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This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:
* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* main/security_events.c: removed unused TIMESTAMP_STR_LEN
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states
Review: https://reviewboard.asterisk.org/r/4526
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4526.patch submitted by dkdegroot (License 6600)
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Assigning a variable to itself isn't super useful. However, the WAV format
modules make use of this in order to perform byte endian checks. This patch
works around the warning by only performing the self assignment if we are
going to do more than just assign it to ourselves. Which is odd, but true.
Review: https://reviewboard.asterisk.org/r/4544/
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4544.patch submitted by dkdegroot (License 6600)
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Clang will treat ((a == b)) as a warning, as it reasonably expects that the
developer may have intended to write (a == b) or ((a = b)). This patch cleans
up all instances where equality, not assignment, was intended between two
parantheses.
Review: https://reviewboard.asterisk.org/r/4531/
ASTERISK-24917
Repoted by: dkdegroot
patches:
rb4531.patch submitted by dkdegroot (License 6600)
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In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by
clang, as it will truncate the -1 to a 1 implicitly.
Instead, we just assign the value a '1'.
Review: https://reviewboard.asterisk.org/r/4537/
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4537.patch submitted by dkdegroot (License 6600)
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This patch fixes clange compiler warnings for initializer overrides.
Specifically:
res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
those enum values, we therefore initialize the value twice to two different
values, "tlsv1" and "default". This patch changes it to just initialize
the index in the array to "tlsv1".
Review: https://reviewboard.asterisk.org/r/4539/
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4539.patch submitted by dkdegroot (License 6600)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes clang compilers warnings for unused functions. Specifically:
* channels/chan_iax2: removed user_ref function
* main/dsp.c: removed goertzel_update function
* main/config.c: made variable_list_switch static
Review: https://reviewboard.asterisk.org/r/4527
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4527.patch submitted by dkdegroot (License 6600)
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Review: https://reviewboard.asterisk.org/r/4503/
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Review: https://reviewboard.asterisk.org/r/4504/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
Patch 4488 includes all functionality needed for SAC's outside connectivity
and some externally accessed features, as well as outbound dialing.
Reported by: Malcolm Davenport
Tested by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/4488/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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eliminated some RAII_VAR() usage.
* Converted the contact_autoexpire container to use the ao2 template hash
and cmp functions. Also made use the OBJ_SEARCH_xxx names instead of the
deprecated names.
* Eliminates several unnecessary uses of RAII_VAR().
Review: https://reviewboard.asterisk.org/r/4524/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.
This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.
ASTERISK-24920 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4532/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Contact expiration object refs were leaked when the module was unloaded.
* Made empty the scheduler of entries before destroying it to release the
object ref held by the scheduler entry.
Review: https://reviewboard.asterisk.org/r/4523/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch updates the kqueue timing module to conform to current timer API.
This fixes issues with using the kqueue timing source on Asterisk 13 on
FreeBSD 10. These issues include:
- Remove support for kevent64(). The values used to support Asterisk timers
fit within 32bits and so can be handled on all platforms via kevent().
- Provide debug logging for, but do not track, unacked events. This matches
the behavior of all other timer implementations.
- Implement continuous mode by triggering and leaving active, a user event.
This ensures that the file descriptor for the timer returns immediately from
poll(), without placing the load of a high speed timer on the kernel.
- In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
type kqueue supports for timers.
- In kqueue_timer_get_event(), assume the caller woke up from poll() and just
return the mode the timer is currently in. This matches all other timer
implementations.
- Adjust the test code now that unacked events are not tracked.
Review: https://reviewboard.asterisk.org/r/4465/
ASTERISK-24857 #close
Reported by: scsiguy
Tested by: Ed Hynan
patches:
rb4465.patch submitted by scsiguy (License 6692)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Use the standard ast_log instead of ast_log_safe for STANDALONE programs.
Review: https://reviewboard.asterisk.org/r/4538/
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This introduces a new logger routine ast_log_safe. This routine should be
used for all error messages in code that can be run as a result of ast_log.
ast_log_safe does nothing if run recursively. All error logging in
astobj2.c, strings.c and utils.h have been switched to ast_log_safe.
This required adding support for raw threadstorage. This provides direct
access to the void* pointer in threadstorage. In ast_log_safe, NULL is used
to signify that this thread is not already running ast_log_safe, (void*)1 when
it is already running. This was done since it's critical that ast_log_safe
do nothing that could log during recursion checking.
