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76
ChangeLog
76
ChangeLog
@@ -1,3 +1,79 @@
|
||||
2012-02-23 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 10.1.3 Released.
|
||||
|
||||
* channels/chan_sip.c: Fix ACK routing for non-2xx responses.
|
||||
|
||||
When we send an ACK for a 2xx response to an INVITE, we are supposed
|
||||
to use the learned route set. However, when we receive a non-2xx
|
||||
final response to an INVITE, we are supposed to send the ACK to the
|
||||
same place we initially sent the INVITE.
|
||||
|
||||
We had been doing this up until the changes went in that would build
|
||||
a route set from provisional responses. That introduced a regression
|
||||
where we would use the learned route set under all circumstances.
|
||||
|
||||
With this change, we now will set the destination of our ACK based on
|
||||
the invitestate. If it is INV_COMPLETED then that means that we have
|
||||
received a non-2xx final response (INV_TERMINATED indicates a 2xx
|
||||
response was received). If it is INV_CANCELLED, then that means the
|
||||
call is being canceled, which means that we should be ACKing a 487
|
||||
response.
|
||||
|
||||
The other change introduced here is setting the invitestate to
|
||||
INV_CONFIRMED when we send an ACK *after* the reqprep instead of
|
||||
before. This way, we can tell in reqprep more easily what the
|
||||
invitestate is prior to sending the ACK.
|
||||
|
||||
(closes issue ASTERISK-19389)
|
||||
reported by Karsten Wemheuer
|
||||
patches:
|
||||
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
|
||||
|
||||
* channels/chan_sip.c: Fix regressions with regards to route-set
|
||||
creation on early dialogs.
|
||||
|
||||
This fixes two main issues:
|
||||
1. Asterisk would send a CANCEL to the route created by the provisional
|
||||
response instead of using the same destination it did in the initial
|
||||
INVITE.
|
||||
2. If a new route set arrives in a 200 OK than was in the 1XX response
|
||||
(perfectly possible if our outbound INVITE gets forked), then the
|
||||
route set in the 200 OK needs to overwrite the route set in the 1XX
|
||||
response.
|
||||
(closes issue ASTERISK-19358)
|
||||
Reported by: Karsten Wemheuer
|
||||
Tested by: Karsten Wemheuer
|
||||
patches:
|
||||
ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
|
||||
ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
|
||||
|
||||
Review: https://reviewboard.asterisk.org/r/1749
|
||||
|
||||
2012-02-09 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 10.1.2 Released.
|
||||
|
||||
* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
|
||||
codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to
|
||||
account for both lowercase alphabetic DTMF events, as well as
|
||||
uppercase alphabetic DTMF events. When this occurred, the comparison
|
||||
of the character buffer containing the event code was changed such
|
||||
that the buffer was first compared against '0' and '9' to determine if
|
||||
it was numeric. Unfortunately, since the first character in the
|
||||
buffer will typically be '1' in the case of non-numeric event codes
|
||||
(10-16), this caused those codes to be converted to a DTMF event of
|
||||
'1'. This patch fixes that, and cleans up handling of both
|
||||
application/dtmf-relay and application/dtmf content types.
|
||||
Review: https://reviewboard.asterisk.org/r/1722/
|
||||
(closes issue ASTERISK-19290) Reported by: Ira Emus
|
||||
Tested by: mjordan
|
||||
|
||||
* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
|
||||
uninitiliazed caller_id storage (closes issue ASTERISK-19311)
|
||||
Reported by: tootai
|
||||
Tested by: rmudgett
|
||||
|
||||
2012-02-06 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 10.1.1 Released.
|
||||
|
||||
@@ -150,6 +150,7 @@ static int parkandannounce_exec(struct ast_channel *chan, const char *data)
|
||||
}
|
||||
|
||||
/* Save the CallerID because the masquerade turns chan into a ZOMBIE. */
|
||||
ast_party_id_init(&caller_id);
|
||||
ast_channel_lock(chan);
|
||||
ast_party_id_copy(&caller_id, &chan->caller.id);
|
||||
ast_channel_unlock(chan);
|
||||
|
||||
@@ -1,119 +0,0 @@
|
||||
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
|
||||
<html xmlns="http://www.w3.org/1999/xhtml">
|
||||
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.1.1</title></head>
|
||||
<body>
|
||||
<h1 align="center"><a name="top">Release Summary</a></h1>
|
||||
<h3 align="center">asterisk-10.1.1</h3>
|
||||
<h3 align="center">Date: 2012-02-06</h3>
|
||||
<h3 align="center"><asteriskteam@digium.com></h3>
|
||||
<hr/>
|
||||
<h2 align="center">Table of Contents</h2>
|
||||
<ol>
|
||||
<li><a href="#summary">Summary</a></li>
|
||||
<li><a href="#contributors">Contributors</a></li>
|
||||
<li><a href="#commits">Other Changes</a></li>
|
||||
<li><a href="#diffstat">Diffstat</a></li>
|
||||
</ol>
|
||||
<hr/>
|
||||
<a name="summary"><h2 align="center">Summary</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
|
||||
<p>The data in this summary reflects changes that have been made since the previous release, asterisk-10.1.0.</p>
|
||||
<hr/>
|
||||
<a name="contributors"><h2 align="center">Contributors</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
|
||||
<table width="100%" border="0">
|
||||
<tr>
|
||||
<td width="33%"><h3>Coders</h3></td>
|
||||
<td width="33%"><h3>Testers</h3></td>
|
||||
<td width="33%"><h3>Reporters</h3></td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
<td>
|
||||
6 rmudgett<br/>
|
||||
5 jrose<br/>
|
||||
2 alecdavis<br/>
|
||||
2 kpfleming<br/>
|
||||
2 russell<br/>
|
||||
2 twilson<br/>
|
||||
1 kmoore<br/>
|
||||
1 mjordan<br/>
|
||||
1 mmichelson<br/>
|
||||
1 seanbright<br/>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
</tr>
|
||||
</table>
|
||||
<hr/>
|
||||
<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
|
||||
<table width="100%" border="1">
|
||||
