asterisk/asterisk.git
8 years agoForce non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabled
Matthew Jordan [Wed, 14 Mar 2012 14:40:44 +0000 (14:40 +0000)]
Force non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabled

In r357272, astobj2 was changed to automatically enable REF_DEBUG when the
TEST_FRAMEWORK flag was enabled.  Unfortunately, some compilers (gcc 4.5.1
at least) will attempt to inline ao2_iterator_destroy in handle_astobj2_test.
This by itself is not a problem; unfortunately, the compiler believes that
there is a code path wherein an object allocated on the stack will be
free'd.  As warnings are treated as errors, this prevents compilation of
astobj2.

This patch works around that by adding the noinline attribue to
ao2_iterator_destroy, but only if the TEST_FRAMEWORK flag is enabled.
Preventing inlining is only needed for the test method defined in astobj2,
which is also only enabled if TEST_FRAMEWORK is enabled.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359306 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix bogus reads/writes of console log levels in asterisk.c
Russell Bryant [Wed, 14 Mar 2012 10:56:53 +0000 (10:56 +0000)]
Fix bogus reads/writes of console log levels in asterisk.c

This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact
that logger.c implements 32 log levels (because of the custom log level stuff).
asterisk.c uses this define to size an array of levels per remote console.

This array is modified in ast_console_toggle_loglevel(), which is called by the
"logger set level" CLI command.  While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to toggle a
custom log level on a remote console, as well.  However, doing so led to an
invalid array index in asterisk.c.

This array is read from any time a log message is written to a console.  So,
all custom log level messages resulted in a bogus read if a remote console
was connected.
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Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359260 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix invalid reads/writes due to incorrect sizeof().
Russell Bryant [Wed, 14 Mar 2012 10:05:07 +0000 (10:05 +0000)]
Fix invalid reads/writes due to incorrect sizeof().

These few places in the code used sizeof() on h_addr in struct hostent.
This is sizeof(char *).  The correct way to get the size of this address is to
use h_length.  This error would result in reads/writes of 8 bytes instead of 4
on 64-bit machines.
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Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359212 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix inaccurate sizeof() in sched.c.
Russell Bryant [Wed, 14 Mar 2012 01:35:30 +0000 (01:35 +0000)]
Fix inaccurate sizeof() in sched.c.

This code just needed sizeof(int), not sizeof(int *).
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Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359162 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix incorrect sizeof() in astman.
Russell Bryant [Wed, 14 Mar 2012 00:45:02 +0000 (00:45 +0000)]
Fix incorrect sizeof() in astman.
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Merged revisions 359116 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359117 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix incorrect usage of sizeof() in res_crypto.
Russell Bryant [Wed, 14 Mar 2012 00:39:23 +0000 (00:39 +0000)]
Fix incorrect usage of sizeof() in res_crypto.

In this case, just remove the memset().  There was a redundant memset that is
done correctly just 2 lines later.
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Merged revisions 359110 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359114 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359115 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix broken usage of sizeof() in res_adsi.
Russell Bryant [Wed, 14 Mar 2012 00:29:47 +0000 (00:29 +0000)]
Fix broken usage of sizeof() in res_adsi.
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Merged revisions 359088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359091 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix incorrect sizeof() usage in features.c.
Russell Bryant [Wed, 14 Mar 2012 00:22:10 +0000 (00:22 +0000)]
Fix incorrect sizeof() usage in features.c.

This didn't actually result in a bug anywhere, luckily.  The only place
where the result of these memcpys was used is in app_dial, and the only
field that it read out of ast_call_feature was the first one, which is an
int, so these memcpys always copied just enough to avoid a problem.
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Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359072 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix incorrect sizeof() on a pointer in MD5Final().
Russell Bryant [Wed, 14 Mar 2012 00:10:37 +0000 (00:10 +0000)]
Fix incorrect sizeof() on a pointer in MD5Final().
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Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359060 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDon't use a buffer after it goes out of scope.
Russell Bryant [Wed, 14 Mar 2012 00:01:40 +0000 (00:01 +0000)]
Don't use a buffer after it goes out of scope.

's' is set to 'workspace'.  Make sure 'workspace' doesn't go out of scope while
the reference to it via 's' is still used.
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Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359057 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlocked revisions 359054
Russell Bryant [Tue, 13 Mar 2012 23:46:55 +0000 (23:46 +0000)]
Blocked revisions 359054

........
Dump cache of published events when a node joins the cluster.

