Fix redirection to /dev/null of cleanup commands. The '2' was being
interpreted as part of the command instead of part of the redirect.
Change-Id: I2e3a591b165e0288c4b82b9ef475fdfd5392a90a
Can't do anonymous http checkout from Security-testsuite.
Need to use same credentials as the gerrit review checkout.
Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05
As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.
ASTERISK-28046 #close
Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc
This new option can be passed for ./configure or
./tests/CI/buildAsterisk.sh to prevent download/install of binary
modules.
Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES
will result in binary modules being enabled even if the build target is
incompatible with those modules. This includes CI scripts which enable
categories before disabling specific modules.
If more binary modules are offered in the future this will help avoid
accidentally downloading them if unwanted or incompatible. Adding a
binary module will only require creating a new menuselect entry similar
to the existing ones, it will not be necessary to modify the CI scripts.
Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166
If the review to be tested is in a project with restricted access,
we need to use the jenkins user's gerrit https credentials when we
do the checkout or the checkout will fail.
Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f
Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`. This
will cause Asterisk to be compiled with coverage support. It also
initializes 'before' coverage data for all sources. Accept
--tested-only to disable modules which are not run by any test.
Enabling coverage also sets tested-only true by default. To build
everything with coverage enabled use `--coverage --tested-only=0`.
./tests/CI/processCoverage.sh is used to process the coverage and
generate HTML reports.
Fix utils/check_expr2 which failed to compiled with coverage enabled.
Add status output 5 times per stage of astobj2_test_perf to ensure
remote CLI does not timeout when compiled with coverage. Remote CLI
disconnects if no output is received for 60 seconds. When coverage is
enabled it takes about 70 seconds for my laptop to run the stages of
this test, so with the change a message is printed every 14 seconds.
Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222
Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree. Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h". In most cases each option
only applies to one or two files. Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds. For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.
Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename. Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.
The only code change is a slight adjustment to how main/astmm.c is
initialized. Initialization functions always exist so main/asterisk.c
can call them unconditionally. Additionally rename the astmm
initialization functions so they are not exported.
Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027
The --test-command argument has now been split, unit tests now use
`--unittest-command` and the testsuite uses --testsuite-command.
This will make it easier to create a script which run everything by
forwarding the same arguments to all CI scripts.
Change-Id: Ia54aa4848eaffbdf13175fcda40fc0b23080ad71
Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.
If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.
According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.
Also added additional functionality to ast_data_buffer, along with some
testing.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
ASTERISK-27810 #close
Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d
Create tests/CI directory and add files used by Jenkins to
build and test Asterisk.
With this commit, Jenkins will run the Asterisk Unit Tests using
the Jenkinsfile at tests/CI/unittests.jenkinsfile. Bash scripts
to do the actual building and testing are also in the same directory.
Output is placed in tests/CI/output so that directory has been
added to .gitignore.
Change-Id: I9448065465e6de2b878634510ace8fd1ef378608
This fixes build warnings found by GCC 8. In some cases format
truncation is intentional so the warning is just suppressed.
ASTERISK-27824 #close
Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84
Replaces the never used opaque data array.
Updated stream tests to include get/set metadata and
stream clone with metadata.
Added stream metadata dump to "core show channel"
Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863
Added unit tests for the data buffer API. These tests include creating a
data buffer, putting payloads into the buffer, resizing the buffer, and
the nominal case for data buffer usage, which consists of adding
the max number of payloads to the buffer, checking to see if the correct
payloads are present, then adding more payloads and checking again to
see if the previous payloads were replaced or not.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Id5b599aa15a5e61d0ec080f97cd0c57bd07e6f8f
* dahdi_chan_name
* dahdi_chan_name_len
* dahdi_chan_mode
* __manager_event
* dialed_interface_info
Added comment about __progname and environ being needed for FreeBSD to
prevent accidental removal in the future.
Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
I've audited all modules that include any header which includes
asterisk/optional_api.h. All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.
In addition ARI dependency declarations have been reworked. Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.
Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
The ability to add to localized storage cannot be supported by
ast_cli_generator. The only calls to ast_cli_generator should be by
functions that need to proxy the CLI generator, for example 'cli check
permissions' or 'core show help'.
* ast_cli_generatornummatches now retrieves the vector of matches and
reports the number of elements (not including 'best' match).
* test_substitution retrieves and iterates the vector.
Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248
Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.
Change-Id: I0123258eafce324249433a69df15a85cc16e509f
test_pbx used raise without explicitly including signal.h. On Mac for
some reason nothing else includes it.
test_logger checked if an unsigned int was negative. Switch the
variable to 'int' so that error check can be effective.
