The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.
ASTERISK-25180 #close
Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
This resolves two observed race conditions.
First, a bit of background on what the Stasis application does:
1a Creates a stasis_app_control structure. This structure is linked into
a global container and can be looked up using a channel's unique ID.
2a Puts the channel in an event loop. The event loop can exit either
because the stasis_app_control structure has been marked done, or
because of some other factor, such as a hangup. In the event loop, the
stasis_app_control determines if any specific ARI commands need to be
run on the channel and will run them from this thread.
3a Checks if the channel is bridged. If the channel is bridged, then
ast_bridge_depart() is called since channels that are added to Stasis
bridges are always imparted as departable.
4a Unlink the stasis_app_control from the container.
When an ARI command is received by Asterisk, the following occurs
1b A thread is spawned to handle the HTTP request
2b The stasis_app_control(s) that corresponds to the channel(s) in the
request is/are retrieved. If the stasis_app_control cannot be
retrieved, then it is assumed that the channel in question has exited
the Stasis app or perhaps was never in Stasis in the first place.
3b A command is queued onto the stasis_app_control, and the channel's
event loop thread is signaled to run the command.
4b While most ARI commands do nothing further, some, such as adding or
removing channels from a bridge, will block until the command they
issued has been completed by the channel's event loop.
The first race condition that is solved by this patch involves a crash
that can occur due to faulty detection of the channel's bridged status
in step 3a. What can happen is that in step 2a, the event loop may run
the ast_bridge_impart() function to asynchronously place the channel
into a bridge, then immediately exit the event loop because the channel
has hung up. In step 3a, we would detect that the channel was not
bridged and would not call ast_bridge_depart(). The reason that the
channel did not appear to be bridged was that the depart_thread that is
spawned by ast_bridge_impart() had not yet started. That is the thread
where the channel is marked as being bridged. Since we did not call
ast_bridge_depart(), the Stasis application would exit, and then the
channel would be destroyed Then the depart_thread would start up and
try to manipulate the destroyed channel, causing a crash.
The fix for this is to switch from using ast_channel_is_bridged() to
checking the NULLity of ast_channel_internal_bridge_channel() to
determine if ast_bridge_depart() needs to be called. The channel's
internal bridge_channel is set when ast_bridge_impart() is called and
is NULLed by the call to ast_bridge_depart(). If the channel's internal
bridge_channel is non-NULL, then the channel must have been imparted
into the bridge and needs to be departed, even if the actual bridging
operation has not yet started. By departing the channel when necessary,
the thread that is running the Stasis application will block until the
bridge gives the okay that the depart_thread has exited.
The second race condition that is solved by this patch involves a leak
of HTTP handler threads. The problem was that step 2b would successfully
retrieve a stasis_app_control structure. Then step 2a would exit the
channel from the event loop due to a hangup. Steps 3a and 4a would
execute, and then finally steps 3b and 4b would. The problem is that at
step 4b, when attempting to add a channel to a bridge, the thread would
block forever since the channel would never execute the queued command
since it was finished with the event loop. This meant that the HTTP
handling thread would be leaked, along with any references that thread
may have owned (in my case, I was seeing bridges leaked).
The fix for this is to hone in better on when the channel has exited the
event loop. The stasis_app_control structure has an is_done field that
is now set at each point where the channel may exit the event loop. If
step 2b retrieves a valid stasis_app_control structure but the control
is marked as done, then the attempted operation exits immediately since
there will be nothing to service the attempted command.
ASTERISK-25091 #close
Reported by Ilya Trikoz
Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
To prevent confusion I am removing the prefetch option until such
time as it is implemented. All other functionality, however, has
been implemented.
ASTERISK-25067
Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895
This event was added some time ago in order to clarify when a channel
took the place of another channel in a parking lot. However, there was
no XML documentation added for the event. This patch adds the XML
documentation.
ASTERISK-24900 #close
Reported by Rusty Newton
Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac
Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject when using an external DNS resolver to process messages for the
transaction.
* Add threadpool API call to get the current serializer associated with
the worker thread.
* Pick a serializer from a pool of default serializers if the caller of
res_pjsip.c:ast_sip_push_task() does not provide one.
This is a simple way to ensure that all outgoing SIP request messages are
processed under a serializer. Otherwise, any place where a pushed task is
done that would result in an outgoing out-of-dialog request would need to
be modified to supply a serializer. Serializers from the default
serializer pool are picked in a round robin sequence for simplicity.
A side effect is that the default serializer pool will limit the growth of
the thread pool from random tasks. This is not necessarily a bad thing.
* Made pjsip_resolver.c use the requesting thread's serializer to execute
the async callback.
* Made pjsip_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.
ASTERISK-25115 #close
Reported by: John Bigelow
Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a
* Fix query_set destruction before we are done kicking the queries off.
* Fixed no queries requested handling.
* Add empty queries request unit test.
* Added missing allocation check in ast_dns_query_set_add().
* Made initial pjsip resolving query vector slightly larger.
ASTERISK-25115
Reported by: John Bigelow
Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2
This patch fixes use-after-free bugs caught by AddressSanitizer.
1. PJSIP transport manager may decide to destroy transport on its own.
For example, when the contact registered via websocket has not renewed
its registration in time. The transport was destoyed, but the websocket
listener thread was still active until the socket closes, and then tried
to call transport_shutdown on transport that has been freed.
Also, the transport destructor accessed wstransport->rdata.tp_info.pool
right after freeing memory that contained wstransport itself.
This patch converts transport to an ao2 object, allowing it to be
refcounted, so that it is available until both websocket listener and
pjsip transport manager are finished with it.
2. The websocket listener deletes the last reference on websocket session
when the tcp connection is closed, and it gets destroyed, but
the transport manager may still use it, for example when disconnect
happens in the middle of a SIP transaction.
