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Author SHA1 Message Date
Asterisk Autobuilder
a510d39d09 Importing release summary for 10.9.0-digiumphones-rc3 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc3@374532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 13:06:30 +00:00
Asterisk Autobuilder
860813a512 Merge r374447 for ASTERISK-20511
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc3@374529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 12:59:28 +00:00
Asterisk Autobuilder
84c3c7a913 Create 10.9.0-digiumphones-rc3
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc3@374526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 12:36:51 +00:00
Asterisk Autobuilder
49c5fdf957 Importing release summary for 10.9.0-digiumphones-rc2 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc2@374299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-03 14:32:51 +00:00
Asterisk Autobuilder
2464c21b3a Merge changes for 10.9.0-digiumphones-rc2
This includes r373879 for ASTERISK-20390; r373179 and r373533 for ASTERISK-20409


git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc2@374296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-03 14:27:48 +00:00
Asterisk Autobuilder
b40457a168 Create 10.9.0-digiumphones-rc2
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc2@374292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-03 13:10:42 +00:00
Asterisk Autobuilder
357590f01c Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc1@373028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-13 18:49:43 +00:00
Asterisk Autobuilder
a7b2e33d54 Importing release summary for 10.9.0-digiumphones-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc1@373027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-13 18:49:32 +00:00
Asterisk Autobuilder
0e6225ee29 Importing files for 10.9.0-digiumphones-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc1@373026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-13 18:49:26 +00:00
Asterisk Autobuilder
1cd48deb04 Creating tag for the release of asterisk-10.9.0-digiumphones-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.9.0-digiumphones-rc1@373023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-13 18:29:19 +00:00
7 changed files with 28227 additions and 26 deletions

3
.lastclean Normal file
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@@ -0,0 +1,3 @@
39

1
.version Normal file
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@@ -0,0 +1 @@
10.9.0-digiumphones-rc3