Also use a more reliable method for stopping the poll() thread.
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Merged revisions 359053 from http://svn.asterisk.org/svn/asterisk/branches/1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359055 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove chan_usbradio and app_rpt.
Russell Bryant [Tue, 13 Mar 2012 23:42:24 +0000 (23:42 +0000)]
Remove chan_usbradio and app_rpt.

These modules are being maintained outside of the tree and have been for a long
time now, so it doesn't make sense to keep them here.

Review: https://reviewboard.asterisk.org/r/1764/
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Merged revisions 359051 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd missing channel_internal.h
Terry Wilson [Tue, 13 Mar 2012 21:24:13 +0000 (21:24 +0000)]
Add missing channel_internal.h

...again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd ability for chan_dahdi ISDN to block connected line updates per span.
Richard Mudgett [Tue, 13 Mar 2012 21:18:31 +0000 (21:18 +0000)]
Add ability for chan_dahdi ISDN to block connected line updates per span.

Added new chan_dahdi.conf colp_send option parameter to block connected
line updates per span.

(closes issue ASTERISK-17025)
Reported by: Michael Smith

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix setting CDR variables in the hangup extension
Terry Wilson [Tue, 13 Mar 2012 20:43:19 +0000 (20:43 +0000)]
Fix setting CDR variables in the hangup extension

A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.

Review: https://reviewboard.asterisk.org/r/1794/
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Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake hints for invalid SIP devices return Unavail, not idle
Terry Wilson [Tue, 13 Mar 2012 20:06:57 +0000 (20:06 +0000)]
Make hints for invalid SIP devices return Unavail, not idle

This patch drastically simplifies the device state aggegation code.
The old method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit test
update is as a result of fixing that bug.

The SIP change stems from a bug introduced by removing a DNS lookup
for hostname-based SIP channels.

(closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/
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Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 358944 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix IMAP storage compilation after opaquification changes
Terry Wilson [Tue, 13 Mar 2012 18:55:14 +0000 (18:55 +0000)]
Fix IMAP storage compilation after opaquification changes

(closes issue ASTERISK-19513)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358908 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFinalize ast_channel opaquification
Terry Wilson [Tue, 13 Mar 2012 18:20:34 +0000 (18:20 +0000)]
Finalize ast_channel opaquification

Review: https://reviewboard.asterisk.org/r/1786/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix crash caused by opaquification change -r356042.
Richard Mudgett [Tue, 13 Mar 2012 17:01:55 +0000 (17:01 +0000)]
Fix crash caused by opaquification change -r356042.

The set_format() function was more subtle in how it modified the
struct ast_channel readtrans/writetrans values.

* Fixed ast_activate_generator() conversion correctly.

(closes issue ASTERISK-19434)
Reported by: Birger Harzenetter
Tested by: rmudgett

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUse struct copy instead of memcpy().
Richard Mudgett [Tue, 13 Mar 2012 16:50:06 +0000 (16:50 +0000)]
Use struct copy instead of memcpy().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnable macros in 1.8 to find the next highest "h" extension in a context, like in...
Tilghman Lesher [Tue, 13 Mar 2012 08:06:20 +0000 (08:06 +0000)]
Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4.

This change restores functionality that was present in 1.4, when AEL macros
were implemented with the Macro dialplan application.  Macros are fraught with
functionality issues, because they consume a large portion of the underlying
application stack.  This limits the ability of AEL users to call many layers
of subroutines, an issue which Gosub does not have (originally tested to
100,000 levels deep).  Therefore, starting in 1.6.0, AEL macros were
implemented with Gosub.

However, there were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is documented in
the related issue.  In particular, the "h" extension is designed to execute not
in the Macro context, but in the topmost calling context.  Due to legacy issues
with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks
in all calling contexts, bubbling up from the deepest level until it finds an
"h" extension.

Since AEL hides the complexity of the underlying dialplan logic from the AEL
programmer, it's reasonable to assume that this behavior should not change in
the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break
working AEL configurations in the transition to Asterisk 1.8 LTS.  This fix
is the result, which implements a search for the "h" extension in all calling
Gosub contexts.

Fixes ASTERISK-19336

Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher
(with slight modifications for 1.8)

Tested by: Johan Wilfer

Review: https://reviewboard.asterisk.org/r/1776/
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Merged revisions 358810 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 358811 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMassive changes in chan_unistim channel driver. Include many fixes in channel driver...
Igor Goncharovskiy [Mon, 12 Mar 2012 17:01:26 +0000 (17:01 +0000)]
Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:
 * Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
 * Added ability for translation on-screen menu to multiple languages. Tested on Russian languages.  Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
 * Other described in CHANGES file

Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.

