Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
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isn't installed
Include the appropriate declarations when not using termcap, but term+curses
and [n]curses do not exist.
(closes issue ASTERISK-22351)
Reported by: A. Iglesias
Patches:
issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674)
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In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.
In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.
(closes issue ASTERISK-22718)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/2925/
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In r337595, additional security events were added for chan_sip
authentication failures. The new IEs added to the existing invalid
password event were defined as required IEs, but existing users of the
event did not set the new IEs and could not since they didn't apply to
existing uses. They are now marked as optional IEs.
(closes issue ASTERISK-22578)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a config object was allocated but one of its global objects was
never encountered, then the global object's defaults were never
applied. Ensure that global objects are initialized properly upon
allocation instead of on configuration.
Review: https://reviewboard.asterisk.org/r/2866/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixed a memory leak discovered in the logger where a temporary string buffer
was not being freed.
(closes issue ASTERISK-22540)
Reported by: John Hardin
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Backported the following as applied to udptl.c:
* -r398020 Fixup udpdl defaults if config file not present.
* -r398533 Fixup improper use of ao2_global_obj_replace().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adjusts '/'-to-'#' replacement to replace all instances of '/'
instead of just the first to ensure that the jitter buffer log file
gets the correct name as per Richard Kenner's suggestion.
(closes issue ASTERISK-21036)
Reported by: Richard Kenner
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When processing the lines under the [applicationmap] context in features.conf, a
segfault occurs from attempting to process a line with an invalid syntax
(basically missing most of the arguments).
Example:
[applicationmap]
automon=*6
* This patch moves the checking for empty arguments to before they are accessed.
* Also, checked the "todo" comment and removed it. Some applications do not
require arguments.
(closes issue ASTERISK-22416)
Reported by: CGI.NET
Tested by: CGI.NET
Patches:
asterisk-22416-check-syntax-first_v2.diff by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2803
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The remote console continued to have issues with its output. In this case CLI
command output would either not show up (if verbose level = 0) or would contain
verbose prefixes (if verbose level > 0) once log messages were sent to the
remote console. The fix now now adds verbose prefix data to all new lines
contained in a verbose log string.
(closes issue ASTERISK-22450)
Reported by: David Brillert
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2825/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.
With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).
This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).
(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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ast_xmldoc_printable returns an allocated block that must be freed by the
caller. Fixed manager.c and res_agi.c to stop leaking these results.
(closes issue ASTERISK-22395)
Reported by: Corey Farrell
Patches:
manager-leaks-11.patch uploaded by coreyfarrell (license 5909)
res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909)
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Fixed a features.c test that leaked a reference to a parked call. This caused
chancount to never reach 0, so graceful shutdown stops. Also added an
unregister test.
(closes issue ASTERISK-22413)
Reported by: Corey Farrell
Patches:
features-TEST_FRAMEWORK.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Refactored cases where a combination of ast_verbose/options_verbose were
present. Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used. Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.
* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().
* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.
* Fixed some formatting in ast_bt_get_symbols().
* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.
* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.
* Moved __dump_backtrace() because of compile issues with the utils
directory.
(closes issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2778/
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The --version-script,asterisk.exports linker flag (and the module
exports) didn't provide _IO_stdin_used in the list of exported symbols.
That causes some kind of libc compatibility mode to kick in, where
stdio file structures (stdout/stderr) land somewhere else. In the
case of the Sparc, they landed on misaligned memory.
This became apparent first after r376428 (Reorder startup sequence)
when a lot of ast_log's were replaced with fprintf's. Writing to
stderr triggered a SIGBUS. (Compared to x86 and amd64 architectures,
the Sparc is very picky about memory alignment.)
(issue ASTERISK-21763)
(issue ASTERISK-21665)
Reported by: Jeremy Kister
Review: https://reviewboard.asterisk.org/r/2760/
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If the file udptl.conf is unavailable at startup, UDPTL will fail to
initialize and while it makes some noise, it isn't immediately
obvious why consumers start to fail when using it. This patch makes
UDPTL load as though an empty config was provided when udptl is
unavailable at startup.
(closes issue ASTERISK-22349)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2773/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch ensures that CLI commands enabled by DEBUG_FD_LEAKS and
DEBUG_THREADLOCALS are cleaned up properly on exit.
(closes issue ASTERISK-22238)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
debug_cli_unregister.patch uploaded by Corey Farrell
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This fixes a single-attribute memory leak that was occurring when the
"required" attribute was not true.
(closes issue ASTERISK-22249)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
xmldoc-free_attr_required.patch uploaded by Corey Farrell
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It is not safe to iterate over a macro'd list of ao2 objects, deref them such
that the item's destructor is called, and leave them in the list. The list
macro to iterate over items requires the item to be a valid allocated object
in order to proceed to the next item; with MALLOC_DEBUG on the corruption of
the linked list is caught in the crash.
This patch fixes the invalid access to free'd memory by removing the ao2 item
from the list before de-refing it.