27909
ChangeLog Normal file

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@@ -0,0 +1,118 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.9.0-digiumphones-rc3</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-10.9.0-digiumphones-rc3</h3>
<h3 align="center">Date: 2012-10-05</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-10.9.0-digiumphones-rc2.</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>
6 root<br/>
2 bebuild<br/>
2 Byron<br/>
2 Karsten<br/>
</td>
<td>
2 alecdavis<br/>
2 Byron Clark<br/>
1 tbsky<br/>
</td>
<td>
1 alecdavis<br/>
1 kristoff<br/>
1 kwemheuer<br/>
1 tbsky<br/>
1 ulugutz<br/>
</td>
</tr>
</table>
<hr/>
<a name="issues"><h2 align="center">Closed Issues</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
<h3>Category: Applications/app_disa</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17493">ASTERISK-17493</a>: [patch] dsp.c sends multiple DTMF key events up to applications<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374508">374508</a><br/>
Reporter: alecdavis<br/>
Testers: alecdavis<br/>
Coders: root<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20442">ASTERISK-20442</a>: dtmf callerid regression <br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374413">374413</a><br/>
Reporter: tbsky<br/>
Testers: tbsky, alecdavis<br/>
Coders: root<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20511">ASTERISK-20511</a>: Directrtpsetup does not wrk in SVN-branch-1.8-r374177<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374474">374474</a><br/>
Reporter: kristoff<br/>
Coders: root<br/>
<br/>
<h3>Category: Resources/res_jabber</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19532">ASTERISK-19532</a>: Asterisk crashed after connecting with jabber server in component mode<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374364">374364</a><br/>
Reporter: kwemheuer<br/>
Testers: Byron Clark<br/>
Coders: Karsten, Byron<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19557">ASTERISK-19557</a>: [Regression] Segfault in res_jabber.c<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374364">374364</a><br/>
Reporter: ulugutz<br/>
Testers: Byron Clark<br/>
Coders: Karsten, Byron<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/10-digiumphones?view=revision&revision=374315">374315</a></td><td>root</td><td>Destroy the generic_monitors container after the core_instances in ccss</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374383">374383</a></td><td>root</td><td>_dsp_init: bring inline with trunk</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374455">374455</a></td><td>root</td><td>Fix DBDelTree error codes for AMI, CLI and AGI</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374526">374526</a></td><td>bebuild</td><td>Create 10.9.0-digiumphones-rc3</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=374529">374529</a></td><td>bebuild</td><td>Merge r374447 for ASTERISK-20511</td>
<td></td></tr></table>
<hr/>
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
<pre>
.version | 2
CHANGES | 3
ChangeLog | 21 +++
asterisk-10.9.0-digiumphones-rc2-summary.html | 61 -----------
asterisk-10.9.0-digiumphones-rc2-summary.txt | 91 -----------------
channels/chan_sip.c | 108 +++++++++++---------
configs/dsp.conf.sample | 36 ++++++
main/ccss.c | 9 -
main/db.c | 30 +++--
main/dsp.c | 138 +++++++++++++++++++-------
res/res_agi.c | 16 +--
res/res_jabber.c | 12 ++
12 files changed, 268 insertions(+), 259 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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@@ -0,0 +1,158 @@
Release Summary
asterisk-10.9.0-digiumphones-rc3
Date: 2012-10-05
<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-10.9.0-digiumphones-rc2.
----------------------------------------------------------------------
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
6 root 2 alecdavis 1 alecdavis
2 bebuild 2 Byron Clark 1 kristoff
2 Byron 1 tbsky 1 kwemheuer
2 Karsten 1 tbsky
1 ulugutz
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Category: Applications/app_disa
ASTERISK-17493: [patch] dsp.c sends multiple DTMF key events up to
applications
Revision: 374508
Reporter: alecdavis
Testers: alecdavis
Coders: root
Category: Channels/chan_dahdi
ASTERISK-20442: dtmf callerid regression
Revision: 374413
Reporter: tbsky
Testers: tbsky, alecdavis
Coders: root
Category: Channels/chan_sip/General
ASTERISK-20511: Directrtpsetup does not wrk in SVN-branch-1.8-r374177
Revision: 374474
Reporter: kristoff
Coders: root
Category: Resources/res_jabber
ASTERISK-19532: Asterisk crashed after connecting with jabber server in
component mode
Revision: 374364
Reporter: kwemheuer
Testers: Byron Clark
Coders: Karsten, Byron
ASTERISK-19557: [Regression] Segfault in res_jabber.c
Revision: 374364
Reporter: ulugutz
Testers: Byron Clark
Coders: Karsten, Byron
----------------------------------------------------------------------
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 |
|----------+---------+--------------------------------------+------------|
| | | Destroy the generic_monitors | |
| 374315 | root | container after the core_instances | |
| | | in ccss | |
|----------+---------+--------------------------------------+------------|
| 374383 | root | _dsp_init: bring inline with trunk | |
|----------+---------+--------------------------------------+------------|
| 374455 | root | Fix DBDelTree error codes for AMI, | |
| | | CLI and AGI | |
|----------+---------+--------------------------------------+------------|
| 374526 | bebuild | Create 10.9.0-digiumphones-rc3 | |
|----------+---------+--------------------------------------+------------|
| 374529 | bebuild | Merge r374447 for ASTERISK-20511 | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
.version | 2
CHANGES | 3
ChangeLog | 21 +++
asterisk-10.9.0-digiumphones-rc2-summary.html | 61 -----------
asterisk-10.9.0-digiumphones-rc2-summary.txt | 91 -----------------
channels/chan_sip.c | 108 +++++++++++---------
configs/dsp.conf.sample | 36 ++++++
main/ccss.c | 9 -
main/db.c | 30 +++--
main/dsp.c | 138 +++++++++++++++++++-------
res/res_agi.c | 16 +--
res/res_jabber.c | 12 ++
12 files changed, 268 insertions(+), 259 deletions(-)
----------------------------------------------------------------------

