George Joseph [Tue, 15 Oct 2024 17:11:28 +0000 (11:11 -0600)]
core_unreal.c: Fix memory leak in ast_unreal_new_channels()
When the channel tech is multistream capable, the reference to
chan_topology was passed to the new channel. When the channel tech
isn't multistream capable, the reference to chan_topology was never
released. "Local" channels are multistream capable so it didn't
affect them but the confbridge "CBAnn" and the bridge_media
"Recorder" channels are not so they caused a leak every time one
of them was created.
Also added tracing to ast_stream_topology_alloc() and
stream_topology_destroy() to assist with debugging.
George Joseph [Fri, 20 Sep 2024 13:47:53 +0000 (07:47 -0600)]
Fix application references to Background
The app is actually named "BackGround" but several references
in XML documentation were spelled "Background" with the lower
case "g". This was causing documentation links to return
"not found" messages.
George Joseph [Mon, 19 May 2025 14:16:53 +0000 (08:16 -0600)]
asterisk.c: Add option to restrict shell access from remote consoles.
UserNote: A new asterisk.conf option 'disable_remote_console_shell' has
been added that, when set, will prevent remote consoles from executing
shell commands using the '!' prefix.
Ben Ford [Tue, 17 Dec 2024 17:42:48 +0000 (11:42 -0600)]
manager.c: Restrict ListCategories to the configuration directory.
When using the ListCategories AMI action, it was possible to traverse
upwards through the directories to files outside of the configured
configuration directory. This action is now restricted to the configured
directory and an error will now be returned if the specified file is
outside of this limitation.
Resolves: #GHSA-33x6-fj46-6rfh
UserNote: The ListCategories AMI action now restricts files to the
configured configuration directory.
app_voicemail: Fix sql insert mismatch caused by cherry-pick
When commit e8c9cb80 was cherry-picked in from master, the
fact that the 20 and 18 branches still had the old "macrocontext"
column wasn't taken into account so the number of named parameters
didn't match the number of '?' placeholders. They do now.
We also now use ast_asprintf to create the full mailbox query SQL
statement instead of trying to calculate the proper length ourselves.
George Joseph [Thu, 8 Aug 2024 16:57:14 +0000 (10:57 -0600)]
manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
If you run an AMI CoreShowChannelMap on a channel that isn't in a
bridge and you're in DEVMODE, you can get a FRACK because the
bridge id is empty. We now simply return an empty list for that
request.
Ben Ford [Tue, 21 May 2024 16:11:26 +0000 (11:11 -0500)]
channel: Add multi-tenant identifier.
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
George Joseph [Mon, 12 Aug 2024 17:58:12 +0000 (11:58 -0600)]
res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
The ub_result pointer passed to unbound_resolver_callback by
libunbound can be NULL if the query was for something malformed
like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result
and return instead of crashing with a SEGV. This causes pjproject
to simply cancel the transaction with a "No answer record in the DNS
response" error. The existing "off nominal" unit test was also
updated to check this condition.
Although not necessary for this fix, we also made
ast_dns_resolver_completed() tolerant of a NULL result.
Mike Bradeen [Wed, 10 Jul 2024 18:58:44 +0000 (12:58 -0600)]
res_stasis: fix intermittent delays on adding channel to bridge
Previously, on command execution, the control thread was awoken by
sending a SIGURG. It was found that this still resulted in some
instances where the thread was not immediately awoken.
This change instead sends a null frame to awaken the control thread,
which awakens the thread more consistently.
George Joseph [Wed, 17 Jul 2024 16:44:17 +0000 (10:44 -0600)]
bridge_softmix: Fix queueing VIDUPDATE control frames
softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else()
with the bridge_channel so the VIDUPDATE control frame isn't echoed back.
softmix_bridge_write_control() was setting bridge_channel to NULL
when calling ast_bridge_queue_everyone_else() for VIDUPDATE control
frames. This was causing the frame to be echoed back to the
channel it came from. In certain cases, like when two channels or
bridges are being recorded, this can cause a ping-pong effect that
floods the system with VIDUPDATE control frames.