Note that this is a backport of r396915 from Asterisk trunk.
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This patch modifies parsing of cookies in Asterisk's http server by doing an
explicit comparison of the "Cookie" header instead of looking at the first
6 characters to determine if the header is a cookie header. This avoids
parsing "Cookie2" headers and overwriting the previously parsed "Cookie"
header.
Note that we probably should be appending the cookies in each "Cookie"
header to the parsed results; however, while clients can send multiple
cookie headers they never really do. While this patch doesn't improve
Asterisk's behavior in that regard, it shouldn't make it any worse either.
Note that the solution in this patch was pointed out on the issue by the
issue reporter, Stuart Henderson.
(closes issue ASTERISK-21789)
Reported by: Stuart Henderson
Tested by: mjordan, Stuart Henderson
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This patch fixes the following memory leaks:
* http.c: The structure containing the addresses to bind to was not being
deallocated when no longer used
* named_acl.c: The global configuration information was not disposed of
* config_options.c: An invalid read was occurring for certain option types.
* res_calendar.c: The loaded calendars on module unload were not being
properly disposed of.
* chan_motif.c: The format capabilities needed to be disposed of on module
unload. In addition, this now specifies the default options for the
maxpayloads and maxicecandidates in such a way that it doesn't cause the
invalid read in config_options.c to occur.
(issue ASTERISK-21906)
Reported by: John Hardin
patches:
http.patch uploaded by jhardin (license 6512)
named_acl.patch uploaded by jhardin (license 6512)
config_options.patch uploaded by jhardin (license 6512)
res_calendar.patch uploaded by jhardin (license 6512)
chan_motif.patch uploaded by jhardin (license 6512)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@392810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two memory leaks:
* When we load a module with the LOAD_PRIORITY flag, we remove its entry from
the load order list. Unfortunately, we don't free the memory associated with
entry in the list. This patch corrects that and properly frees the memory
for the module in the list.
* When adding a custom format (such as SILK or CELT), the routine for adding
the format was leaking a reference. RAII_VAR cleans this up properly.
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Check the returned bridged pointer for NULL to avoid a crash. It looks
like chan_agent is returning a NULL pointer when it probably should be
returning a pointer to the channel the Agent channel is pretending to be.
(closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles
Patches:
jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Rodrigo P. Telles
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk shuts down and shuts down the loggin gsubsystem, any
messages currently in flight will not get logged. This patch prevents the
loop writing messages from breaking out prematurely, such that all of the
messages are logged.
(closes issue ASTERISK-21716)
Reported by: Corey Farrell
patches:
logger-process-all-messages.patch uploaded by Corey Farrell (license 5909)
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If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying
to access a possible NULL t->track object. A NULL check has been added before
trying to access the memory.
(closes issue ASTERISK-21724)
Reported by: Corey Farrell
Fixed by: Corey Farrell
Patches:
ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909)
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The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.
(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CALL-ID (ie [C-00000074]) is missing when logging to syslog. This was just
an oversight when this feature was added.
* Add CALL-IDs when using syslog
(closes issue ASTERISK-21430)
Reported by: Nikola Ciprich
Tested by: Nikola Ciprich, Michael L. Young
Patches:
asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2526/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Way back when in the dark days of Asterisk 1.8.9, blind transferring a call
in a context that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad thing". The fix
was to properly check for the softhangup flags on the channel and only execute
the 'h' extension logic (and, in later versions, hangup handler logic) if the
channel was well and truly dead (Jim).
Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected
that the channel was leaving the bridge (but not to die) caused some crucial
snippet of CDR code, lying in ambush in the middle of the bridging code, to
not get executed. This had the effect of blowing away one of the CDRs that is
typically created during a blind transfer.
While we live and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and still manages to
not run the 'h' extension during a blind transfer (at least not when it's
supposed to).
Thanks to Steve Davies for diagnosing this and providing a fix.
Review: https://reviewboard.asterisk.org/r/2476
(closes issue ASTERISK-21394)
Reported by: Ishfaq Malik
Tested by: Ishfaq Malik, mjordan
patches:
fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012)
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If a system allows for its usage, Asterisk will use glob to help parse
Asterisk .conf files. The config file loading routine was leaking the memory
allocated by the glob() routine when the config file was in an unmodified
or invalid state.
This patch properly calls globfree in those off nominal paths.
(closes issue ASTERISK-21412)
Reported by: Corey Farrell
patches:
config_glob_leak.patch uploaded by Corey Farrell (license 5909)
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This patch cleans up two things features:
* It properly unregisters the CLI commands that features registered
* It cancels and performs a pthread_join on the created parking thread. This
not only properly joins a non-detached thread, but also prevents disposing
of the parking lots prior to the parking thread completely exiting.
(closes issue ASTERISK-21407)
Reported by: Corey Farrell
patches:
features_shutdown-r2.patch uploaded by Corey Farrell (License 5909)
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The two party bridging loops were changing the bridge peer pointers
without the channel locks held. Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.
