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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
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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
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into 13
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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
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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
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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
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Find and unlink the specified sorcery object type to complement
ast_sorcery_object_register(). Without this function you cannot
completely unload individual modules that use sorcery for configuration.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88
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* The AMI version is bumped to 2.8.0.
ASTERISK-25189 #close
Reported by: John Hardin
Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac
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Added checks when a unit test is registered to see that the summary and
description strings do not end with a new-line '\n' for consistency.
The check generates a warning message and will cause the
/main/test/registrations unit test to fail.
* Updated struct ast_test_info member doxygen comments.
Change-Id: I295909b6bc013ed9b6882e85c05287082497534d
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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
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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
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Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976
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Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
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So this issue is a bit complicated. Since it is possible to pass values to AMI
that contain a '\r\n' (or other similar sequences) these values need to be
escaped. One way to solve this is to escape the values and then pass the escaped
values to the AMI variable parameter string building function. However, this
puts the onus on the pre-build function to escape all string values. This
potentially requires a fair amount of changes along with a lot of string
allocations/freeing for all values.
Surely there is a way to push this complexity down a level into the string
building function itself? This of course is possible, but ends up requiring a
way to distinguish between strings that need to be escaped and those that don't.
The best way to handle this is by introducing a new format specifier in the
format string. For instance a %s (no escape) and %S (escape). However, that is
a bit weird and unexpected.
So faced with those possibilities this patch implements a limited version of the
first option. Instead of attempting to escape all string values this patch only
escapes those values that make sense. This approach limits the number of changes
and doesn't suffer from the odd format specifier problem.
ASTERISK-24934 #close
Reported by: warren smith
Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0
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The length of frames retured by sample functions was twice as large as
real, what caused global buffer overflow caught by AddressSanitizer.
ASTERISK-24717 #close
Reported by: Badalian Vyacheslav
Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6
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This reverts commit 35c699086ae2fd81b2473307ccb2ae79ad32375a.
Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7
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When an endpoint was created, it's messages were being forwarded to
both the tech endpoint topic and the all endpoints topic. Since
the tech topic was also forwarded to all, this was resulting in
duplicate messages whenever an endpoint published. This patch
causes the endpoint to only forward to the tech topic and lets
the tech topic forward to all.
To accomplish this, the existing stasis_cp_single_create function
(which both creates and forwards) was cloned and split into 2
functions, one that creates the topic and one that sets up the
forwarding. This allows endpoint_internal_create to create
the topic from the endpoint_all cache without forwarding it there,
then allows it to do the forward to the tech's topic.
ASTERISK-25137 #close
Reported-by: Vitezslav Novy
ASTERISK-25116 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c
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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>
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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
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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>
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resets" into 13
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Currently, everytime a sample rate change occurs (on read or write) the
associated factory buffers are reset. If the requested sample rate on a
read differed from that of a write then the buffers are continually reset
on every read and write. This has the side effect of emptying the buffer,
thus there being no data to read and then write to a file in the case of
call recording.
This patch fixes it so that an audiohook_list's rate always maintains the
maximum sample rate among hooks and formats. Audiohook sample rates are
only overwritten by this value when slin native compatibility is turned on.
Also, the audiohook sample rate can only overwrite the list's sample rate
when its rate is greater than that of the list or if compatibility is
turned off. This keeps the rate from constantly switching/resetting.
ASTERISK-24944 #close
Reported by: Ronald Raikes
Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f
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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
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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
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Currently you can 'apply' a wizard to an object type but the wizard
always goes at the end of the object type's wizard list. This patch
adds a new ast_sorcery_insert_wizard_mapping function that allows
you to insert a wizard anyplace in the list. I.E. You could
add a caching wizard to an object type and place it before all
wizards.
ast_sorcery_get_wizard_mapping_count and
ast_sorcery_get_wizard_mapping were added to allow examination
of the mapping list.
ast_sorcery_remove_mapping was added to remove a mapping by name.
As part of this patch, the object type's wizard list was converted
from an ao2_container to an AST_VECTOR_RW.
A new test was added to test_sorcery for this capability.
ASTERISK-25044 #close
Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57
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Based on feedback from Corey Farrell and Y Ateya, a few new
macros have been added...
AST_VECTOR_REMOVE which takes a parameter to indicate if
order should be preserved.
AST_VECTOR_ADD_SORTED which adds an element to
a sorted vector.
AST_VECTOR_RESET which cleans all elements from the vector
leaving the storage intact.
Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14
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termination" into 13
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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
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After using the new vector stuff for real I found...
A bug in AST_VECTOR_INSERT_AT that could cause a seg fault.