(closes issue ASTERISK-16890)

Review: https://reviewboard.asterisk.org/r/1243/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTransition app_page to using app_confbridge internally for the conference bridge...
Joshua Colp [Sat, 10 Mar 2012 20:06:46 +0000 (20:06 +0000)]
Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles.

Review: https://reviewboard.asterisk.org/r/1754/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoResolve a few more cases of variable shadowing.
Sean Bright [Thu, 8 Mar 2012 17:48:14 +0000 (17:48 +0000)]
Resolve a few more cases of variable shadowing.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEliminate a bunch of shadow warnings.
Sean Bright [Thu, 8 Mar 2012 17:02:52 +0000 (17:02 +0000)]
Eliminate a bunch of shadow warnings.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd some underscores in a few of our llist macros to reduce name collisions.
Sean Bright [Thu, 8 Mar 2012 17:00:22 +0000 (17:00 +0000)]
Add some underscores in a few of our llist macros to reduce name collisions.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake transfer not ignore port information with SIP.
Jonathan Rose [Thu, 8 Mar 2012 16:59:30 +0000 (16:59 +0000)]
Make transfer not ignore port information with SIP.

Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail
because port would be cut from the host string and ignored. This simply keeps chan_sip
from cutting off the port number during these kinds of transfers.

(closes issue ASTERISK-19321)
Reported by: Federico Alves
Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header
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Merged revisions 358644 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358645 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd --enable-dev-mode=strict to configure.
Sean Bright [Thu, 8 Mar 2012 16:21:49 +0000 (16:21 +0000)]
Add --enable-dev-mode=strict to configure.

Passing -Wshadow to gcc enables shadow warnings.  From the gcc manual:

    Warn whenever a local variable or type declaration shadows another
    variable, parameter, type, or class member (in C++), or whenever a
    built-in function is shadowed.

Asterisk will not currently compile with this option set, but a number of bugs
have been discovered by enabling this flag on specific files.  The long-term
goal is to eliminate all of the suspect code that causes this warning to be
emitted.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoWhitespace only change to the Makefile
Sean Bright [Thu, 8 Mar 2012 16:12:11 +0000 (16:12 +0000)]
Whitespace only change to the Makefile

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358609 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoHandle numeric columns for eventtype properly in cel_odbc
Terry Wilson [Wed, 7 Mar 2012 21:28:55 +0000 (21:28 +0000)]
Handle numeric columns for eventtype properly in cel_odbc

Patch also implements correct handling of datetime2 and datetimeoffset new
datatypes in SQL Server 2008 and 2008 R2.

(closes issue ASTERISK-17548)

Review: https://reviewboard.asterisk.org/r/1160/
Review: https://reviewboard.asterisk.org/r/1804/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock().
Richard Mudgett [Wed, 7 Mar 2012 18:33:12 +0000 (18:33 +0000)]
Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock().

Update to:
(issue ASTERISK-19372)
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Merged revisions 358530 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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8 years agoReturn g729 and g723.1 frames with the number of samples set properly.
Sean Bright [Wed, 7 Mar 2012 16:16:03 +0000 (16:16 +0000)]
Return g729 and g723.1 frames with the number of samples set properly.

If the wctc4xxp returns more than a single packet, we need to update the number
of samples in the returned frame accordingly.

Acked-by: Shaun Ruffell <sruffell@digium.com>
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8 years agoSet snarkiness = 0 in cdr_adaptive_odbc.conf.sample
Terry Wilson [Wed, 7 Mar 2012 15:19:05 +0000 (15:19 +0000)]
Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample
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8 years agoAdd detection for ODBC WCHAR fields
Terry Wilson [Wed, 7 Mar 2012 15:08:08 +0000 (15:08 +0000)]
Add detection for ODBC WCHAR fields

Without detecting these types, cel_odbc blows up when the character
set for the table is utf8. This also wraps cdr_adaptive_odbc's use of
those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the
code.
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8 years agoFix ring cadance setup for outgoing calls on FXS ports.
Richard Mudgett [Tue, 6 Mar 2012 17:47:40 +0000 (17:47 +0000)]
Fix ring cadance setup for outgoing calls on FXS ports.

* Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence().

Thanks to Sean Bright for compiling with -Wshadow and finding this bug.
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8 years agoAdd dialtone_detect option for analog incoming calls.
Richard Mudgett [Tue, 6 Mar 2012 01:56:10 +0000 (01:56 +0000)]
Add dialtone_detect option for analog incoming calls.