(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
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In messages.c, there are several places in the code where we create a
tmp_tech_holder and pass that into an ao2_find call. Unfortunately, we
weren't initializing the rwlock on the tmp_tech_holder, which the hash
function was locking. It's apparently harmless, but still not the best
code.
This patch extracts all that copy/pasted code into two functions,
msg_find_by_tech and msg_find_by_tech_name, which properly initialize
and destroy the rwlock on the tmp_tech_holder.
Review: https://reviewboard.asterisk.org/r/2454/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in,
then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event,
so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE.
(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2452/
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While looking at the security vulnerability in ASTERISK-20967, Walter noticed
a file descriptor leak and some other issues in off nominal code paths. This
patch corrects them.
Note that this patch is not related to the vulnerability in ASTERISK-20967,
but the patch was placed on that issue.
(closes issue ASTERISK-20967)
Reported by: wdoekes
patches:
issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674)
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AST-2012-014, fixed in January of this year, contained a fix for Asterisk's
HTTP server for a remotely-triggered crash. While the fix put in place fixed
the possibility for the crash to be triggered, a denial of service vector still
exists with that solution if an attacker sends one or more HTTP POST requests
with very large Content-Length values. This patch resolves this by capping
the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with
Content-Length greater than this cap will not result in any memory allocation.
The POST will be responded to with an HTTP 413 "Request Entity Too Large"
response.
This issue was reported by Christoph Hebeisen of TELUS Security Labs
(closes issue ASTERISK-20967)
Reported by: Christoph Hebeisen
patches:
AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
AST-2013-002-10.diff uploaded by mmichelson (License 5049)
AST-2013-002-11.diff uploaded by mmichelson (License 5049)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
r375757 attempted to resolve a race condition between multiple submissions of
CDRs while in batch mode from attempting to destroy the scheduled batch
submission by extending the batch CDR lock. Unfortunately, this causes a
deadlock between the pending CDR lock and the batch CDR lock. This patch
resolves the intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is kept to protect
manipulation of the batch CDR settings, but has been placed such that it
is not held when the pending lock is held.
Thanks to Chase Venters for providing lock analysis on the issue.
(issue ASTERISK-21162)
Reported by: Chase Venters
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AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
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Several new IEs were not given types (or names), causing the comparison
function to improperly succeed. This adds those.
(closes issue AST-1128)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.
Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.
(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
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The incorrect callid was being written to the "data1" field in queue_log table
for transfer events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to" number and
places that in the "data1" field of the queue_log table when a transfer event
is triggered.
(closes issue ASTERISK-19960)
Reported by: vladimir shmagin
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This patch fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.
(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
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While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.
Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything). If it is different then
store/replace and log, otherwise just leave as is. Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.
(issue ASTERISK-21014)
(closes issue ASTERISK-21014)
Report by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2324/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was introduced which removed automatic fallback behavior from
the PBX. This behavior was used by call parking (or at least documented as
how the feature works) in order to select an extension when the flat channel
extension wasn't available from the comebackcontext. Parking now handles
the fallbacks internally in order to keep behavior matching with how it is
documented.
(closes issue ASTERISK-20716)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2296/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch came about due to a problem observed where wav files had an
empty header. The header is supposed to be updated in wav_close(). It
turns out that this was broken when the cache_record_files option from
asterisk.conf was enabled. The cleanup code was moving the file to its
final destination *before* running the close() method of the file
destructor, so the header didn't get updated.
Another problem here is that the move was being done before actually
closing the FILE *.
Finally, the last bug fixed here is that I noticed that wav_close()
checks for stream->filename to be non-NULL. In the previous cleanup
order, it's checking a pointer to freed memory. This doesn't actually
cause anything to break, but it's treading on dangerous waters. Now the
free() of stream->filename is happening after the format module's
close() method gets called, so it's safer.
Review: https://reviewboard.asterisk.org/r/2286/
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When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
* Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console cannot connect
to a running instance of Asterisk.
In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.
Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.
(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
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When Asterisk forks itself into the background via a call to daemon, it must
re-set the pid value of the new process. Otherwise, astcanary gets the pid
value of the process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer communicate
with the proverbial canary in the coal mine.
This patch ensures that the correct process identifier is used by astcanary.
Note that this is getting committed to 10 as a regression fix.
(closes issue ASTERISK-20947)
Reported by: Jakob Hirsch
Tested by: mjordan
patches:
asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113)
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XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
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This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
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When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
* issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
* issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
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Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.
(issue ASTNOW-284)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Adds the following CLI commands to control MALLOC_DEBUG reporting of
unreleased malloc memory when Asterisk is shut down.
memory atexit list on
memory atexit list off
memory atexit summary byline
memory atexit summary byfunc
memory atexit summary byfile
memory atexit summary off
* Made check all remaining allocated region blocks atexit for fence
violations.
* Increased the allocated region hash table size by about three times. It
still isn't large enough considering the number of malloced blocks
Asterisk uses.
