[asterisk-commits] lmadsen: tag 1.6.0.14 r214607 - /tags/1.6.0.14/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Aug 28 10:54:29 CDT 2009


Author: lmadsen
Date: Fri Aug 28 10:54:25 2009
New Revision: 214607

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=214607
Log:
Importing release summary for 1.6.0.14 release.

Added:
    tags/1.6.0.14/asterisk-1.6.0.14-summary.html   (with props)
    tags/1.6.0.14/asterisk-1.6.0.14-summary.txt   (with props)

Added: tags/1.6.0.14/asterisk-1.6.0.14-summary.html
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.14/asterisk-1.6.0.14-summary.html?view=auto&rev=214607
==============================================================================
--- tags/1.6.0.14/asterisk-1.6.0.14-summary.html (added)
+++ tags/1.6.0.14/asterisk-1.6.0.14-summary.html Fri Aug 28 10:54:25 2009
@@ -1,0 +1,1105 @@
+<!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-1.6.0.14</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-1.6.0.14</h3>
+<h3 align="center">Date: 2009-08-28</h3>
+<h3 align="center">&lt;asteriskteam at 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-1.6.0.10.</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>
+28 mmichelson<br/>
+25 dvossel<br/>
+24 tilghman<br/>
+23 kpfleming<br/>
+16 russell<br/>
+11 jpeeler<br/>
+11 rmudgett<br/>
+8 dbrooks<br/>
+7 seanbright<br/>
+5 file<br/>
+4 alecdavis<br/>
+4 qwell<br/>
+3 mnicholson<br/>
+3 Nick<br/>
+2 klaus3000<br/>
+2 lmadsen<br/>
+2 mvanbaak<br/>
+2 p<br/>
+1 adomjan<br/>
+1 araasch<br/>
+1 atis<br/>
+1 Benjamin<br/>
+1 chappell<br/>
+1 contactmayankjain<br/>
+1 dimas<br/>
+1 edantie<br/>
+1 ghenry<br/>
+1 Jamuel<br/>
+1 jthurman<br/>
+1 Kristijan<br/>
+1 lacoursj<br/>
+1 lasko<br/>
+1 latinsud<br/>
+1 leobrown<br/>
+1 loloski<br/>
+1 makoto<br/>
+1 mattf<br/>
+1 murf<br/>
+1 rain<br/>
+1 twilson<br/>
+1 vrban<br/>
+1 xvisor<br/>
+</td>
+<td>
+12 dvossel<br/>
+5 aragon<br/>
+4 alecdavis<br/>
+3 dbrooks<br/>
+3 klaus3000<br/>
+3 scottbmilne<br/>
+2 mmichelson<br/>
+2 mnicholson<br/>
+2 Nick_Lewis<br/>
+2 rmudgett<br/>
+1 alexh<br/>
+1 amilcar<br/>
+1 awk<br/>
+1 caspy<br/>
+1 CGMChris<br/>
+1 contactmayankjain<br/>
+1 ffloimair<br/>
+1 gentian<br/>
+1 Jamuel<br/>
+1 jmacz<br/>
+1 jpeeler<br/>
+1 jthurman<br/>
+1 Kristijan<br/>
+1 lacoursj<br/>
+1 lasko<br/>
+1 legart<br/>
+1 lmadsen<br/>
+1 madkins<br/>
+1 murf<br/>
+1 p_lindheimer<br/>
+1 pprindeville<br/>
+1 rain<br/>
+1 rue_mohr<br/>
+1 seanbright<br/>
+1 slavon<br/>
+1 suretec<br/>
+1 Takehiko_Ooshima<br/>
+1 tilghman<br/>
+1 timeshell<br/>
+1 tkarl<br/>
+1 toc<br/>
+1 volivier<br/>
+1 vrban<br/>
+</td>
+<td>
+6 alecdavis<br/>
+5 Nick_Lewis<br/>
+4 klaus3000<br/>
+3 pkempgen<br/>
+3 rain<br/>
+2 araasch<br/>
+2 aragon<br/>
+2 ffloimair<br/>
+2 ibc<br/>
+2 mvanbaak<br/>
+2 p_lindheimer<br/>
+2 pprindeville<br/>
+2 slavon<br/>
+2 sodom<br/>
+1 adomjan<br/>
+1 afosorio<br/>
+1 agupta<br/>
+1 alexh<br/>
+1 atis<br/>
+1 avinoash<br/>
+1 bcnit<br/>
+1 Benjamin Kluck<br/>
+1 caspy<br/>
+1 CGMChris<br/>
+1 chappell<br/>
+1 chris-mac<br/>
+1 contactmayankjain<br/>
+1 corruptor<br/>
+1 davidw<br/>
+1 dazza76<br/>
+1 deuffy<br/>
+1 dimas<br/>
+1 edantie<br/>
+1 fdecher<br/>
+1 flefoll<br/>
+1 fvdb<br/>
+1 gentian<br/>
+1 greenfieldtech<br/>
+1 huangtx2009<br/>
+1 Jamuel<br/>
+1 JimDickenson<br/>
+1 jpiszcz<br/>
+1 jsmith<br/>
+1 jthurman<br/>
+1 lacoursj<br/>
+1 lasko<br/>
+1 latinsud<br/>
+1 legart<br/>
+1 leobrown<br/>
+1 lmadsen<br/>
+1 loic<br/>
+1 macogeek<br/>
+1 madkins<br/>
+1 makoto<br/>
+1 markd<br/>
+1 mbrancaleoni<br/>
+1 mnnojd<br/>
+1 Nik Soggia<br/>
+1 noahisaac<br/>
+1 okrief<br/>
+1 pj<br/>
+1 rue_mohr<br/>
+1 samy<br/>
+1 schmidts<br/>
+1 scottbmilne<br/>
+1 seadweller<br/>
+1 st<br/>
+1 suretec<br/>
+1 Takehiko Ooshima<br/>
+1 timeshell<br/>
+1 timking<br/>
+1 tobias_e<br/>
+1 toc<br/>
+1 travisghansen<br/>
+1 trendboy<br/>
+1 umberto71<br/>
+1 vbcrlfuser<br/>
+1 viraptor<br/>
+1 volivier<br/>
+1 wtca<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: Addons/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15269">#15269</a>: [patch] memory leak in asterisk some bug fixing and removing Redundant condition<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=201682">201682</a><br/>
+Reporter: contactmayankjain<br/>
+Testers: contactmayankjain, dvossel<br/>
+Coders: contactmayankjain, dvossel<br/>
+<br/>
+<h3>Category: Applications/General</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204581">204581</a><br/>
