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.
Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632
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.
ASTERISK-28140
Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621
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.
ASTERISK-28169
Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993
New dialplan function PJSIP_PARSE_URI added to parse an URI and return
a specified part of the URI.
This is useful when need to get part of the URI instead of cutting it
using a CUT function.
For example to get 'user' part of Remote URI
${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)}
ASTERISK-28144 #close
Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a
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.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
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.
Change-Id: If5ee905a15821296c355bb84cda38950ad8edc45
(cherry picked from commit a335f4c9ad)
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.
Change-Id: Idd656085f854afede813ac241b9e312b31358160
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.
Change-Id: Id64fae94a6a1e940ddf38fde622dcd4391635382
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.
Change-Id: Iee8a3613cdb711fa7e7d217c5a775a575907ae22
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.
Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f
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.
ASTERISK-28129 #close
Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc
We cannot use need_el_end and SIGURG when restarting. Instead we need
to run el_end within the SIGHUP restartnow handler.
ASTERISK-28158
Change-Id: Ia852276363c81bdcf1aa29eb4558c5c2fa1218a0
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)
Change-Id: I4d35d6c431c6c6836cb61d37b1378cc47f0b414d
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.
Change-Id: I057e2fcbe97f8a175e50890be89c28c20490a20f
Merge storage for the stats object and name string into the main
allocation for struct ast_taskprocessor.
Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1
* Ignore console=yes configuration option in remote console processes.
* Use new flag to tell consolethread to run el_end and exit when needed.
ASTERISK-28158
Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438
pack_string crashed on non-NULL strings returned when s->has_error was
true if the string was the result of 's' format without '#', '%' or '+'.
Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0
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.
ASTERISK-28146 #close
Change-Id: Iaac810c0a7c4d9b1bf8989fcc041cdb910ef08a0
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.
Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836
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.
This patch replaces the pipe with a comma.
ASTERISK-28150
Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719
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.
Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd
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.
Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a
When a call pickup is performed using and invite with replaces header
the ast_do_pickup method is attempted and a PICKUP stasis message is sent.
ASTERISK-28081 #close
Reported-by: Luit van Drongelen
Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a
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.
ASTERISK-28136
Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d
Joshua C. Colp
George Joseph
Sungtae Kim
Alexei Gradinari
Corey Farrell
Torrey Searle
Robert Cripps
Kevin Harwell
Chris-Savinovich
Jasper Hafkenscheid
Pascal Cadotte Michaud