* Made CLI "memory show allocations anomalies" use
regions_check_all_fences().
Review: https://reviewboard.asterisk.org/r/2196/
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* Fixed CLI "memory show allocations" misspelling of anomalies option.
The command will still accept the original misspelling.
* Miscellaneous tweaks to CLI "memory show allocations" command output
format.
* Made CLI "memory show summary" summarize by line number instead of by
function if a filename is given.
* Made CLI "memory show summary" sort its output by filename or
function-name/line-number depending upon request.
* Miscellaneous tweaks to CLI "memory show summary" command output format.
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Prior to this patch, challenge would yield a multiple logins error if used
without providing the username (which isn't really supposed to be an argument
to challenge) if allowmultiplelogin was set to no because allowmultiplelogin
finds a user with a zero length login name. This check is simply disabled for
the challenge action when the username is empty by this patch.
(closes issue ASTERISK-20677)
Reported by: Vladimir
Patches:
challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182)
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The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.
* Made the old exten matching code consistently ignore '-' chars.
* Made the old exten matching code consistently handle case in the
matching.
* Made ignore empty character sets.
* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented. The only
user of it in pbx_lua.c was testing for -1. It was originally returning
the strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of characters
and start with the same character. Character set [0-9] now sorts before
[02-9a] as originally intended.
* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.
(closes issue ASTERISK-19205)
Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2201/
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Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
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Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup. This can occur when a recursive mutex is obtained prior to the
daemon call executing. Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex. If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid. Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.
This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information. This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.
This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
localtime monitoring thread. This thread is now spawned after Asterisk has
fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
startup. This limits the potential of threading primitives being accessed
by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to daemon,
as calls to ast_log may access recursive mutexes that store thread related
information.
d) It reorganizes when thread local storage is created for storing lock
information during the creation of threads. Prior to this patch, the
read/write lock protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being initialized;
this patch prevents that.
On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.
Review: https://reviewboard.asterisk.org/r/2197
(closes issue ASTERISK-19463)
Reported by: mjordan
Tested by: mjordan
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generator.
This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.
(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
from the function which retrieves headers from an action indicates that the header has a
value. This is incorrect and the contents must be checked to see if they are blank.
(closes issue ASTERISK-20628)
Reported by: jkroon
Patches:
asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
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* Makes malloc() behave like calloc(). It will return a memory block
filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.
* Puts the freed memory block into a circular array so it is not reused
immediately.
* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.
* Made the astmm_log message wording better.
* Made crash if the DO_CRASH menuselect option is enabled and something is
found.
* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.
* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().
* Updated handle_memory_show() CLI usage help.
Review: https://reviewboard.asterisk.org/r/2182/
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.
* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.
(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
Modified to jira_asterisk_20492_v1.8.patch
jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2169/
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Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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Manager's tcp/tls objects have a periodic function that purge old manager
sessions periodically. During shutdown, the underlying container holding
those sessions can be disposed of and set to NULL before the tcp/tls periodic
function is stopped. If the periodic function fires, it will attempt to
iterate over a NULL container.
This patch checks for whether or not the sessions container exists before
attempting to purge sessions out of it. If the sessions container is NULL,
we simply return.
Note that this error was also caught by the Asterisk Test Suite.
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This patch does two things:
1) It properly unregisters the manager CLI commands
2) It cleans up AMI users on exit. Prior to this patch, the AMI users
were not being disposed of properly, resulting in a memory leak.
(closes issue ASTERISK-20646)
Reported by: Corey Farrell
patches:
manager_shutdown.patch uploaded by Corey Farrell (license 5909)
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The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3
database. These statements should be finalized during Asterisk shutdown so
that the SQLite3 database can be properly closed. Failure to finalize the
statements results in a memory leak and a failure when closing the database.
This patch fixes those issues by ensuring that all prepared statements are
properly finalized at shutdown.
(closes issue ASTERISK-20647)
Reported by: Corey Farrell
patches:
astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909)
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This patch fixes two memory leaks:
1) When building XML documentation items, the 'name' attribute was extracted
from XML elements but not properly freed after being copied into the item
being built.
2) When unloading XML documentation, the doctree container objects were not
properly freed.
This patch corrects these memory leaks. Note that this patch was modified
slightly for this commmit, as the case where the 'name' attribute doesn't
exist also wasn't handled in the item construction. This patch also checks
for that attribute not existing.
(closes issue ASTERISK-20648)
Reported by: Corey Farrell
Tested by: mjordan
patches:
xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Asterisk Test Suite caught an error condition where a scheduled CDR batch
write can be deleted twice if two channels attempt to post their CDRs at the
same time. The batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually scheduling the
CDR write. As such, two threads can attempt to remove the currently scheduled
batch write at the same time, resulting in an assertion error.
This patch extends the time that the mutex is locked to encompass actually
scheduling the write. This prevents two threads from unscheduling the
currently scheduled write at the same time.
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Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
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This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.
This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.
Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.
Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.