+Reporter: greenfieldtech<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Applications/app_chanspy</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15660">#15660</a>: ChanSpy "whisper" is broken in 1.4.26<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=214198">214198</a><br/>
+Reporter: corruptor<br/>
+Testers: dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Applications/app_fax</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15355">#15355</a>: app_fax does not compile with iaxmodem 1.2.0<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=202184">202184</a><br/>
+Reporter: deuffy<br/>
+Coders: seanbright<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15480">#15480</a>: [patch] Not all fixes from #14849 are committed<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=205771">205771</a><br/>
+Reporter: dimas<br/>
+Coders: dimas<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15606">#15606</a>: app_fax.c is not compiling under OpenBSD<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210239">210239</a><br/>
+Reporter: mvanbaak<br/>
+Coders: kpfleming<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15610">#15610</a>: T.38 re-INVITE received after T.38 already negotiated fails<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210818">210818</a><br/>
+Reporter: huangtx2009<br/>
+Coders: file<br/>
+<br/>
+<h3>Category: Applications/app_meetme</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15493">#15493</a>: [patch] contrib/scripts/meetme.sql doesn't contain all fields<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206585">206585</a><br/>
+Reporter: lasko<br/>
+Coders: lasko<br/>
+<br/>
+<h3>Category: Applications/app_milliwatt</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15386">#15386</a>: [patch] Milliwatt() is off by -11dbm<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=209840">209840</a><br/>
+Reporter: rue_mohr<br/>
+Testers: rue_mohr<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: Applications/app_mixmonitor</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=201449">201449</a><br/>
+Reporter: travisghansen<br/>
+Testers: dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15699">#15699</a>: [patch] using ast_free instead of mixmonitor_free<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213135">213135</a><br/>
+Reporter: edantie<br/>
+Coders: edantie<br/>
+<br/>
+<h3>Category: Applications/app_queue</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14536">#14536</a>: [patch] After a caller is processed by app_queue the queue_log logs the hangup as TRANSFER<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=211960">211960</a><br/>
+Reporter: aragon<br/>
+Testers: aragon, mnicholson<br/>
+Coders: mnicholson<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205351">205351</a><br/>
+Reporter: latinsud<br/>
+Coders: latinsud<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15664">#15664</a>: [patch] QUEUE_MEMBER_LIST() returns member names instead of interfaces<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=211044">211044</a><br/>
+Reporter: rain<br/>
+Coders: rain<br/>
+<br/>
+<h3>Category: Applications/app_stack</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15557">#15557</a>: [patch] Gosub() dequotes once more than Macro()<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210909">210909</a><br/>
+Reporter: rain<br/>
+Testers: rain<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15617">#15617</a>: [patch] crash in LOCAL() if Gosub stack is allocated but empty<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=211233">211233</a><br/>
+Reporter: rain<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Applications/app_voicemail</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=203818">203818</a><br/>
+Reporter: lacoursj<br/>
+Testers: lacoursj<br/>
+Coders: lacoursj<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=199632">199632</a><br/>
+Reporter: jpiszcz<br/>
+Testers: seanbright<br/>
+Coders: seanbright<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=203722">203722</a><br/>
+Reporter: markd<br/>
+Coders: dbrooks<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15333">#15333</a>: [patch] add FILE_STORAGE to Voicemail Build Options<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=200945">200945</a><br/>
+Reporter: mvanbaak<br/>
+Coders: mvanbaak<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15720">#15720</a>: opendir() return code is not checked in last_message_index()<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=212628">212628</a><br/>