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352956">352956</a></td><td>rmudgett</td><td>Audit of ao2_iterator_init() usage for v1.8.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352965">352965</a></td><td>jrose</td><td>Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19249">ASTERISK-19249</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=352992">352992</a></td><td>rmudgett</td><td>Audit of ao2_iterator_init() usage for v10.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353039">353039</a></td><td>rmudgett</td><td>Audit of ao2_iterator_init() usage for v10. Missed one.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353078">353078</a></td><td>russell</td><td>Update ast_set_default_eid() to find more network interfaces.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353127">353127</a></td><td>kpfleming</td><td>Add 'L16-256' MIME subtype alias for slin16.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353176">353176</a></td><td>russell</td><td>Find even more network interfaces.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353261">353261</a></td><td>kpfleming</td><td>Clarify log WARNING message when port-zero SDP 'm' lines received.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353321">353321</a></td><td>alecdavis</td><td>RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353369">353369</a></td><td>alecdavis</td><td>prevent debug messsges displaying -ve Cseq numbers. Missed in R353320</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353397">353397</a></td><td>twilson</td><td>Re-link peers by IP when dnsmgr changes the IP</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19106">ASTERISK-19106</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353463">353463</a></td><td>rmudgett</td><td>Fix memory leak in error paths for action_originate().</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353503">353503</a></td><td>twilson</td><td>Allow res_calendar to be unloaded</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16744">ASTERISK-16744</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353551">353551</a></td><td>mjordan</td><td>Added clarification for the VERBOSITY setting to etc_default_asterisk</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17030">ASTERISK-17030</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353599">353599</a></td><td>seanbright</td><td>Resolve an overlap in the ast_audiohook_flags values.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19246">ASTERISK-19246</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353721">353721</a></td><td>jrose</td><td>Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16930">ASTERISK-16930</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353771">353771</a></td><td>jrose</td><td>Fix sip show peers port output, align columns, and fix ami port output.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16930">ASTERISK-16930</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19281">ASTERISK-19281</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353820">353820</a></td><td>mmichelson</td><td>Fix TLS port binding behavior as well as reload behavior:</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16959">ASTERISK-16959</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19201">ASTERISK-19201</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19204">ASTERISK-19204</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353868">353868</a></td><td>rmudgett</td><td>Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888)</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19176">ASTERISK-19176</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353916">353916</a></td><td>kmoore</td><td>Ensure entering T.38 passthrough does not cause an infinite loop</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18951">ASTERISK-18951</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=353962">353962</a></td><td>jrose</td><td>Fixes a segfault occuring when performing attended transfer with FAXOPT(gateway)=yes</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19184">ASTERISK-19184</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354000">354000</a></td><td>jrose</td><td>Fixes deadlocks occuring in chan_agent due to r335976</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19285">ASTERISK-19285</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354119">354119</a></td><td>rmudgett</td><td>Add missing headers to AMI UnParkedCall event to uniquely identify the call.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19240">ASTERISK-19240</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>
|
||||
apps/app_chanspy.c | 13 -
|
||||
apps/app_queue.c | 34 +-
|
||||
channels/chan_agent.c | 171 +++++++++++---
|
||||
channels/chan_dahdi.c | 57 ++++
|
||||
channels/chan_iax2.c | 16 -
|
||||
channels/chan_sip.c | 412 +++++++++++++++++++++++-------------
|
||||
channels/sig_pri.c | 98 +++++++-
|
||||
channels/sig_pri.h | 6
|
||||
channels/sip/include/dialog.h | 4
|
||||
channels/sip/include/sip.h | 16 -
|
||||
configs/http.conf.sample | 3
|
||||
configs/manager.conf.sample | 3
|
||||
contrib/init.d/etc_default_asterisk | 3
|
||||
include/asterisk/audiohook.h | 11
|
||||
include/asterisk/channel.h | 16 -
|
||||
include/asterisk/dnsmgr.h | 27 ++
|
||||
include/asterisk/indications.h | 3
|
||||
include/asterisk/manager.h | 1
|
||||
main/dnsmgr.c | 31 ++
|
||||
main/features.c | 9
|
||||
main/http.c | 11
|
||||
main/indications.c | 2
|
||||
main/manager.c | 85 ++++---
|
||||
main/netsock.c | 32 ++
|
||||
main/pbx.c | 1
|
||||
main/rtp_engine.c | 1
|
||||
main/taskprocessor.c | 2
|
||||
res/res_calendar.c | 42 ++-
|
||||
res/res_fax.c | 1
|
||||
res/res_monitor.c | 35 +--
|
||||
res/res_odbc.c | 5
|
||||
res/res_srtp.c | 46 ++--
|
||||
res/snmp/agent.c | 9
|
||||
tests/test_format_api.c | 1
|
||||
34 files changed, 869 insertions(+), 338 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
@@ -1,210 +0,0 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-10.1.1
|
||||
|
||||
Date: 2012-02-06
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Other Changes
|
||||
4. Diffstat
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Summary
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This release includes only bug fixes. The changes included were made only
|
||||
to address problems that have been identified in this release series.
|
||||
Users should be able to safely upgrade to this version if this release
|
||||
series is already in use. Users considering upgrading from a previous
|
||||
release series are strongly encouraged to review the UPGRADE.txt document
|
||||
as well as the CHANGES document for information about upgrading to this
|
||||
release series.
|
||||
|
||||
The data in this summary reflects changes that have been made since the
|
||||
previous release, asterisk-10.1.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Contributors
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This table lists the people who have submitted code, those that have
|
||||
tested patches, as well as those that reported issues on the issue tracker
|
||||
that were resolved in this release. For coders, the number is how many of
|
||||
their patches (of any size) were committed into this release. For testers,
|
||||
the number is the number of times their name was listed as assisting with
|
||||
testing a patch. Finally, for reporters, the number is the number of
|
||||
issues that they reported that were closed by commits that went into this
|
||||
release.
