Richard Mudgett [Wed, 17 Oct 2018 21:08:19 +0000 (16:08 -0500)]
res_rtp_asterisk.c: Add conditional module dependency to res_pjproject
* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.
This patch is not in the upstream pjproject and does unsafe things with
the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset()
outside of the timer heap lock. pj_timer_entry_reset() is also called for
timers that are not about to be rescheduled in a few places.
Corey Farrell [Wed, 10 Oct 2018 09:37:23 +0000 (05:37 -0400)]
refdebug: Create refstats.py script.
This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.
* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.
Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.
Add comments to explain thread safety of the 'struct __priv_data'
bitfields.
Alexei Gradinari [Fri, 12 Oct 2018 17:14:03 +0000 (13:14 -0400)]
res_pjsip: set callerid_tag to empty string
This patch sets the callerid_tag to empty string by default.
If the callerid_tag is set to NULL then the tag does not
become part of a connected line update.
For example:
Alice's tag is "Alice".
Bob's tag is empty.
Charlie's tag is "Charlie".
Alice calls Bob and then does attended transfer to Charlie.
When Alice hangs up the CONNECTEDLINE(tag) is "Alice"
on the interception routine on the Charlie's channel, but should be empty.
Ths patch also fix memory leaks if there are more then one options
"callerid", "callerid_tag", "voicemail_extension" and "contact_user"
in the pjsip.conf endpoint definition.
Corey Farrell [Thu, 11 Oct 2018 11:24:40 +0000 (07:24 -0400)]
threadpool: Eliminate pointless AO2 usage.
thread_worker_pair, set_size_data and task_pushed_data structures are
allocated with AO2 objects, passed to a taskprocessor, then released.
They never have multiple owners or use locking so AO2 only adds
overhead.
Corey Farrell [Fri, 12 Oct 2018 17:21:24 +0000 (13:21 -0400)]
main/astfd: Fix GCC8 format-truncation warning.
The field used to store call arguments was not large enough to hold the
arguments string that can be constructed for 'open'. Expand it to
prevent this warning/error.
Richard Mudgett [Tue, 9 Oct 2018 21:18:49 +0000 (16:18 -0500)]
res_statsd.c: Fix returned reload status.
The return status when there was no change in statsd.conf was incorrect.
This resulted in the wrong status message on the CLI when reloading the
module.
* Fixed cleanup on initial load if initializing statsd failed.
neutrino88 [Wed, 3 Oct 2018 21:51:49 +0000 (17:51 -0400)]
core/frame: generate correct T.140 payload in ast_sendtext_data()
ast_sendtext_data() would create an incorrect T.140 text frame which
length include the null terminator byte. It causes ultimately RTP
packets to be send with this trailing 0. The proposed fix just set the
correct length to the text frame
Corey Farrell [Thu, 4 Oct 2018 23:33:25 +0000 (19:33 -0400)]
loader: Flag module as declined in all cases where it fails to load.
This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts
startup, but it's possible for this code to be returned on manual load
of a module after startup.
It is an error for a module to not have a load callback but this is not
a fatal system error. In this case flag the module as declined, return
AST_MODULE_LOAD_FAILURE only if a required module is broken.
Expand doxygen documentation for AST_MODULE_LOAD_*.
Richard Mudgett [Tue, 2 Oct 2018 21:15:37 +0000 (16:15 -0500)]
res_smdi.c: Fix module ref counting and inverted test.
I think this module is so screwed up that it doesn't work anymore. Even
with these attempts to fix things it still won't gracefully shut down.
The module refs will not go to zero to allow unloading the module.
* Fix module ref counting dealing with the SMDI interface object. There
were several off-nominal paths that unbalanced the module ref count. Also
the destructor freed the ao2 object itself which is bad. Made the
smdi_read thread not hold its own ref to the SMDI interface object so when
all refs go away the destructor will stop the listener thread.
* Fixed the smdi_load() return code of 1 concerning the number of
listeners. The test was inverted.
Richard Mudgett [Tue, 2 Oct 2018 21:23:31 +0000 (16:23 -0500)]
res_smdi.c: Made use defaults if the smdi.conf file does not exist.
