ast_format_get_sample_rate(.) returns an unsigned type. The difference of a
substraction between two unsigned types does not get implicitly converted to a
signed type. Therefore, using abs(.) did not make sense.
ASTERISK-27549
Change-Id: Ib904d9ee0d46b6fdd1476fbc464fbbf813304017
Some compiler optimizers seem to assume that dlopen will not use
__attribute__((constructor)) functions to call back to the program.
This was causing resource_being_loaded to be optimized away completely.
ASTERISK-27531 #close
Tested By: abelbeck
Change-Id: If17a3b889e06811a0e7119f0539d052494d6ece9
(cherry picked from commit 92fb393cab)
Those SIP messages that create dialogs require a contact header to be present.
If the contact header was missing from the message it could cause Asterisk to
crash.
This patch checks to make sure SIP messages that create a dialog contain the
contact header. If the message does not and it is required Asterisk now returns
a "400 Missing Contact header" response. Also added NULL checks when retrieving
the contact header that were missing as a "just in case".
ASTERISK-27480 #close
Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe
(cherry picked from commit 53799318bc)
* listen uses the variable `s` for the result from ast_poll() then
overwrites it with the result of accept(). Create a separate variable
poll_result to avoid confusion since ast_poll does not return a file
descriptor.
* Resolve fd leak that would occur if setsockopt failed in listen.
* Reserve an extra byte while processing completion results from remote
daemon. This fixes a bug where completion processing used strstr() on
a string that was not '\0' terminated. This was no risk to the Asterisk
daemon, the bug was only reachable the remote console process.
* Resolve leak in handle_showchan when the channel is not found.
* Multiple leaks and a deadlock in pbx_config CLI completion.
* Fix leaks in "manager show command".
Change-Id: I8f633ceb1714867ae30ef4e421858f77c14485a9
When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before
close the file descriptor. Close the FD twice will hangs the asterisk
under heavy load.
ASTERISK-27299 #close
Reported-by: Aaron An
Tested-by: AaronAn
Change-Id: I870a072d73fd207463ac116ef97100addbc0820a
When a channel that is on hold gets added to a bridge by
the Bridge AMI action or the dialplan application of the same name,
music continues to play, causing "robotic sound".
This commit adds a call to ast_moh_stop to stop the music.
Also, it makes the AMI Park action use the right MOH class when the
channel gets parked.
Reported by: Zane Conkle
ASTERISK-25079 #close
Change-Id: I4b129c5a20c15e63968842460ac5a1a85903cf9f
In change_redirecting_information variables we use ast_strlen_zero to
see if a value should be saved. In the case where the value is not NULL
but is a zero length string we leaked.
handle_response_subscribe leaked a reference to the ccss monitor
instance.
Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f
chan_console supports multiple devices but the CLI only works on a
single device. 'console set active' selects this device.
Sadly that CLI picks the wrong command-line parameter and will only
work for a device called 'active'.
ASTERISK-27490 #close
Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d
Some variables are set and never changed, making them constant. This
means that code in the 'false' block of the conditional is unreachable.
In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.
Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
* Fix small leaks in from error condition in translate.c.
* Check new file descriptor is less than 0, not less than or equal.
Change-Id: Id7782775486175c739e0c4bf3ea5e17e3f452a99
* ast_linear_stream would leak a file descriptor if it failed to allocate
lin.
* ast_control_tone leaked zone and ts if ast_playtones_start failed.
Additionally added whitespace to ast_linear_stream, pulled assignments
out of conditionals for improved readability.
Change-Id: I6d1a10cf9161b1529d939b9b2d63ea36d395b657
This module uses AST_DEFINE_APP_ARGS_TYPE to define struct's instead of
directly using AST_DECLARE_APP_ARGS. Initialize the variables declared
in this way.
Change-Id: If97fbdd8d63a204e2efd498a192effc14e90fb31
* mwi_sub_event_cb: mwist leaked on separate_mailbox failure.
* add_email_attachment: A reference to sox_gain_tmpdir was used
after the storage was out of scope.
Change-Id: I6282c542ff7b82fa091177a912d11234a8b00a30
The completion generator is missing a return so typing "core set debug
all off <tab>" causes the command to actually execute.
Change-Id: Ibf6462088a74eee66967732b50445783ebefc20b
The mute/unmute sounds are only played when the
action is initiated using the DTMF menu.
ASTERISK-24756
Change-Id: I55b3dd5bc166096bf5e2f547ddd0ce355f36e3dc
The Local channel has never supported app_transfer
from what I can see so remove it from the documentation.
ASTERISK-25649
Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9
Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and
disable_multi_domain=no results in a misleading empty endpoint name
message. The message should say the endpoint was not found.
* Added missing endpoint not found message.
* Added more information to the empty endpoint name msgs if available.
* Eliminated RAII_VAR in request().
Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4