For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered.  If dialtone is
detected, the call is hung up.
no:       Disabled. (Default)
yes:      Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always:   Look for dialtone for the entire call.  Dialtone may return
          if the far end hangs up first.

dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always

(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper

Review: https://reviewboard.asterisk.org/r/1737/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDrop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION.
Richard Mudgett [Mon, 5 Mar 2012 22:32:48 +0000 (22:32 +0000)]
Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION.

SS7 is a trunk protocol and should clear a failed call as soon as
possible.

* Made SS7 hangup a call immediately if it has not connected yet for
INCOMPLETE/BUSY/CONGESTION causes.  Otherwise, play an appropriate inband
tone.

(closes issue ASTERISK-19372)
Reported by: Igor Nikolaev
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8 years agoMake usage of DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.
Richard Mudgett [Mon, 5 Mar 2012 21:55:28 +0000 (21:55 +0000)]
Make usage of DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSetup DSP when SS7 call is connected or early media is available.
Richard Mudgett [Mon, 5 Mar 2012 21:48:32 +0000 (21:48 +0000)]
Setup DSP when SS7 call is connected or early media is available.

Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.

* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.

* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.

(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
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8 years agoEliminate double close of file descriptor in manager.c
Jonathan Rose [Mon, 5 Mar 2012 19:06:46 +0000 (19:06 +0000)]
Eliminate double close of file descriptor in manager.c

The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.

(closes issue ASTERISK-18453)
Reported By: Jaco Kroon
Review: https://reviewboard.asterisk.org/r/1793/
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8 years agoDefer sending the connected line reinvite if a reinvite is already in progress.
Joshua Colp [Mon, 5 Mar 2012 16:44:16 +0000 (16:44 +0000)]
Defer sending the connected line reinvite if a reinvite is already in progress.

(issue ASTERISK-19355)
Reported by: tomaso

(closes issue AST-825)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure Asterisk acknowledges ACKs to 4xx on Replaces errors
Kinsey Moore [Mon, 5 Mar 2012 16:00:32 +0000 (16:00 +0000)]
Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors

Asterisk was not setting pendinginvite in the upper half of
handle_request_invite such that the 4xx was retransmitted repeatedly even
though an ack was received for every retransmission.

(closes issue ASTERISK-19303)
Reported by: Jon Tsiros
Patches:
  fix-19303.patch uploaded by Jeremiah Gowdy (license 6358)

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8 years agoTab to spaces and text change.
Sean Bright [Mon, 5 Mar 2012 11:20:00 +0000 (11:20 +0000)]
Tab to spaces and text change.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix unused-but-set-variable warnings
Terry Wilson [Fri, 2 Mar 2012 23:29:53 +0000 (23:29 +0000)]
Fix unused-but-set-variable warnings

All of these were pretty obviously unused. Some were unused because
the code that used them was #if 0'd. In those cases, I just commented
out the unused-but-set variables.
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8 years agoCorrect some set-but-unused variable warnings in the mISDN library.
Terry Wilson [Fri, 2 Mar 2012 23:25:10 +0000 (23:25 +0000)]
Correct some set-but-unused variable warnings in the mISDN library.

(from kpfleming's commit to trunk r356292)
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8 years agoMake chan_usbradio compile under dev mode
Terry Wilson [Fri, 2 Mar 2012 22:36:28 +0000 (22:36 +0000)]
Make chan_usbradio compile under dev mode

x=++x and x=x=1? Really?
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8 years agoFix case-sensitivity for device-specific event subscriptions and CCSS
Kinsey Moore [Fri, 2 Mar 2012 21:06:12 +0000 (21:06 +0000)]
Fix case-sensitivity for device-specific event subscriptions and CCSS

This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive.  This should also preserve the original case of
the device string as passed in to the event system.  CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.

The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.

This adds a unit test to verify that the event system works as expected.

(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
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8 years agoRemove ISDN hold restriction for non-bridged calls.
Richard Mudgett [Fri, 2 Mar 2012 18:38:49 +0000 (18:38 +0000)]
Remove ISDN hold restriction for non-bridged calls.

The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive.  The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application.  The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.

* Remove ISDN hold restriction for calls connected to applications.

* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.

(closes issue ASTERISK-19388)
Reported by: Birger Harzenetter
Tested by: rmudgett
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8 years agoAdds a transfer callee on hangup option (like with Dial option F) to queues.
Jonathan Rose [Fri, 2 Mar 2012 16:57:12 +0000 (16:57 +0000)]
Adds a transfer callee on hangup option (like with Dial option F) to queues.

