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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* 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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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
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
https://origsvn.digium.com/svn/asterisk/branches/10
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 333947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed
before ast_cond_wait called to wait this
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r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines
security fix. really drop call if signalling addr is not same as socket
addr
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r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines
Merged revisions 331146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines
move ast_cond_signal for admitted call after all data filled/freed
clear all log channels by pointed number not only first
free allocated callToken in ooh323_answer
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r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines
Setup IP proto version for call in GK mode
Added additional check for IP semantics before parse destination
by ast_parse_args due to it can parse numeric as IP.
(closes issue ASTERISK-18218)
Reported by: slesru
Patch: ASTERISK-18218.patch
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r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines
Fix a crash in the MySQL() application.
This code was not handling channel datastores safely. The channel
must be locked.
(closes issue #17964)
Reported by: wuwu
Patches:
issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
Tested by: wuwu
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r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines
Don't duplicate our data on the stack and just use the MYSQL_ROW directly.
With large result sets we were blowing out the stack.
(closes issue #19090)
Reported by: mickecarlsson
Patches:
issue19090_trunk_svn.patch uploaded by seanbright (license 71)
Tested by: mickecarlsson
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IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313482 65c4cc65-6c06-0410-ace0-fbb531ad65f3