|
||||
|
||||
Coders Testers Reporters
|
||||
6 rmudgett
|
||||
5 jrose
|
||||
2 alecdavis
|
||||
2 kpfleming
|
||||
2 russell
|
||||
2 twilson
|
||||
1 kmoore
|
||||
1 mjordan
|
||||
1 mmichelson
|
||||
1 seanbright
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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 |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 352956 | rmudgett | Audit of ao2_iterator_init() | |
|
||||
| | | usage for v1.8. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Make failed PauseMonitor and | |
|
||||
| 352965 | jrose | UnpauseMonitor with no valid | ASTERISK-19249 |
|
||||
| | | channel not close AMI | |
|
||||
| | | session. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 352992 | rmudgett | Audit of ao2_iterator_init() | |
|
||||
| | | usage for v10. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353039 | rmudgett | Audit of ao2_iterator_init() | |
|
||||
| | | usage for v10. Missed one. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Update ast_set_default_eid() | |
|
||||
| 353078 | russell | to find more network | |
|
||||
| | | interfaces. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353127 | kpfleming | Add 'L16-256' MIME subtype | |
|
||||
| | | alias for slin16. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353176 | russell | Find even more network | |
|
||||
| | | interfaces. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Clarify log WARNING message | |
|
||||
| 353261 | kpfleming | when port-zero SDP 'm' lines | |
|
||||
| | | received. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | RFC3261 Section 8.1.1.5. The | |
|
||||
| 353321 | alecdavis | sequence number value MUST | |
|
||||
| | | be expressible as a 32-bit | |
|
||||
| | | unsigned integer | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | prevent debug messsges | |
|
||||
| 353369 | alecdavis | displaying -ve Cseq numbers. | |
|
||||
| | | Missed in R353320 | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353397 | twilson | Re-link peers by IP when | ASTERISK-19106 |
|
||||
| | | dnsmgr changes the IP | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Fix memory leak in error | |
|
||||
| 353463 | rmudgett | paths for | |
|
||||
| | | action_originate(). | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353503 | twilson | Allow res_calendar to be | ASTERISK-16744 |
|
||||
| | | unloaded | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Added clarification for the | |
|
||||
| 353551 | mjordan | VERBOSITY setting to | ASTERISK-17030 |
|
||||
| | | etc_default_asterisk | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 353599 | seanbright | Resolve an overlap in the | ASTERISK-19246 |
|
||||
| | | ast_audiohook_flags values. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Use | |
|
||||
| 353721 | jrose | ast_sockaddr_stringify_fmt | ASTERISK-16930 |
|
||||
| | | wrappers for various | |
|
||||
| | | functions in chan_sip | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Fix sip show peers port | ASTERISK-16930, |
|
||||
| 353771 | jrose | output, align columns, and | ASTERISK-19281 |
|
||||
| | | fix ami port output. | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Fix TLS port binding | ASTERISK-16959, |
|
||||
| 353820 | mmichelson | behavior as well as reload | ASTERISK-19201, |
|
||||
| | | behavior: | ASTERISK-19204 |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Restore the 'w' modifier | |
|
||||
| 353868 | rmudgett | support for ISDN spans. | ASTERISK-19176 |
|
||||
| | | Dial(DAHDI/g0/1234w888) | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Ensure entering T.38 | |
|
||||
| 353916 | kmoore | passthrough does not cause | ASTERISK-18951 |
|
||||
| | | an infinite loop | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Fixes a segfault occuring | |
|
||||
| 353962 | jrose | when performing attended | ASTERISK-19184 |
|
||||
| | | transfer with | |
|
||||
| | | FAXOPT(gateway)=yes | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| 354000 | jrose | Fixes deadlocks occuring in | ASTERISK-19285 |
|
||||
| | | chan_agent due to r335976 | |
|
||||
|----------+------------+------------------------------+-----------------|
|
||||
| | | Add missing headers to AMI | |
|
||||
| 354119 | rmudgett | UnParkedCall event to | ASTERISK-19240 |
|
||||
| | | uniquely identify the call. | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
apps/app_chanspy.c | 13 -
|
||||
apps/app_queue.c | 34 +-
|
||||
channels/chan_agent.c | 171 +++++++++++---
|
||||
channels/chan_dahdi.c | 57 ++++
|
||||
channels/chan_iax2.c | 16 -
|
||||
channels/chan_sip.c | 412 +++++++++++++++++++++++-------------
|
||||
channels/sig_pri.c | 98 +++++++-
|
||||
channels/sig_pri.h | 6
|
||||
channels/sip/include/dialog.h | 4
|
||||
channels/sip/include/sip.h | 16 -
|
||||
configs/http.conf.sample | 3
|
||||
configs/manager.conf.sample | 3
|
||||
contrib/init.d/etc_default_asterisk | 3
|
||||
include/asterisk/audiohook.h | 11
|
||||
include/asterisk/channel.h | 16 -
|
||||
include/asterisk/dnsmgr.h | 27 ++
|
||||
include/asterisk/indications.h | 3
|
||||
include/asterisk/manager.h | 1
|
||||
main/dnsmgr.c | 31 ++
|
||||
main/features.c | 9
|
||||
main/http.c | 11
|
||||
main/indications.c | 2
|
||||
main/manager.c | 85 ++++---
|
||||
main/netsock.c | 32 ++
|
||||
main/pbx.c | 1
|
||||
main/rtp_engine.c | 1
|
||||
main/taskprocessor.c | 2
|
||||
res/res_calendar.c | 42 ++-
|
||||
res/res_fax.c | 1
|
||||
res/res_monitor.c | 35 +--
|
||||
res/res_odbc.c | 5
|
||||
res/res_srtp.c | 46 ++--
|
||||
res/snmp/agent.c | 9
|
||||
tests/test_format_api.c | 1
|
||||
34 files changed, 869 insertions(+), 338 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
137
asterisk-10.1.3-summary.html
Normal file
137
asterisk-10.1.3-summary.html
Normal file
@@ -0,0 +1,137 @@
|
||||
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
|
||||
<html xmlns="http://www.w3.org/1999/xhtml">
|
||||
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-10.1.3</title></head>
|
||||
<body>
|
||||
<h1 align="center"><a name="top">Release Summary</a></h1>
|
||||
<h3 align="center">asterisk-10.1.3</h3>
|
||||
<h3 align="center">Date: 2012-02-23</h3>
|
||||