Change-Id: Ie1db5dd1818ac25cc2ae41b644f848b5865b1362
menuselect detects compiler support for multiple styles of weak
functions. This is a remnant from 2013 when OPTIONAL_API required weak
functions. It is no longer correct for menuselect to switch
dependencies from optional to required based on lack of weak function
support.
Note an issue remains - dependencies should switch from optional to
required based on OPTIONAL_API being enabled or disabled. I don't think
this is possible. menuselect needs to know at startup if OPTIONAL_API
is enabled or disabled, so the only way to fix this is to remove
OPTIONAL_API from menuselect and create a configure option. I've left
the code that switches in place but it's preprocessed out.
Additionally removed:
- WEAKREF variable from Asterisk makeopts.in.
- Related disabled code from test_utils.
- Pointless AC_REVISION call from menuselect/configure.ac.
Change-Id: Ifa702e5f98eb45f338b2f131a93354632a8fb389
On second run the config_hook test was unexpectedly failing to load
test_config.conf because it was still unmodified since the last load.
This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial
loads, only using it when we are tested that a reload of unmodified
files do not initiate the hook.
ASTERISK-25960
Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856
Use temporary variable to prevent multiple evaluations of elem argument.
This resolves a memory leak in res_pjproject startup.
ASTERISK-27317 #close
Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d
* ast_channel_request_stream_topology_change() must not be called with any
channel locks held.
* ast_channel_stream_topology_changed() must be called with only the
passed channel lock held.
ASTERISK-27212
Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691
When the iostream code went in it introduced a conditional that made it so the
hook event was not being raised even if a hook is present. This patch adds a
check to see if a hook is present in astman_append. If so then call into the
send_string function, which in turn raises the even for specified hook.
Also updated the ami hooks unit test, so the test could be automated.
ASTERISK-27200 #close
Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36
* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.
Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31
This adds support for parsing timelen values from config files. This
includes support for all flags which apply to PARSE_INT32. Support for
this parser is added to ACO via the OPT_TIMELEN_T option type.
Fixes an issue where extra characters provided to ast_app_parse_timelen
were ignored, they now cause an error.
Testing is included.
ASTERISK-27117 #close
Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554
* Update SDP unit tests to test negotiating with declined streams.
Generation of declined m= lines created and responded tested.
Change-Id: I5cb99f5010994ab0c7d9cf2d395eca23fab37b98
The SDP offer/answer model requires an answer to an offer before a new SDP
can be processed. This allows our local SDP creation to be deferred until
we know that we need to create an offer or an answer SDP. Once the local
SDP is created it won't change until the SDP negotiation is restarted.
An offer SDP in an initial SIP INVITE can receive more than one answer
SDP. In this case, we need to merge each answer SDP with our original
offer capabilities to get the currently negotiated capabilities. To
satisfy this requirement means that we cannot update our proposed
capabilities until the negotiations are restarted.
Local topology updates from ast_sdp_state_update_local_topology() are
merged together until the next offer SDP is created. These accumulated
updates are then merged with the current negotiated capabilities to create
the new proposed capabilities that the offer SDP is built.
Local topology updates are merged in several passes to attempt to be smart
about how streams from the system are matched with the previously
negotiated stream slots. To allow for T.38 support when merging, type
matching considers audio and image types to be equivalent. First streams
are matched by stream name and type. Then streams are matched by stream
type only. Any remaining unmatched existing streams are declined. Any
new active streams are either backfilled into pre-merge declined slots or
appended onto the end of the merged topology. Any excess new streams
above the maximum supported number of streams are simply discarded.
Remote topology negotiation merges depend if the topology is an offer or
answer. An offer remote topology negotiation dictates the stream slot
ordering and new streams can be added. A remote offer can do anything to
the previously negotiated streams except reduce the number of stream
slots. An answer remote topology negotiation is limited to what our offer
requested. The answer can only decline streams, pick codecs from the
offered list, or indicate the remote's stream hold state.
I had originally kept the RTP instance if the remote offer SDP changed a
stream type between audio and video since they both use RTP. However, I
later removed this support in favor of simply creating a new RTP instance
since the stream's purpose has to be changing anyway. Any RTP packets
from the old stream type might cause mischief for the bridged peer.
* Added ast_sdp_state_restart_negotiations() to restart the SDP
offer/answer negotiations. We will thus know to create a new local SDP
when it is time to create an offer or answer.
* Removed ast_sdp_state_reset(). Save the current topology before
starting T.38. To recover from T.38 simply update the local topology to
the saved topology and restart the SDP negotiations to get the offer SDP
renegotiating the previous configuration.
* Allow initial topology for ast_sdp_state_alloc() to be NULL so an
initial remote offer SDP can dictate the streams we start with. We can
always update the local topology later if it turns out we need to offer
SDP first because the remote chose to defer sending us a SDP.