A new reference to websocket session has been added that is released
with the transport to prevent this.
ASTERISK-25096 #close
Reported by: Josh Kitchens
ASTERISK-24963 #close
Reported by: Badalian Vyacheslav
Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b
* Add some type casting so tv_usec can really be a long, instead of
some strange platform specific type.
* Add some .dylib style files to .gitignore.
* Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
versions of GCC, when compiling the Homebrew formula for Asterisk,
are not properly passing the -Xlinker options to the linker. Given
that -Wl, does exactly the [same thing][], and does it properly, this
patch changes the -Xlinker options to use -Wl, instead.
[reasons unknown]: http://bit.ly/1SUbEYx
[same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html
Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd
This change implements the expire_on_reload option for memory caches.
If enabled and a reload is performed all objects within the cache
will be expired and the cache emptied.
ASTERISK-25067
Reported by: Matt Jordan
Change-Id: Id46aa1957d660556700e689e195eed57c989b85e
This change adds a CLI command which can perform memory cache thrashing as well
as unit tests which perform thrashing under the following configurations:
1. Low number of unique objects that go stale after 1 second
2. Low number of unique objects that expire after 1 second
3. Low number of unique objects which are constantly updated
4. Large number of unique objects which exceed a defined cache size
5. Large number of unique objects which exceed a defined cache size
that also expire and go stale rapidly
6. Large number of unique objects which expire and go stale rapidly
7. Large number of unique objects
For all of the above there are a large number of threads constantly
attempting to retrieve random objects and each test runs for a few
seconds.
ASTERISK-25067
Reported by: Matt Jordan
Change-Id: I8c8ceff977332c80ed4a31f10d694d48552b2f78
This change adds a testsuite event for when a refresh occurs.
This is useful as it provides a guaranteed mechanism of knowing when
it has occurred instead of waiting an arbitrary amount of time.
ASTERISK-25067
Reported by: Matt Jordan
Change-Id: Iaa6b8d2d6bab7f99ee08e1c8908b8272a8987e65
It is possible to receive incoming requests or responses after the channel
on an ast_sip_session has been destroyed and NULLed out. Handlers of these
sorts of requests or responses need to be prepared for the possibility
that the channel is NULL or else they could cause a crash.
While several places have been amended to deal with NULL channels, there
were still a couple of places that needed updating.
res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
return early if there is no channel on the session.
res_pjsip_session.c: When handling a 302 response, we need to stop the
redirecting attempt if there is no channel on the session.
ASTERISK-25148 #close
reported by Mark Michelson
Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9
contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status
to force the creation of a contact_status object whenever a new
contact is added but it didn't unref the returned object.
Added an ao2_cleanup(status) to plug the leak.
ASTERISK-25141
Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40
Reported-by: Corey Farrell
app_control_register_rule and app_control_unregister_rule lock/unlock
the queue, which is a mutating operation according to the
ao2_lock/_unlock prototype. Depending on the specific (implicit) casts
in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only
callers of those functions do not have the const, get consistent results
by just dropping it.
Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb
When the remote peer requires authentication for in-dialog requests then
re-INVITEs to the peer cause the call to be disconnected and other
in-dialog requests to the peer like MESSAGE just don't go through.
* Made session_inv_on_tsx_state_changed() handle in-dialog authentication
for re-INVITEs and other methods. Initial INVITEs cannot be handled here
because the INVITE transaction must be restarted earlier.
* Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in
preparation for removing the file. The generic outbound authentication
code did not work as well as anticipated.
* Created outbound_invite_auth() to only handle initial outbound INVITEs.
Re-INVITEs cannot be handled here. The re-INVITE transaction is still in
progress and the PJSIP library cannot handle the overlapping INVITE
transactions. Other method types should not be handled here as this code
only works on outgoing calls and we need to handle incoming and outgoing
calls.
ASTERISK-25131 #close
Reported by: Richard Mudgett
Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0
The loop to find the first available contact of an endpoint grabbed
contact from the iterator, then checked for offline state. This
caused the first contact after the state was found to leak a reference.
ASTERISK-25141
Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08
When permanent_uri_handler was creating the contact status
object for each contact, it wasn't unreffing it at the
end of the loop.
ASTERISK-25141 #close
Reported-by: Corey Farrell
Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12
This change adds the following CLI commands and AMI actions:
sorcery memory cache show
sorcery memory cache dump
sorcery memory cache expire
sorcery memory cache stale
SorceryMemoryCacheExpire
SorceryMemoryCacheExpireObject
SorceryMemoryCacheStale
SorceryMemoryCacheStaleObject
These allow both examination and manipulation of sorcery memory
caches from external sources.
Cached objects can be explicitly expired from a cache or marked
as stale. If expired they are immediately removed. If marked as
stale they will be background refreshed when next retrieved.
ASTERISK-25067
Reported by Matt Jordan
Change-Id: I68e03cfd8c34b5e07f4b6ee4fd93a3f4a00a3d9e
This change introduces a check of object_lifetime_stale when retrieving
cached objects. If the amount of time the object has been in the cache
exceeds the lifetime, then a task is scheduled to update the cached
object based on an object retrieved from other sorcery wizards instead.
To prevent the cached object from being retrieved during a refresh,
thread-local storage is used to mark the thread as being a stale object
update. This results in the cache returning no object, leading to
sorcery querying other wizards for the object instead.
A test has been added for stale objects as well. This test ensures that
stale objects are retrieved the same as freshly-cached objects. The test
also ensures that after an object is stale, changes in the backend are
reflected in the cache, to include if the object has been deleted from
the backend.
ASTERISK-25067
Reported by Matt Jordan
Change-Id: I9bd7c049adf6939bfe2899f393c2bfbbf412d217
Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Use function PQescapeStringConn for escaping the name of the table and
schema instead of doing it manually.