<h3 align="center"><asteriskteam@digium.com></h3>
|
||||
<hr/>
|
||||
<h2 align="center">Table of Contents</h2>
|
||||
<ol>
|
||||
<li><a href="#summary">Summary</a></li>
|
||||
<li><a href="#contributors">Contributors</a></li>
|
||||
<li><a href="#commits">Other Changes</a></li>
|
||||
<li><a href="#diffstat">Diffstat</a></li>
|
||||
</ol>
|
||||
<hr/>
|
||||
<a name="summary"><h2 align="center">Summary</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
|
||||
<p>The data in this summary reflects changes that have been made since the previous release, asterisk-10.1.2.</p>
|
||||
<hr/>
|
||||
<a name="contributors"><h2 align="center">Contributors</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
|
||||
<table width="100%" border="0">
|
||||
<tr>
|
||||
<td width="33%"><h3>Coders</h3></td>
|
||||
<td width="33%"><h3>Testers</h3></td>
|
||||
<td width="33%"><h3>Reporters</h3></td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
<td>
|
||||
13 seanbright<br/>
|
||||
8 rmudgett<br/>
|
||||
4 pabelanger<br/>
|
||||
3 kmoore<br/>
|
||||
3 mmichelson<br/>
|
||||
2 file<br/>
|
||||
2 qwell<br/>
|
||||
2 twilson<br/>
|
||||
1 alecdavis<br/>
|
||||
1 may<br/>
|
||||
1 mjordan<br/>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
</tr>
|
||||
</table>
|
||||
<hr/>
|
||||
<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
|
||||
<table width="100%" border="1">
|
||||
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354656">354656</a></td><td>kmoore</td><td>Make the config parser remove escaping backslashes</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17121">ASTERISK-17121</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354703">354703</a></td><td>kmoore</td><td>Fix parsing of SIP headers where compact and non-compact headers are mixed</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17192">ASTERISK-17192</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354750">354750</a></td><td>twilson</td><td>Note that CDRs are immutable once a bridge is torn down</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16923">ASTERISK-16923</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354836">354836</a></td><td>rmudgett</td><td>Fix AMI Redirect ExtraChannel not redirecting to the same exten and context.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16974">ASTERISK-16974</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354890">354890</a></td><td>qwell</td><td>Fix a voicemail memory leak with heard/deleted messages.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354938">354938</a></td><td>file</td><td>Don't try to play sound files that do not exist.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19188">ASTERISK-19188</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=354959">354959</a></td><td>rmudgett</td><td>Fix reconnecting to pgsql database after connection loss.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16982">ASTERISK-16982</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355010">355010</a></td><td>file</td><td>Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355057">355057</a></td><td>rmudgett</td><td>Fix occasional incorrectly delayed call-file execution.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19081">ASTERISK-19081</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355137">355137</a></td><td>may</td><td>call manager_event only if there is not null channel structure</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19298">ASTERISK-19298</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355183">355183</a></td><td>seanbright</td><td>Clear the high order bit from the destination call number before sending.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355229">355229</a></td><td>qwell</td><td>Don't enable sqlite3 CDRs by default in sample configs.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355271">355271</a></td><td>mmichelson</td><td>Properly invert the return of a strncmp call.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355320">355320</a></td><td>rmudgett</td><td>Fix lock typo that should be unlock in cel_sqlite_custom reload.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19356">ASTERISK-19356</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355375">355375</a></td><td>rmudgett</td><td>Fix voicemail problems when using ogg/vorbis.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16926">ASTERISK-16926</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355449">355449</a></td><td>seanbright</td><td>Use TRUNK_CALL_START as originally intended.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355467">355467</a></td><td>seanbright</td><td>Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355530">355530</a></td><td>seanbright</td><td>When IAX2 debugging is enabled, make sure to log 'apathetic' messages too.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355575">355575</a></td><td>rmudgett</td><td>Fix AMI Monitor action without File header converting channel name into filename.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355620">355620</a></td><td>rmudgett</td><td>Fix compile problem when old version of libvorbisfile v1.1.2 is used.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19370">ASTERISK-19370</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355623">355623</a></td><td>seanbright</td><td>Revert a change to audio_audiohook_write_list that had no affect.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355733">355733</a></td><td>mmichelson</td><td>Fix regressions with regards to route-set creation on early dialogs.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19358">ASTERISK-19358</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355747">355747</a></td><td>seanbright</td><td>Pass the correct value to ast_timer_set_rate() for IAX2 trunking.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355794">355794</a></td><td>seanbright</td><td>Don't allow trunkfreq to be greater than 1000ms.