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Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
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Each time the dial plan is reloaded, a lot of logs like these are generated:
"Added extension 'XXXXX' priority 1 to YYYYYYYYYYY"
This patch changes the log level for those logs.
ASTERISK-27084
Change-Id: I5662902161c50890997ddc56835d4cafb456c529
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pbx_extension_helper has a check for q->swo.exec == NULL but it doesn't
actually return so we would still run the function. Fix the return.
Move the 'int res' variable into the only scope which uses it.
Also fix a copy-paste error in ast_pbx_init which could result in a
crash on allocation failure (we exit with a normal error instead).
Change-Id: I0693af921fdc7f56b6a72a21fb816ed08b960a69
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Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
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This resolves potentials leaks on AST_VECTOR_APPEND error in:
* ast_context_add_include2
* ast_context_add_switch2
* ast_context_add_ignorepat2
Change-Id: Ib60e95c4f622fa3b832d87227c0523a695d736b6
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This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.
Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83
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When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.
ASTERISK-26789
Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10
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Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core
to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE
is passed to these functions, the calling thread will be blocked until
the newly created channel has been hung up.
After this patch, we run the dial on the current thread rather than
spawning a new one. The only in-tree code that passes
AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced
thread usage if you are using .call files.
Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913
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Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range. e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.
The buffer overwrite is fixed two ways in this patch.
1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens. Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set. Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.
2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.
ASTERISK-26668
Change-Id: I555d97411140e47e0522684062d174fbe32aa84a
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* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension. Ran into this when
developing a testsuite test. The AMI event ExtensionStatus came out with
the hint header value containing garbage. The AMI event PresenceStatus
also had the same issue.
* manager.c:action_extensionstate() no need to completely initialize the
hint[]. Only initialize the first element.
* pbx.c:ast_add_hint() Remove unnecessary assignment.
* chan_sip.c: Eliminate an unneeded hint[] local variable. We only care
about the return value of ast_get_hint() there.
Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
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In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
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Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.
This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.
ASTERISK-26658 #close
Reported by: Jonathan R. Rose
Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30
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Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call. The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works. Have also tested both 'exten'
and 'app' versions of app_originate.
Opened by: dkerr
Patch by: dkerr
Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57
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ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
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Previously, if context A was including context B and context B was including
context A, i.e. if there was a circular dependency between contexts, then
calling manager_show_dialplan_helper could lead to an infinite recursion,
resulting in a crash.
This commit applies the same solution as the one implemented in the
show_dialplan_helper function. The manager_show_dialplan_helper and
show_dialplan_helper functions contain lots of code in common, but the former
was missing the "infinite recursion avoidance" code.
ASTERISK-26226 #close
Change-Id: I1aea85133c21787226f4f8442253a93000aa0897
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Do not check registrar of the first extension head. We should only check
the registrar when we match the priority.
Additionally fix a couple calls to strcmp which used the input callerid
instead of the clean version ex.cidmatch.
ASTERISK-26233
Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1
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This adds a two strings to ast_exten. name to go with exten and
cidmatch_display to go with cidmatch. The new fields contain input used
to add the extension in the first place. The existing fields now
contain stripped input that excludes insignificant spaces and dashes.
These stripped fields should always be used for comparisons. The
unstripped fields should normally be used for display, but displaying
stripped values will not cause runtime errors.
Note the actual string is only stored twice if it contains dashes. If
no dashes are found then both 'char *' fields point to the same memory.
So this change has a minimum effect on memory usage.
The existing functions ast_get_extension_name and
ast_get_extension_cidmatch return unstripped values as they did before
this change. Other similar bugs likely still exist where unstripped
extensions are saved outside pbx.c then passed back in.
ASTERISK-26233 #close
Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f
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We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity. Otherwise, we could never
execute dangerous functions.
ASTERISK-25996 #close
Reported by: Andrew Nagy
Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba
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Merge code found in both branches of a conditional in
ast_add_extension2_lockopt.
The updated code initializes peer_table and peer_label_table of the
extension before linking it to the context.
Change-Id: Ic759e27cdc9906c6877df41d28ee9c5be8f41c20
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This changes context switches from a linked list to a vector, makes
'struct ast_sw' opaque to pbx.c.
