The current settings AST_PBX_MAX_STACK is 128 entries which is
too low for some FreePBX installations with complex parking
arrangements. Increased to 512 if LOW_MEMORY is not defined.
ASTERISK-28300
Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299
To prevent one subsystem's taskprocessors from causing others
to stall, new capabilities have been added to taskprocessors.
* Any taskprocessor name that has a '/' will have the part
before the '/' saved as its "subsystem".
Examples:
"sorcery/acl-0000006a" and "sorcery/aor-00000019"
will be grouped to subsystem "sorcery".
"pjsip/distributor-00000025" and "pjsip/distributor-00000026"
will bn grouped to subsystem "pjsip".
Taskprocessors with no '/' have an empty subsystem.
* When a taskprocessor enters high-water alert status and it
has a non-empty subsystem, the subsystem alert count will
be incremented.
* When a taskprocessor leaves high-water alert status and it
has a non-empty subsystem, the subsystem alert count will be
decremented.
* A new api ast_taskprocessor_get_subsystem_alert() has been
added that returns the number of taskprocessors in alert for
the subsystem.
* A new CLI command "core show taskprocessor alerted subsystems"
has been added.
* A new unit test was addded.
REMINDER: The taskprocessor code itself doesn't take any action
based on high-water alerts or overloading. It's up to taskprocessor
users to check and take action themselves. Currently only the pjsip
distributor does this.
* A new pjsip/global option "taskprocessor_overload_trigger"
has been added that allows the user to select the trigger
mechanism the distributor uses to pause accepting new requests.
"none": Don't pause on any overload condition.
"global": Pause on ANY taskprocessor overload (the default and
current behavior)
"pjsip_only": Pause only on pjsip taskprocessor overloads.
* The core pjsip pool was renamed from "SIP" to "pjsip" so it can
be properly grouped into the "pjsip" subsystem.
* stasis taskprocessor names were changed to "stasis" as the
subsystem.
* Sorcery core taskprocessor names were changed to "sorcery" to
match the object taskprocessors.
Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56
Event type filtering is now enabled, and configurable per application. An app is
now able to specify which events are sent to the application by configuring an
allowed and/or disallowed list(s). This can be done by issuing the following:
PUT /applications/{applicationName}/eventFilter
And then enumerating the allowed/disallowed event types as a body parameter.
ASTERISK-28106
Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b
Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a
little easier to retrieve a string field from the JSON object.
Also added an 'ast_strings_equal' function that safely checks (checks for NULLs)
for equality between two strings.
Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b
When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.
This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.
This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.
ASTERISK-28277
Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6
Testing revealed that the cache added no benefit but that it could
consume excessive memory.
Two new index related functions were created:
ast_sounds_get_index_for_file() and ast_media_index_update_for_file()
which restrict index updating to specific sound files.
The original ast_sounds_get_index() and ast_media_index_update()
calls are still available but since they no longer cache the results
internally, developers should re-use an index they may already have
instead of calling ast_sounds_get_index() repeatedly. If information
for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().
The media_index directory scan code was elimininated in favor of
using the existing ast_file_read_dirs() function.
Since there's no more cache, ast_sounds_index_init now only
registers the sounds cli commands instead of generating the
initial index and subscribing to stasis format register/unregister
messages.
ast_sounds_reindex() is now a no-op but left for backwards
compatibility.
loader.c no longer registers "sounds" as a special reload target.
Both the sounds cli commands and the sounds ari resources were
refactored to only call ast_sounds_get_index() once per invocation
and to use ast_sounds_get_index_for_file() when a specific sound
file is requested.
Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d
A bug in GCC causes TEST_CEL to return
failure under the following conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline. The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()
Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7
Previously both AMI and ARI used a default route on
their stasis message router to handle some of the
messages for publishing out their respective
connection. This caused messages to be given to
their subscription that could not be formatted
into AMI or JSON.
This change adds an API call to the stasis message
router which allows a default route to be set as well
as formatters that the default route is expecting.
This allows both AMI and ARI to specify that their
default route only wants messages of their given
formatter. By doing so stasis can more intelligently
filter at publishing time so that they do not receive
messages which will not be turned into AMI or JSON.
ASTERISK-28244
Change-Id: I65272819a53ce99f869181d1d370da559a7d1703
This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to have the same
performance benefits as version 16.
By default old behavior, i.e. the ContactStatus event will be sent when a
device refreshes its registration.
Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46
During Bridging of two channels if masquerade operation is performed on a
channel (clone channel) which was created with endpoint details
(ast_channel_alloc_with_endpoint()) and the original channel which is created
without endpoint details (ast_channel_alloc()) then both the channels must
exchange their endpoint details or else after masquerade when clone channel
is being destroyed the endpoint cleanup callbacks will be destroyed too and
after call completion unique_id of original channel will still be there in
ast_endpoint structure's channel_ids container.
ASTERISK-28197
Change-Id: Ied0451f378a3f2a36acc8c0984959a69895efa17
This prevents use-after-scope issues when unwinding the stack,
which happens in reverse order. The varname variable needs to
remain alive for the destruction to be able to access it.
Issue was found using clang + address-sanitizer.
ASTERISK-28232 #close
Change-Id: I00811c34ae910836a5fb6d22304528aef92624db
This change adds statistics gathering to Stasis topics,
subscriptions, and message types. These can be viewed using
CLI commands and provide insight into how Stasis is used
and how long certain operations take to execute.
These are only available when Asterisk is compiled in
developer mode and do not have any impact under normal
operation.
ASTERISK-28117
Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f
A subscriber can now indicate that it only wants messages
that have formatters of a specific type. For instance,
manager can indicate that it only wants messages that have a
"to_ami" formatter. You can combine this with the existing
filter for message type to get only messages with specific
formatters or messages of specific types.
ASTERISK-28186
Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c
Some platforms provide an implementation of socket() and pipe2() that allow the
caller to specify that the resulting file descriptors should be non-blocking.
Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous
calls to fcntl() to set the O_NONBLOCK flag afterwards.
In passing, change ast_alertpipe_init() to use pipe2() directly instead of the
wrapper if it is available.
Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0
We've had multiple opportunities where Richard Mudgett's
malloc_trim patch has been useful. Let's get it
pushed up to gerrit and merged.
Since malloc_trim is only available in libc, an entry is
added to configure.ac to create a definition for
HAVE_MALLOC_TRIM.
Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.
ao2_container_alloc is now restricted to modules only and is being
removed from Asterisk 17.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
When a subscribe or unsubscribe occurs a message is published
containing this information. This change makes it so that the
message no longer uses stringfields or a lock, as both are not
really needed for the message.
Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632
We've been seeing crashes in libbfd when we attempt to generate
a stack trace from multiple threads. It turns out that libbfd
is NOT thread-safe. It can cache the bfd structure and give it to
multiple threads without protecting itself. To get around this,
we've added a global mutex around the bfd functions and also have
refactored the use of those functions to be more efficient and
to provide more information about inlined functions.
Also added a few more tests to test_pbx.c. One just calls
ast_assert() and the other calls ast_log_backtrace(). Neither are
run by default.
WARNING: This change necessitated changing the return value of
ast_bt_get_symbols() from an array of strings to a VECTOR of
strings. However, the use of this function outside Asterisk is not
likely.
ASTERISK-28140
Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
As mentioned in the comment I've added in the code there is no
ability to unsubscribe all subscribers from a topic and explicitly
destroy it. This is not currently a problem as we have two types of
topics:
Long lived topics which exist for the lifetime of the system.
Ephemeral topics which feed a long lived topic.
In the case of the ephemeral topics there is no subscriber which does
not have its lifetime managed by the same entity that has created
the topic. This ensures that when the topic is being unreferenced the
subscribers are also unsubscribed and destroyed, allowing the topic
to ultimately be destroyed as well.
Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a
This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.
The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
Change-Id: I16af967815efd904597ec2f033337e4333d097cd
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are still available for use but only in modules. Only
ao2_container_alloc remains due to it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
__ast_mutex_logger used the variable `canlog` without accepting it as a
argument. Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement. This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.
Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
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.
Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b
Use json_vsprintf from versions which contain fix for va_copy leak.
Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.
Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539
When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.
This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.
Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.
Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0
There's been a long standing leak when using topic pools. The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically. If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.
* Added stasis_topic_pool_delete_topic() so modules can clean up
topics from pools.
* Registered the topic pool containers so it can be examined from
the CLI when AO2_DEBUG is enabled. They'll be named
"<topic_pool_name>-pool".
Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25
This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.
ASTERISK-28059
Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189
Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.
* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
output.
* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
in brackets.
* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
to also set pjsip_rx_data.pkt_info.src_addr.
Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8
As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.
ASTERISK-28046 #close
Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc
Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address
needs to be rendered to a string and then parsed into the correct
structure. This also involves a call to getaddrinfo(3). The same is true
for the inverse operation.
Instead, because we know the internal structure of both ast_sockaddr and
pj_sockaddr, we can translate directly between the two without the
need for an intermediate string.
Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761