(closes issue ASTERISK-20212)
reported by Phil Ciccone
Review: https://reviewboard.asterisk.org/r/2123
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Similar to r369351, the billing duration can be skewed when batch mode is
enabled. This happened much more rarely than the duration, as it only
occured when the call was answered (thereby indicating an actual answer
time) and immediately hung up on (indicating a billsec of 0). Since
a billing time of '0' can either mean that the call immediately ended
or that the CDR was improperly answered, we have to use additional information
to know whether or not we can trust the CDR billsec value. Prior to this
patch, we looked to see if we had a valid answer time. If we did, and
billsec was zero, we used the current time to calculate what billsec value
we could from the CDR being written. If batch mode is enabled, this will
incorrectly report a billsec value being much greater than the actual
duration of the call.
Instead of relying on the presence of an answer time to know whether or not
we can re-calculate the billsec for the CDR, we now also use the presence
of the CDR's end time to know if we need to re-calculate or whether we can
trust the billsec value that we have. This prevents erroneous jumps in the
billsec value, while still making sure that in the worst case, some billing
time will be calculated.
(closes issue AST-1016)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
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During testing, it was discovered that having chan_sip
export global symbols was problematic.
The biggest problem was that load order was affected.
Trying to use realtime could be problematic since in
all likelihood the necessary realtime driver(s) would
not be loaded before chan_sip.
In addition, it was found that it was impossible to
use the Digium Phone Module for Asterisk since it
must be loaded before chan_sip since it must hook
into chan_sip's configuration parsing.
The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like
app_voicemail.
(closes issue ASTERISK-20545)
Reported by: kmoore
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to usage of ast_hook_send_action, AMI action handling code should
be able to handle a NULL mansession->session. This would cause a crash
on NULL dereference if action_originate was called from
ast_hook_send_action.
(closes issue ASTERISK-20544)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes trivial build error on Solaris:
acl.c: In function `get_local_address':
acl.c:196: error: `best_score' undeclared (first use in this function)
acl.c:196: error: (Each undeclared identifier is reported only once
acl.c:196: error: for each function it appears in.)
make[2]: *** [acl.o] Error 1
(issue ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang
patches:
0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch by Shaun Ruffell (license 5417)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the channel technology agnostic application/AMI command MessageSend,
the "From" field is technically optional for the SIP channel driver. However,
if being sent by the XMPP resource module (either res_xmpp or res_jabber), the
"From" field is necessary, and must correspond to a defined account. This
patch updates the documentation for this application/AMI command to reflect
this.
(closes issue ASTERISK-20405)
Reported by: Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines
Improve AMI long line error handling
In AMI's parser, when it receives a long line (> 1024 characters), it discards
that line, but continues to process the message normally.
Typically, this is not a problem because a) who has lines that long and b)
usually a discarded line results in an invalid message. But if that line is
specifying an optional field, then the message will be processed, you get a
'Response: Success', but things don't work the way you expected them to.
This patch changes the behavior when a line-too-long parse error occurs.
* Changes the log message to avoid way-too-long (and truncated anyways) log
messages
* Adds a 'parsing' status flag to Response: Success
* Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long
* Responds with an appropriate error if parsing != MESSAGE_OKAY
(closes issue AST-961)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2142/
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r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
I've committed too much. Reverting part of r374570.
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The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.
This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).
* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
results in successful result
(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
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Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.
Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.
Power level difference between frequencies for different Administrations/RPOAs
NTT = Max. 5 dB
AT&T = 4dB(reverse) to 8dB(normal)
Danish = Max. 6 dB
Australian = Max. 10 dB
Brazilian = Max. 9 dB
ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)
Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications
Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2141/
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For each item in core_instances disposed of in the shutdown of ccss, any
generic monitor instances referenced by the objects will be removed from
generic_monitors during their destruction. Hilarity ensues if
generic_monitors no longer exists.
Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly
when it ran into this.
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Richard pointed out two problems with the check-in from r374177:
* The ast_msg_shutdown function declaration doesn't match the prototype
in main/message.c.
* The ref/alloc function usage in astobj2 (in trunk) can use the ao2_t_*
variants of the functions to allow the REF_DEBUG flag to enable/disable
their debug counterparts.
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Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case. This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.
The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.
As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.
Review: https://reviewboard.asterisk.org/r/2136/
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in res_xmpp on unload.
This patch fixes an issue where hangup flags were not being reset on a
channel, affecting subsequent use of that channel. The patch also adds some
additional cleanup to res_xmpp to fix an issue with reloading the module.
(closes ASTERISK-20360)
Reported by: Noah Engelberth
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2134/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it
is possible for Asterisk to skip calling the modules's .load function.
Asterisk was loading and linking the module via load_dynamic_module() but was
not adding the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules in the heap.
Now use load_resource() instead of load_dynamic_module() for non-optional
requirement. This will add the module to the resource_heap so the module can
be properly initialized in the correct order.
This is required if there are any module global data structures initialized in
the .load() callback for the module on platforms which do not support weak
references.