This should (and does in my testing) act just like the Dial option of the same name.
This allows a queue member to be transfered to the next priority (no args), or to
a context/extension/priority similar to goto (with args context^extension^priority)
when a caller hangs up on them.

(closes issue ASTERISK-19283)
Reported by: To
Patches:
queue_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1785/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove bad usage of goto in ChanSpy next_channel().
Richard Mudgett [Fri, 2 Mar 2012 16:26:01 +0000 (16:26 +0000)]
Remove bad usage of goto in ChanSpy next_channel().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBeef up the IAX2 sample configuration a bit and fix some formatting issues.
Sean Bright [Fri, 2 Mar 2012 16:19:53 +0000 (16:19 +0000)]
Beef up the IAX2 sample configuration a bit and fix some formatting issues.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357821 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix channel reference leak in ChanSpy.
Richard Mudgett [Fri, 2 Mar 2012 16:03:38 +0000 (16:03 +0000)]
Fix channel reference leak in ChanSpy.

* Fix next_channel() channel reference leak in ChanSpy.

(closes issue ASTERISK-19461)
Reported by: Irontec
Patches:
      app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec

(issue ASTERISK-17515)
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8 years agoFix compile error from latest channel opaquification change.
Richard Mudgett [Fri, 2 Mar 2012 16:01:05 +0000 (16:01 +0000)]
Fix compile error from latest channel opaquification change.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357814 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoThe default value for mohinterpret is the empty string, so when resetting to
Sean Bright [Fri, 2 Mar 2012 16:00:41 +0000 (16:00 +0000)]
The default value for mohinterpret is the empty string, so when resetting to
default values don't explicitly set the value to "default."
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8 years agoFix race condition that can cause important control frames (such as a hangup) to...
Mark Michelson [Fri, 2 Mar 2012 01:33:06 +0000 (01:33 +0000)]
Fix race condition that can cause important control frames (such as a hangup) to be missed.

This takes two actions.

1. Move the reading of the alertpipe in __ast_read() to immediately before the
removal of frames from the readq. This means we won't do something silly like
read from the alertpipe, then ignore the fact that there's a frame to get from
the readq since channel's fdno is the AST_TIMING_FD.

2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the
channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because
if the rate is 0 and the timingfunc is NULL, it means that the channel's timing
fd is being invalidated, so any pending reads should not occur.

This may actually solve more issues than the referenced one below, but it's not
known at this time for sure.

(closes issue ASTERISK-19223)
reported by Frank-Michael Wittig

Review: https://reviewboard.asterisk.org/r/1779
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8 years agoFix compilation error due to typo during channel opaquification.
Mark Michelson [Fri, 2 Mar 2012 01:25:36 +0000 (01:25 +0000)]
Fix compilation error due to typo during channel opaquification.

s/ast_channel_fd_set/ast_channel_internal_fd_set/g

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357774 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOpaquify ast_channel typedefs, fd arrays, and softhangup flag
Terry Wilson [Thu, 1 Mar 2012 22:09:18 +0000 (22:09 +0000)]
Opaquify ast_channel typedefs, fd arrays, and softhangup flag

Review: https://reviewboard.asterisk.org/r/1784/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent outbound SIP NOTIFY packets from displaying a port of 0
Kinsey Moore [Thu, 1 Mar 2012 14:22:01 +0000 (14:22 +0000)]
Prevent outbound SIP NOTIFY packets from displaying a port of 0

In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the
behavior of ast_find_ourip such that port number was wiped out.  This caused
the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be
0.  This change causes ast_find_ourip to be port-preserving again.

(closes issue ASTERISK-19430)
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8 years agoUpdate stringfield documentation for removed second va_list in favor of va_copy.
Walter Doekes [Wed, 29 Feb 2012 20:41:38 +0000 (20:41 +0000)]
Update stringfield documentation for removed second va_list in favor of va_copy.

In r320946, the second va_list that was passed to ast_string_field_build_va
and friends, was removed. This patch updates the documentation to reflect that.
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8 years agoAdd IPv6 support to FastAGI.
Sean Bright [Wed, 29 Feb 2012 20:31:48 +0000 (20:31 +0000)]
Add IPv6 support to FastAGI.

Review: https://reviewboard.asterisk.org/r/1774/
Reviewed by: Simon Perreault, Mark Michelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix copying of CDR(accountcode) to local channels.
Walter Doekes [Wed, 29 Feb 2012 19:48:33 +0000 (19:48 +0000)]
Fix copying of CDR(accountcode) to local channels.

