Age | Commit message (Collapse) | Author |
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Per the bluez API, in order to bind to the first available port, the rc_channel
field of the socket addressing structure used to bind the socket should be set
to 0. Previously, Asterisk had set the rc_channel field set to 1, causing it
to connect to whatever happens to be on port 1.
We could probably not explicitly set rc_channel to 0 since we memset the struct
earlier, but explicitly setting it will hopefully prevent someone from coming
in and setting it to some explicit port in the future.
(closes issue ASTERISK-16357)
Reported by: challado
Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin, eliafino, David van Geyn
patches:
ASTERISK-16357.diff uploaded by Nikolay Ilduganov (license 6253)
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The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.
The SMSSRC should now populate correctly.
(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
fixSMSSRC.patch uploaded by jonax (license 6320)
(closes issue ASTERISK-19153)
Reported by: Panos Gkikakis
patches:
sms-sender-fix.diff uploaded by roeften (license 5884)
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From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.
(issue ASTERISK-20658)
Reported by: wdoekes
patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
it is effectively a noop. No channels can attach a reference to that
module.
* Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
The caller of unload_module() has already called it.
* Removed redundant channel module references in pbx_dundi.c. The
registered dialplan function callback dispatchers for the read/read2/write
callbacks already reference the module before calling.
* pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
functions to the first thing the unload_module() does. This will reduce
the chance of new channels using DUNDi services while the module is being
torn down.
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Update and extend the configuration_file group and enable linking.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Update title that was left behind many years ago. Used revision 6596 as my guide for what it should be.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Start adding configuration file linking and pages. Add module loading doxygen block.
Breaking up commits to keep it easy to track
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
(closes issue ASTERISK-19308)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-19308.patch
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(close issue ASTERISK-20094)
Patches:
ASTERISK-20094-2.patch
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this was broken in rev 369602
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to H.323 Gatekeeper (GkClient state)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(close issue ASTERISK-20094)
Patches:
ASTERISK-20094.patch
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This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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This patch allows you to specify a port number for the MySQL server.
It's useful if a MySQL server is running on a non-standard port.
Even though this module is deprecated in favor of func_odbc, someone
asked for this feature and it seems pretty harmless to add.
It has been tested using a number of combinations of with/without a
port number specified in the dialplan and changing the port number
for mysqld.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
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options are documented in config sample
sample config rename to proper name - ooh323.conf
To change media address ooh323 send empty TCS if there was
completed TCS exchange or send facility forwardedelements
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.
If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange
Review: https://reviewboard.asterisk.org/r/1607/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/1767
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue ASTERISK-19298)
Reported by:
Dmitry Melekhov
Patches:
ASTERISK-18322-2.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(rtp voice, video, udptl except rtcp)
(closes issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury
Patches:
19179-ooh323-ast10.patch
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This patch fixes a potential crash in mp3_read() by not assuming that
dbuf has enough data to finish filling up the output buffer. The patch
also makes sure that the dbuf state gets reset after we know we read
everything out of it already.
In passing, this patch includes some other cleanups of this module,
including stripping trailing whitespace, formatting fixes based on
coding guidelines, and removing a number of unused members from the
private state struct.
(closes issue ASTERISK-19761)
Reported by: Chris Maciejewsk
Tested by: Chris Maciejewsk
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from value to pointer per functions that use this.
(close issue ASTERISK-19670)
Reported by: Matt Jordan
Patches:
ASTERISK-19670.patch (License #5415)
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instead of without data buffer
(close issue ASTERISK-19674)
Reported by: Matt Jordan
Patches:
ASTERISK-19674.patch (License #5415)
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(issue ASTERISK-19663)
Reported by: Matt Jordan
Patches:
ASTERISK-19663-ooh323.patch (License #5415)
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Review: https://reviewboard.asterisk.org/r/1732/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array
of size 16) would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte integers),
which would still leave it within the allocated memory of struct hfp. This
would corrupt other elements in that struct but not necessarily cause any
further issues.
* app_sms: The array imsg is of size 250, while the array (ud) that the data
is copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This would corrupt
the user data header (array udh) adjacent to ud.
* chan_unistim: A number of invalid memmoves are corrected. These would move
data (which may or may not be valid) into the ends of these buffers.
* asterisk: ast_console_toggle_loglevel does not check that the console log
level being set is less then or equal to the allowed log levels of 32.
* format_pref: In ast_codec_pref_prepend, if any occurrence of the specified
codec is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array.
* jitterbuf: If the element being placed into the jitter buffer lands in the
last available slot in the jitter history buffer, the insertion sort attempts
to move the last entry in the buffer into one slot past the maximum length
of the buffer. Note that this occurred for both the min and max jitter
history buffers.
* tdd: If a read from fsk_serial returns a character that is greater then 32,
an attempt to read past one of the statically defined arrays containing the
values that character maps to would occur.
* localtime: struct ast_time and tm are not the same size - ast_time is larger,
although it contains the elements of tm within it in the same layout. Hence,
when using memcpy to copy the contents of tm into ast_time, the size of tm
should be used, as opposed to the size of ast_time.
* extconf: this treats ast_timing's minmask array as if it had a length of 48,
when it has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48.
(issue ASTERISK-19668)
Reported by: Matt Jordan
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The important parts of the patch were already applied through other updates.
(closes issue ASTERISK-19445)
Reported by: Makoto Dei
Patches:
memset-memcpy-length.patch uploaded by Makoto Dei (license 5027)
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(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
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Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8
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(closes issue ASTERISK-19540)
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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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Review: https://reviewboard.asterisk.org/r/1784/
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Review: https://reviewboard.asterisk.org/r/1773/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(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
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Review: https://reviewboard.asterisk.org/r/1770/
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Review: https://reviewboard.asterisk.org/r/1753/
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This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes issue ASTERISK-19298)
Reported by: robinfood
Patches:
issue19298.patch uploaded by may213 (License #5415)
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Review: https://reviewboard.asterisk.org/r/1733/
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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Review: https://reviewboard.asterisk.org/r/1707/
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sounds).
(Closes issue ASTERISK-19233)
Reported by: Matt Behrens
Patches:
chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
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Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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