Although ast_walk_context_switches is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_switches_count (AST_VECTOR_SIZE)
* ast_context_switches_get (AST_VECTOR_GET)
As with ast_walk_context_switches callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the switches, they have been converted to use the new functions.
Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998
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This changes context ignore patterns from a linked list to a vector,
makes 'struct ast_ignorepat' opaque to pbx.c.
Although ast_walk_context_ignorepats is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_ignorepats_count (AST_VECTOR_SIZE)
* ast_context_ignorepats_get (AST_VECTOR_GET)
As with ast_walk_context_ignorepats callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the ignorepats, they have been converted to use the new functions.
Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a
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This changes context includes from a linked list to a vector, makes
'struct ast_include' opaque to pbx.c.
Although ast_walk_context_includes is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_includes_count (AST_VECTOR_SIZE)
* ast_context_includes_get (AST_VECTOR_GET)
As with ast_walk_context_includes callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the includes, they have been converted to use the new functions.
const have been applied where possible to parameters for ast_include
functions.
Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60
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Create include_free to run ast_destroy_timing and ast_free, use that in
all places that freed an ast_include structure. This fixes a couple of
paths that previously did not run ast_destroy_timing.
ASTERISK-26196 #close
Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838
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This eliminates some casts that I made a note saying v10 and above
would no longer need them.
Better late than never :)
Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
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* Pull out a loop invariant.
* Convert an else-if ladder to a switch statement.
Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95
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Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e
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This change introduces the concept of autohints. These are hints
which are created as a result of device state changes occurring within
the core. When this happens a hint will be created (if it does not
exist already) using the device name as the extension.
For example if a device state change is received for "PJSIP/bob"
and autohints are enabled on a context then a hint will exist in
that context for "bob" with a device of "PJSIP/bob".
For virtual or custom device states the name after the type will
be used. For example if the device state of "Custom:bob" changes
then a hint will exist in that context for "bob" with a device of
"Custom:bob".
This functionality can be enabled in extensions.conf by placing
"autohints=yes" in a context.
ASTERISK-25881 #close
Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e
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Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5)
introduced the possibility of a deadlock. Due to the mentioned modifications
ast_change_hints now needs to keep both merge/delete and state callbacks from
occurring while it executes. Unfortunately, sometimes ast_change_hints can be
called with the contexts container locked. When this happens it's possible for
another thread to grab the context_merge_lock before the thread calling into
ast_change_hints does and then try to obtain the contexts container lock. This
of course causes a deadlock between the two threads. The thread calling into
ast_change_hints waits for the other thread to release context_merge_lock and
the other thread is waiting on that one to release the contexts container lock.
Unfortunately, there is not a great way to fix this problem. When hints change,
the subsequent state callbacks cannot run at the same time as a merge/delete,
nor when the usual state callbacks do. This patch alleviates the problem by
having those particular callbacks (the ones run after a hint change) occur in a
serialized task. By moving the context_merge_lock to a task it can now safely be
attempted or held without a deadlock occurring.
ASTERISK-25640 #close
Reported by: Krzysztof Trempala
Change-Id: If2210ea241afd1585dc2594c16faff84579bf302
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This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves hangup handler management functions to their own source.
Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104
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This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves dialplan application management functions to their own source.
Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c
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This is the fifth patch in a series meant to reduce the bulk of pbx.c.
This moves ast_switch functions to their own source.
Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e
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This is the fourth patch in a series meant to reduce the bulk of pbx.c.
This moves pbx timing functions to their own source.
Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6
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This is the third patch in a series meant to reduce the bulk of pbx.c.
This moves channel and global variable routines to their own source.
Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6
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This is the second patch in a series meant to reduce the bulk of pbx.c.
This moves custom function management routines to their own source.
Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177
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We joked about splitting pbx.c into multiple files but this first step was
fairly easy. All of the pbx_builtin dialplan applications have been moved
into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
is called by asterisk.c just after load_pbx().
A few functions were renamed and are cross-exposed between the 2 source files.
Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a
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When changing a hint extension without removing the hint first the
device state and presence state is not updated. This causes the state
of the hint to be that of the previous extension and not the current
one. This state is kept until a state change occurs as a result of
something (presence state change, device state change).
