The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
When a line is the maximum length "\n" is found at sizeof(buf) - 2 since
the last character is actually the null terminator. In addition if a
line was exactly 8190 plus a multiple of 8192 characters long the config
parser would skip the following line.
Additionally fix comment in voicemail.conf sample config. It previously
stated that emailbody can only contain up to 512 characters which is
always wrong. The buffer is normally 8192 characters unless LOW_MEMORY
is enabled then it is 512 characters. The updated comment states that
the line can be up to 8190 or 510 characters since the line feed and
NULL terminator each use a character.
ASTERISK-26688 #close
Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015
When an RTP packet is received by an RTP engine it has to map the
payload into the Asterisk format. The code was incorrectly checking
our own static list for ALL payloads if it couldn't find a negotiated one.
This included dynamic payloads. If the payload mapped to a format
of a different type (for example receiving a video packet on an audio
RTP instance) then the core stream code could cause a crash if a legacy
channel driver was in use as no stream would be present.
To provide further protection the core stream code will no longer assume
that a video or audio frame will always have a stream for legacy channel
drivers. If no stream is present the frame is dropped.
ASTERISK-27488
Change-Id: I022556f524ad8379ee73f14037040af17ea3316a
This will make the source filename match the 'module reload sounds'
command. This will allow conversion to a built-in module in Asterisk 16
without needing to redefine AST_MODULE.
Change-Id: Ifb8e489575b27eb33d8c0b6a531f266670557f6e
Expand locking to include full reload process for extconfig to ensure
nothing can read the config mappings between clearing and reloading.
Change-Id: I378316bad04f1b599ea82d0fef62b8978a644b92
* Replace ad-hoc array management with macro's from vector.h.
* Remove redundent logger messages.
* Use normal Asterisk allocators instead of directly using libc
allocators.
* Free memory when an API has no implementation or users.
Change-Id: Ic6ecb31798d4a78e7df39ece86a68b60eac05bf5
Add an AMI events Load and Unload for notify when the
module has been loaded and unloaded.
ASTERISK-27661
Change-Id: Ib916c41eddd63651952998f2f49c57c42ef87a64
Jansson is thread safe for all read-only functions and reference
counting starting v2.11. This allows simplification of our code and
removal of locking around reference counting and dumping.
Change-Id: Id985cb3ffa6681f9ac765642e20fcd187bd4aeee
This removes the embedded copy of editline from the Asterisk source
tree, making a system copy of libedit mandatory in Asterisk 16+.
ASTERISK-27634 #close
Change-Id: Iedb64ad92acb78419f3caefedaa2bb7cd2a1a33f
Need to remove all CDR's listed by a CDR object from the active_cdrs_all
container including the root/master record.
ASTERISK-27656
Change-Id: I48b4970663fea98baa262593d2204ef304aaf80e
* Changed to create ami_event string only when the given blob is not
json_null().
* Fixed bad expression.
ASTERISK-27621
Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f
ast_str_append_event_header() could potentially leak and corrupt memory if
the ast_str needed to expand to add the AMI event header.
* Fixed to return error if the ast_str_append() failed.
Change-Id: I92f36b855540743b208d76e274152ee2d758176d
* Made not allocate memory if the channel snapshot is an internal channel.
* Free memory earlier when no longer needed.
Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
This addresses all performance issues with 'module load' completion. In
addition to using ast_cli_completion_add we stop using libedit's
filename_completion_function, instead using ast_file_read_dir. This
ensures all results are produced from a single call to opendir.
Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134
The previous fix broke the case
HAVE_SYSINFO = no
HAVE_SYSCTL = yes
HAVE_SWAPCTL = no
which occurs on FreeBSD 11.1 for example.
ASTERISK-26563
Change-Id: If77c39bc75f0b83a6c8a24ecb2fa69be8846160a
Move initialization of units which do not require configuration to occur
before preload modules. This leaves only units which load config between
module preload and regular load stages.
Change-Id: I1d15384acad16a22c3498124421af474fa517478
This change causes the configure script to throw an error if neither
__sync nor __atomic builtin functions are available.
ASTERISK-27619
Change-Id: Ie01a281e0f5c41dfeeb5f250c1ccea8752f56ef9
The code which handled loading modules had too many situations which
would result in halting Asterisk startup. Treat most errors as declines
instead of failures. The exception is when the module load function
returns AST_MODULE_LOAD_FAILURE or an invalid code.
Clear the missingdeps vector when appropriate to ensure the next loop
starts clean.
ASTERISK-27620
Change-Id: I45547d9641fd45bd86d80250224417625631ad84
* Copy more than one character at a time when there is nothing to
substitute.
* Fix off by one error if a '}' or ']' is missing.
* Eliminated the requirement that the "used" parameter had to point to a
variable. The current callers were always declaring a variable to meet
the requirement and discarding the value put into that variable. Now it
can be NULL.
* In ast_str_substitute_variables_full() fixed using the bogus channel to
evaluate a function. We were not using the bogus channel we just created
to help evaluate a subexpression.
Change-Id: Ia83d99f4f16abe47f329eb39b6ff2013ae7c9854
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