(issue ASTERISK-20439)
Reported by: sruffell
Patches:
0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417)
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As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.
Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.
Review: https://reviewboard.asterisk.org/r/2113/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* ASTERISK-20383
Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.
* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up. In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.
Regression because of the named call pickup group feature.
* See ASTERISK-20386 for the implementation improvements. These are the
changes in channel.c and channel.h.
* Fixed some locking issues in CHANNEL().
(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2112/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
portions of the SSL library. Asterisk calls SSL_library_init and
SSL_load_error_strings during SSL initialization; collectively this
obviates the need for calling any of the following during initialization
or client connection handling:
* ERR_load_crypto_strings (handled by SSL_load_error_strings)
* OpenSSL_add_all_algorithms (synonym for SSL_library_init)
* SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
the SSL library for TLS clients. This included not freeing the SSL_CTX
object in the SIP channel driver, as well as not clearing the error
stack when the TLS client exited.
Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.
(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
(bugAST-889.patch) by Thomas Arimont (license 5525)
Review: https://reviewboard.asterisk.org/r/2105
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ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!
This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.
(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
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I didn't realize that libasteriskssl.c was still compiled, even when you
disable asteriskssl; it simple gets statically linked into asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.
This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.
(closes issue AST-937)
Reported by Jason Parker
Patches:
AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
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When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.
(issue ASTERISK-20335)
Reported by: aragon
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chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of
using chan_motif and res_xmpp. They are a feature-equivalent
replacement and are written to be more easily maintainable.
(closes issue ASTERISK-20298)
Review: https://reviewboard.asterisk.org/r/2082/
Reported-by: Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.
(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
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........
Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.
Without this flag, those files will compile with the system installed
OpenSSL headers (if they exist). This is a real bummer if a different
path was specified using --with-ssl=
(closes issue ASTERISK-20392)
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The previous fix still would look in the static_RTP_PT table, which
is inappropriate since we specifically want to find a codec that has
been negotiated.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
Patches:
codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary;
1. reseting of hits=0, when no signal, only need to set it once.
2. incrementing of hits, when the hit is the same as the current hit.
3. setting of lasthit, when it's the same as before.
Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3
& 3 spelling mistakes
(closes issue ASTERISK-19610)
alecdavis (license 585)
Reported by: Jean-Philippe Lord
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2085/
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The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As noted by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
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The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
(closes issue ASTERISK-20240)
Reported by: Egor Gorlin
Patches:
lock.c.patch uploaded by Egor Gorlin (license 6416)
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When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(closes issue ASTERISK-20090)
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In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.
(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373)
Reported-by: Benjamin (bulkorok)
Reported-by: Rob Gagnon (rgagnon)
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When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
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This patch adds the feature "Private representation of caller, connected
and redirecting party ids", as previously discussed with us (DATUS) and
Digium.
1. Feature motivation
Until now it is quite difficult to modify a party number or name which can
only be seen by exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on one
channel is spread over several channels are supplementary services like
call transfer or pickup. To implement these features Asterisk internally
copies caller and connected ids from one channel to another. Another
example are extension subscriptions. The monitoring entities (watchers)
are notified of state changes and - if desired - of party numbers or names
which represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e. where a
party name shall be exclusively signaled to only one particular user, is a
private user-specific name resolution for party numbers. A lookup in a
private destination-dependent telephone book shall provide party names
which cannot be seen by any other user at any time.
2. Feature Description
This feature comes along with the implementation of additional private
party id elements for caller id, connected id and redirecting ids inside
Asterisk channels.
The private party id elements can be read or set by the user using
Asterisk dialplan functions.
When a technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting update
event, it merges the corresponding public id with the private id to create
an effective party id. The effective party id is then used for protocol
signaling.
The channel technologies which initially support the private id
representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and
PRI (chan_dahdi).
Once a private name or number on a channel is set and (implicitly) made
valid, it is generally used for any further protocol signaling until it is
rewritten or invalidated.
To simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all connected/redirecting
update events which are generated by technology channels -- receiving
regarding protocol information - automatically trigger the invalidation of
private ids.
If not using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting related
functions, the current characteristic of Asterisk is not affected by the
new extended functionality.
3. User interface Description
To grant access to the private name and number representation from the
Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan
functions are extended by the following data types. The formats of these
data types are equal to the corresponding regular 'non-private' already
existing data types:
CALLERID:
priv-all
priv-name priv-name-valid priv-name-charset priv-name-pres
priv-num priv-num-valid priv-num-plan priv-num-pres
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
REDIRECTING:
priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset
priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd
priv-orig-tag
priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan
priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd
priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset
priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan
priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2030/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
You cannot unref a pointer and then expect to ref it again later.
* Fix potential NULL pointer deref if the call pickup search fails.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Quote from review board:
This patch extends the extension state callbacks so that monitoring channels
(as chan_sip) get more information of the devices which are responsible for
an extension state change. The additional information is needed by chan_sip
to present names/numbers of the caller and callee in an early-state SIP
notification. Users of extenstion state callback not interested in the
additional information are not affected by the changes.