ASTERISK-25132 #close
Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
Change-Id: I302a263f7210d20925f14716b508b081998b7608
Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves
truncated before passing to pjsip_tpmgr_receive_packet, but the length
was passed unaltered, thus causing memory corruption and segfault.
ASTERISK-25122 #close
Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab
Many uses of stasis_unsubscribe in modules can be reached through unload.
These have been switched to stasis_unsubscribe_and_join.
Some subscription callbacks do nothing, for these I've created a noop
callback function in stasis.c. This is used by some modules that monitor
MWI topics in order to enable cache, since the callback does not become
invalid after dlclose it is safe to use stasis_unsubscribe on these, even
during module unload.
ASTERISK-25121 #close
Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
In addition to specifying lists of 'presence' and 'message-summary',
users can also create lists of type 'dialog'. These should be treated in
the same fashion as 'presence'.
Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e
When a SUBSCRIBE request is made to a dialplan hint that doesn't exist,
the current NOTICE message informing users of this swaps the context and
extension parameters. This can cause a bit of confusion.
Thanks to CptBurger in #asterisk for helping to point this out.
Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43
Prior to this patch, when a WebSocket connection is made, ARI would not
be informed of the connection until after the WebSocket layer had
accepted the connection. This created a brief race condition where the
ARI client would be notified that it was connected, a channel would be
sent into the Stasis dialplan application, but ARI would not yet have
registered the Stasis application presented in the HTTP request that
established the WebSocket.
This patch resolves this issue by doing the following:
* When a WebSocket attempt is made, a callback is made into the ARI
application layer, which verifies and registers the apps presented in
the HTTP request. Because we do not yet have a WebSocket, we cannot
have an event session for the corresponding applications. Some
defensive checks were thus added to make the application objects
tolerant to a NULL event session.
* When a WebSocket connection is made, the registered application is
updated with the newly created event session that wraps the WebSocket
connection.
ASTERISK-24988 #close
Reported by: Joshua Colp
Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636
This patch refactors the transaction timeout processing to eliminate
calling the lower level public pjsip functions and reverts to calling
pjsip_endpt_send_request again. This is the result of me noticing
a possible incompatibility with pjproject-2.4 which was causing
contact status flapping.
The original version of this feature used the lower level calls to
get access to the tsx structure in order to cancel the transaction
when our own timer expires. Since we no longer have that access,
if our own timer expires before the pjsip timer, we call the callbacks
and just let the pjsip transaction take it's own course. When the
transaction ends, it discovers the callbacks have already been run
and just cleans itself up.
A few messages in pjsip_configuration were also added/cleaned up.
ASTERISK-25105 #close
Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
This makes the "object_lifetime_maximum" option operational.
On the addition of an object to an empty memory cache a scheduled
task is created which, when invoked, expires objects from the cache
which have exceeded their lifetime. If more objects have been added
the remaining life of the oldest object is used to schedule the
next invocation of the scheduled task.
If the oldest object is removed from the cache before it can be
expired automatically the scheduled task is cancelled, if possible,
and the lifetime of the next oldest is used to schedule the task.
If during these two operations no additional objects exist in the
cache then no task is scheduled.
An additional unit test has been added which verifies this
functionality.
ASTERISK-25067
Reported by: Matt Jordan
Change-Id: I87409674674a508e7717ee20739ca15cec6ba7b6
When an inbound call is received the To header is checked
for the "line" option. Some remote servers will place this
in the request URI instead. This adds an additional check for
the option in the request URI.
ASTERISK-25072 #close
Reported by: Dmitriy Serov
Change-Id: Id4e44debbb80baad623b914a88574371575353c8
This makes the "maximum_objects" option operational.
A heap has been added alongside the hash table in the cache. When
objects are added to the cache, they are also added to the heap.
Similarly, when objects are removed from the cache, they are removed
from the heap.
The heap's use comes into play when an item is to be added to a "full"
cache. When the cache is full, the oldest item is removed from the
cache, using the heap to determine the oldest item.
A unit test has been added that verifies that the maximum_objects option
works as expected and that the oldest object is removed from the cache
when an object beyond the maximum is added.
ASTERISK-25067 #close
Reported by Matt Jordan
Change-Id: I490658830e9c4cbf0b3051e4cdc4913cf9f1b73a
This change adds a basic res_sorcery_memory_cache module which implements
configuration option parsing, configuration file parsing for threading,
sorcery interface implementation, and unit tests.
Objects can be added, updated, deleted, and retrieved from the memory
cache. Automatic expiration and stale handling will be added in the
future.
Note that unit tests exist within the module itself in case the
threading done as a result of expiration results in asynchronous
actions (which it likely will). Providing access and a notification
mechanism for an external test module would be complicated and
not worth it.
ASTERISK-25067 #close
Reported by: Matt Jordan
Change-Id: Id8a6a357ef5a83d466f81eee56a67d13eeb118b9
Use ast_manager_register_xml for res_mwi_external_ami manager
actions. This ensures the module is held open while any of
the actions are being run.
ASTERISK-25117 #close
Reported by: Corey Farrell
Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7
This patch updates http_websocket and its corresponding implementation
with a pre-session established callback. This callback allows for
WebSocket server consumers to be notified when a WebSocket connection is
attempted, but before we accept it. Consumers can choose to reject the
connection, if their application specific logic allows for it.
As a result, this patch pulls out the previously private
websocket_protocol struct and makes it public, as
ast_websocket_protocol. In order to preserve backwards compatibility
with existing modules, the existing APIs were left as-is, and new APIs
were added for the creation of the ast_websocket_protocol as well as for
adding a sub-protocol to a WebSocket server.
