George Joseph [Mon, 19 Nov 2018 17:59:07 +0000 (10:59 -0700)]
CI: Get job timeouts from environment
The job timeouts were hard coded in the jenkinsfiles which
means changes had to go through gerrit. Now they are taken
from the following environment variables (and their defaults) that
can be set in Jenkins configuration...
This replaces the inline functions with macros. This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.
George Joseph [Thu, 8 Nov 2018 15:53:44 +0000 (08:53 -0700)]
backtrace: Refactor ast_bt_get_symbols so it doesn't crash
We've been seeing crashes in libbfd when we attempt to generate
a stack trace from multiple threads. It turns out that libbfd
is NOT thread-safe. It can cache the bfd structure and give it to
multiple threads without protecting itself. To get around this,
we've added a global mutex around the bfd functions and also have
refactored the use of those functions to be more efficient and
to provide more information about inlined functions.
Also added a few more tests to test_pbx.c. One just calls
ast_assert() and the other calls ast_log_backtrace(). Neither are
run by default.
WARNING: This change necessitated changing the return value of
ast_bt_get_symbols() from an array of strings to a VECTOR of
strings. However, the use of this function outside Asterisk is not
likely.
Joshua C. Colp [Sun, 18 Nov 2018 23:53:14 +0000 (19:53 -0400)]
stasis: Remove stringfields and lock from change message.
When a subscribe or unsubscribe occurs a message is published
containing this information. This change makes it so that the
message no longer uses stringfields or a lock, as both are not
really needed for the message.
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
George Joseph [Sat, 17 Nov 2018 19:07:32 +0000 (12:07 -0700)]
CI: Pass work directory to runTestsuite
The testsuite can now use a user-specified work directory for
all it's temp files. This allows the docker containers to use
a tmpfs backed directory for the temp files instead of it's
own write-layer image.
* runTestsuite.sh now accepts a --work-dir command line argument
that gets exported as AST_WORK_DIR before running the testsuite.
* gates.jenkinsfile now specifies --work-dir to be
<testsuite_dir>/astroot.
Since the Asterisk CI docker hosts now mount /srv/jenkins/workspace
on a tmpfs, asterisk should be compiled and the testsuite run all in
memory.
Sungtae Kim [Sat, 17 Nov 2018 02:33:20 +0000 (03:33 +0100)]
res/res_ari: Fix null endpoint handle
The res_ari(POST /channels/create handler) deos not check the endpoint
parameter length. And it causes core
dump.
Fixed it to check the parameter length. Also fixed memory leak.
George Joseph [Thu, 15 Nov 2018 17:41:44 +0000 (10:41 -0700)]
CI: Allow runUnittests to use 'expect' to run the tests
There seems to be a race condition between starting the asterisk
daemon and attempting to use 'asterisk -r' that can cause the
control socket file to not be created. Since all of the Jenkins
slaves have 'expect' installed, the runUnittests script can use
it to start asterisk in the forground and issue the commands
interactively. This is much more reliable and it can also make
startup errors more visible since they'll be in the Jenkins console
output.
If 'expect' isn't installed, the original daemon/asterisk -r
process is used.
Also added a "core show settings" before running the tests
and added "notice,warning,error" to the console log.
Corey Farrell [Mon, 12 Nov 2018 18:23:34 +0000 (13:23 -0500)]
taskprocessor: Prevent race creating new taskprocessor.
Task processors are retrieved using a 'get or create' pattern. The
singleton container was unlocked between the get and create steps so
it's possible that two threads could create task processors with the
same name at the same time.
Corey Farrell [Fri, 16 Nov 2018 12:20:11 +0000 (07:20 -0500)]
pjproject-bundled: Use AST_DEVMODE for conditional compilation.
We previously allowed resample and g711 codecs to be built when
TEST_FRAMEWORK was enabled. This could cause errors if the testsuite
was run without this option enabled. Switch the build system to allow
those codecs to be built when --enable-dev-mode is used. This removes a
chance for strange testsuite errors from use of an inadequate pjsua
binary.
Corey Farrell [Thu, 15 Nov 2018 20:47:50 +0000 (15:47 -0500)]
res_pjsip_caller_id: Use static pj_str_t for fromto header names.
PJSIP assumes that these header names are not allocated, does not clone
the name strings when reusing headers.
Block unload of res_pjsip_caller_id until shutdown to ensure static
memory stays valid. It was previously unsafe to unload while any
sessions are active.
Torrey Searle [Wed, 24 Oct 2018 12:38:37 +0000 (14:38 +0200)]
res/res_pjsip_nat: Fix logic for REINVITES
The presence of Record-Route in re-invites is optional, thus it is
important to make sure the dialog doesn't have a routset before
rewriting the contact header.