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@@ -307,11 +307,13 @@ static int local_devicestate(void *data)
res = AST_DEVICE_NOT_INUSE;
it = ao2_iterator_init(locals, 0);
while ((lp = ao2_iterator_next(&it))) {
while ((lp = ao2_iterator_next(&it)) && (res == AST_DEVICE_NOT_INUSE)) {
if (!strcmp(exten, lp->exten) && !strcmp(context, lp->context) && lp->owner) {
res = AST_DEVICE_INUSE;
ao2_ref(lp, -1);
break;
ao2_lock(lp);
if (ast_test_flag(lp, LOCAL_LAUNCHED_PBX)) {
res = AST_DEVICE_INUSE;
}
ao2_unlock(lp);
}
ao2_ref(lp, -1);
}

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@@ -30353,7 +30353,8 @@ static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
if (!(opp_chan = ast_bridged_channel(chan))) {
return NULL;
} else if ((opp_chan->tech != &sip_tech) || (!(opp = opp_chan->tech_pvt))) {
} else if (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
(!(opp = opp_chan->tech_pvt))) {
return NULL;
}
@@ -30429,21 +30430,22 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
return AST_RTP_GLUE_RESULT_FORBID;
}
if (!(opp_chan = ast_bridged_channel(chan))) {
return AST_RTP_GLUE_RESULT_FORBID;
} else if ((opp_chan->tech != &sip_tech) || (!(opp = opp_chan->tech_pvt))) {
if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
(!(opp = opp_chan->tech_pvt)))) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
while (sip_pvt_trylock(opp)) {
while (opp && sip_pvt_trylock(opp)) {
sip_pvt_unlock(p);
usleep(1);
sip_pvt_lock(p);
}
if (!(p->rtp)) {
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
@@ -30453,7 +30455,7 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, opp, "audio")) {
if (opp && !apply_directmedia_ha(p, opp, "audio")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
} else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
@@ -30462,7 +30464,9 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
res = AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
if (p->srtp) {
res = AST_RTP_GLUE_RESULT_FORBID;
@@ -30484,21 +30488,22 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru
return AST_RTP_GLUE_RESULT_FORBID;
}
if (!(opp_chan = ast_bridged_channel(chan))) {
return AST_RTP_GLUE_RESULT_FORBID;
} else if ((opp_chan->tech != &sip_tech) || (!(opp = opp_chan->tech_pvt))) {
if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
(!(opp = opp_chan->tech_pvt)))) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
while (sip_pvt_trylock(opp)) {
while (opp && sip_pvt_trylock(opp)) {
sip_pvt_unlock(p);
usleep(1);
sip_pvt_lock(p);
}
if (!(p->vrtp)) {
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
@@ -30508,12 +30513,14 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, opp, "video")) {
if (opp && !apply_directmedia_ha(p, opp, "video")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
}
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
sip_pvt_unlock(p);
return res;
@@ -30530,21 +30537,22 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
return AST_RTP_GLUE_RESULT_FORBID;
}
if (!(opp_chan = ast_bridged_channel(chan))) {
return AST_RTP_GLUE_RESULT_FORBID;
} else if ((opp_chan->tech != &sip_tech) || (!(opp = opp_chan->tech_pvt))) {
if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
(!(opp = opp_chan->tech_pvt)))) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
while (sip_pvt_trylock(opp)) {
while (opp && sip_pvt_trylock(opp)) {
sip_pvt_unlock(p);
usleep(1);
sip_pvt_lock(p);
}
if (!(p->trtp)) {
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
@@ -30554,12 +30562,14 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, opp, "text")) {
if (opp && !apply_directmedia_ha(p, opp, "text")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
}
sip_pvt_unlock(opp);
if (opp) {
sip_pvt_unlock(opp);
}
sip_pvt_unlock(p);
return res;