This module is an optional dependency of a couple of other modules. If it
declines to load, it then forces other modules that can optionally use
this module to also decline.
* Made use the default configuration if the config file does not exist and
simplified some of the logic.
Sean Bright [Wed, 3 Oct 2018 12:56:34 +0000 (08:56 -0400)]
http.c: Reload TLS even if http.conf hasn't changed
There is currently no way to indicate to Asterisk that TLS certificates
and/or keys have been updated other than by modifying http.conf or
restarting Asterisk.
There is already code in main/tcptls.c that determines if a reload is
actually necessary based on the hashes of the certicate and dependent
files, so this change merely gives us a way to request a reload without
explicitly modifying http.conf.
Richard Mudgett [Tue, 2 Oct 2018 18:29:59 +0000 (13:29 -0500)]
res_statsd.c: Made use defaults if the statsd.conf file does not exist.
This module is an optional dependency of many modules. If it declines to
load it then forces other modules that can optionally use this module to
also decline.
* Made use default configuration if there is a config error or the config
file does not exist.
* Display list of unavailable dependencies when they cause another
module to fail loading.
* When a module declines to load find all modules which depend on it so
they can be declined and listed together.
* Prevent retry of declined modules during startup.
* When a module fails to dlopen try loading it with RTLD_LAZY so we can
attempt to display the list of missing dependencies.
These changes are meant to reduce logger spam that is caused when a
module has many dependencies and declines to load. This also fixes some
error paths which failed to recognize required modules.
Module load/start errors are delayed until the end of loader startup.
core/frame: Fix ast_frdup() and ast_frisolate() for empty text frames
If a channel creates an AST_TEXT_FRAME with datalen == 0, the ast_frdup()
and ast_frisolate() functions could create a clone frame with an invalid
data.ptr which would cause a crash. The proposed fix is to make sure that
for such empty text frames, ast_frdup() and ast_frisolate() return cloned
text frames with a valid data.ptr.
Corey Farrell [Mon, 1 Oct 2018 04:11:44 +0000 (00:11 -0400)]
astobj2: Record lock usage to refs log when DEBUG_THREADS is enabled.
When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock
was ever used, so it can be recorded in the REF_DEBUG destructor entry.
Create contrib/scripts/reflocks.py to process locking used by
allocator. This can be used to identify places where
AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage.
res_pjsip: improve realtime performance on CLI 'pjsip show contacts'
CLI command 'pjsip show contacts' inefficiently make a lot of DB requests.
For example if there are 10k aors then asterisk requests these 10k records
of aor and then does 10k requests of contact - one request per aor.
Even if use 'like <pattern>' the asterisk requests all aor's and contact's
records and then filters them by itself.
This patch gathers contact's container by
- retrieving all dynamic contacts by regex (filtered by reg_server)
- retrieving all aors with permanent contacts
- finally filters container by regex
Add a volatile flag to lock tracking structures so we only need to use
the global lock when first initializing tracking.
Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by
astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is
not defined.
George Joseph [Thu, 27 Sep 2018 18:19:28 +0000 (12:19 -0600)]
app_confbridge: Use bridge join hook to send join and leave events
The first attempt at publishing confbridge events to participants
involved publishing them at the same time stasis events were
created. This caused issues with bridge and channel locks. The
second attempt involved publishing them when the stasis events
were received by the code that published the confbridge AMI events.
This caused timing issues because, depending on resources available,
the event could be received before channels actually joined the
bridge and would therefore fail to send messages to the participant.
This attempt reverts to the original mechanism with one exception.
The join and leave events are published via bridge join and leave
hooks. This guarantees the states of the channels and bridge and
provides deterministic timing for event publishing.
When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log
before we shutdown astobj2_container. This caused the AO2_DEBUG
container registration container to be reported as a leak.
app_queue: Fix Attended transfer hangup with removing pending member.
This issue related to setting of holdtime, announcements, member delays.
It works well if we set the member delays to "0" and no announcements
and no holdtime.This issue will happen if we set member delays to "1",
"2"... or announcements or holdtime and hangs up the call during
processing it.
And here is the reason:
(At the step of answering a phone.)