George Joseph [Mon, 22 Jul 2024 14:05:03 +0000 (08:05 -0600)]
manager.c: Add entries to Originate blacklist
Added Reload and DBdeltree to the list of dialplan application that
can't be executed via the Originate manager action without also
having write SYSTEM permissions.
Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan
functions that can't be executed via the Originate manager action
without also having write SYSTEM permissions.
If the Queue application is attempted to be run by the Originate
manager action and an AGI parameter is specified in the app data,
it'll be rejected unless the manager user has either the AGI or
SYSTEM permissions.
George Joseph [Tue, 9 Apr 2024 13:23:36 +0000 (07:23 -0600)]
app_voicemail_odbc: Allow audio to be kept on disk
This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
which when set causes the ODBC variant of app_voicemail to leave
the message and greeting audio files on disk and only store the
message metadata in the database. This option came from a concern
that the database could grow to large and cause remote access
and/or replication to become slow. In a clustering situation
with this option, all asterisk instances would share the same
database for the metadata and either use a shared filesystem
or other filesystem replication service much more suitable
for synchronizing files.
The changes to app_voicemail to implement this feature were actually
quite small but due to the complexity of the module, the actual
source code changes were greater. They fall into the following
categories:
* Tracing. The module is so complex that it was impossible to
figure out the path taken for various scenarios without the addition
of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
code that's not related to the functional change. Making this worse
was the fact that many "if" statements in this module didn't use
braces. Since the tracing macros add multiple statements, many "if"
statements had to be converted to use braces.
* Excessive use of PATH_MAX. Previous maintainers of this module
used PATH_MAX to allocate character arrays for filesystem paths
and SQL statements as though they cost nothing. In fact, PATH_MAX
is defined as 4096 bytes! Some functions had (and still have)
multiples of these. One function has 7. Given that the vast
majority of installations use the default spool directory path
`/var/spool/asterisk/voicemail`, the actual path length is usually
less than 80 bytes. That's over 4000 bytes wasted. It was the
same for SQL statement buffers. A 4K buffer for statement that
only needed 60 bytes. All of these PATH_MAX allocations in the
ODBC related code were changed to dynamically allocated buffers.
The rest will have to be addressed separately.
* Bug fixes. During the development of this feature, several
pre-existing ODBC related bugs were discovered and fixed. They
had to do with leaving orphaned files on disk, not preserving
original message ids when moving messages between folders,
not honoring the "formats" config parameter in certain circumstances,
etc.
UserNote: This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
George Joseph [Tue, 9 Apr 2024 20:49:36 +0000 (14:49 -0600)]
logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
If you're tracing a large function that may call another function
multiple times in different circumstances, it can be difficult to
see from the trace output exactly which location that function
was called from. There's no good way to automatically determine
the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT
simply print out a trace line before and after the call.
The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
that SCOPE_CALL ignores the function's return value (if any) where
SCOPE_CALL_WITH_RESULT allows you to specify the type of the
function's return value so it can be assigned to a variable.
SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
and the "int" return type.
Resequencing is a process that occurs when we open a voicemail folder
and discover that there are gaps between messages (e.g. `msg0000.txt`
is missing but `msg0001.txt` exists). Resequencing involves shifting
the existing messages down so we end up with a sequential list of
messages.
Currently, this process stops after reaching a threshold based on the
message limit (`maxmsg`) configured on the current folder. However, if
`maxmsg` is lowered when a voicemail folder contains more than
`maxmsg + 10` messages, resequencing will not run completely leaving
the mailbox in an inconsistent state.
We now resequence up to the maximum number of messages permitted by
`app_voicemail` (currently hard-coded at 9999 messages).
Ivan Poddubny [Sun, 5 May 2024 12:53:11 +0000 (14:53 +0200)]
asterisk.c: Fix sending incorrect messages to systemd notify
Send "RELOADING=1" instead of "RELOAD=1" to follow the format
expected by systemd (see sd_notify(3) man page).