ASTERISK-24155 #close
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/4502/
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* Add ast_register_cleanup to utils/clicompat.c to deal with
any utils that copy sources from main.
* Asterisk 13+: remove unused variables from core_local.c.
Review: https://reviewboard.asterisk.org/r/4534/
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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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* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
crashes in some cases. In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.
ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.
ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
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* In res/res_sorcery_realtime.c: Broke long line.
* In main/bucket.c: Eliminated unnecessary NULL check as
ast_sorcery_unref() is NULL tolerant and set the global object to NULL
after unref in the system shutdown bucket_cleanup().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
setting was never actually processed. This was due to not properly copying
over the global settings to the client settings when applying the
configuration to the run-time object.
Review: https://reviewboard.asterisk.org/r/4496/
ASTERISK-14233
ASTERISK-24780 #close
Reported by: Simon Arlott
patches:
asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
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messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to
(long) when printing members of certain time structs.
Review: https://reviewboard.asterisk.org/r/4507
ASTERISK-24879 #close
Reported by: snuffy
Tested by: snuffy
patches:
openbsd-time64.diff uploaded by snuffy (License 5024)
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This patch addresses compilation issues for OpenBSD. Specifically, it
addresses:
* It allows including <sys/vmmeter.h> in asterisk.c
* Provides a needed (size_t) cast in xmldoc.c
In 13+, it also addresses a conditional inclusion in loader.c.
Review: https://reviewboard.asterisk.org/r/4506
ASTERISK-24880 #close
Reported by: snuffy
Tested by: snuffy
patches:
misc-openbsd.diff uploaded by snuffy (License 5024)
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Valgrind found some memory leaks associated with
ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending
responses to OPTIONS requests, processing MESSAGE requests, and
res_pjsip supplements implementing the incoming_request callback.
* Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in
res/res_pjsip.c:supplement_on_rx_request(),
res/res_pjsip/pjsip_options.c:send_options_response(),
res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
res/res_pjsip_messaging.c:send_response().
* Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
res/res_pjsip_nat.c:nat_on_rx_message().
* Fixed inconsistent but benign return value in
res/res_pjsip/pjsip_options.c:options_on_rx_request().
Review: https://reviewboard.asterisk.org/r/4511/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Valgrind found a memory leak and invalid access.
* Fix invalid access by sscanf() being fed a non-nul terminated string of
digits in res/res_pjsip_sdp_rtp.c:get_codecs().
* Fix memory leak in main/sorcery.c:sorcery_object_field_destructor().
* Fix potential NULL pointer dereference in
main/xmldoc.c:xmldoc_get_syntax_config_option().
Review: https://reviewboard.asterisk.org/r/4513/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When r432935 was merged, it did correctly fix a situation where a FILE read
operation on the middle of a file buffer would not read the requested length
in the parameters passed to the FILE function. Unfortunately, it would also
allow the FILE function to append more bytes than what was available in the
buffer if the length exceeded the end of the buffer length.
This patch takes the minimum of the remaining bytes in the buffer along with
the calculated length to append provided by the original patch, and uses
that as the length to append in the return result. This patch also updates
the unit tests with the scenarios that were originally pointed out in
ASTERISK-21765 that the original implementation treated incorrectly.
ASTERISK-21765
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When logger.conf is missing or invalid enable console logging and display
an error message.
ASTERISK-24817 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4497/
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* Replace functions for ref/undef of dialogs and peers with macro's
to call ao2_t_bump/ao2_t_cleanup.
* Enable passthough of REF_DEBUG caller information to sip_alloc and
find_call.
ASTERISK-24882 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4189/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Release the scheduler reference to the dialog for reinvite timeout during
dialog_unlink_all.
ASTERISK-24876 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4491/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Using DEBUG_CHAOS several instances of a null
pointer crash, and one uninitialized variable
were uncovered and fixed. Also added details
on why Asterisk failed to initialize.
Review: https://reviewboard.asterisk.org/r/4468/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Locate potential crashes by exercising seldom
used code paths. This patch introduces a new
define DEBUG_CHAOS, and mechanism to randomly
return an error condition from functions that
will seldom do so. Functions that handle the
allocation of memory get the first treatment.
Review: https://reviewboard.asterisk.org/r/4463/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Valgrind found some memory leaks associated with ast_sockaddr_resolve().