+Reporter: tobias_e<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Applications/app_voicemail/IMAP</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14496">#14496</a>: [patch] IMAP crash multiple callers / callers hangup at beep<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210568">210568</a><br/>
+Reporter: vbcrlfuser<br/>
+Testers: lmadsen, mmichelson, dbrooks<br/>
+Coders: lmadsen<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14597">#14597</a>: greetings can not be retrieved from IMAP<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213411">213411</a><br/>
+Reporter: wtca<br/>
+Testers: jpeeler<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14950">#14950</a>: [patch] Greetings are stored as IMAP messages even when imapgreetings=no<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213837">213837</a><br/>
+Reporter: noahisaac<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: CDR/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15751">#15751</a>: [patch] Core dump in ast_bridge_call features.c line  2772<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213346">213346</a><br/>
+Reporter: atis<br/>
+Coders: atis<br/>
+<br/>
+<h3>Category: Channels/General</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=201443">201443</a><br/>
+Reporter: okrief<br/>
+Testers: dbrooks<br/>
+Coders: dbrooks<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205729">205729</a><br/>
+Reporter: avinoash<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Channels/chan_dahdi</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=13917">#13917</a>: [patch] fxo modules incorrectly believes channel is answered, if telco reverses line polarity at off hook.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=203698">203698</a><br/>
+Reporter: alecdavis<br/>
+Testers: alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14383">#14383</a>: priexclusive parameter ignored if pri = pri_cpe ?<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=203910">203910</a><br/>
+Reporter: mbrancaleoni<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14434">#14434</a>: [patch] Dahdi does not wait for wink on outbound calls before dialing DTMF with Signalling type = em_w<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207636">207636</a><br/>
+Reporter: araasch<br/>
+Coders: araasch<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14434">#14434</a>: [patch] Dahdi does not wait for wink on outbound calls before dialing DTMF with Signalling type = em_w<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207860">207860</a><br/>
+Reporter: araasch<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14477">#14477</a>: pseudo channel disappears after dahdi restart<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=203855">203855</a><br/>
+Reporter: timking<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14696">#14696</a>: reload in console overwrites priindication=outofband setting<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=208384">208384</a><br/>
+Reporter: fdecher<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=207157">207157</a><br/>
+Reporter: lmadsen<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15248">#15248</a>: [patch] Multiple Groups Not working<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=199228">199228</a><br/>
+Reporter: gentian<br/>
+Testers: gentian<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205729">205729</a><br/>
+Reporter: alecdavis<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15655">#15655</a>: [patch] Dialplan starts execution before call is accepted<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210647">210647</a><br/>
+Reporter: alecdavis<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Channels/chan_iax2</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=201997">201997</a><br/>
+Reporter: ffloimair<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15404">#15404</a>: [patch] Unrequired Debug Message<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=203716">203716</a><br/>
+Reporter: leobrown<br/>
+Coders: leobrown<br/>
+<br/>
+<h3>Category: Channels/chan_misdn</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204836">204836</a><br/>
+Reporter: fvdb<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=12113">#12113</a>: [patch] asterisk crash at reload chan_misdn.so<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=212507">212507</a><br/>
+Reporter: agupta<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=206555">206555</a><br/>
+Reporter: sodom<br/>
+Testers: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=206555">206555</a><br/>
