------------------+------------|
+ | 206867 | dvossel | avoid segfault caused by user | |
+ | | | error | |
+ |----------+------------+-----------------------------------+------------|
+ | 207033 | dvossel | sip option flags handled | #15376 |
+ | | | incorrectly | |
+ +------------------------------------------------------------------------+
+
+ ----------------------------------------------------------------------
+
+ Diffstat Results
+
+ [Back to Top]
+
+ This is a summary of the changes to the source code that went into this
+ release that was generated using the diffstat utility.
+
+ Makefile | 14
+ UPGRADE.txt | 10
+ apps/app_chanspy.c | 20
+ apps/app_dial.c | 3
+ apps/app_meetme.c | 16
+ apps/app_mixmonitor.c | 98 +++-
+ apps/app_queue.c | 9
+ apps/app_voicemail.c | 2
+ build_tools/cflags-devmode.xml | 2
+ build_tools/make_version_h | 6
+ channels/chan_dahdi.c | 99 +++-
+ channels/chan_iax2.c | 63 ++-
+ channels/chan_misdn.c | 785 +++++++++++++++++++++++----------------
+ channels/chan_sip.c | 362 +++++++++++++----
+ channels/misdn/isdn_lib.c | 155 ++++---
+ channels/misdn/isdn_lib.h | 3
+ channels/misdn/isdn_lib_intern.h | 1
+ codecs/codec_dahdi.c | 12
+ config.guess | 168 +++++---
+ config.sub | 160 ++++++-
+ configs/chan_dahdi.conf.sample | 5
+ configs/iax.conf.sample | 2
+ contrib/scripts/safe_asterisk | 8
+ doc/asterisk-conf.txt | 4
+ funcs/func_cut.c | 11
+ funcs/func_odbc.c | 11
+ funcs/func_realtime.c | 2
+ funcs/func_strings.c | 26 -
+ include/asterisk.h | 12
+ include/asterisk/audiohook.h | 4
+ include/asterisk/cdr.h | 19
+ include/asterisk/channel.h | 36 +
+ include/asterisk/devicestate.h | 81 +++-
+ include/asterisk/frame.h | 11
+ include/asterisk/linkedlists.h | 39 +
+ include/asterisk/pbx.h | 9
+ include/asterisk/rtp.h | 17
+ include/asterisk/time.h | 2
+ include/asterisk/utils.h | 6
+ main/asterisk.c | 17
+ main/astmm.c | 8
+ main/audiohook.c | 12
+ main/autoservice.c | 17
+ main/callerid.c | 4
+ main/cdr.c | 36 +
+ main/channel.c | 394 ++++++++++++++-----
+ main/cli.c | 1
+ main/devicestate.c | 108 ++++-
+ main/file.c | 25 -
+ main/frame.c | 125 ++----
+ main/loader.c | 81 ++++
+ main/pbx.c | 138 +++---
+ main/rtp.c | 71 ++-
+ main/say.c | 505 ++++++++++++++-----------
+ main/slinfactory.c | 30 -
+ pbx/pbx_spool.c | 6
+ res/res_convert.c | 4
+ res/res_crypto.c | 46 ++
+ res/res_features.c | 45 +-
+ res/res_jabber.c | 16
+ res/res_musiconhold.c | 20
+ res/res_smdi.c | 4
+ 62 files changed, 2676 insertions(+), 1330 deletions(-)
+
+ ----------------------------------------------------------------------
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 14:26:23 +0000