Motivation: to present the involved party's name/number in an early-state
nofification (used by the notified device as a pickup offer) one after another
so that a user can see which call he will pick up in an undirected pickup.
Such a pickup offer to a user shall indicate the same call (number/name-A calls
number/name-B) as the call which would be picked up when an undirected pickup
is executed.
Users interested in additional state info must use the new functions
ast_extension_state_add_extended() resp.
ast_extension_state_add_destroy_extended() to register an extended state
callback. When the callback is registered this way, an extra member
device_state_info of struct ast_state_cb_info is passed to the callback in
addition to the aggregated extension state. This container holds an object for
every device of the monitored extension hint consisting of the device name, the
device state and a channel reference to the channel which (presumably) caused
the device state.
The information is used by chan_sip for early-state notifications. When the
state of a device changes and the new state contains AST_EVENT_RINGING, an
early-state notification is sent to the subscribed devices with the
caller/callee names/numbers of the oldest ringing channel of the monitored
extension. The notified user may then invoke a direct pickup, which will pickup
exactly this channel.
Users of the old non-extended callbacks will only be called when the aggregated
state did change (same behavior as before). Users of the extended callback will
also be called when the state is unchanged but does contain AST_EVENT_RINGING.
That could be the case if two channels are ringing at one device and one of
them hangs up, so the aggregated state does not change. This way the monitoring
channel can create a new early-state notification with the now ringing
party-ids.
Review: https://reviewboard.asterisk.org/r/2048
This contribution comes from Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the work done by Olle Johansson on review board.
The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.
(closes issue ASTERISK-18644)
Reported by Olle Johansson
Review: https://reviewboard.asterisk.org/r/1472
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implementation of a dialplan function for checking manager accounts. Right now
it only returns the number of logged in sessions for a manager account, but
other attributes can be added later.
Patch by: Olle Johansson
Review: https://reviewboard.asterisk.org/r/421/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.
This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.
Review: https://reviewboard.asterisk.org/r/1412
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, updating the device state cache was done by the thread
that originated the event. It would update the cache and then queue the event
up for another thread to dispatch. This thread moves the cache updating part
to be in the same thread as event dispatching.
I was working with someone on a heavily loaded Asterisk system and while
reviewing backtraces of the system while it was having problems, I noticed that
there were a lot of threads contending for the lock on the event cache. By
simply moving this into a single thread, this helped performance *a lot* and
alleviated some deadlock-like symptoms.
Review: https://reviewboard.asterisk.org/r/2066/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new CLI command to the res_corosync module. It is primarily
used as a debugging tool. It lets you fire off an event which will cause
res_corosync on other nodes in the cluster to place messages into the logger if
everything is working ok. It verifies that the corosync communication is
working as expected.
I didn't put anything in the CHANGES file for this, because this module is new
in Asterisk 11. There is already a generic "res_corosync new module" entry in
there so I figure that covers it just fine.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous change to the build system for using a system-provided editline
library was missing a crucial include directory for building against the
copy of the library in the Asterisk source tree.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the build system to refer to the embedded editline directory
using an absolute path, which will resolve a problem seen on the CentOS
automated build agents.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) There is no such function as ast_ref()
2) The patch was originally credited as the one uploaded by Guenther
Kelleter (license 6372) via issue AST-921, but the patch committed
was not the patch referenced on the issue.
3) Guenther Kelleter's patch was actually correct. It moved the
ast_free above the presencechange_cleanup label. I am not
committing his change as it is not technically necesary--calling
ast_free(NULL) is perfectly safe and I worry that moving the
ast_free outside of the label could lead to future bugs if
someone ever adds another failure conditional and expects
'goto presencechange_cleanup;' to clean up after everything.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
items (separated by commas), and items in the rule can be negated by prefixing
them with '!'. This simplifies Asterisk Realtime configurations, since it is no
longer necessray to control the order that the 'permit' and 'deny' columns are
returned from queries.
Review: https://reviewboard.asterisk.org/r/1592/
Initial patch contributed by Tilghman Lesher
Unit tests written by Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes the following:
* Unit tests for the abstract Jitter Buffer API. This includes both fixed
and adaptive flavors, testing nominal creation, frame input, frame retrieval,
resyncing; off nominal frame input overflow, out of order, and others.
* Tweaks to the abstract_jb API to remove the unnecessary resync_threshold
parameter from the create function (resync_threshold is already in the
struct passed into the create function)
* Ensure the fixed jitter buffer is empty before destroying it, to avoid an
ASSERT
* Don't "resync" the adaptive jitter buffer. The mechanism that was being
used actually causes the jitter buffer to think its being overflowed by going
around the jitterbuf API and attempting to 'resynch' it improperly. If a
resync is needed, the jitter buffer will do it properly by itself. Note that
this is only an optimization needed for trunk, as the worst that happens is
the loss of three voice packets before the adaptive jitter buffer will resync
anyway.