Most of the leaks had already been fixed by earlier memory leak hunt
patches. This patch performs an audit of ast_sockaddr_resolve() and found
one more.
* Fix ast_sockaddr_resolve() memory leak in
apps/app_externalivr.c:app_exec().
* Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs
parameter for safety so the pointer will never be uninitialized on return.
The same goes for res/res_pjsip_acl.c:extract_contact_addr().
* Made functions that call ast_sockaddr_resolve() with RAII_VAR()
controlling the addrs variable use ast_free instead of ast_free_ptr to
provide better MALLOC_DEBUG information.
Review: https://reviewboard.asterisk.org/r/4509/
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Updated some documentation stating that endpoint identifiers registered without
a name are place at the front of the lookup list. Also renamed register method
'ast_sip_register_endpoint_identifier_by_name' to
'ast_sip_register_endpoint_identifier_with_name'
ASTERISK-24840
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.
ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In realtime, it is normal to have a database with both 'allow' and 'disallow'
columns in the schema. It is perfectly valid to have an 'allow' value of
'!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files,
you can't *not* provide the disallow value. Thus, the empty disallow value
causes a spurious WARNING message, which is kind of annoying.
This patch makes it so that a 'disallow' value with no ... value ... is
ignored. Granted, you can still screw this up as well, as technically
specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no
'allow' value in your database. But really, why would you do that? WHY?
ASTERISK-16779 #close
Reported by: Atis Lezdins
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This code originally kept a lock held when performing the HTTP
request to ensure that the options provided to curl remain valid.
This doesn't seem to be necessary these days and holding the lock
caused requests to happen sequentially instead of in parallel.
ASTERISK-18708 #close
Reported by: Dave Cabot
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The "core set debug channel" CLI command mistakenly had source filenames
added to its tab completion. This occurred because the CLI generator fell back
to the "core set debug" command which permits setting debug at a source
filename level.
ASTERISK-21038 #close
Reported by: Richard Kenner
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The loop that reads in a file was not correctly using the offset when
determining what bytes to append to the output. This patch corrects
the logic such that the correct portion of the file is extracted when an
offset is specified.
ASTERISK-21765
Reported by: John Zhong
Tested by: Matt Jordan, Di-Shi Sun
patches:
file_read_390821.patch uploaded by Di-Shi Sun (License 5076)
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This patch corrects the documentation for the AMD application. Specifically:
* It documents the maximum_word_length option, which limits the maximum allowed
length of a single utterance.
* It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH
was documented as MAXWORDS, while MAXWORDS was undocumented.
Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues.
ASTERISK-19470 #close
Reported by: Frank DiGennaro
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Also fixed similar problem with AMI action PJSIPShowEndpoints.
ASTERISK-24872 #close
Reported by: Dmitriy Serov
Review: https://reviewboard.asterisk.org/r/4487/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The res_pjsip modules were manually checking both name and number
presentation values when there is a function that determines the combined
presentation for a party ID struct. The function takes into account if
the name or number components are valid while the manual code rarely
checked if the data was even valid.
* Made use ast_party_id_presentation() rather than manually checking party
ID presentation values.
* Ensure that set_id_from_pai() and set_id_from_rpid() will not return
presentation values other than what is pulled out of the SIP headers. It
is best if the code doesn't assume that AST_PRES_ALLOWED and
AST_PRES_USER_NUMBER_UNSCREENED are zero.
* Fixed copy paste error in add_privacy_params() dealing with RPID
privacy.
* Pulled the id->number.valid test from add_privacy_header() and
add_privacy_params() up into the parent function add_id_headers() to skip
adding PAI/RPID headers earlier.
* Made update_connected_line_information() not send out connected line
updates if the connected line number is invalid. Lower level code would
not add the party ID information and thus the sent message would be
unnecessary.
* Eliminated RAII_VAR usage in send_direct_media_request().
Review: https://reviewboard.asterisk.org/r/4472/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.
ASTERISK-24840
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When an audiohook is created (which is used by the various Spy applications
and Snoop channel in Asterisk 13+), it initially is given a sample rate of
8kHz. It is expected, however, that this rate may change based on the media
that passes through the audiohook. However, the read/write operations on the
audiohook behave very differently.
When a frame is written to the audiohook, the format of the frame is checked
against the internal sample rate. If the rate of the format does not match
the internal sample rate, the internal sample rate is updated and a new SLIN
format is chosen based on that sample rate. This works just fine.