In particular, the following new API calls were added:
* ast_websocket_add_protocol2 - add a protocol to the core WebSocket
server
* ast_websocket_server_add_protocol2 - add a protocol to a specific
WebSocket server
* ast_websocket_sub_protocol_alloc - allocate a sub-protocol object.
Consumers can populate this with whatever callbacks they wish to
support, then add it to the core server or a specified server.
ASTERISK-24988
Reported by: Joshua Colp
Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2
This patch fixes a number of errors and warning messages in the doxygen
log. Specifically, it addresses:
* A number of files incorrectly places a '\brief' tag immediately after
a '\file' tag. Doing so emits a warning, as '\file' takes an optional
argument specifying which file the doxygen comment is for. As '\brief'
is not a file, doxygen was unamused.
* A grouping of Stasis Topics and Messages in rtp_engine.h was
incorrectly terminated. We now correctly terminate the grouping, which
prevents members of rtp_engine.h from showing up in the wrong group.
* Group indicators which are not part of the Stasis Topics and Messages
group were removed. Group indicators without an \addtogroup or
\ingroup have no meaning.
Change-Id: Ia1415ffec6767e27233ae1cae5ed5970de5656d4
The config wizard was always pulling the first occurrence of
a variable from an ast_variable list but this gets the template
value from the list instead of any overridden value. This patch
creates ast_variable_find_last_in_list() in config.c and updates
res_pjsip_config_wizard to use it instead of
ast_variable_find_in_list. Now the overridden values, where they
exist, are used instead of template variables.
Updated test_config to test the new API.
ASTERISK-25089 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4
There are 3 ways that calls directly to standard allocator functions can
be dealt with:
1. Block their use, cause them to generate an error. This is the default.
2. Replace them with the Asterisk equivalent function calls.
3. Leave them alone.
This change allows one of these 3 options to be selected by any source.
The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT,
or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK
is the correct option, so it is default when ASTMM_LIBC is not defined.
In some cases when building 3rd party code it is desirable to have it use
Asterisk functions, without changing the whole source - ASTMM_REDIRECT
accomplishes this. When using 3rd party libraries sometimes a static
inline function will make use of malloc or free. In these cases it may
be unsafe to replace the allocator in the header, as it's possible the
memory could be freed by the library using standard allocators. For
those cases ASTMM_IGNORE is needed.
Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7
First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC
https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values.
Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31
astmm.h includes defines that are meant to cause error's when standard
allocators (malloc, calloc, free, etc) are used. It actually only
causes a warning, which is not always caught on certain sources. In
modules this unknown symbol is not detected until runtime, where the
module fails to load. This modifies the define's so that using one
of the blocked functions will cause a compile error regardless of
CFLAGS.
Moved spandsp header includes to before asterisk.h so the static inline
functions can continue using malloc and free. Although these functions
are never called and optimized away, the updated replacement macro's
would still cause a failure.
Change-Id: I532640aca0913ba9da3b18c04a0f010ca1715af5
While trying to get WebRTC working with chan_pjsip, I was running
into the following error:
Attempted to set an invalid DTLS-SRTP configuration on RTP
instance...
Josh helpfully pointed out that res_srtp.so might not be loaded, and
sure enough, it wasn't. This patch adds a ERROR indiciating as much
to hopefully help others having a similar problem.
Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f
The res_pjsip_exten_state module currently has a race condition between
processing the extension state callback from the PBX core and processing
the subscription shutdown callback from res_pjsip_pubsub. There is currently
no synchronization between the two. This can present a problem as while
the SIP subscription will remain valid the tree it points to may not.
This is in particular a problem as a task to send a NOTIFY may get queued
which will try to use the tree that may no longer be valid.
This change does the following to fix this problem:
1. All access to the subscription tree is done within the task that
sends the NOTIFY to ensure that no other thread is modifying or
destroying the tree. This task executes on the serializer for the
subscriptions.
2. A reference to the subscription serializer is kept to ensure it
remains valid for the lifetime of the extension state subscription.
3. The NOTIFY task has been changed so it will no longer attempt
to send a NOTIFY if the subscription has already been terminated.
ASTERISK-25057 #close
Reported by: Matt Jordan
Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643
Creating a snoop channel in ARI and spying only on a single direction (in or
out) results in CPU utilization continually increasing until the CPU is fully
consumed. This occurs because frames are being put in the opposing direction's
slin factory queue, but not being removed.
Fixed the problem by always reading and disposing of frames from the opposite
queue of the direction selected.
ASTERISK-24938 #closes
Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60
Missed this module in the previous commit. res_ari_bridges uses symbols
from res_stasis_playback and res_stasis_recording.
ASTERISK-25027 #close
Reported by: Corey Farrell
Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f
* Pass module to ast_cli_register and ast_cli_register_multiple.
* Add a module reference before executing any CLI callback, remove
the reference when complete.
ASTERISK-25049 #close
Reported by: Corey Farrell
Change-Id: I7aafc7c9f2b912918f28fe51d51e9e8a755750e3
ast_module_info->self is often needed to register items with the core. Many
modules have ad-hoc code to make this pointer available to auxiliary sources.
This change updates the module build process to make the needed information
available to all sources in a module.
ASTERISK-25056 #close
Reported by: Corey Farrell
Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815
Apply the negative connection cache setting to all connections,
even those that are not pooled. This ensures that the connection
will not be re-established before the negative connection cache
time is met.
ASTERISK-22708 #close
Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271
Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options.
This extra space prevented any of the dependencies from being seen by
menuselect, so building with default options would fail if PJSIP was
not installed.
This also makes the tool that extracts information for menuselect
tolerant of multiple spaces in the future.
ASTERISK-25033 #close
Reported by: Peter Whisker
Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698
The res_ari_device_states module depends on res_stasis_device_state,
not res_stasis_device_states.
Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de
This patch has two main purposes:
1) Improve warning messages when ACLs are configured improperly.