In r203638, during the addition of the Channel Event Logging, in mid-2009, this
got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the
CDR(accountcode) from the calling channel is available to dialed channels again
as well as showing up properly in the CDR's.

(closes issue ASTERISK-19384)
Reported by: jamicque
Patches: accountcode.patch (License #6033) by jamicque
Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett
........

Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 357576 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357577 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOpaquify ast_channel structs and lists
Terry Wilson [Wed, 29 Feb 2012 16:52:47 +0000 (16:52 +0000)]
Opaquify ast_channel structs and lists

Review: https://reviewboard.asterisk.org/r/1773/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in...
Jonathan Rose [Tue, 28 Feb 2012 22:31:24 +0000 (22:31 +0000)]
Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE

(issue ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
........

Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd additional character type types to supported data types for cdr_adaptive_odbc
Jonathan Rose [Tue, 28 Feb 2012 22:01:45 +0000 (22:01 +0000)]
Add additional character type types to supported data types for cdr_adaptive_odbc

The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to the list of types
cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really
UTF8 characters as much as it was a failure to respond to the exact type that was
declared/in use on that database.

(closes issue ASTERISK-19334)
Reported By: Igor Nikolaev
Patches:
cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236)
........

Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357458 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357460 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrectly reset the dialplan priority.
Tilghman Lesher [Tue, 28 Feb 2012 21:26:34 +0000 (21:26 +0000)]
Correctly reset the dialplan priority.

When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns.  However, if we just want to restore the
priority, then we need to subtract 1 before setting it.  Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan.  This is because when we return from an application, the PBX
increments the priority for us.
........

Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357421 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUse more reasonable cause code when rejecting incoming call waiting calls.
Richard Mudgett [Tue, 28 Feb 2012 21:01:09 +0000 (21:01 +0000)]
Use more reasonable cause code when rejecting incoming call waiting calls.

(closes issue ASTERISK-19397)
Reported by: Birger Harzenetter
Patches:
      nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter
........

Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357408 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357409 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agorevision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3
Jonathan Rose [Tue, 28 Feb 2012 20:43:49 +0000 (20:43 +0000)]
revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3

(issue ASTERISK-19352)
reported by: jamicque
........

Merged revisions 357405 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357406 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix REF_DEBUG compile errors.
Richard Mudgett [Tue, 28 Feb 2012 20:34:11 +0000 (20:34 +0000)]
Fix REF_DEBUG compile errors.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMoves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12
Jonathan Rose [Tue, 28 Feb 2012 20:33:53 +0000 (20:33 +0000)]
Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12

(issue ASTERISK-19352)
reported by: jamicque
........

Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357400 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357403 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdds UPGRADE.txt notes to r357266 indicating changes to transport option
Jonathan Rose [Tue, 28 Feb 2012 20:04:15 +0000 (20:04 +0000)]
Adds UPGRADE.txt notes to r357266 indicating changes to transport option

(issue ASTERISK-19352)
Reported by: jamicque
........

Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 357357 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDocumentation update. There is no AST_SOCKADDR_UNSPEC.
Sean Bright [Tue, 28 Feb 2012 19:55:32 +0000 (19:55 +0000)]
Documentation update.  There is no AST_SOCKADDR_UNSPEC.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357355 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove dupliate 'i' option table entry in app_page.c.
Richard Mudgett [Tue, 28 Feb 2012 19:37:04 +0000 (19:37 +0000)]
Remove dupliate 'i' option table entry in app_page.c.

(closes issue ASTERISK-19310)
Reported by: Makoto Dei
Patches:
      app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei
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Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357353 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd a security event for the case where fake authentication challenge is sent.
Mark Michelson [Tue, 28 Feb 2012 18:52:13 +0000 (18:52 +0000)]
Add a security event for the case where fake authentication challenge is sent.
........

Merged revisions 357318 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoConvert struct ast_tcptls_session_instance to finally use the ao2 object lock.
Richard Mudgett [Tue, 28 Feb 2012 18:46:34 +0000 (18:46 +0000)]
Convert struct ast_tcptls_session_instance to finally use the ao2 object lock.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChanges transport option in sip.conf so that using multiple instances doesn't stack.
Jonathan Rose [Tue, 28 Feb 2012 18:23:02 +0000 (18:23 +0000)]
Changes transport option in sip.conf so that using multiple instances doesn't stack.

Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.

(closes ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
........

Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357271 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357288 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAstobj2 locking enhancement.
Richard Mudgett [Tue, 28 Feb 2012 18:15:34 +0000 (18:15 +0000)]
Astobj2 locking enhancement.