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355851">355851</a></td><td>alecdavis</td><td>push 'outgoing' flag from sig_XXX up to chan_dahdi</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19316">ASTERISK-19316</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355895">355895</a></td><td>pabelanger</td><td>Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355896">355896</a></td><td>pabelanger</td><td>Revert commit</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355902">355902</a></td><td>seanbright</td><td>Set the length of the ast_sockaddr, so that we can set it's port later.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355905">355905</a></td><td>seanbright</td><td>Add some boilerplate documentation for IAXVAR and IAXPEER.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355950">355950</a></td><td>seanbright</td><td>Change some debug messages from LOG_DEBUG to ast_debug.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355953">355953</a></td><td>seanbright</td><td>This was a LOG_NOTICE, so roll it back.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=355998">355998</a></td><td>seanbright</td><td>Remove spurious warning when 'qualifyfreqnotok' is set successfully.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17176">ASTERISK-17176</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356074">356074</a></td><td>kmoore</td><td>Add missing newline to ccss state change notification</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356108">356108</a></td><td>seanbright</td><td>Make 'iax2 show callnumber usage' output make sense when an IP is passed in.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356215">356215</a></td><td>mjordan</td><td>Fix potential buffer overrun and memory leak when executing "sip show peers"</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19231">ASTERISK-19231</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19361">ASTERISK-19361</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356297">356297</a></td><td>twilson</td><td>Track module use count for res_calendar</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356428">356428</a></td><td>pabelanger</td><td>Multiple revisions 356290,356335,356337</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356431">356431</a></td><td>pabelanger</td><td>Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356476">356476</a></td><td>mmichelson</td><td>Fix ACK routing for non-2xx responses.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19389">ASTERISK-19389</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=356522">356522</a></td><td>rmudgett</td><td>Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.</td>
|
||||
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19322">ASTERISK-19322</a></td></tr></table>
|
||||
<hr/>
|
||||
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
|
||||
<pre>
|
||||
addons/chan_ooh323.c | 7
|
||||
apps/app_confbridge.c | 8
|
||||
apps/app_rpt.c | 69 +----
|
||||
apps/app_voicemail.c | 23 -
|
||||
autoconf/ast_c_declare_check.m4 | 31 ++
|
||||
cel/cel_sqlite3_custom.c | 2
|
||||
channels/chan_dahdi.c | 10
|
||||
channels/chan_iax2.c | 62 +++--
|
||||
channels/chan_sip.c | 115 +++++----
|
||||
channels/sig_analog.c | 16 -
|
||||
channels/sig_analog.h | 1
|
||||
channels/sig_pri.c | 16 -
|
||||
channels/sig_pri.h | 1
|
||||
channels/sig_ss7.c | 14 -
|
||||
channels/sig_ss7.h | 1
|
||||
configs/cdr_sqlite3_custom.conf.sample | 8
|
||||
configs/extconfig.conf.sample | 4
|
||||
configs/iax.conf.sample | 3
|
||||
configure.ac | 5
|
||||
formats/format_ogg_vorbis.c | 399 +++++++++++----------------------
|
||||
funcs/func_cdr.c | 4
|
||||
include/asterisk/autoconfig.h.in | 16 -
|
||||
include/asterisk/calendar.h | 2
|
||||
main/audiohook.c | 4
|
||||
main/ccss.c | 2
|
||||
main/config.c | 4
|
||||
main/features.c | 81 +++---
|
||||
main/loader.c | 4
|
||||
main/manager.c | 7
|
||||
pbx/pbx_config.c | 10
|
||||
pbx/pbx_spool.c | 41 ++-
|
||||
res/res_calendar.c | 2
|
||||
res/res_config_pgsql.c | 66 +++++
|
||||
res/res_monitor.c | 22 +
|
||||
34 files changed, 567 insertions(+), 493 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
257
asterisk-10.1.3-summary.txt
Normal file
257
asterisk-10.1.3-summary.txt
Normal file
@@ -0,0 +1,257 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-10.1.3
|
||||
|
||||
Date: 2012-02-23
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Other Changes
|
||||
4. Diffstat
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Summary
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This release includes only bug fixes. The changes included were made only
|
||||
to address problems that have been identified in this release series.
|
||||
Users should be able to safely upgrade to this version if this release
|
||||
series is already in use. Users considering upgrading from a previous
|
||||
release series are strongly encouraged to review the UPGRADE.txt document
|
||||
as well as the CHANGES document for information about upgrading to this
|
||||
release series.
|
||||
|
||||
The data in this summary reflects changes that have been made since the
|
||||
previous release, asterisk-10.1.2.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Contributors
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This table lists the people who have submitted code, those that have
|
||||
tested patches, as well as those that reported issues on the issue tracker
|
||||
that were resolved in this release. For coders, the number is how many of
|
||||
their patches (of any size) were committed into this release. For testers,
|
||||
the number is the number of times their name was listed as assisting with
|
||||
testing a patch. Finally, for reporters, the number is the number of
|
||||
issues that they reported that were closed by commits that went into this
|
||||
release.
|
||||
|
||||
Coders Testers Reporters
|
||||
13 seanbright
|
||||
8 rmudgett
|
||||
4 pabelanger
|
||||
3 kmoore
|
||||
3 mmichelson
|
||||
2 file
|
||||
2 qwell
|
||||
2 twilson
|
||||
1 alecdavis
|
||||
1 may
|
||||
1 mjordan
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Commits Not Associated with an Issue
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all changes that went into this release that did not
|
||||
directly close an issue from the issue tracker. The commits may have been
|
||||
marked as being related to an issue. If that is the case, the issue
|
||||
numbers are listed here, as well.