+Reporter: sodom<br/>
+Testers: rmudgett<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204248">204248</a><br/>
+Reporter: flefoll<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=12434">#12434</a>: Handle wrong at offer/answer in sdp in media description(m=)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207425">207425</a><br/>
+Reporter: mnnojd<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=12869">#12869</a>: [patch] 'context' doesn't change when 'sip reload' issued when driven from realtime<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213094">213094</a><br/>
+Reporter: bcnit<br/>
+Testers: lasko<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204302">204302</a><br/>
+Reporter: p_lindheimer<br/>
+Coders: p<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=13623">#13623</a>: Asterisk segfaults when using SIP session timers<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=205117">205117</a><br/>
+Reporter: Nik Soggia<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14239">#14239</a>: [patch] 491-request pending is sent out of dialog<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=208389">208389</a><br/>
+Reporter: klaus3000<br/>
+Testers: klaus3000<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14464">#14464</a>: [patch] lock during simple call processing<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=202338">202338</a><br/>
+Reporter: pj<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14575">#14575</a>: BYE to 408 Request Timeout<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=208589">208589</a><br/>
+Reporter: chris-mac<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=202675">202675</a><br/>
+Reporter: klaus3000<br/>
+Testers: dvossel, klaus3000<br/>
+Coders: klaus3000, dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=202338">202338</a><br/>
+Reporter: schmidts<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15283">#15283</a>: [patch] CLI NOTIFY always tries to use UDP, even if the peer is connected via TCP<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=199821">199821</a><br/>
+Reporter: jthurman<br/>
+Testers: jthurman, dvossel<br/>
+Coders: jthurman<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=202338">202338</a><br/>
+Reporter: aragon<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15349">#15349</a>: Deadlock in do_monitor() of chan_sip<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=202338">202338</a><br/>
+Reporter: samy<br/>
+Testers: aragon<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15362">#15362</a>: [patch] log message output is truncated<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=214200">214200</a><br/>
+Reporter: klaus3000<br/>
+Coders: klaus3000<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15376">#15376</a>: SIP option (SIP_OPT_ flag) is not handled correctly<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207032">207032</a><br/>
+Reporter: Takehiko Ooshima<br/>
+Testers: dvossel, Takehiko_Ooshima<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15403">#15403</a>: [patch] Session timer is not activated<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206775">206775</a><br/>
+Reporter: makoto<br/>
+Coders: makoto<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=13958">#13958</a>: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv'<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=200724">200724</a><br/>
+Reporter: toc<br/>
+Testers: toc<br/>
+Coders: kpfleming<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=206947">206947</a><br/>
+Reporter: Nick_Lewis<br/>
+Testers: Nick_Lewis, dvossel<br/>
+Coders: Nick, dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14584">#14584</a>: [patch] Asterisk does not stop retransmission<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=202344">202344</a><br/>
+Reporter: klaus3000<br/>
+Testers: klaus3000<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205777">205777</a><br/>
+Reporter: ibc<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205879">205879</a><br/>
+Reporter: ibc<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15158">#15158</a>: [patch] Message: "Unable to handle indication 3"<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=200362">200362</a><br/>
+Reporter: madkins<br/>
+Testers: madkins<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15442">#15442</a>: [patch] Asterisk cannot handle SIP 183 "Session Progress" if no SDP is contained in it<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=208264">208264</a><br/>