Add the ability to specify what kind of locking an ao2 object has when it
is allocated.  The locking could be one of: MUTEX, RWLOCK, or none.

New API:

ao2_t_alloc_options()
ao2_alloc_options()
ao2_t_container_alloc_options()
ao2_container_alloc_options()

ao2_rdlock()
ao2_wrlock()
ao2_tryrdlock()
ao2_trywrlock()

The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning
change.  They no longer mean that the object is protected by an external
mechanism.  They mean the lock associated with the object has already been
manually obtained by one of the ao2_lock calls.  This change is necessary
for RWLOCK support since they are not reentrant.  Also an operation on an
ao2 container may require promoting a read lock to a write lock by
releasing the already held read lock to re-acquire as a write lock.

Replaced API calls:

ao2_t_link_nolock()
ao2_link_nolock()
ao2_t_unlink_nolock()
ao2_unlink_nolock()

with the respective

ao2_t_link_flags()
ao2_link_flags()
ao2_t_unlink_flags()
ao2_unlink_flags()

API calls to be more flexible and to allow an anticipated enhancement to
control linking duplicate objects into a container.

The changes to format.c and format_cap.c are taking advantange of the new
ao2 locking options to simplify the use of the format capabilities
containers.

Review: https://reviewboard.asterisk.org/r/1554/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake COMPILE_DOUBLE magic actually work.
Kevin P. Fleming [Tue, 28 Feb 2012 14:47:23 +0000 (14:47 +0000)]
Make COMPILE_DOUBLE magic actually work.

The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.
........

Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357213 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357214 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTrailing whitespace cleanup.
Kevin P. Fleming [Tue, 28 Feb 2012 14:15:33 +0000 (14:15 +0000)]
Trailing whitespace cleanup.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357178 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd ability to clone ao2 containers.
Richard Mudgett [Tue, 28 Feb 2012 00:42:38 +0000 (00:42 +0000)]
Add ability to clone ao2 containers.

Occasionally there is a need to put all objects in one container also into
another container.

Some reasons you might need to do this:

1) You need to reconfigure a container.  You would do this by creating a
new container with the new configuration and ao2_container_dup the old
container into it.  Then replace the old container with the new.  Then
destroy the old container.

2) You need the contents of a container to remain stable while operating
on all of the objects.  You would do this by creating a cloned container
of the original with ao2_container_clone.  The cloned container is a
snapshot of the objects at the time of the cloning.  When done, just
destroy the cloned container.

Review: https://reviewboard.asterisk.org/r/1746/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357145 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix ast_channel allocation init setting priority to -1 instead of 1.
Richard Mudgett [Tue, 28 Feb 2012 00:17:19 +0000 (00:17 +0000)]
Fix ast_channel allocation init setting priority to -1 instead of 1.

* Fix opaquification conversion error.

(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
      asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357101 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix callerid of Originated calls.
Richard Mudgett [Mon, 27 Feb 2012 23:42:12 +0000 (23:42 +0000)]
Fix callerid of Originated calls.

Thanks to Matt Riddell for tracking this down.

(closes issue ASTERISK-19385)
Reported by: ornix
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Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 357095 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357096 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoConverts locking for odbc containers from ast_mutex_lock to ao2_locks.
Jonathan Rose [Mon, 27 Feb 2012 19:55:14 +0000 (19:55 +0000)]
Converts locking for odbc containers from ast_mutex_lock to ao2_locks.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAddress comments from Mark Michelson
Sean Bright [Mon, 27 Feb 2012 17:03:46 +0000 (17:03 +0000)]
Address comments from Mark Michelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDeprecated macro usage for connected line, redirecting, and CCSS
Kinsey Moore [Mon, 27 Feb 2012 16:50:19 +0000 (16:50 +0000)]
Deprecated macro usage for connected line, redirecting, and CCSS

This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated.  This also adds
deprecation warnings for those features when used and in documentation.

Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoConvert netsock.h over to use ast_sockaddrs rather than sockaddr_in and update
Sean Bright [Mon, 27 Feb 2012 16:31:24 +0000 (16:31 +0000)]
Convert netsock.h over to use ast_sockaddrs rather than sockaddr_in and update
chan_iax2 to pass in the correct types.

chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk
at this point, so this feels like a safe change to make.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdds an option to sip.conf that prevents diversion headers from being added.
Jonathan Rose [Mon, 27 Feb 2012 16:24:17 +0000 (16:24 +0000)]
Adds an option to sip.conf that prevents diversion headers from being added.

send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.