|
||||
|
||||
+------------------------------------------------------------------------+
|
||||
|Revision|Author |Summary |Issues |
|
||||
| | | |Referenced |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|354656 |kmoore |Make the config parser remove |ASTERISK-17121 |
|
||||
| | |escaping backslashes | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Fix parsing of SIP headers where | |
|
||||
|354703 |kmoore |compact and non-compact headers are |ASTERISK-17192 |
|
||||
| | |mixed | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|354750 |twilson |Note that CDRs are immutable once a |ASTERISK-16923 |
|
||||
| | |bridge is torn down | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Fix AMI Redirect ExtraChannel not | |
|
||||
|354836 |rmudgett |redirecting to the same exten and |ASTERISK-16974 |
|
||||
| | |context. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|354890 |qwell |Fix a voicemail memory leak with | |
|
||||
| | |heard/deleted messages. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|354938 |file |Don't try to play sound files that |ASTERISK-19188 |
|
||||
| | |do not exist. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|354959 |rmudgett |Fix reconnecting to pgsql database |ASTERISK-16982 |
|
||||
| | |after connection loss. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Only allow one 'dialplan reload' to | |
|
||||
|355010 |file |execute at a time as otherwise they | |
|
||||
| | |would share the same common local | |
|
||||
| | |context list. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355057 |rmudgett |Fix occasional incorrectly delayed |ASTERISK-19081 |
|
||||
| | |call-file execution. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355137 |may |call manager_event only if there is |ASTERISK-19298 |
|
||||
| | |not null channel structure | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Clear the high order bit from the | |
|
||||
|355183 |seanbright|destination call number before | |
|
||||
| | |sending. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355229 |qwell |Don't enable sqlite3 CDRs by default| |
|
||||
| | |in sample configs. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355271 |mmichelson|Properly invert the return of a | |
|
||||
| | |strncmp call. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355320 |rmudgett |Fix lock typo that should be unlock |ASTERISK-19356 |
|
||||
| | |in cel_sqlite_custom reload. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355375 |rmudgett |Fix voicemail problems when using |ASTERISK-16926 |
|
||||
| | |ogg/vorbis. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355449 |seanbright|Use TRUNK_CALL_START as originally | |
|
||||
| | |intended. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Only use maxtrunkcall and | |
|
||||
|355467 |seanbright|maxnontrunkcall in chan_iax2 if | |
|
||||
| | |IAX_OLD_FIND is specified. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |When IAX2 debugging is enabled, make| |
|
||||
|355530 |seanbright|sure to log 'apathetic' messages | |
|
||||
| | |too. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Fix AMI Monitor action without File | |
|
||||
|355575 |rmudgett |header converting channel name into | |
|
||||
| | |filename. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355620 |rmudgett |Fix compile problem when old version|ASTERISK-19370 |
|
||||
| | |of libvorbisfile v1.1.2 is used. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Revert a change to | |
|
||||
|355623 |seanbright|audio_audiohook_write_list that had | |
|
||||
| | |no affect. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355733 |mmichelson|Fix regressions with regards to |ASTERISK-19358 |
|
||||
| | |route-set creation on early dialogs.| |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Pass the correct value to | |
|
||||
|355747 |seanbright|ast_timer_set_rate() for IAX2 | |
|
||||
| | |trunking. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355794 |seanbright|Don't allow trunkfreq to be greater | |
|
||||
| | |than 1000ms. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355851 |alecdavis |push 'outgoing' flag from sig_XXX up|ASTERISK-19316 |
|
||||
| | |to chan_dahdi | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355895 |pabelanger|Fix -Werror=unused-but-set-variable | |
|
||||
| | |compiler error (gcc 4.6.2) | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355896 |pabelanger|Revert commit | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355902 |seanbright|Set the length of the ast_sockaddr, | |
|
||||
| | |so that we can set it's port later. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355905 |seanbright|Add some boilerplate documentation | |
|
||||
| | |for IAXVAR and IAXPEER. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355950 |seanbright|Change some debug messages from | |
|
||||
| | |LOG_DEBUG to ast_debug. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|355953 |seanbright|This was a LOG_NOTICE, so roll it | |
|
||||
| | |back. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Remove spurious warning when | |
|
||||
|355998 |seanbright|'qualifyfreqnotok' is set |ASTERISK-17176 |
|
||||
| | |successfully. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|356074 |kmoore |Add missing newline to ccss state | |
|
||||
| | |change notification | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Make 'iax2 show callnumber usage' | |
|
||||
|356108 |seanbright|output make sense when an IP is | |
|
||||
| | |passed in. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Fix potential buffer overrun and |ASTERISK-19231,|
|
||||
|356215 |mjordan |memory leak when executing "sip show|ASTERISK-19361 |
|
||||
| | |peers" | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|356297 |twilson |Track module use count for | |
|
||||
| | |res_calendar | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|356428 |pabelanger|Multiple revisions | |
|
||||
| | |356290,356335,356337 | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|356431 |pabelanger|Fix -Werror=unused-but-set-variable | |
|
||||
| | |compiler error (gcc 4.6.2) | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
|356476 |mmichelson|Fix ACK routing for non-2xx |ASTERISK-19389 |
|
||||
| | |responses. | |
|
||||
|--------+----------+------------------------------------+---------------|
|
||||
| | |Fix blind transfer parking issues if| |
|
||||
|356522 |rmudgett |the dialed extension is not |ASTERISK-19322 |
|
||||
| | |recognized as a parking extension. | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Diffstat Results
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a summary of the changes to the source code that went into this
|
||||
release that was generated using the diffstat utility.
|
||||
|
||||
addons/chan_ooh323.c | 7
|
||||
apps/app_confbridge.c | 8
|
||||
apps/app_rpt.c | 69 +----
|
||||
apps/app_voicemail.c | 23 -
|
||||
autoconf/ast_c_declare_check.m4 | 31 ++
|
||||
cel/cel_sqlite3_custom.c | 2
|
||||
channels/chan_dahdi.c | 10
|
||||
channels/chan_iax2.c | 62 +++--
|
||||
channels/chan_sip.c | 115 +++++----
|
||||
channels/sig_analog.c | 16 -
|
||||
channels/sig_analog.h | 1
|
||||
channels/sig_pri.c | 16 -
|
||||
channels/sig_pri.h | 1
|
||||
channels/sig_ss7.c | 14 -
|
||||
channels/sig_ss7.h | 1
|
||||
configs/cdr_sqlite3_custom.conf.sample | 8
|
||||
configs/extconfig.conf.sample | 4
|
||||
configs/iax.conf.sample | 3
|
||||
configure.ac | 5
|
||||
formats/format_ogg_vorbis.c | 399 +++++++++++----------------------
|
||||
funcs/func_cdr.c | 4
|
||||
include/asterisk/autoconfig.h.in | 16 -
|
||||
include/asterisk/calendar.h | 2
|
||||
main/audiohook.c | 4
|
||||
main/ccss.c | 2
|
||||
main/config.c | 4
|
||||
main/features.c | 81 +++---
|
||||
main/loader.c | 4
|
||||
main/manager.c | 7
|
||||
pbx/pbx_config.c | 10
|
||||
pbx/pbx_spool.c | 41 ++-
|
||||
res/res_calendar.c | 2
|
||||
res/res_config_pgsql.c | 66 +++++
|
||||
res/res_monitor.c | 22 +
|
||||
34 files changed, 567 insertions(+), 493 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
@@ -1283,7 +1283,7 @@ static int auto_congest(const void *arg);
|
||||
static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method);
|
||||
static void free_old_route(struct sip_route *route);
|
||||
static void list_route(struct sip_route *route);
|
||||
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
|
||||
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp);
|
||||
static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
|
||||
struct sip_request *req, const char *uri);
|
||||
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
|
||||
@@ -8023,7 +8023,7 @@ static void forked_invite_init(struct sip_request *req, const char *new_theirtag
|
||||
ast_string_field_set(p, our_contact, original->our_contact);
|
||||
ast_string_field_set(p, fullcontact, original->fullcontact);
|
||||
parse_ok_contact(p, req);
|
||||
build_route(p, req, 1);
|
||||
build_route(p, req, 1, 0);
|
||||
|
||||
transmit_request(p, SIP_ACK, p->ocseq, XMIT_UNRELIABLE, TRUE);
|
||||
transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
|
||||
@@ -10473,7 +10473,15 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
|
||||
snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);
|
||||
|
||||
add_header(req, "Via", p->via);
|
||||
if (p->route) {
|
||||
/*
|
||||
* Use the learned route set unless this is a CANCEL on an ACK for a non-2xx
|
||||
* final response. For a CANCEL or ACK, we have to send to the same destination
|
||||
* as the original INVITE.