2) Prevent misconfigured ACLs from allowing potentially unwanted
traffic.
To acomplish point (2) in most cases, whatever configuration object that
the ACL belonged to was not allowed to load.
The one exception is res_pjsip_acl. In that case, ACLs are their own
configuration object. Furthermore, the module loading code has no
indication that a ACL configuration had a failure. So the tactic taken
here is to create an ACL that just blocks everything.
ASTERISK-24969
Reported by Corey Farrell
Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae
When the PJSIP pjsip_regc_send function is invoked and an error
status returned the caller currently decrements the reference count
of the client state that it just incremented, assuming the
registration callback would not have been invoked. In practice
this is not correct. If the failure happens after the transaction
has been set up the callback will still be invoked. This will
cause the reference count to be incorrectly decremented twice, once
by the registration callback and second by the caller of
pjsip_regc_send.
This change makes it so that whether the callback is invoked or
not is known by the caller of pjsip_regc_send. Depending on
this it can know whether it is responsible for decrementing the
reference count of the client state or not.
ASTERISK-25037 #close
Reported by: Joshua Colp
Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de
A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so
a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits
per second. This reverts all or some of those patches since according to the
v.27ter standard a rate of 2400 bits per second is also supported.
One of the original patches also added 9600 bits per second support for v.27.
This patch also removes that since v.27ter only supports 2400/4800 bits per
second.
Also, since Asterisk specifically supports v.27ter the enum was renamed to
better reflect this.
ASTERISK-24955 #close
Reported by: Matt Jordan
Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733
Odd behaviors have been observed during outbound registrations. The most
common problem witnessed has been one where a request with
authentication credentials cannot be created after receiving a 401
response. Other behaviors include apparently processing an incorrect SIP
response.
Inspecting the code led to an apparent issue with regards to how we
handle transactions in outbound registration code. When a response to a
REGISTER arrives, we save a pointer to the transaction and then push a
task onto the registration serializer. Between the time that we save the
pointer and push the task, it's possible for the transaction to be
destroyed due to a timeout. It's also possible for the address to be
reused by the transaction layer for a new transaction.
To allow for authentication of a REGISTER request to be authenticated
after the transaction has timed out, we now hold a reference to the
original REGISTER request instead of the transaction. The function for
creating a request with authentication has been altered to take the
original request instead of the transaction where the original request
was sent.
ASTERISK-25020
Reported by Mark Michelson
Change-Id: I756c19ab05ada5d0503175db9676acf87c686d0a
ARI modules that are generated by 'make ari-stubs' are all dependent on
res_ari_model. Additionally some of the same modules depend on one or more
res_stasis_* modules.
ASTERISK-25027 #close
Reported by: Corey Farrell
Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153
This switches files used to generate other sources to use the new
ASTERISK_REGISTER_FILE macro.
ASTERISK-25026 #close
Reported by: Corey Farrell
Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740
Sections Exist in pjsip.conf
This patch modifies the current loading strategy of the pjsip configuration. If
duplicate sections (e.g. sections containing the same [id/type]) are defined in
[pjsip.conf], the loader will consider the configuration for the given type as
invalid when the duplicate section is encountered. The entire configuration
(including what was previously loaded) for the duplicate [id/type] sections
will be rejected and destroyed, an error message is logged and the load
processing for the given stops.
ASTERISK-24996
Reported By: Ashley Sanders
Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef
When problems occur regarding outbound registrations, it currently
is difficult to debug. Most off-nominal paths had warning messages,
but sometimes we want to know what's going on before hitting the
off-nominal path. This patch adds lots of debugging output that
should give a clearer picture of what is happening with regards
to outbound registrations.
ASTERISK-25020
Reported by Mark Michelson
Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45
ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created.
The resources are linked into a table, but the original alloc refs
are never released. ast_strdup leak in rtp_engine.c. If
ast_rtp_dtls_cfg_copy() is called twice on the same destination struct,
a pointer to an alloc'd string is overwritten before the string is free'd.
ASTERISK-25022
Reported by: one47
Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b
* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
Permanent contacts that hadn't been qualified yet were missing
their contact_status entries causing SEGVs when running CLI
commands.
This patch makes sure that contact_statuses are created for
both dynamic and permanent contacts when they are created.
It also adds checks in the CLI code to make sure there's a
contact_status, just in case.
ASTERISK-25018 #close
Reported-by: Ivan Poddubny
Tested-by: Ivan Poddubny
Tested-by: George Joseph
Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029
The way PJSIP generates an authenticated request is to use a previous
request as a template. This means that the authenticated request will
have the same Call-ID, From header (including tag), and CSeq as the
original request. PJSIP generates a new branch on the Via header to
indicate that this is a new transaction, though.
There are some SIP implementations, though, that do not notice the
change in the branch and therefore will match the authed request to the
original request's transaction. Since the CSeq is the same, the server
will repeat the response it sent to the original request.
This patch aids interoperability by increasing the CSeq of the authed
request by one.
ASTERISK-24845 #close
Reported by: Carl Fortin
Tested by: Carl Fortin
Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01
clang can warn about a so called tautological-compare, when it finds
comparisons which are logically always true, and are therefor deemed
unnecessary.
Exanple:
unsigned int x = 4;
if (x > 0) // x is always going to be bigger than 0
Enum Case:
Each enumeration is its own type. Enums are an integer type but they
do not have to be *signed*. C leaves it up to the compiler as an
implementation option what to consider the integer type of a particu-
lar enumeration is. Gcc treats an enum without negative values as
an int while clang treats this enum as an unsigned int.
rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
The cast does have an effect. For gcc, which seems to treat all enums
as int, the cast to unsigned int will eliminate the possibility of
negative values being allowed. For clang, which seems to treat enums
without any negative members as unsigned int, the cast will have no
effect. If for some reason in the future a negative value is ever
added to the enum the assert will still catch the negative value.