When a frame is read, however, we do something quite different. If the format
rate matches the internal sample rate, all is fine. However, if the rates
don't match, the audiohook attempts to "fix up" the number of samples that
were requested. This can result in some seriously large number of samples
being requested from the read/write factories.
Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of
audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)).
However, if the audiohook is still expecting an internal sample rate of 8000,
we'll attempt to "fix up" the requested samples to:
samples_converted = samples * (ast_format_get_sample_rate(format) /
(float) audiohook->hook_internal_samp_rate);
which is:
92160 = 3840 * (192000 / 8000)
This results in us attempting to read 92160 samples from our factories, as
opposed to the 3840 that we actually wanted. On a 64-bit machine, this
miraculously survives - despite allocating up to two buffers of length 92160
on the stack. The 32-bit machines aren't quite so lucky. Even in the case where
this works, we will either (a) get way more samples than we wanted; or (b) get
about 3840 samples, assuming the timing is pretty good on the machine.
Either way, the calculation being performed is wrong, based on the API users
expectations.
My first inclination was to allocate the buffers on the heap. As it is,
however, there's at least two drawbacks with doing this:
(1) It's a bit complicated, as the size of the buffers may change during the
lifetime of the audiohook (ew).
(2) The stack is faster (yay); the heap is slower (boo).
Since our calculation is flat out wrong in the first place, this patch fixes
this issue by instead updating the internal sample rate based on the format
passed into the read operation. This causes us to read the correct number of
samples, and has the added benefit of setting the audihook with the right
SLIN format.
Note that this issue was caught by the Asterisk Test Suite as a result of
r432195 in the 13 branch. Because this issue is also theoretically possible
in Asterisk 11, the change is being made here as well.
Review: https://reviewboard.asterisk.org/r/4475/
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RAII_VAR, which is used extensively in Asterisk to manage reference counted
resources, uses a GCC extension to automatically invoke a cleanup function
when a variable loses scope. While this functionality is incredibly useful
and has prevented a large number of memory leaks, it also prevents Asterisk
from being compiled with clang.
This patch updates the RAII_VAR macro such that it can be compiled with clang.
It makes use of the BlocksRuntime, which allows for a closure to be created
that performs the actual cleanup.
Note that this does not attempt to address the numerous warnings that the clang
compiler catches in Asterisk.
Much thanks for this patch goes to:
* The folks on StackOverflow who asked this question and Leushenko for
providing the answer that formed the basis of this code:
http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
* Diederik de Groot, who has been extremely patient in working on getting this
patch into Asterisk.
Review: https://reviewboard.asterisk.org/r/4370/
ASTERISK-24133
ASTERISK-23666
ASTERISK-20399
ASTERISK-20850 #close
Reported by: Diederik de Groot
patches:
RAII_CLANG.patch uploaded by Diederik de Groot (License 6600)
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Done as a separate commit from a finding in
https://reviewboard.asterisk.org/r/4467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When a type=global section is not defined in pjsip.conf the global
defaults are not applied. As a result the mandatory Max-Forwards header
is not added to SIP messages for res_pjsip/chan_pjsip.
The handling of pjsip.conf type=global objects has several problems:
1) If the global object is missing the defaults are not applied.
2) If the global object is missing the default_outbound_endpoint's default
value is not returned by ast_sip_global_default_outbound_endpoint().
3) Defines are needed so default values only need to be changed in one
place.
* Added a sorcery instance observer callback to check if there were any
type=global sections loaded. If there were more than one then issue an
error message. If there were none then apply the global defaults.
* Fixed ast_sip_global_default_outbound_endpoint() to return the
documented default when no type=global object is defined.
* Made defines for the global default values.
* Increased the default_useragent[] size because SVN version strings can
get lengthy and 128 characters may not be enough.
* Fixed an off-nominal code path ref leak in global_alloc() if the string
fields fail to initialize.
* Eliminated RAII_VAR in get_global_cfg() and
ast_sip_global_default_outbound_endpoint().
ASTERISK-24807 #close
Reported by: Anatoli
Review: https://reviewboard.asterisk.org/r/4467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Setting pjsip.conf useragent to an empty string results in an empty SIP
header being sent.
* Made not add an empty SIP header item to the global SIP headers list.
Review: https://reviewboard.asterisk.org/r/4467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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