If two channels enter different native rtp bridges at the same time it is
possible that the framehook interface data pointer can be corrupted
because the struct variable was declared static.
* Fixed the reentrancy corruption by changing the framehook interface
struct static variable to a stack local variable.
* Moved the hook.data assignment outside of the channel lock. It did not
need the lock's protection. It probably was giving a false sense of
security.
The testsuite
channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up
test caught this with MALLOC_DEBUG and DO_CRASH enabled.
Corey Farrell [Sun, 24 Dec 2017 04:51:13 +0000 (23:51 -0500)]
loader: Add volatile to resource_being_loaded.
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.
Kevin Harwell [Wed, 20 Dec 2017 22:17:40 +0000 (16:17 -0600)]
AST-2017-014: res_pjsip - Missing contact header can cause crash
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".
Corey Farrell [Tue, 19 Dec 2017 02:12:47 +0000 (21:12 -0500)]
CLI: Address multiple issues.
* 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".
Aaron An [Tue, 19 Dec 2017 20:15:29 +0000 (15:15 -0500)]
res_rtp_asterisk: Avoid close the rtp/rtcp fd twice.
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
Ivan Poddubny [Sat, 16 Dec 2017 13:51:16 +0000 (14:51 +0100)]
bridge: Stop music on hold on adding an arbitrary channel to a bridge
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.
Corey Farrell [Tue, 19 Dec 2017 00:59:57 +0000 (19:59 -0500)]
chan_sip: Fix memory leaks.
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.
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.
Corey Farrell [Tue, 19 Dec 2017 00:04:56 +0000 (19:04 -0500)]
app_voicemail: Fix memory management issues.
* 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.
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.
Corey Farrell [Tue, 21 Nov 2017 05:10:58 +0000 (00:10 -0500)]
loader: Rework of load_dynamic_module.
* Split off load_dlopen to perform actual dlopen, check results and log
warnings when needed.
* Use flags which minimize number of calls to dlopen required. First
attempt always uses RTLD_GLOBAL when global_symbols_only is enabled,
RTLD_LOCAL when it is not.
This patch significantly reduces the number of dlopen's performed. With
299 modules my system ran dlopen 857 times before this patch, 655 times
after this patch.
Corey Farrell [Wed, 22 Nov 2017 02:34:56 +0000 (21:34 -0500)]
loader: Minor fix to module registration.
This protects the module loader itself against crashing if dlopen is
called on a module from outside loader.c.
* Expand scope of lock inside ast_module_register to include reading of
resource_being_loaded.
* NULL check resource_being_loaded.
* Set resource_being_loaded NULL as soon as dlopen returns. This fixes
some error paths where it was not NULL'ed.
* Create module_destroy function to deduplicate code from
ast_module_unregister and modules_shutdown.
* Resolve leak that occured if a module did not successfully register.
* Simplify checking for successful registration.
Corey Farrell [Tue, 12 Dec 2017 18:55:12 +0000 (13:55 -0500)]
aco: Minimize use of regex.
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Corey Farrell [Tue, 12 Dec 2017 18:36:39 +0000 (13:36 -0500)]
aco: Create ways to minimize use of regex.
ACO uses regex in many situations where it is completely unneeded. In
some cases this doubles the total processing performed by
aco_process_config.
* Create ACO_IGNORE category type for use in place of skip_category
regex source string.
* Create additional aco_category_op values to allow specifying category
filter using either a single plain string or a NULL terminated array
of plain strings.
* Create ACO_PREFIX to allow matching option names to case insensitive
prefixes.
Richard Mudgett [Tue, 12 Dec 2017 00:20:06 +0000 (18:20 -0600)]
res_rtp_asterisk.c: Disable packet flood detection for video streams.
We should not do flood detection on video RTP streams. Video RTP streams
are very bursty by nature. They send out a burst of packets to update the
video frame then wait for the next video frame update. Really only audio
streams can be checked for flooding. The others are either bursty or
don't have a set rate.
* Added code to selectively disable packet flood detection for video RTP
streams.
Corey Farrell [Wed, 13 Dec 2017 15:50:59 +0000 (10:50 -0500)]
CLI: Remove special handling of 'core set verbose' from rasterisk.
rasterisk does not need to handle setting verbose levels locally, it
should just tell the daemon what it wants and print what it is given.
Just max out the verbose level on the local client so all filtering
happens on the daemon.
Joshua Colp [Thu, 30 Nov 2017 16:12:55 +0000 (16:12 +0000)]
AST-2017-012: Place single RTCP report block at beginning of report.
When the RTCP code was transitioned over to Stasis a code change
was made to keep track of how many reports are present. This count
controlled where report blocks were placed in the RTCP report.
If a compound RTCP packet was received this logic would incorrectly
place a report block in the wrong location resulting in a write
to an invalid location.
This change removes this counting logic and always places the report
block at the first position. If in the future multiple reports are
supported the logic can be extended but for now keeping a count
serves no purpose.
Joshua Colp [Wed, 6 Dec 2017 14:24:03 +0000 (14:24 +0000)]
pjsip: Ignore state changes from old transactions.
When we fail over to a new target we create a new transaction
and it becomes the current INVITE transaction. This does not
prevent the previous transaction from raising state changes
and causing the session to be prematurely disconnected if a
transport error occurs immediately.
This change backports a fix from PJSIP that eliminates the
incorrect state change and reduces when they would be raised
in the first place.
Ivan Poddubny [Tue, 12 Dec 2017 21:38:01 +0000 (22:38 +0100)]
app_queue: Fix extension state subscriptions removed on dialplan reload
The approach with having a single global subscription to all extension
state changes has one issue: dynamically created hints don't have any
watchers and are therefore garbage collected on the first dialplan
reload.
This change creates a state subscription for every queue member with a
hint as state_interface, thus increasing the count of watches for
hints, so they are not destroyed prematurely anymore.
There are 2 side effects:
1. The state change callback in app_queue is not executed when
there are no members referring to the extension.
2. The callback is called multiple times for the same hint if it's
associated with more than one queue member.
Corey Farrell [Tue, 12 Dec 2017 21:16:38 +0000 (16:16 -0500)]
menuselect: Tweak check for recently run configure.
Recently menuselect has randomly produced an error stating that
configure was just run and make had to be restarted. I believe this is
due to an incorrect menuselect/Makefile rule. The original rule
produced an error if makeopts or autoconfig.h were older than
makeopts.in or autoconfig.h.in. I believe this can create an issue if
makeopts is older than autoconfig.h.in or if autoconfig.h is older than
makeopts.in. The new rules compare files independently.
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
projects / thirdparty / asterisk.git / log
