Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM,
which would unload a module even if it was in use.
* Changed unload mode to proper mode
ASTERISK-25173
Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533
Having a debug message tell us that we attempted to look up an item but
failed is nice in circumstances when it isn't clear if the wizard was
queried correctly or not.
Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7
This patch fixes some bad default value handling in the following
settings:
* The 'message_context' and 'accountcode' settings are not mandatory. As
such, we can allow their stringfield values to be empty.
* The 'media_encryption' setting applies a default value of 'none' to
the setting, which it then can't parse or understand. Since the value
is documented to be 'no', this will now apply that as the default
value.
Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83
The crash fix for ASTERISK-25183 backported some code from master to try
to make sure that a BYE response is processed by the same serializer used
by the BYE request. The identified race condition causing that backport
was the BYE request code had not finished processing after sending the BYE
before the BYE response came in for processing under a different thread.
Unfortunately, there is still a race condition. Now the race condition is
between destroying the call session's serializer in
ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a
reference to the serializer for a BYE response. Even worse, the new race
condition is a design limitation of the taskprocessor implementation that
didn't matter in versions before v12. Back then, taskprocessors were only
destroyed when a module unloaded. Now res_pjsip can destroy them when a
call ends.
However, as noted on the ASTERISK-25183 commit,
session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED.
This is a tad too soon because our BYE request transaction has not
completed yet.
* Split session_end() that is called by session_inv_on_state_changed() to
hold off session destruction until the BYE transaction timeout occurs or a
failed initial INVITE transaction timeout occurs in
session_inv_on_tsx_state_changed().
ASTERISK-25201 #close
Reported by: Matt Jordan
Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961
An http request can be sent to reload an Asterisk module. If the
module can not be reloaded or is not already loaded, an error
response will be returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, based
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be reloaded through http requests
ASTERISK-25173
Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1
An http request can be sent to unload an Asterisk module. If the
module can not be unloaded or is already unloaded, an error response
will be returned.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, depending
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be unloaded through http requests
ASTERISK-25173
Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57
An http request can be sent to load an Asterisk module. If the
module can not be loaded or is loaded already, an error response
will be returned.
The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be loaded through http requests
ASTERISK-25173
Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33
An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on a single module can now be retrieved
ASTERISK-25173
Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463
Again, tests now need to not end with a newline. This patch makes it so
the tests can register again, unit tests will actually pass, and we can
stop wasting time trying to figure out why builds are failing when they
really aren't failing.
Change-Id: Ide519fbeba89f413c733446c5ff7b224fc4ce840
An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending on configuration)
can be run in the terminal to access this new functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on modules can now be retrieved
Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0
Gerrit is complaining of conflicts when trying to create a patch series
of all of the cherry-picked master commits, so I have instead squashed
it all into one commit.
ASTERISK-25067 #close
Reported by: Matt Jordan
Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9
This change fixes a bug where the DTLS timeout timer would be
initialized to 0 if DTLS was not used for an RTP session.
ASTERISK-25103
Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac
This change moves logic for setting up the DTLS SSL contexts to
when the SDP is done being processed instead of when ICE negotiation
completes. It also stops handshakes from being initiated when we
are acting as a server.
Manipulating the SSL context when ICE negotiation has completed
is problematic as the SSL context is not protected and if acting
as a client the remote side may have started DTLS negotiation
already.
The retransmission timeout timer code has also been split up
and simplified some. Both RTP and RTCP now have their own timers
and the points at which the timer is stopped and started is now
more specific. When a packet is sent the timer is started. When
a response is received but before it is processed the timer is
stopped. This provides a guarantee that the timeout is not
occurring while the response is processed.
ASTERISK-22805 #close
ASTERISK-24550 #close
ASTERISK-24651 #close
ASTERISK-24832 #close
ASTERISK-25103 #close
ASTERISK-25127 #close
Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91
Calling t38_change_state() sets the t38 state so it makes little sense to
then check the state right after the call for something else.
* Made the code in t38_interpret_parameters() reject or exit T.38 mode as
intended but not implemented.
Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2
MWI subscriptions can crash or corrupt memory when using the subscription
datastore to access the MWI subscription object because the datastore is
not holding a reference to the object.
* Give the subscription datastore a ref to the MWI subscription object.
It is unfortunate that the ref causes a circular ref chain that must be
explicitly broken to allow the memory to get released. The loop is broken
when the subscription is shutdown and if the subscription setup fails.
ASTERISK-25168 #close
Reported by: Carl Fortin
Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f
When res_pjsip body generator modules were generating XML or XPIDF
response bodies, there was a chance that the generated body would be the
exact size of the supplied buffer. Adding the nul string terminator would
then write beyond the end of the buffer and potentially corrupt memory.
* Fix MALLOC_DEBUG high fence violations caused by adding a nul string
terminator on the end of a buffer for XML or XPIDF response bodies.
* Made calls to pj_xml_print() safer if the XML prolog is requested. Due
to a bug in pjproject, the return value could be -1 _or_
AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
* Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
return value of pj_xml_print() when the supplied buffer is not large
enough.
ASTERISK-25168
Reported by: Carl Fortin
Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de
When a caller calls a FAX number and then hangs up right after the call is
answered then the T.38 re-INVITE automatic reject timer may still be
running after the channel goes away.
* Added session NULL channel checks on the code paths that get executed by
t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
automatic reject timer expires.
ASTERISK-25168
Reported by: Carl Fortin
Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403
All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject and res_pjsip.
* 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_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.