* Made the ast_sdp_state_alloc() initial topology limit to max_streams,
limit to configured codecs, handle declined streams, and discard
unsupported types.
* Convert struct ast_sdp to ao2 object. Needed to easily save off a
remote SDP to refer to later for various reasons such as generating
declined m= lines in the local SDP.
* Improve converting remote SDP streams to a topology including stream
state. A stream state of AST_STREAM_STATE_REMOVED indicates the stream is
declined/dead.
* Improve merging streams to take into account the stream state.
* Added query for remote hold state.
* Added maximum streams allowed SDP config option.
* Added ability to create new streams as needed. New streams are created
with configured default audio, video, or image codecs depending on stream
type.
* Added global locally_held state along with a per stream local hold
state. Historically, Asterisk only has a global locally held state
because when the we put the remote on hold we do it for all active
streams.
* Added queries for a rejected offer and current SDP negotiation role.
The rejected query allows the using module to know how to respond to a
failed remote SDP set. Should the using module respond with a 488 Not
Acceptable Here or 500 Internal Error to the offer SDP?
* Moved sdp_state_capabilities.connection_address to ast_sdp_state. There
seems no reason to keep it in the sdp_state_capabilities struct since it
was only used by the ast_sdp_state.proposed_capabilities instance.
* Callbacks are now available to allow the using module some customization
of negotiated streams and to complete setting up streams for use. See the
typedef doxygen for each callback for what is allowable and when they are
called.
* Added topology answerer modify callback.
* Added topology pre and post apply callbacks.
* Added topology offerer modify callback.
* Added topology offerer configure callback.
* Had to rework the unit tests because I changed how SDP topologies are
merged. Replaced several unit tests with new negotiation tests.
Change-Id: If07fe6d79fbdce33968a9401d41d908385043a06
* changes:
SDP: Set the remote c= line in RTP instance.
SDP: Add t= line in sdp_create_from_state()
stream: Ignore declined streams for some topology calls.
* Made ast_format_cap_from_stream_topology() not include any formats from
declined streams.
* Made ast_stream_topology_get_first_stream_by_type() ignore declined
streams to return the first active stream of the type.
* Updated unit tests to check these changes have the expected effect.
Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df
This change adds a deferred queue to bridging. If a bridge
technology determines that a frame can not be written and
should be deferred it can indicate back to bridging to do so.
Bridging will then requeue any deferred frames upon a new
channel joining the bridge.
This change has been leveraged for T.38 request negotiate
control frames. Without the deferred queue there is a race
condition between the bridge receiving the T.38 request
negotiate and the second channel joining and being in the
bridge. If the channel is not yet in the bridge then the T.38
negotiation fails.
A unit test has also been added that confirms that a T.38
request negotiate control frame is deferred when no other
channel is in the bridge and that it is requeued when a new
channel joins the bridge.
ASTERISK-26923
Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415
Some of the test names were actually reserved words (true, false,
int, null, string, bool). When the jenkins test results analyzer
does its thing it tries to create a map using the test names as
keys and fails because they're reserved words.
Added "type_" to those test names.
Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b
...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'
This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.
To allow this a new member was added to the ast_test_info
structure named 'explicit_only'. If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.
Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed
Added functions that convert a string to an unsigned integer or unsigned long.
A couple of unit test were also created to test the routines. The reasons for
adding these conversion utilities (and hopefully eventually more) are as
follows:
* Conversion routines are functionally contained with consistent and
better error checking
* The function names offer a better description of what is happening
* It encourages code reuse for easier bug fixing at a single source
* It's simpler to use
* It's unit testable
For instance, currently in a lot of places when converting to an integer or
similar the "sscanf" function is used. When using "sscanf" it may not be
immediately clear what's happening as it lacks semantic naming. Limited error
checking is usually done as well. For example, most of the time a check is done
to make sure the value converted, but does not check for overflows or negative
valued conversions when converting unsigned numbers.
Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the
built in error handling that "strtoul" has. For instance "strtoul" contains
overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly
more complex in its use. And maybe a bit controversial, but it may be ("big if")
potentially slower than "strtoul" in some cases.
Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb
* changes:
SDP: Make process possible multiple fmtp attributes per rtpmap.
SDP: Explicitly stop a RTP instance before destoying it.
SDP: Rework merge_capabilities().
SDP: Update ast_get_topology_from_sdp() to keep RTP map.
George Joseph
Sean Bright
Corey Farrell
Jenkins2
Walter Doekes
Ben Ford
Alexander Traud
Richard Mudgett
Joshua Colp
Kevin Harwell