+Reporter: ffloimair<br/>
+Testers: tkarl, ffloimair<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14344">#14344</a>: [patch] Outbound proxy not used for registrations<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206017">206017</a><br/>
+Reporter: Nick_Lewis<br/>
+Testers: dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14366">#14366</a>: [patch] Registration expiry not compatible with some ITSP<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=211952">211952</a><br/>
+Reporter: Nick_Lewis<br/>
+Testers: mnicholson<br/>
+Coders: mnicholson, Nick<br/>
+<br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205843">205843</a><br/>
+Reporter: Jamuel<br/>
+Testers: Jamuel<br/>
+Coders: Jamuel, dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15539">#15539</a>: [patch] Register request line contains wrong address when domain and registrar host differ<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213727">213727</a><br/>
+Reporter: Nick_Lewis<br/>
+Testers: Nick_Lewis, dvossel<br/>
+Coders: Nick, dvossel<br/>
+<br/>
+<h3>Category: Channels/chan_sip/T.38</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14849">#14849</a>: [patch] SendFax function not working as expected on > 1.6.0.7<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=205697">205697</a><br/>
+Reporter: afosorio<br/>
+Coders: kpfleming<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15182">#15182</a>: [patch] T.38 invite does not always comply with RFC 2327<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=209133">209133</a><br/>
+Reporter: CGMChris<br/>
+Testers: CGMChris<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Channels/chan_sip/TCP-TLS</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=13865">#13865</a>: [patch] SIP/TLS enabled - just one call possible - 481 Call/Transaction Does Not Exist<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=200992">200992</a><br/>
+Reporter: st<br/>
+Testers: mmichelson, Kristijan, vrban, jmacz, dvossel<br/>
+Coders: Kristijan, mmichelson, vrban, dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=14452">#14452</a>: in "_sip_tcp_helper_thread" Buffer is filled with dirty bytes<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=203780">203780</a><br/>
+Reporter: umberto71<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Video</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15121">#15121</a>: [patch] Video support in SIP channel driver appears to be totally broken<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=211348">211348</a><br/>
+Reporter: jsmith<br/>
+Coders: file<br/>
+<br/>
+<h3>Category: Core/BuildSystem</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15697">#15697</a>: most cleaner alaw don't compile<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213180">213180</a><br/>
+Reporter: slavon<br/>
+Coders: qwell<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15698">#15698</a>: [patch] If enable DEBUG_FD_LEAKS - h323 can't start.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213561">213561</a><br/>
+Reporter: slavon<br/>
+Testers: slavon, tilghman<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15714">#15714</a>: [patch] Asterisk won't build with curl unless curl_config is present<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=214519">214519</a><br/>
+Reporter: pprindeville<br/>
+Testers: pprindeville<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Core/Channels</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14723">#14723</a>: ERROR[5003]: channel.c:2043 __ast_read: ast_read() called with no recorded file descriptor.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207362">207362</a><br/>
+Reporter: seadweller<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: Core/Configuration</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14509">#14509</a>: [patch] users.conf (and other .conf files) have incorrect whitespacing<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=202754">202754</a><br/>
+Reporter: timeshell<br/>
+Testers: awk, timeshell<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14730">#14730</a>: [patch] Fix runlevels in Debian rc files<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213901">213901</a><br/>
+Reporter: pkempgen<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15273">#15273</a>: [patch] german time (20:01:00 oh clock) is announced wrong<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=214073">214073</a><br/>
+Reporter: Benjamin Kluck<br/>