ASTERISK-24917
Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62
When Asterisk originates a channel to an application, the channel is
hung up once the application finishes executing. When the application
in question is SendFax, the Asterisk PJSIP code will attempt to reinvite
the T.38 session to audio after the FAX completes. The hangup of the
channel happens in the midst of this reinvite transaction. In most
circumstances, this works out okay because the BYE is delayed until the
reinvite transaction can complete.
However, if the reinvite that Asterisk sends receives a 401/407
response, then Asterisk's attempt to re-send the reinvite with
authentication will fail. This is because the session supplement in
res_pjsip_t38 makes the assumption that the channel on the session will
always be non-NULL. Since the channel has been hung up, though, the
channel is now NULL. Attempting to operate on the channel causes a
crash.
This patch fixes the issue by ensuring that the channel on the session
is not NULL before attempting to mess with the T.38 framehook.
This patch also contains some corrections for comments that were
incorrect and really confused me when I first started looking at the
code.
ASTERISK-25004 #close
Reported by Mark Michelson
Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0
Currently we use pjsip_parse_hdr to validate contact uris but it
appears that it allows uris without a scheme if there's a port
supplied. I.E myexample.com will fail but myexample.com:5060 will
pass even though it has no scheme. This causes SEGVs later on
whenever the uri is used.
To prevent this, permanent_contact_validate has been updated to check
that the scheme is either 'sip' or 'sips'.
2 uses of possibly-null endpoint have also been fixed in
create_out_of_dialog_request.
ASTERISK-24999
Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2
Reported-by: Brad Latus
clang can warn about a so called tautological-compare, when it finds
comparisons which are logically always true, and are therefor deemed
unnecessary.
Exanple:
unsigned int x = 4;
if (x > 0) // x is always going to be bigger than 0
Enum Case:
Each enumeration is its own type. Enums are an integer type but they
do not have to be *signed*. C leaves it up to the compiler as an
implementation option what to consider the integer type of a particu-
lar enumeration is. Gcc treats an enum without negative values as
an int while clang treats this enum as an unsigned int.
rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
The cast does have an effect. For gcc, which seems to treat all enums
as int, the cast to unsigned int will eliminate the possibility of
negative values being allowed. For clang, which seems to treat enums
without any negative members as unsigned int, the cast will have no
effect. If for some reason in the future a negative value is ever
added to the enum the assert will still catch the negative value.
ASTERISK-24917
Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a
On some systems, res_corosync isn't compatible with the installed version of
corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE,
and Asterisk terminates. The work around has been to remember to add
res_corosync as a noload in modules.conf. A better solution though is to have
res_corosync check for its config file before attempting to call corosync apis
and return LOAD_DECLINE if there's no config file. This lets Asterisk loading
continue.
If you have a res_corosync.conf file and res_corosync fails, you get the same
behavior as today and the fatal error tells you something is wrong with the
install.
ASTERISK-24998
Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889
Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon
a mailbox state change (such as a new message being left, or one being deleted).
In practice this is not sufficient to keep clients aware of the current MWI status.
This change makes the module send unsolicited MWI NOTIFY on startup so that
clients are guaranteed to have the most up to date MWI information. It also makes
clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware
of the current MWI status they receive it.
ASTERISK-24982 #close
Reported by: Joshua Colp
Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58
When SUBSCRIBE dialogs were established, we never associated
the endpoint that created the subscription with the dialog
we end up creating. In most cases, this ended up not causing
any problems.
The actual bug that was observed was that when a device that
was behind NAT established a subscription with Asterisk, Asterisk
would end up sending in-dialog NOTIFY requests to the device's
private IP addres instead of the public address of the NAT router.
When Asterisk receives the initial SUBSCRIBE from the device,
res_pjsip_nat rewrites the contact to the public address on which the
SUBSCRIBE was received. This allows for the dialog to have its target
address set to the proper public address. Asterisk then would send a 200
OK response to the SUBSCRIBE, then a NOTIFY with the initial
subscription state. The device would then send a 200 OK response to
Asterisk's NOTIFY.
Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat
did not rewrite the address in the Contact header. Then, when the PJSIP
dialog layer processed the 200 OK, PJSIP would perform a comparison
between the IP address in the Contact header and its saved target
address for the dialog. Since they differed, PJSIP would update the
target dialog address to be the address in the Contact header. From this
point, if Asterisk needed to send a NOTIFY to the device, the result was
that the NOTIFY would be sent to the private address that the device
placed in the Contact header.
The reason why res_pjsip_nat did not rewrite the address when it
received the 200 OK response was that it could not associate the
incoming response with a configured endpoint. This is because on a
response, the only way to associate the response to an endpoint is by
finding the dialog that the response is associated with and then finding
the endpoint that is associated with that dialog. We do not perform
endpoint lookups on responses. res_pjsip_pubsub skipped the step of
associating the endpoint with the dialog we created, so res_pjsip_nat
could not find the associated endpoint and therefore couldn't rewrite
the contact.
This commit message is like 50x longer than the actual fix.
ASTERISK 24981 #close
Reported by Mark Michelson
Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd
Contact status rtt is an int64_t and needs the PRId64 macro to
properly create the format specifier on 32-bit systems.
Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7
The "Add qualify_timeout processing and eventing" patch introduced
an issue where contacts that had qualify_frequency set to 0 were
showing Unavailable instead Unknown. This patch checks for
qualify_frequency=0 and create an "Unknown" contact_status
with an RTT = 0.
Previously, the lack of contact_status implied Unknown but since
we're now changing endpoint state based on contact_status, I've
had to add new UNKNOWN status so that changes could trigger the
appropriate contact_status observers.
