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The threadpool_auto_increment test fails infrequently for a couple of
reasons
* The threadpool listener was notified of fewer tasks being pushed than
were actually pushed
* The "was_empty" flag was set to an unexpected value.
The problem is that the test pushes three tasks into the threadpool.
Test expects the threadpool to essentially gather those three tasks, and
then distribute those to the threadpool threads. It also expects that as
the tasks are pushed in, the threadpool listener is alerted immediately
that the tasks have been pushed. In reality, a task can be distributed
to the threadpool threads quicker than expected, meaning that the
threadpool has already emptied by the time each subsequent task is
pushed. In addition, the internal threadpool queue can be delayed so
that the threadpool listener is not alerted that a task has been pushed
even after the task has been executed.
From the test's point of view, there's no way to be able to predict
exactly the order that task execution/listener notifications will occur,
and there is no way to know which listener notifications will indicate
that the threadpool was previously empty.
For this reason, the test has been updated to only check the things it
can check. It ensures that all tasks get executed, that the threads go
idle after the tasks are executed, and that the listener is told the
proper number of tasks that were pushed.
Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c
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res_odbc.exports.in was missing a few symbols.
Changed to wildcards.
Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c
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res_statsd.export.in was missing the _va variations of the log
functions causing Asterisk to crash in res_pjsip if OPTIONAL_API
wasn't enabled.
ASTERISK-25727 #close
Reported-by: Gergely Dömsödi
Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b
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If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid
or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify
the header added by the dialplan function. Since the header added by the
dialplan function is generic string, there are no virtual functions to parse
the uri and we get a segfault when we try. Since the modify, was really only
an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER
and recreate it.
This raises a question for another time though: What should happen with
duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups
so if it's session supplement is loaded after res_pjsip_caller_id's (or any
other module that adds headers), there'll be dups in the message.
ASTERISK-25337 #close
Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa
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It is possible when processing a SIP REGISTER request to have two
threads end up creating contact_status structures in sorcery.
contact_status is created using a "find or create" function. If two
threads call into this at the same time, each thread will fail to find
an existing contact_status, and so both will end up creating a new
contact status.
During testing, we would see sporadic failures because the
PJSIP_CONTACT() dialplan function would operate on a different
contact_status than what had been updated by res_pjsip/pjsip_options.
The fix here is two-fold:
1) The "find or create" function for contact_status now has a lock
around the entire operation. This way, if two threads attempt the
operation simultaneously, the first to get there will create the object,
and the second will find the object created by the first thread.
2) res_sorcery_memory has had its create callback updated so that it
will not allow for objects with duplicate IDs to be created.
Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97
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A problem arose when testing the AMI subscription listing actions where it
was possible for a subscription that had not been fully initialized to be
listed. This was problematic as the underlying listing code would crash.
This change makes it so the subscription tree is fully set up before it is
added to the list of subscriptions. This ensures that when the listing actions
get the subscription it is valid.
ASTERISK-25738 #close
Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48
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listener() opens the same code block in two places (#if and #else). This
confuses some folding editors causing it to think that an extra code block
was opened. Folding in 'geany' causes all code after listener() to be
folded as if it were part of that procedure.
ASTERISK-24813 #close
Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961
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load_module was just too hairy with every step having to clean up all
previous steps on failure.
Some of the pjproject init calls have now been moved to a separate
load_pjsip function and the unload_pjsip function was enhanced to clean
up everything if an error happened at any stage of the load process.
In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
and ast_threadpool_shutdowns were also corrected.
Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302
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* heap-use-after-free happens when we free "cfg"
but then use "value" which refers to it
* A memory leak occurs because in some cases
it is not released "defaults"
ASTERISK-25721 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469
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ASTERISK-25272 #close
Reported by: Etienne Lessard
patches:
AST-25272.patch submitted by Etienne Lessard (license #6394)
Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17
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realtime" into 13
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Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
increments the lock on the returned dialog. To account for this, configure.ac
now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
the original call or the new one. If the new one was used, the ref count is
decremented before returning.
ASTERISK-25751 #close
Reported-by Josh Colp
Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8
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In older versions of the compiler was not sanitizes.
Compilers other than GCC can not support the Usan and TSAN
or have other options for *FLAGS.
ASTERISK-25767 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916
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USAN can be used together with other sanitizers.
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f
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FD_SET contains a conditional statement to protect against buffer
overruns. The statement was overly complicated and prevented use
of the last array element of ast_fdset. We now just verify the fd
is less than ast_FDMAX.
Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40
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When terminating the threads thrashing a sorcery memory cache each
would be told to stop and then we would wait on them. During at
least one thrashing test this was problematic due to the specific
usage pattern in use. It would take some time for termination of the
thread to occur.
This would occur due to contention between the threads retrieving
and the threads updating the cache. As the retrieving threads are
given priority it may be some time before the updating threads
are able to proceed.
This change makes it so all threads are told to stop and then each
are joined to ensure they stop. This way all the threads should
stop at around the same time instead of waiting for one to stop,
the next to stop, then the next, and so on. As a result of this
the execution time for each thrash test is much closer to their
expected value than previously seen as well.
Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827
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Attempting to load a transport from realtime was forcing asterisk into an
infinite recursion loop. The first thing transport_apply did was to do a
sorcery retrieve by id for an existing transport of the same name. For files,
this just returns the previous object from res_sorcery_config's internal
container, if any. For realtime, the res_sourcery_realtime driver looks in the
database and finds the existing row but now it has to rehydrate it into a
sorcery object which means calling... transport_apply. And so it goes.
The main issue with loading from realtime (apart from the loop) was that
transport stores structures and pointers directly in the ast_sip_transport
structure instead of the separate ast_transport_state structure. This patch
separates those items into the ast_sip_transport_state structure. The pattern
is roughly the same as res_pjsip_outbound_registration.
Although all current usages of ast_sip_transport and ast_sip_transport_state
were modified to use the new ast_sip_get_transport_state API, the original
items are left in ast_sip_transport and kept updated to maintain ABI
compatability for third-party modules. They are marked as deprecated and
noted that they're now in ast_sip_transport_state.
ASTERISK-25606 #close
Reported-by: Martin Moučka
Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19
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A user cannot set new bridge options after the conference is created by
the first user. Attempting to do so is documented as undefined behavior.
This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.
Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
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Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98
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* changes:
app_confbridge: Add ability to get the muted conference state.
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
app_confbridge: Make non-admin users join a muted conference muted.
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The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.
This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.
Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
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ps_systems needed disable_tcp_switch
ps_registrations needed line and endpoint
ASTERISK-25737 #close
Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19
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Device state subscription lifetimes were governed by when the
subscription was established and unsubscribed from. However, it is
possible that at the time of unsubscription, there could be device state
events still in flight. When those device state events occur, the device
state callback could attempt to dereference a freed pointer. Crash.
This change ensures that the lifetime of the device state subscription
does not end until the underlying stasis subscription has confirmed that
its final message has been sent.
Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2
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During ICE negotiation the IPs of the local interfaces are sent to the remote
peer as host candidates. In many cases Asterisk is behind a static one-to-one
NAT, so these host addresses will be internal IP addresses.
To help in hiding the topology of the internal network, this patch adds the
ability to override the host candidates by matching them against a
user-defined list of replacements.
Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f
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overflow." into 13
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Sending UDPTL packets to Asterisk with the right amount of missing
sequence numbers and enough redundant 0-length IFP packets, can make
Asterisk crash.
ASTERISK-25603 #close
Reported by: Walter Doekes
ASTERISK-25742 #close
Reported by: Torrey Searle
Change-Id: I97df8375041be986f3f266ac1946a538023a5255
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This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.
The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.
ASTERISK-24972 #close
Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
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Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times. These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.
NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.
* The overflow is now detected and the previous timeout time is
calculated.
ASTERISK-25397 #close
Reported by: Alexander Traud
Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
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test_dlinklists doesn't need to NOTICE everyone that every macro worked.
res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or
provider was registered.
res_odbc was missing a newline at the end of one message.
Change-Id: I6c06361518ef3711821795e535acd439782a995e
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asterisk's own ip." into 13
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A regression was introduced where searching for realtime PJSIP objects
by regex by starting the regex with a leading "^" would cause no items
to be returned.
This was due to a change which attempted to drop the requirement for a
leading "^" to be present due to how some CLI commands formulate their
regexes. However, the change, rather than simply eliminating the
requirement, caused any regexes that did begin with "^" to end up not
returning the expected results.
This change fixes the problem by inspecting the regex and formulating
the realtime query differently depending on if it begins with "^".
ASTERISK-25702 #close
Reported by Nic Colledge
Patches:
realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691
Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693
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The module res_xmpp does not accept usernames in the form used in component
mode (XEP-0114). In component mode there is no @something in the name.
In component mode the connection is now not dropped anymore.
If the xmpp server sends out a "stream" tag before handshake is finished,
the connection gets dropped in res_xmpp. Now this tag will be ignored and
the connection will be established.
After connecting there will be an exchange of presence states. This does
not work as expected in component mode. The responsible function
"xmpp_pak_presence" is left before the states get sent out. Sending
presence states in component mode is now moved to the top of the function.
ASTERISK-25735 #close
Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c
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Fix some warnings found with clang.
Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd
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The "clean" target was attempting to clean res/ari from inside
the res directory which doesn't remove anything. Removed the res/
prefix.
Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6
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A recent commit set qualify_timeout to Decimal which isn't supported.
This path corrects it to Float.
Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf
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When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
asterisk to include the same value for its own ip in both cases a) and b),
but it seems a) produces a contact header like Contact:
<sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
<sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf
My guess is that manager_sipnotify should call
ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
because after applying this patch, both cases a) and b) produce
the contact header that I expect: <sip:asterisk@192.168.1.227:8060>
Reported by: Stefan Engström
Tested by: Stefan Engström
Change-Id: I86af5e209db64aab82c25417de6c768fb645f476
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