+Coders: Benjamin<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15649">#15649</a>: T38 Faxing failing on 1.6.1 svn<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210993">210993</a><br/>
+Reporter: dazza76<br/>
+Coders: kpfleming<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15667">#15667</a>: LOGGER WARNING : error executing after rotate<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=212578">212578</a><br/>
+Reporter: loic<br/>
+Coders: seanbright<br/>
+<br/>
+<h3>Category: Core/Internationalization</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204471">204471</a><br/>
+Reporter: volivier<br/>
+Testers: volivier<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Core/ManagerInterface</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15397">#15397</a>: [patch] segfault in action_coreshowchannels() at manager.c<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=210915">210915</a><br/>
+Reporter: caspy<br/>
+Testers: caspy<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Core/PBX</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=199301">199301</a><br/>
+Reporter: p_lindheimer<br/>
+Testers: p_lindheimer, dvossel<br/>
+Coders: dvossel, p<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15242">#15242</a>: [patch] log does not indicate which function is missing closing parenthesis<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213972">213972</a><br/>
+Reporter: Nick_Lewis<br/>
+Coders: dbrooks, loloski<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15303">#15303</a>: new_find_extension arguments in wrong order<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=199994">199994</a><br/>
+Reporter: JimDickenson<br/>
+Coders: dbrooks<br/>
+<br/>
+<h3>Category: Documentation</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15518">#15518</a>: iax.conf, IP-based access control<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206876">206876</a><br/>
+Reporter: pkempgen<br/>
+Coders: dvossel<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15755">#15755</a>: Description in queues.conf on call recording is slightly misleading<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=213495">213495</a><br/>
+Reporter: trendboy<br/>
+Coders: qwell<br/>
+<br/>
+<h3>Category: Functions/func_callerid</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15476">#15476</a>: callerid(num) is wrong when username is missing<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206705">206705</a><br/>
+Reporter: viraptor<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Functions/func_devstate</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=204754">204754</a><br/>
+Reporter: legart<br/>
+Testers: dvossel, legart, amilcar<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Functions/func_iconv</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15169">#15169</a>: When building with uClibc, configure script mistakenly assumes iconv is always available<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=214153">214153</a><br/>
+Reporter: pprindeville<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Functions/func_realtime</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15517">#15517</a>: [patch] memory leak in func_realtime<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=206809">206809</a><br/>
+Reporter: adomjan<br/>
+Coders: adomjan<br/>
+<br/>
+<h3>Category: Functions/func_uri</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15439">#15439</a>: [patch] URIENCODE() throws a warning when passed an empty string<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=207947">207947</a><br/>
+Reporter: pkempgen<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: General</h3><br/>
+<a href="https://issues.asterisk.org/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/1.6.0?view=revision&revision=205729">205729</a><br/>
+Reporter: scottbmilne<br/>
+Testers: scottbmilne, alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15571">#15571</a>: [patch] 'received' typos in trunk, in 6 files<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=209221">209221</a><br/>
+Reporter: alecdavis<br/>
+Coders: dbrooks<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15595">#15595</a>: [patch] fix spelling for typos, mainly in comments.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=209555">209555</a><br/>
+Reporter: alecdavis<br/>
+Coders: dbrooks<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15595">#15595</a>: [patch] fix spelling for typos, mainly in comments.<br/>

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