ASTERISK-24977: #close
Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
Three fax related tests started failing as a result of changes made for
ASTERISK-24841:
tests/fax/pjsip/gateway_t38_g711
tests/fax/sip/gateway_mix1
tests/fax/sip/gateway_mix3
Historically, ast_channel_make_compatible() did nothing if the channels
were already "compatible" even if they had a sub-optimal translation path
already setup. With the changes from ASTERISK-24841 this is no longer
true in order to allow the best translation paths to always be picked. In
res_fax.c:fax_gateway_framehook() code manually setup the channels to go
through slin and then called ast_channel_make_compatible(). With the
previous version of ast_channel_make_compatible() this was always a
no-operation.
* Remove call to ast_channel_make_compatible() in fax_gateway_framehook()
that now undoes what was just setup when the framehook is attached.
* Fixed locking around saving the channel formats in
fax_gateway_framehook() to ensure that the formats that are saved are
consistent.
* Fix copy pasta errors in fax_gateway_framehook() that confuses read and
write when dealing with saved channel formats.
ASTERISK-24841
Reported by: Matt Jordan
Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d
A potential problem that can arise is the following:
* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.
If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.
Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.
The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:
* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.
This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:
* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.
The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.
Address review feedback on gerrit.
* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c
ASTERISK-24958 #close
Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
* changes:
res_pjsip: Add global option to limit the maximum time for initial qualifies
pjsip_options: Add qualify_timeout processing and eventing
res_pjsip: Refactor endpt_send_request to include transaction timeout
Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
This change makes the send_notify of the sub_tree
not happen when the sub_tree has been deleted due
to the notify call failing, which avoids a crash.
ASTERISK-24970 #close
Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf
This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
Since we currently have no control over pjproject transaction timeout, this
patch pulls the pjsip_endpt_send_request function out of pjproject and into
res_pjsip/endpt_send_transaction in order to implement that capability.
Now when the transaction is initiated, we also schedule our own pj_timer with
our own desired timeout.
If the transaction completes before either timeout, pjproject cancels its timer,
and calls our tsx callback where we cancel our timer and run the app callback.
If the pjproject timer times out first, pjproject calls our tsx callback where
we cancel our timer and run the app callback.
If our timer times out first, we terminate the transaction which causes
pjproject to cancel its timer and call our tsx callback where we run the app
callback.
Regardless of the scenario, pjproject is calling the tsx callback inside the
group_lock and there are checks in the callback to make sure it doesn't run
twice.
As part of this patch ast_sip_send_out_of_dialog_request was created to replace
its similarly named private function. It takes a new timeout argument in
milliseconds (<= 0 to disable the timeout).
ASTERISK-24863 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
This change adds the following:
1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked.
2. Unit tests for the query set implementation.
3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups.
For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV
are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A,
with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit
transport has been provided. Configured transports on the system are taken into account to
eliminate resolved addresses which have no hope of completing.
ASTERISK-24947 #close
Reported by: Joshua Colp
Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e
This new macro allows a single line to add all additional
sources to a module. This helps prevent modules from
missing steps, and makes future changes easier since
they can be made in a single place.
ASTERISK-24960 #close
Reported by: Corey Farrell
Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can add local ignores to the .git/info/exclude file
without having to do a commit.
Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.
Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph
Prior to this patch, the far_max_datagram value on the UDPTL structure would
remain -1 if the remote endpoint fails to provide the SDP media attribute
T38FaxMaxDatagram. This can result in the INVITE request being rejected. With
this patch, we will now properly initialize the value with either the default
value or with the value provided by pjsip.conf's t38_udptl_maxdatagram
parameter.
Review: https://reviewboard.asterisk.org/r/4589
ASTERISK-24928 #close
Reported by: Juergen Spies
Tested by: Juergen Spies
patches:
pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698)
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With this patch, chan_pjsip/res_pjsip now sets the native formats to the
codecs negotiated by a call.
* The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
formats to include all the negotiated audio codecs instead of only the
initial preferred audio codec and later the currently received audio
codec.
* The audio frame handling in channel.c:ast_read() is more streamlined and
will automatically adjust to changes in received frame formats. The new
policy is to remove translation and pass the new frame format to the
receiver except if the translation was to a signed linear format. A more
long winded version is commented in ast_read() along with some caveats.
* The audio frame handling in channel.c:ast_write() is more streamlined
and will automatically adjust any needed translation to changes in the
frame formats sent. Frame formats sent can change for many reasons such
as a recording is being played back or the bridged peer changed the format
it sends. Since it is a normal expectation that sent formats can change,
the codec mismatch warning message is demoted to a debug message.
* Removed the short circuit check in
channel.c:ast_channel_make_compatible_helper(). Two party bridges need to
make channels compatible with each other. However, transfers and moving
channels among bridges can result in otherwise compatible channels having
sub-optimal translation paths if the make compatible check is short
circuited. A result of forcing the reevaluation of channel compatibility
is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
options take effect consistently now. It is unfortunate that these two
options are enabled by default and negate some of the benefits to the
changes in channel.c:ast_read() by forcing translation through signed
linear on a two party bridge.
* Improved the softmix bridge technology to better control the translation
of frames to the bridge. All of the incoming translation is now normally
handled by ast_read() instead of splitting any translation steps between
ast_read() and the slin factory. If any frame comes in with an unexpected
format then the translation path in ast_read() is updated for the next
frame and the slin factory handles the current frame translation.
This is the final patch in a series of patches aimed at improving
translation path choices. The other patches are on the following reviews:
https://reviewboard.asterisk.org/r/4600/https://reviewboard.asterisk.org/r/4605/
ASTERISK-24841 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4609/
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This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.
Review: https://reviewboard.asterisk.org/r/4438
ASTERISK-24706 #close
Reported by: yaron nahum
patches:
yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
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While investigating other unload issues I realized that the load/unload process
for the config wizard was pretty ugly so I've refactored it as follows...