Review: https://reviewboard.asterisk.org/r/2035
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The value "none" is specified in the config file as a valid value for
the "video_mode" option. The code prior to the ACO conversion did not
check for "none", but just ignored it and relied on the default zero
value. The parsing with ACO is more strict, so without handling
"none" specifically, parsing would fail.
When parsing failed, but the module loaded anyway, the config info
would never be stored, and one place in the code did not check for
this case and would segfault. It was also possible that the
aco_info struct's internals would be destroyed and used as well.
This patch keeps the module from loading after parse failures, adds
the "none" option to "video_mode", registers CLI functions only
after parsing has completed, checks the config data for NULL before
accessing it, and returns -1 on some allocation failures when
initializing.
(closes issue ASTERISK-20159)
Reported by: Birger "WIMPy" Harzenetter
Tested by: Birger "WIMPy" Harzenetter
Patches:
confbridge_fix3.txt uploaded by Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.
(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's harder to correlate the Newchannel and Hangup AMI events without
specifying "AccountCode" in both.
(closes issue ASTERISK-19963)
Reported by: Oleg A. Arkhangelsky
Patches:
hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A number of applications/AMI commands in Asterisk have specific behavioral
differences depending on the resource or channel technology those
applications are executed on. For example, the MessageSend application/
command is technology agnostic, but how the channel drivers that support
that functionality behave is dependant on the protocols and channel
driver implementation. Prior to this patch, those details were either
documented in the application/command documentation itself, or were left
undocumented.
This patch adds a new element to the documentation schema, <info/>. An info
node is essentially a piece of technology specific reference information that
can be included by any top level XML documentation node. For example, the
MessageSend application can now include XMPP/SIP specific information, where
that technology specific information can be defined in chan_motif/res_xmpp/
chan_sip. Likewise, that information can also be included in the MessageSend
AMI command.
Review: https://reviewboard.asterisk.org/r/2049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
To fix a memory leak in CEL, a channel datastore was introduced whose
destruction function pointer was pointed to the ast_free macro. Without
MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free.
With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a
different place then utils.h, and became undefined. This patch resolves this
by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
ast_free, which is defined to be free.
(issue AST-916)
Reported by: Thomas Arimont
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.
The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.
(closes issue AST-916)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2053/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds Named ACL functionality to Asterisk. This allows system
administrators to define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control lists.
It also includes updates to all core supported consumers of ACLs. That includes
manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk
by Olle E. Johansson and provides a subset of the Named ACL functionality
implemented in that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki.
Review: https://reviewboard.asterisk.org/r/1978/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because some of the manager events are defined in the top of the source, due
to the macro calls not containing all necessary information to have the
documentation colocated with the call itself, several include statements were
failing when built with 'make'. While this did not cause any problems in
compilation or validation, it did result in a number of warnings being dumped
to stderr.
This patch changes those references such that they always resolve, regardless
of the documentation build options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file. If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file. Without the python scripts, these XML fragments will not validate.
This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds some basic documentation for a number of modules. This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD
has also been updated to allow referencing of AMI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the OpenSSL duplicate initialization issues were resolved in r351447,
res_curl could fail to load if it checked SSL_library_init after SSL
initialization completed. This is due to the SSL_library_init stub returning
a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of
1 to indicate success - in fact, SSL_library_init is documented to always return
1. Interestingly, the CURL libraries actually checked the return value - the fact
that nothing else that depends on OpenSSL was having problems loading probably means
they don't check the return value.
(closes issue AST-924)
Reported by: Guenther Kelleter
patches:
(AST-924.patch license #6372 uploaded by Guenther Kelleter)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis.
(closes issue ASTERISK-20107)
Reported by: Malcolm Davenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function.
The code being removed is superfluous because the SSL_CTX
structures for servers will be properly freed when ast_ssl_teardown
is called.
(closes issue ASTERISK-20074)
Reported by Trevor Helmsley
Patches:
ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
Testers:
Trevor Helmsley
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The opaquing the ast_channel struct no longer requires .cleancount to be
changed when the struct is changed.
* Bump .cleancount value one last time because of struct ast_channel for
old times sake.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows the setting of flags via the config options api.
For example, code like this:
#define OPT1 1 << 0
#define OPT2 1 << 1
#define OPT3 1 << 2
struct thing {
unsigned int flags;
};
and a config like this:
[blah]
opt1=yes
opt2=no
opt3=yes
Review: https://reviewboard.asterisk.org/r/2004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.
review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jonathan Rose
Kevin Harwell
Walter Doekes
Kinsey Moore
Richard Mudgett
Michael L. Young
David M. Lee
Matthew Jordan
John Bigelow
Alec L Davis
Jason Parker
Tzafrir Cohen
Mark Michelson
Russell Bryant
Tilghman Lesher
Brent Eagles
Joshua Colp
Sean Bright
Terry Wilson
Kevin P. Fleming