(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoThere isn't much point in saving off and restoring a value that we never use again.
Sean Bright [Mon, 27 Feb 2012 16:12:51 +0000 (16:12 +0000)]
There isn't much point in saving off and restoring a value that we never use again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356966 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCopy CDR variables when set during a bridge
Terry Wilson [Mon, 27 Feb 2012 16:08:28 +0000 (16:08 +0000)]
Copy CDR variables when set during a bridge

This patch makes sure amaflags, accountcode, and userfield get copied
to the bridge CDR when set during a bridge (like via a custom feature).

(closes issue ASTERISK-16990)
Review: https://reviewboard.asterisk.org/r/1721/
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Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356964 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356965 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove possible segfaults from res_odbc by adding locks around usage of odbc handle
Jonathan Rose [Mon, 27 Feb 2012 15:35:10 +0000 (15:35 +0000)]
Remove possible segfaults from res_odbc by adding locks around usage of odbc handle

(closes issue ASTERISK-19011)
Reported by: Walter Doekes
Patches:
issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674)
review: https://reviewboard.asterisk.org/r/1719/
review: https://reviewboard.asterisk.org/r/1622/
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Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356961 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake ast_netsock_set_qos() delegate to ast_set_qos().
Sean Bright [Mon, 27 Feb 2012 14:57:23 +0000 (14:57 +0000)]
Make ast_netsock_set_qos() delegate to ast_set_qos().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356916 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrect typo in deprecation comment.
Sean Bright [Mon, 27 Feb 2012 14:15:24 +0000 (14:15 +0000)]
Correct typo in deprecation comment.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrefer ast_set_qos() over ast_netsock_set_qos()
Sean Bright [Mon, 27 Feb 2012 14:13:58 +0000 (14:13 +0000)]
Prefer ast_set_qos() over ast_netsock_set_qos()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove trailing whitespace
Sean Bright [Mon, 27 Feb 2012 13:45:10 +0000 (13:45 +0000)]
Remove trailing whitespace

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356881 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd support change gatekeeper mode or ip per ooh323 reload command
Alexandr Anikin [Sun, 26 Feb 2012 18:25:23 +0000 (18:25 +0000)]
Add support change gatekeeper mode or ip per ooh323 reload command

(issue ASTERISK-19298)
Reported by: Dmitry Melekhov
Patches:
        change_gk_on_reload-1.patch (License #5415)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix crash in app_voicemail during close_mailbox
Matthew Jordan [Sat, 25 Feb 2012 17:22:55 +0000 (17:22 +0000)]
Fix crash in app_voicemail during close_mailbox

In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers.  However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL.  In that case, an invalid free would be attempted,
which could crash app_voicemail.  As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers.  This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
........

Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356798 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoastobj2.h comment tweaks.
Richard Mudgett [Fri, 24 Feb 2012 23:40:23 +0000 (23:40 +0000)]
astobj2.h comment tweaks.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356765 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoastobj2.h documentation updates.
Richard Mudgett [Fri, 24 Feb 2012 20:47:12 +0000 (20:47 +0000)]
astobj2.h documentation updates.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix worker thread resource leak in SIP TCP/TLS.
Richard Mudgett [Fri, 24 Feb 2012 18:33:04 +0000 (18:33 +0000)]
Fix worker thread resource leak in SIP TCP/TLS.

The SIP TCP/TLS worker threads were created joinable but noone could join
them if they died on their own.

* Fix the SIP TCP/TLS worker threads to not be created joinable.

* _sip_tcp_helper_thread() only needs one parameter since the pvt
parameter is only passed in as NULL and never used.

(closes issue ASTERISK-19203)
Reported by: Steve Davies

Review: https://reviewboard.asterisk.org/r/1714/
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Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356690 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove srtp_shutdown from res_srtp
Matthew Jordan [Fri, 24 Feb 2012 17:43:26 +0000 (17:43 +0000)]
Remove srtp_shutdown from res_srtp

The patch for ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload.  Unfortunately, not all distributions
have the srtp_shutdown call.  As such, this patch removes calling
srtp_shutdown.
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Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356651 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAllow SRTP policies to be reloaded
Matthew Jordan [Fri, 24 Feb 2012 15:10:35 +0000 (15:10 +0000)]
Allow SRTP policies to be reloaded

Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place.  Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp.  This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.

This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library.  From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately.  This
was changed to account for the differences in handling remote and local
policies in libsrtp.

Review: https://reviewboard.asterisk.org/r/1741/

(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
  srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
  (with some small modifications for this check-in)
........

Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 356605 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356606 65c4cc65-6c06-0410-ace0-fbb531ad65f3