It takes care any holdtime, announcements, member delays,
or other options after a call has been answered if it exists.
Normally, After the call has been aswered,
and we wait for the processing one of the cases of the member delays
or hold time or announcements finished, "if (ast_check_hangup(peer))"
will be not executed, then queue will be updated at update_queue().
Here, pending member will be removed.
However, after the call has been aswered,
if we hangs up the call during one of the cases of the member delays
or hold time or announcements, "if (ast_check_hangup(peer))"
will be executed.
outgoing = NULL and at hangupcalls, pending members will not be removed.
* This fixed patch will remove the pending member from container
before hanging up the call with outgoing is NULL.
ASTERISK-27920
Reported by: Cao Minh Hiep
Tested by: Cao Minh Hiep
Moritz Fain [Tue, 26 Jun 2018 14:17:37 +0000 (16:17 +0200)]
res_stasis: Fix stale data in ARI bridges
Fixed an issue that resulted in "Allocation failed" each time an ARI
request was made to start playing MOH on a bridge.
In bridge_moh_create() we were attaching the after bridge callbacks to
chan which is the ;1 channel of the unreal channel pair. We should have
attached them to the ;2 channel which is pushed into the bridge by
ast_unreal_channel_push_to_bridge(). The callbacks are called when the
specific channel leaves the bridging system. Since the ;1 channel is
never put into a bridge the callbacks never get called. The callbacks
then never remove the moh_wrapper from the app_bridges_moh container. As
a result we cannot find the channel associated with the wrapper to start
MOH because it has hungup. This is the reason causing the reported issue.
* Rather than using after bridge callbacks to cleanup, we now have
moh_channel_thread() doing the cleanup when the channel hangs up.
* Fixed moh_channel_thread() accumulating control frames on the stasis
bridge MOH channel until MOH is stopped. Control frames are no longer
accumulated while MOH is playing.
* Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may
or may not return a channel ref. As a result ast_ari_bridges_start_moh()
wouldn't know it may have a channel ref to release.
stasis_app_bridge_moh_channel() will now return a ref with the channel it
returns.
Ben Ford [Mon, 10 Sep 2018 16:28:09 +0000 (11:28 -0500)]
res_rtp_asterisk.c: Add "seqno" strictrtp option
When networks experience disruptions, there can be large gaps of time
between receiving packets. When strictrtp is enabled, this created
issues where a flood of packets could come in and be seen as an attack.
Another option - seqno - has been added to the strictrtp option that
ignores the time interval and goes strictly by sequence number for
validity.
On SQL error there is not diagnostic information about this error.
There is only
WARNING res_odbc.c: SQL Execute error -1!
The function ast_odbc_print_errors calls a SQLGetDiagField to get the number
of available diagnostic records, but the SQLGetDiagField returns 0.
However SQLGetDiagRec could return one diagnostic records in this case.
Looking at many example of getting diagnostics error information
I found out that the best way it's to use only SQLGetDiagRec
while it returns SQL_SUCCESS.
Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR
to res_config_odbc.
chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI
With tls and udp enabled asterisk generates a warning about sending
message via udp instead of tls.
sip notify command via cli works as expected and without warning.
asterisk has to set the connection information accordingly to connection
and not on presumption
George Joseph [Mon, 24 Sep 2018 22:56:07 +0000 (16:56 -0600)]
configure.ac: Check for unbound version >= 1.5
In order to do this and provide good feedback, a new macro was
created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and
path setups for the library then compiles, links and runs a supplied
code fragment to do the final determination. In this case, the
final code fragment compares UNBOUND_VERSION_MAJOR
and UNBOUND_VERSION_MINOR to determine if they're greater than or
equal to 1.5.
Since we require version 1.5, some code in res_resolver_unbound
was also simplified.
res_rtp_asterisk: Raise event when RTP port is allocated
This change raises a testsuite event to provide what port
Asterisk has actually allocated for RTP. This ensures that
testsuite tests can remove any assumption of ports and instead
use the actual port in use.
* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
safe to just use ast_json_ref.
Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped. This was previously
avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null(). The
"?" is a new feature of ast_json_pack starting with Asterisk 16.
projects / thirdparty / asterisk.git / log