Do not send STOPPING=1 in remote console mode:
attempting to execute "asterisk -rx" by the main process leads to
a warning if NotifyAccess=main (the default) or to a forced termination
if NotifyAccess=all.
George Joseph [Tue, 23 Apr 2024 20:15:20 +0000 (14:15 -0600)]
tcptls/iostream: Add support for setting SNI on client TLS connections
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
George Joseph [Sat, 27 Apr 2024 20:40:28 +0000 (14:40 -0600)]
make_buildopts_h: Always include DETECT_DEADLOCKS
Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must
always be included in buildopts.h instead of only when
ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise.
George Joseph [Tue, 2 Apr 2024 20:28:35 +0000 (14:28 -0600)]
rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
rtp_engine.c and stun.c were calling ast_register_cleanup which
is skipped if any loadable module can't be cleanly unloaded
when asterisk shuts down. Since this will always be the case,
their cleanup functions never get run. In a practical sense
this makes no difference since asterisk is shutting down but if
you're in development mode and trying to use the leak sanitizer,
the leaks from both of those modules clutter up the output.
George Joseph [Mon, 1 Apr 2024 12:30:54 +0000 (06:30 -0600)]
Fix incorrect application and function documentation references
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
George Joseph [Mon, 26 Feb 2024 23:16:37 +0000 (16:16 -0700)]
Rename dialplan_functions.xml to dialplan_functions_doc.xml
When using COMPILE_DOUBLE, dialplan_functions.xml is mistaken
for the source for an embedded XML document and gets compiled
to dialplan_functions.o. This causes dialplan_functions.c to
be ignored making its functions unavailable and causing chan_pjsip
to fail to load.
Mike Bradeen [Wed, 31 Jan 2024 15:55:04 +0000 (08:55 -0700)]
app_chanspy: Add 'D' option for dual-channel audio
Adds the 'D' option to app chanspy that causes the input and output
frames of the spied channel to be interleaved in the spy output frame.
This allows the input and output of the spied channel to be decoded
separately by the receiver.
If the 'o' option is also set, the 'D' option is ignored as the
audio being spied is inherently one direction.
Fixes: #569
UserNote: The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
Naveen Albert [Tue, 26 Dec 2023 17:14:11 +0000 (17:14 +0000)]
manager.c: Fix regression due to using wrong free function.
Commit 424be345639d75c6cb7d0bd2da5f0f407dbd0bd5 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.
George Joseph [Tue, 31 Oct 2023 21:08:14 +0000 (15:08 -0600)]
chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
Mike Bradeen [Thu, 7 Sep 2023 17:21:32 +0000 (11:21 -0600)]
res_speech: allow speech to translate input channel
* Allow res_speech to translate the input channel if the
format is translatable to a format suppored by the
speech provider.
Resolves: #129
UserNote: res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
Mike Bradeen [Mon, 2 Oct 2023 18:35:32 +0000 (12:35 -0600)]
res_stasis: signal when new command is queued
res_statsis's app loop sleeps for up to .2s waiting on input
to a channel before re-checking the command queue. This can
cause delays between channel setup and bridge.
This change is to send a SIGURG on the sleeping thread when
a new command is enqueued. This exits the sleeping thread out
of the ast_waitfor() call triggering the new command being
processed on the channel immediately.
Resolves: #362
UserNote: Call setup times should be significantly improved
when using ARI.
George Joseph [Thu, 5 Oct 2023 14:18:17 +0000 (08:18 -0600)]
logger.h: Add ability to change the prefix on SCOPE_TRACE output
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else. Full documentation in logger.h.
George Joseph [Thu, 21 Sep 2023 17:12:31 +0000 (11:12 -0600)]
Add libjwt to third-party
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.
A few tweaks were also made to the third-party infrastructure as
a whole. The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.