|
||||
*/
|
||||
if (sipmethod == SIP_CANCEL ||
|
||||
(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED))) {
|
||||
set_destination(p, ast_strdupa(p->uri));
|
||||
} else if (p->route) {
|
||||
set_destination(p, p->route->hop);
|
||||
add_route(req, is_strict ? p->route->next : p->route);
|
||||
}
|
||||
@@ -13708,15 +13716,15 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
|
||||
{
|
||||
struct sip_request resp;
|
||||
|
||||
if (sipmethod == SIP_ACK) {
|
||||
p->invitestate = INV_CONFIRMED;
|
||||
}
|
||||
|
||||
reqprep(&resp, p, sipmethod, seqno, newbranch);
|
||||
if (sipmethod == SIP_CANCEL && p->answered_elsewhere) {
|
||||
add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
|
||||
}
|
||||
|
||||
if (sipmethod == SIP_ACK) {
|
||||
p->invitestate = INV_CONFIRMED;
|
||||
}
|
||||
|
||||
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
|
||||
}
|
||||
|
||||
@@ -14285,8 +14293,9 @@ static void list_route(struct sip_route *route)
|
||||
}
|
||||
}
|
||||
|
||||
/*! \brief Build route list from Record-Route header */
|
||||
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards)
|
||||
/*! \brief Build route list from Record-Route header
|
||||
\param resp the SIP response code or 0 for a request */
|
||||
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp)
|
||||
{
|
||||
struct sip_route *thishop, *head, *tail;
|
||||
int start = 0;
|
||||
@@ -14304,8 +14313,11 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward
|
||||
p->route = NULL;
|
||||
}
|
||||
|
||||
/* We only want to create the route set the first time this is called */
|
||||
p->route_persistent = 1;
|
||||
/* We only want to create the route set the first time this is called except
|
||||
it is called from a provisional response.*/
|
||||
if ((resp < 100) || (resp > 199)) {
|
||||
p->route_persistent = 1;
|
||||
}
|
||||
|
||||
/* Build a tailq, then assign it to p->route when done.
|
||||
* If backwards, we add entries from the head so they end up
|
||||
@@ -19024,7 +19036,8 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
|
||||
|
||||
/* Need to check the media/type */
|
||||
if (!strcasecmp(c, "application/dtmf-relay") ||
|
||||
!strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
|
||||
!strcasecmp(c, "application/vnd.nortelnetworks.digits") ||
|
||||
!strcasecmp(c, "application/dtmf")) {
|
||||
unsigned int duration = 0;
|
||||
|
||||
if (!p->owner) { /* not a PBX call */
|
||||
@@ -19033,44 +19046,55 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
|
||||
return;
|
||||
}
|
||||
|
||||
/* Try getting the "signal=" part */
|
||||
if (ast_strlen_zero(c = get_body(req, "Signal", '=')) && ast_strlen_zero(c = get_body(req, "d", '='))) {
|
||||
ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid);
|
||||
transmit_response(p, "200 OK", req); /* Should return error */
|
||||
return;
|
||||
/* If dtmf-relay or vnd.nortelnetworks.digits, parse the signal and duration;
|
||||
* otherwise use the body as the signal */
|
||||
if (strcasecmp(c, "application/dtmf")) {
|
||||
const char *msg_body;
|
||||
|
||||
if ( ast_strlen_zero(msg_body = get_body(req, "Signal", '='))
|
||||
&& ast_strlen_zero(msg_body = get_body(req, "d", '='))) {
|
||||
ast_log(LOG_WARNING, "Unable to retrieve DTMF signal for INFO message on "
|
||||
"call %s\n", p->callid);
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
}
|
||||
ast_copy_string(buf, msg_body, sizeof(buf));
|
||||
|
||||
if (!ast_strlen_zero((msg_body = get_body(req, "Duration", '=')))) {
|
||||
sscanf(msg_body, "%30u", &duration);
|
||||
}
|
||||
} else {
|
||||
ast_copy_string(buf, c, sizeof(buf));
|
||||
/* Type is application/dtmf, simply use what's in the message body */
|
||||
get_msg_text(buf, sizeof(buf), req);
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero((c = get_body(req, "Duration", '=')))) {
|
||||
duration = atoi(c);
|
||||
}
|
||||
if (!duration) {
|
||||
duration = 100; /* 100 ms */
|
||||
}
|
||||
|
||||
|
||||
/* An empty message body requires us to send a 200 OK */
|
||||
if (ast_strlen_zero(buf)) {
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
}
|
||||
|
||||
if ('0' <= buf[0] && buf[0] <= '9') {
|
||||
event = buf[0] - '0';
|
||||
} else if (buf[0] == '*') {
|
||||
if (!duration) {
|
||||
duration = 100; /* 100 ms */
|
||||
}
|
||||
|
||||
if (buf[0] == '*') {
|
||||
event = 10;
|
||||
} else if (buf[0] == '#') {
|
||||
event = 11;
|
||||
} else if (buf[0] == '!') {
|
||||
event = 16;
|
||||
} else if ('A' <= buf[0] && buf[0] <= 'D') {
|
||||
event = 12 + buf[0] - 'A';
|
||||
} else if ('a' <= buf[0] && buf[0] <= 'd') {
|
||||
event = 12 + buf[0] - 'a';
|
||||
} else if (buf[0] == '!') {
|
||||
event = 16;
|
||||