This is a cherry-pick from master.
**** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a
NOTE: session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED.
Unfortunately this is a tad too soon because our BYE request transaction
has not completed yet.
ASTERISK-25183 #close
Reported by: Matt Jordan
Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a
This change makes it so that when accepting a WebSocket
connection the HTTP response is sent as one packet instead of
fragmented. Browsers don't like it when you send it fragmented.
ASTERISK-25103
Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
This fixes so a failure to get a timer file descriptor does not cascade
to closing FD 0.
On error, both res_timing_kqueue and res_timing_timerfd would call the
destructor before setting the file handle. The file handle had been
initialized to 0, causing FD 0 to be closed. This in turn, resulted in
floods of "CLI>" messages and an unusable terminal.
ASTERISK-19277 #close
Reported by: Barry Chern
For the 13 branch, this was already fixed. This patch only ensures that
we do not attempt to close a negative file descriptor.
Change-Id: I147d7e33726c6e5a2751928d56561494f5800350
This prevents a leak of a sorcery object type when realtime sorcery
objects are retrieved by fields or when multiple objects are retrieved.
The extent of this leak is that sorcery object types would be leaked.
These are allocated whenever an object type is registered with sorcery,
meaning that on module shutdown, these objects would be leaked. This
could be problematic if many reloads were performed, but it is not as
severe as if every sorcery object retrieved from realtime were being
leaked.
ASTERISK-25165 #close
Reported by Corey Farrell
Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8
Returns a 'failure' from the module load routine indicates to Asterisk
that it should abort loading completely. This is rarely - in fact,
really, never - a good option. Aborting load of Asterisk from a dynamic
module implies that the core, and the rest of the dynamic modules, don't
matter: we should abandon all processing.
res_corosync is really not that important.
This patch updates the module such that, if it fails to load, it
politely declines (emitting ERROR messages along the way), and allows
Asterisk to continue to function.
Note that this issue was keeping Asterisk unit tests from running on
certain build agents.
Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f
A previous change made the contact only get rewritten if the dialog's
route set was not marked frozen. Unfortunately, while the intent of this
is correct, the dialog's route set actually gets marked as frozen
earlier than expected, especially for UAS dialogs.
Instead, the idea is that the contact needs to not be rewritten if there
is a pre-existing route set on the dialog. This is now accomplished by
checking the dialog's route set list instead of checking if the route
set is frozen.
Doing this causes some broken tests to begin passing again.
ASTERISK-25196
Reported by Mark Michelson
Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e
The client_state objects contain a serializer used to send the outbound
REGISTER messages. Once all those message transactions are complete then
the module can shutdown.
ASTERISK-24907 #close
Reported by: Kevin Harwell
Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547
res_pjsip_refer will attempt to add Referred-By or Replaces headers to
outbound INVITEs at times. If the INVITE gets challenged for
authentication, then we will resend the INVITE. Prior to this patch, the
Referred-By or Replaces header would be re-added to the outbound INVITE,
resulting in duplicated headers.
ASTERISK-25204 #close
Reported by Mark Michelson
Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d
When performing some provider testing, the rewrite_contact option was
interfering with proper construction of a route set when sending an ACK
after receiving a 200 OK response to an INVITE.
The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
header with URI sip:bar. In addition, the 200 OK had Record-Route
headers for sip:baz and sip:foo, in that order. Since the Record-Route
headers had the lr parameter, the result should have been:
* Set R-URI of the ACK to sip:bar.
* Add Route headers for sip:foo and sip:baz, in that order.
However, the rewrite_contact option resulted in our rewriting the
Contact header on the 200 OK to sip:foo. The result was:
* R-URI remained sip:foo.
* We added Route headers for sip:foo and sip:baz, in that order.
The result was that sip:bar was not indicated in the ACK at all, so the
far end never received our ACK. The call eventually dropped.
The intention of rewrite_contact is to rewrite the most immediate
destination of our SIP request to be the same address on which we
received a request or response. In the case of processing a SIP response
with Record-Route headers, this means that instead of rewriting the
Contact header, we should instead rewrite the bottom-most Record-Route
header. In the case of processing a SIP request with Record-Route
headers, this means we rewrite the top-most Record-route header.
Like when we rewrite the Contact header, we also ensure to update
the dialog's route set if it exists.
ASTERISK-25196 #close
Reported by Mark Michelson
Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
A module trying to unload needs to wait for all serializers it creates and
uses to complete processing before unloading.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059
* handle_client_state_destruction() must always be passed a ref to
client_state because it will always unref client_state.
handle_registration_response() was not passing a client_state ref.
* Made the final un-REGISTER message get sent normally using the pjproject
register control structure in handle_client_state_destruction(). The
previous code attempted to short circuit the response handling for the
module to unload. That doesn't work for a couple reasons. One,
pjsip_regc_send() may call the registered callback before it returns and
unbalance the client_state ref count. Two, the registered callback
handles any authentication for the un-REGISTER message.
* Made the distinction between internal registration state and external
registration status with sip_outbound_registration_status_str(). This is
necessary to avoid altering documented AMI messages with internal
changes.
* Removed references to client_state->client outside of the serializer
thread. When handle_client_state_destruction() destroys the pjproject
register control structure that memory is freed and cannot be referenced
anymore. These accesses were to provide information for debug and
off-nominal warning messages.
* In sip_outbound_registration_timer_cb() you should not access entry->id
after unrefing client_state because the passed in entry is normally
pointing to the timer entry in the client_state object.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