Mike Bradeen [Tue, 26 Sep 2023 19:19:27 +0000 (13:19 -0600)]
res_pjsip: update qualify_timeout documentation with DNS note
The documentation on qualify_timeout does not explicitly state that the timeout
includes any time required to perform any needed DNS queries on the endpoint.
If the OPTIONS response is delayed due to the DNS query, it can still render an
endpoint as Unreachable if the net time is enough for qualify_timeout to expire.
George Joseph [Wed, 13 Sep 2023 15:18:04 +0000 (09:18 -0600)]
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
George Joseph [Mon, 11 Sep 2023 14:27:41 +0000 (08:27 -0600)]
file.c: Add ability to search custom dir for sounds
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory. If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory. For performance
reasons, the "sounds_search_custom_dir" defaults to "false".
Resolves: #315
UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.
George Joseph [Wed, 13 Sep 2023 20:08:02 +0000 (14:08 -0600)]
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options. In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.
OK so why would we want to include them? Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.
So...
* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
which tells make_buildopts_h to include the non-ABI-breaking
flags in buildopts.h as well as the ABI-breaking ones. The default
is disabled to preserve current behavior. As before though,
only the ABI-breaking flags appear in AST_BUILDOPTS and only
those are used to calculate AST_BUILDOPT_SUM.
A new AST_BUILDOPT_ALL define was created to capture all of the
flags.
* make_version_c was streamlined to use buildopts.h and also to
create asterisk_build_opts_all[] and ast_get_build_opts_all(void)
* "core show settings" now shows both AST_BUILDOPTS and
AST_BUILDOPTS_ALL.
UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
George Joseph [Mon, 11 Sep 2023 16:19:41 +0000 (10:19 -0600)]
safe_asterisk: Change directory permissions to 755
If the safe_asterisk script detects that the /var/lib/asterisk
directory doesn't exist, it now creates it with 755 permissions
instead of 770. safe_asterisk needing to create that directory
should be extremely rare though because it's normally created
by 'make install' which already sets the permissions to 755.
Using the Set dialplan application does not actually
delete channel or global variables. Instead the
variables are set to an empty value.
This change adds two dialplan functions,
GLOBAL_DELETE and DELETE which can be used to
delete global and channel variables instead
of just setting them to empty.
There is also no ability within the dialplan to
determine if a global or channel variable has
actually been set or not.
This change also adds two dialplan functions,
GLOBAL_EXISTS and VARIABLE_EXISTS which can be
used to determine if a global or channel variable
has been set or not.
Resolves: #289
UserNote: Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.
George Joseph [Tue, 5 Sep 2023 15:51:19 +0000 (09:51 -0600)]
ari-stubs: Fix broken documentation anchors
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
George Joseph [Mon, 26 Jun 2023 12:55:49 +0000 (06:55 -0600)]
rest-api: Updates for new documentation site
The new documentation site uses traditional markdown instead
of the Confluence flavored version. This required changes in
the mustache templates and the python that generates the files.
George Joseph [Fri, 18 Aug 2023 16:20:47 +0000 (10:20 -0600)]
download_externals: Fix a few version related issues
* Fixed issue with the script not parsing the new tag format for
certified releases. The format changed from certified/18.9-cert5
to certified-18.9-cert5.
* Fixed issue where the asterisk version wasn't being considered
when looking for cached versions.
George Joseph [Fri, 18 Aug 2023 15:23:12 +0000 (09:23 -0600)]
Remove .lastclean and .version from source control
Historically these were checked in for certified releases but
since the move to github and the unified release process,
they are no longer needed and cause issues.
Joshua C. Colp [Wed, 9 Aug 2023 10:57:33 +0000 (07:57 -0300)]
manager: Tolerate stasis messages with no channel snapshot.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
app_queue: Add support for applying caller priority change immediately.
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.
This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.
Resolves: #202
UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
George Joseph [Mon, 10 Jul 2023 13:52:32 +0000 (07:52 -0600)]
app.h: Move declaration of ast_getdata_result before its first use
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c). The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.
projects / thirdparty / asterisk.git / log