George Joseph [Thu, 25 Oct 2018 15:25:58 +0000 (09:25 -0600)]
AST-2018-010: Fix length of buffer needed for SRV and NAPTR results
When dn_expand was being called on SRV and NAPTR results, the
return value was being used to calculate the size of the buffer
needed to store the host names. Since dn_expand returns the
length of the COMPRESSED name the buffer could be too short
to hold the EXPANDED name. The expanded name is NULL terminated
so using strlen() is the correct way to determine the length
actually needed for the buffer.
ASTERISK-28127
Reported by: Jan Hoffmann
patches:
patch.diff submitted by janhoffmann (license 6986)
Corey Farrell [Tue, 13 Nov 2018 16:51:00 +0000 (11:51 -0500)]
test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy.
It's possible for a 4th task to be spawned before we cancel. This
results in a write to the already freed test_data1. Wait long enough to
verify success of the cancelation before freeing test_data1.
Corey Farrell [Fri, 2 Nov 2018 11:38:19 +0000 (07:38 -0400)]
pbx_config: Only the first [globals] section is seen.
If multiple [globals] sections are used (for example via separate
included files), only the first one is processed. This can result in
lost global variables when using a modular extensions.conf.
res_pjsip: Send a 503 response when overload state if reliable transport.
When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.
Kevin Harwell [Tue, 6 Nov 2018 22:35:30 +0000 (16:35 -0600)]
res_pjsip: formatting error in documentation
The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.
res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue
The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests. Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.
Longer running tasks with the round-robin method can delay processing
tasks.
* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.
Joshua Colp [Mon, 5 Nov 2018 14:30:54 +0000 (14:30 +0000)]
stasis: Clarify lifetime of topics.
As mentioned in the comment I've added in the code there is no
ability to unsubscribe all subscribers from a topic and explicitly
destroy it. This is not currently a problem as we have two types of
topics:
Long lived topics which exist for the lifetime of the system.
Ephemeral topics which feed a long lived topic.
In the case of the ephemeral topics there is no subscriber which does
not have its lifetime managed by the same entity that has created
the topic. This ensures that when the topic is being unreferenced the
subscribers are also unsubscribed and destroyed, allowing the topic
to ultimately be destroyed as well.
contrib/sip_to_pjsip: add a --quiet option to avoid prints
Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.
Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.
Torrey Searle [Tue, 2 Oct 2018 12:31:43 +0000 (14:31 +0200)]
res_pjsip_session: add new flag use_callerid_contact
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
Corey Farrell [Wed, 10 Oct 2018 12:09:15 +0000 (08:09 -0400)]
chan_sip deprecation.
This officially deprecates chan_sip in Asterisk 17+. A warning is
printed upon startup or module load to tell users that they should
consider migrating. chan_sip is still built by default but the default
modules.conf skips loading it at startup.
Very important to note we are not scheduling a time where chan_sip will
be removed. The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.
Richard Mudgett [Thu, 18 Oct 2018 00:34:37 +0000 (19:34 -0500)]
logger.c: Fix default console logging when no logger.conf available.
Default logging was not setup correctly when there was no logger.conf.
This resulted in many expected log messages not actually getting out to
the console.
George Joseph [Tue, 16 Oct 2018 12:02:19 +0000 (06:02 -0600)]
bridge_softmix: Add SDP "label" attribute to streams
Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel. Only bridge_softmix has that
data so now it's set when the bridge topology is changed.
Nick French [Wed, 18 Jul 2018 12:45:26 +0000 (07:45 -0500)]
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
This change implements a few different generic things which were brought
on by Google Voice SIP.
1. The concept of flow transports have been introduced. These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target. These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity). When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.
2. Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.
3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module. If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.
4. Configurable outbound extension support has been added to the outbound
registration module. When set the extension will be placed in the
Supported header.
5. Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.
6. Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.
All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
lock: Replace __ast_mutex_logger with private log_mutex_error.
__ast_mutex_logger used the variable `canlog` without accepting it as a
argument. Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement. This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.
Richard Mudgett [Wed, 17 Oct 2018 21:08:19 +0000 (16:08 -0500)]
res_rtp_asterisk.c: Add conditional module dependency to res_pjproject
* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.
Corey Farrell [Sun, 14 Oct 2018 12:58:59 +0000 (08:58 -0400)]
taskprocessor: Warn on unused result from pushing task.
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
This patch is not in the upstream pjproject and does unsafe things with
the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset()
outside of the timer heap lock. pj_timer_entry_reset() is also called for
timers that are not about to be rescheduled in a few places.
projects / thirdparty / asterisk.git / log