The callbacks needed to be closer to ao2_callback in behavior
WRT to CMP_MATCH and CMP_STOP behavior and the ability to return
a vector of matched entries.
A pre-existing issue with APPEND and REPLACE was also fixed.
I also added a new macro to test.h that acts like ast_test_validate
but also accepts a return code variable and a cleanup label. As well
as printing the error, it sets the rc variable to AST_TEST_FAIL and
does a goto to the specified label on error. I had a local version
of this in test_vector so I just moved it.
ASTERISK-25045
Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc
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Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really
does replace not insert. The few users of AST_VECTOR_INSERT were
refactored. Because these are macros, there should be no ABI
compatibility issues.
Added AST_VECTOR_INSERT_AT that actually inserts an element into the
vector at a specific index pushing existing elements to the right.
Added AST_VECTOR_GET_CMP that can retrieve from the vector based
on a user-provided compare function.
Added AST_VECTOR_CALLBACK function that will execute a function
for each element in the vector. Similar to ao2_callback and
ao2_callback_data functions although the vector callback can take
a variable number of arguments. This should allow easy migration
to a vector where a container might be too heavy.
Added read/write locked vector and lock manipulation macros.
Added unit tests.
ASTERISK-25045 #close
Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0
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A few cases exist where headers of optional_api provders are included but
not needed. This causes unneeded calls to ast_optional_api_use.
* Don't include optional_api.h from sip_api.h.
* Move 'struct ast_channel_monitor' to channel.h.
* Don't include monitor.h from chan_sip.c, channel.c or features.c.
The move of struct ast_channel_monitor is needed since channel.c depends on
it. This has no effect on users of monitor.h since channel.h is included
from monitor.h.
ASTERISK-25051 #close
Reported by: Corey Farrell
Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478
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Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3
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Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which
checks compiler requirements for RAII:
gcc: -fnested-functions support
clang: -fblocks (and if required -lBlocksRuntime)
The original check was implemented in configure.ac and now has it's
own file. This function also sets C_COMPILER_FAMILY to either gcc or
clang for use by makefile
Created autoconf/ast_check_strsep_array_bounds.m4 (contains
AST_CHECK_STRSEP_ARRAY_BOUNDS):
which checks if clang is able to handle the optimized strsep & strcmp
functions (linux). If not, the standard libc implementation should be
used instead. Clang + the optimized macro's work with:
strsep(char *, char []), but not with strsepo(char *, char *).
Instead of replacing all the occurences throughout the source code,
not using the optimized macro version seemed easier
See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h':
llvm-comment: Normally, this array-bounds warning are suppressed for
macros, so that unused paths like the one that accesses __s1[3] are
not warned about. But if you preprocess manually, and feed the
result to another instance of clang, it will warn about all the
possible forks of this particular if statement. Instead of switching
of this optimization, another solution would be to run the preproces-
sing step with -frewrite-includes, which should preserve enough
information so that clang should still be able to suppress the diag-
nostic at the compile step later on.
See also "https://llvm.org/bugs/show_bug.cgi?id=20144"
See also "https://llvm.org/bugs/show_bug.cgi?id=11536"
Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning
suppressions:
-Wno-unused-value
-Wno-parentheses-equality
In an earlier review (reviewboard: 4550 and 4554), they were deemed a
nuisace and less than benefitial.
configure.ac:
Added AST_CHECK_RAII() see earlier
Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier
Removed moved content
ASTERISK-24917
Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb
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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
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This is the Asterisk 13 version of a change to master that allows for
registration responses to be processed successfully potentially after
the original transaction has timed out. The main difference between this
and the master change is that the master version has API changes that
are unacceptable for 13. For 13, this is worked around by adding a new
API call that the outbound registration code uses instead.
The following is the text from the master version of this commit:
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 also 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: If1ee5f601be839479a219424f0358a229f358f7c
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into 13
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- When you need to refer to 'variable XXX' outside a block, it needs
to be declared as '__block XXX', otherwise it will not be available with-
in the block, making updating that variable hard to do, and ast_free
lead to issues.
- Removed the #error message
because it creates complications when compiling external projects
against asterisk For example when using a different compiler than the
one used to compile asterisk. The warning/error should be generated
during the configure process not the compilation process
ASTERISK-24917
Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2
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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
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When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be
called as a function. This causes a compile error with raw threadstorage as
it uses NULL for cleanup. This fix uses a macro that provides NULL when
DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);"
with "{};" when DEBUG_THREADLOCALS is enabled.
ASTERISK-24975 #close
Reported by: Ashley Sanders
Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402
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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
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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>
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