When the res_pjsip sorcery instance is created the config_wizard bumps it's own
module reference to prevent it from unloading while the sorcery instance is
still active. When res_pjsip unloads and it's sorcery instance is destroyed,
the config wizard unrefs itself which then allows itself to unload cleanly.
Since the config wizard now can't load after res_pjsip or unload before it
(which should have been the correct behavior all along), I was able to remove
the chunks of code in both load_module and unload_module that handled that case.
Ran the testsuite tests to insure there were no functional changes and REF_DEBUG
to insure that Asterisk was shutting down cleanly with no FRACKs or leaks.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4610/
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When the ChannelHold event was added, the 'musicclass' parameter was
erroneously removed. This caused the ChannelHold events to be rejected as
they failed model validation. This patch updates the Swagger schema such that
it now properly reflects the event that is being created.
Hooray for tests that catch things like this.
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res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to
a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the
OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator.
This plugged the leak but exposed an unload order issue between
res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip.
res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip.
Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it
unloads, it's objects are still in the sorcery instance. When res_pjsip
unloads, it destroys all its objects including res_pjsip_phoneprov_provider's.
The phoneprov destructor then attempts to unregister the extension from
res_phoneprov but because res_phoneprov is already cleaned up, its users
container is gone and we get a FRACK.
Simple solution, check for the NULL users container before attempting to remove
the entry. Duh.
Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in
res_pjsip_phoneprov_provider and no FRACKs.
Reported-by: Corey Farrell
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4608/
ASTERISK-24935 #close
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This patch adds a new session supplement that handles in-dialog OPTIONS
requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
for the OPTIONS request would already have been done by the time the
session supplement receives the inbound request.
ASTERISK-24862 #close
Reported by: yaron nahum
patches:
res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)
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This fixes autological comparison warnings in the following:
* chan_skinny: letohl may return a signed or unsigned value, depending on the
macro chosen
* func_curl: Provide a specific cast to CURLoption to prevent mismatch
* cel: Fix enum comparisons where the enum can never be negative
* enum: Fix comparison of return result of dn_expand, which returns a signed
int value
* event: Fix enum comparisons where the enum can never be negative
* indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
negative
* presencestate: Use the actual enum value for INVALID state
* security_events: Fix enum comparisons where the enum can never be negative
* udptl: Don't bother to check if the return value from encode_length is less
than 0, as it returns an unsigned int
* translate: Since the parameters are unsigned int, don't bother checking
to see if they are negative. The cast to unsigned int would already blow
past the matrix bounds.
* res_pjsip_exten_state: Use a temporary value to cache the return of
ast_hint_presence_state
* res_stasis_playback: Fix enum comparisons where the enum can never be
negative
* res_stasis_recording: Add an enum value for the case where the recording
operation is in error; fix enum comparisons
* resource_bridges: Use enum value as opposed to -1
* resource_channels: Use enum value as opposed to -1
Review: https://reviewboard.asterisk.org/r/4533
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4533.patch submitted by dkdegroot (License 6600)
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Without this patch, if a PJSIP endpoint with udptl enabled and authentication
set attempted to use sendFax, the FAX session would fail during setup. This
was because the invite issued in response to being auth challenged would cause
the PJSIP channel performing the FAX to receive a second T38 framehook and
this would cause frames to be consumed in an inappropriate manner.
ASTERISK-24933 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4577/
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This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
evaluate to 'true'. This patch changes the evaluation to use
ast_strlen_zero.
* app_queue:
- Fixed evaluation of qe->parent->monfmt, which always evaluates to
true. Instead, we just check to see if the dereferenced pointer
evaluates to true.
- Fixed evaluation of mem->state_interface, wrapping it with a call to
ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
Review: https://reviewboard.asterisk.org/r/4541
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4541.patch submitted by dkdegroot (License 6600)
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For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.
One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.
In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.
Review: https://reviewboard.asterisk.org/r/4549/
ASTERISK-24922 #close
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When setting the configuration option 'timers' equal to 'no' the bit flag was
not properly negated. This patch clears all associated flags and only sets the
specified one. pjsip will handle any necessary flag combinations. Also went
ahead and did similar for the '100rel' option.
ASTERISK-24910 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/4582/
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This is a change to align behavior with that of Asterisk 11 and previous versions.
In those versions, if a parked call were retrieved, and the call ended, the parked
call retriever would be hung up after the ParkedCall application ran. Prior to this
patch, in Asterisk 13, the same situation would result in the parked call retriever
falling through to additional priorities in the extension where the ParkedCall
application was called. With this patch, the behavior between Asterisk 11 and 13
aligns.
ASTERISK-24899 #close
Reported by Malcolm Davenport
Patches:
ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
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Outbound SIP MESSAGEs had the potential to be sent out
of order from how they were specified in a set of
dialplan steps.
This change creates a serializer for sending outbound
MESSAGE requests on. This ensures that the MESSAGEs are
sent by Asterisk in the same order that they were sent
from the dialplan.
ASTERISK-24937 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4579
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This change adds support for parsing SRV records and consuming their values
in an easy fashion. It also adds automatic sorting of SRV records according
to RFC 2782.
Tests have also been included which cover parsing, sorting, and off-nominal
cases where the record is corrupted.
ASTERISK-24931 #close
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/4528/
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Richard Mudgett
Joshua Colp
Matt Jordan
Mark Michelson
Kevin Harwell
Ivan Poddubny
David M. Lee
George Joseph
Joerg Sonnenberger
Corey Farrell
Rodrigo Ramírez Norambuena
demon-ru
John Bigelow
Yousf Ateya
Sean Bright
Martin Tomec
Ashley Sanders
Steve Davies
Diederik de Groot
Scott Griepentrog
Jonathan Rose