} else {
|
||||
/* Unknown digit */
|
||||
event = 0;
|
||||
} else if ((sscanf(buf, "%30u", &event) != 1) || event > 16) {
|
||||
ast_log(AST_LOG_WARNING, "Unable to convert DTMF event signal code to a valid "
|
||||
"value for INFO message on call %s\n", p->callid);
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
}
|
||||
|
||||
if (event == 16) {
|
||||
/* send a FLASH event */
|
||||
struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
|
||||
@@ -19087,56 +19111,8 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
|
||||
f.subclass.integer = '*';
|
||||
} else if (event == 11) {
|
||||
f.subclass.integer = '#';
|
||||
} else if (event < 16) {
|
||||
f.subclass.integer = 'A' + (event - 12);
|
||||
}
|
||||
f.len = duration;
|
||||
ast_queue_frame(p->owner, &f);
|
||||
if (sipdebug) {
|
||||
ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
|
||||
}
|
||||
}
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
} else if (!strcasecmp(c, "application/dtmf")) {
|
||||
/*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
|
||||
unsigned int duration = 0;
|
||||
|
||||
if (!p->owner) { /* not a PBX call */
|
||||
transmit_response(p, "481 Call leg/transaction does not exist", req);
|
||||
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
|
||||
return;
|
||||
}
|
||||
|
||||
get_msg_text(buf, sizeof(buf), req);
|
||||
duration = 100; /* 100 ms */
|
||||
|
||||
if (ast_strlen_zero(buf)) {
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
}
|
||||
event = atoi(buf);
|
||||
if (event == 16) {
|
||||
/* send a FLASH event */
|
||||
struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH }, };
|
||||
ast_queue_frame(p->owner, &f);
|
||||
if (sipdebug) {
|
||||
ast_verbose("* DTMF-relay event received: FLASH\n");
|
||||
}
|
||||
} else {
|
||||
/* send a DTMF event */
|
||||
struct ast_frame f = { AST_FRAME_DTMF, };
|
||||
if (event < 10) {
|
||||
f.subclass.integer = '0' + event;
|
||||
} else if (event == 10) {
|
||||
f.subclass.integer = '*';
|
||||
} else if (event == 11) {
|
||||
f.subclass.integer = '#';
|
||||
} else if (event < 16) {
|
||||
f.subclass.integer = 'A' + (event - 12);
|
||||
} else {
|
||||
/* Unknown digit. */
|
||||
f.subclass.integer = '0';
|
||||
f.subclass.integer = 'A' + (event - 12);
|
||||
}
|
||||
f.len = duration;
|
||||
ast_queue_frame(p->owner, &f);
|
||||
@@ -19146,7 +19122,6 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
|
||||
}
|
||||
transmit_response(p, "200 OK", req);
|
||||
return;
|
||||
|
||||
} else if (!strcasecmp(c, "application/media_control+xml")) {
|
||||
/* Eh, we'll just assume it's a fast picture update for now */
|
||||
if (p->owner) {
|
||||
@@ -20325,7 +20300,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
|
||||
* */
|
||||
parse_ok_contact(p, req);
|
||||
if (!reinvite) {
|
||||
build_route(p, req, 1);
|
||||
build_route(p, req, 1, resp);
|
||||
}
|
||||
if (!req->ignore && p->owner) {
|
||||
if (get_rpid(p, req)) {
|
||||
@@ -20375,7 +20350,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
|
||||
* */
|
||||
parse_ok_contact(p, req);
|
||||
if (!reinvite) {
|
||||
build_route(p, req, 1);
|
||||
build_route(p, req, 1, resp);
|
||||
}
|
||||
if (!req->ignore && p->owner) {
|
||||
struct ast_party_redirecting redirecting;
|
||||
@@ -20401,7 +20376,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
|
||||
* */
|
||||
parse_ok_contact(p, req);
|
||||
if (!reinvite) {
|
||||
build_route(p, req, 1);
|
||||
build_route(p, req, 1, resp);
|
||||
}
|
||||
if (!req->ignore && p->owner) {
|
||||
if (get_rpid(p, req)) {
|
||||
@@ -20501,7 +20476,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
|
||||
parse_ok_contact(p, req);
|
||||
/* Save Record-Route for any later requests we make on this dialogue */
|
||||
if (!reinvite)
|
||||
build_route(p, req, 1);
|
||||
build_route(p, req, 1, resp);
|
||||
|
||||
if(set_address_from_contact(p)) {
|
||||
/* Bad contact - we don't know how to reach this device */
|
||||
@@ -23086,7 +23061,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
|
||||
*recount = 1;
|
||||
|
||||
/* Save Record-Route for any later requests we make on this dialogue */
|
||||
build_route(p, req, 0);
|
||||
build_route(p, req, 0, 0);
|
||||
|
||||
if (c) {
|
||||
ast_party_redirecting_init(&redirecting);
|
||||
@@ -25051,7 +25026,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
|
||||
if (sipdebug)
|
||||
ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
|
||||
check_via(p, req);
|
||||
build_route(p, req, 0);
|
||||
build_route(p, req, 0, 0);
|
||||
} else if (req->debug && req->ignore)
|
||||
ast_verbose("Ignoring this SUBSCRIBE request\n");
|
||||
|
||||
|
||||
Reference in New Issue
Block a user