This change updates the hint with the current device and presence
state of the new extension when it is changed. Any state callbacks
which may have been added before the hint extension is changed are
also informed of the new device and presence state if either have
changed.
ASTERISK-25394 #close
Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f
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When issuing the "core show hints" CLI command a combination of both
the hint extension and context is created. This uses a fixed size
buffer expecting that the extension will not exceed maximum extension
length. When the extension is actually a pattern match this constraint
does not hold true, and the extension may exceed the maximum extension
length. In this case extra characters are written past the end of the
fixed size buffer.
This change makes it so the construction of the combined hint extension
and context can not exceed the size of the buffer.
ASTERISK-25367 #close
Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499
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ASTERISK-25256 #close
Reported by: Richard Mudgett
Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3
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When 8297136f was merged for ASTERISK-25040, a regression was introduced
surrounding the case sensitivity of device names within hints.
Previously, device names - such as 'sip/foo' - were compared in a case
insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
As a result, some dialplan hints stopped working.
This patch re-introduces case insensitive matching for device names in
hints.
ASTERISK-25040
ASTERISK-25202 #close
Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c
(cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03)
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Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
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The PBX core maintains two hash tables for hints: a container of the
actual hints (hints), along with a container of devices that are watching that
hint (hintdevices). When a dialplan reload occurs, each hint in the hints
container is destroyed; this requires a lookup in the container of devices to
find the device => hint mapping object. In the current code, this performs an
ao2_callback, iterating over each of the device to hint objects in the
hintdevices container. For a large number of hints, this is extremely
expensive: dialplan reloads with 20000 hints could take several minutes
in just this phase.
This patch improves the performance of this step in the dialplan reloads
by caching which devices are watching a hint on the hint object itself.
Since we don't want to create a circular reference, we just cache the
name of the device. This allows us to perform a smarter ao2_callback on
the hintdevices container during hint removal, hashing on the name of the
device and returning an iterator to the matching names. The overall
performance improvement is rather large, taking this step down to a number of
seconds as opposed to minutes.
In addition, this patch also registers the hint containers in the PBX
core with the astobj2 library. This allows for reasonable debugging to
hash collisions in those containers.
ASTERISK-25040 #close
Reported by: Matt Jordan
Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360
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When a PBX registrar is unloaded, it will fail to remove its extension from
the context root_table if a dialplan application used by that extension is
still loaded. This can be the case for AGI, which can be unloaded after several
of the standard PBX providers. Often, this is harmless; however, if the
extension's priorities are removed during the failed unloading *and* the
dialplan application later unregisters, it leaves a ticking timebomb for the
next PBX provider that attempts to iterate over the extensions. When that
occurs, the peer_table pointer on the extension will already be set to NULL.
The current code does not check to see if the pointer is NULL before passing
it to a hashtab function this is not NULL tolerant.
Since it is possible for the peer_table to be NULL when we normally would not
expect that to be the case, the solution in this patch is to simply skip over
processing an extension's priorities if peer_table is NULL.
Prior to this patch, the tests/pbx/callerid_match test would crash during
module unload. With this patch, the test no longer crashes after running.
ASTERISK-24774 #close
Reported by: Corey Farrell
Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40
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Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
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Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups. Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe. ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.
Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.
ASTERISK-24142 #close
Reported by: David Brillert
ASTERISK-24683 #close
Reported by: Peter Katzmann
ASTERISK-24805 #close
Reported by: Badalian Vyacheslav
ASTERISK-24881 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Switch logger callid's from AO2 objects to simple integers.
This helps in two ways. Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead. This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.
ASTERISK-24833 #comment Committed callid conversion to trunk.
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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These memory leaks were found and fixed by John Hardin. I'm just
committing them for him.
ASTERISK-24736 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4389
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Merged revisions 431468 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When a hint gets created, any subsequent device or presence
state changes result in extension status events getting sent
out to interested parties. However, at the time of hint creation,
no such event gets sent out, so watchers of extension state are
potentially left in the dark until the first state change after
hint creation.
Patch contributed by John Hardin (License #6512)
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Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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