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Error response code descriptions may contain wiki markup that need to be
escaped. Without this patch, Confluence will reject the document being sent
and the responsible script will raise an exception.
Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5
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When ab803ec342 was committed, it accidentally forgot to actually *add* the
HOLD_INTERCEPT function. This highlights two interesting points:
* Gerrit forces you to put the patch as it is going to into the repo up for
review, which Review Board did not. Yay Gerrit.
* No one apparently bothered to use this feature, or else they don't know about
it. I'm going to go with the latter explanation.
ASTERISK-24922
Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396
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This patch adds the functions
ast_cdr_modifier_register()
ast_cdr_modifier_unregister()
That work much like ast_cdr_register() and ast_cdr_unregister().
Modules registered will be given a chance to modify (or to do whatever
they want) CDR fields just before they are passed to registered engines.
Thus, for instance, if a module change the "userfield" field of a CDR,
the modified value will be passed to every registered CDR backend for
logging.
ASTERISK-25479 #close
Change-Id: If11d8fd19ef89b1a66ecacf1201e10fcf86ccd56
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This patch adds some minor tweaks for autosupport to update it for Asterisk 13.
This includes:
* Finally removing most references to Zaptel
* Adding support for some additional 'core' commands, and fixing nomenclature
that generally hasn't been used for some time
* Adding some PJSIP/SIP commands to gather endpoints/peers and active channels
Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1
(cherry picked from commit 9fc9777fa34753fb38991d42d8dbed516e907ca2)
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ASTERISK-25308 fixed a memory leak in res_ari_events.c, but
this file is regenerated by a template and the template was
not fixed.
Change-Id: Ied4c6deae89d21f87f9cf99676b1d055aa83b38b
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* changes:
config.c: Fix off-nominal memory leak.
config.c: Fix potential memory corruption after [section](+).
config.c: Fix #include after [section](+).
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On v13, loading several thousand PJSIP endpoints on Asterisk start causes
a deadlock most of the time.
Thanks to mdu113 for discovering that there was a call to pgsql_exec() not
protected by the pgsql_lock reentrancy lock.
{quote}
I believe a code path exists that attempts to use pgsql connection without
locking pgsql_lock. I believe what happens during that deadlock that I
see is two concurrent threads are both attempting to send query to pgsql,
one of the thread is using a code path without locking pgsql_lock. If
they managed to send queries at the same time, it seems postgres ignores
one of the queries and replies only to the one of them. If it happens so
that the thread holding the lock didn't receive the reply it will wait for
it (and hold the lock) forever (or at least for very long time), thus
completely blocking all access to db.
{quote}
* Added missing reentrancy locking around pgsql_exec() in find_table().
* Moved unlock of pgsql_lock in unload_module() to avoid locking inversion
between the psql_tables list lock and the pgsql_lock.
ASTERISK-25455 #close
Reported by: mdu113
Patches:
res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113
Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2
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To quote Olle:
"When issuing a hangup due to RTP timeouts the cause code is not set. I have
selected 44 based on Cisco's implementation..."
ASTERISK-25135 #close
Reported by: Olle Johansson
patches:
rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267)
Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
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Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0
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The memory corruption could happen if the [section](+) is the last section
in the file with trailing comments. In this case process_text_line() has
left *last_cat is set to newcat and newcat is destroyed.
Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93
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An #include right after a [section](+) would associate any variable
assignments before a new section in the #include with the wrong section.
* Fix section association by setting the current section to the appended
section.
* Fix '+' and '!' section flag interaction corner case depending upon
which flag came first. If the '!' came first then it would be ignored.
If the '!' came after then it would affect the appended section. The '!'
will now no longer be ignored.
ASTERISK-25461 #close
Reported by: Sean Pimental
Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3
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This patch adds menuselect options for building Asterisk with
various sanitizers provided by gcc and clang.
When one of *SANITIZER flags is set in menuselect, the appropriate
option is added to CFLAGS ad LDFLAGS for the build.
Information on sanitizers in the project wiki:
https://github.com/google/sanitizers/wiki
GCC Manual:
https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html
Clang Compiler User's Manual:
http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation
ASTERISK-24718 #close
Reported by: Badalian Vyacheslav
Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0
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Versions of libunbound before v1.4.21 do not compile with Asterisk.
However, since v1.4.21 has a configure script bug that fails to detect the
ldns library (which is fixed in v1.4.22) and v1.4.22 is not an easily
detectable version we will require v1.5.0 as a minimum version of the
library to work with Asterisk.
ASTERISK-25108 #close
Reported by: Richard Mudgett
Change-Id: Ieb228bfb01467573fc121c7356a9dde27128894d
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Wrote the skeleton framework for the Asterisk StatsD dialplan
application. This includes a load function, unload function, a
callback for execution, and XML documentation.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363
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The struct send_request_wrapper has a pjsip lock associated with it that
is created non-recursive. There is a code path for the struct
send_request_wrapper lock that will attempt to lock it recursively. The
reporter's deadlock showed that the thread calling endpt_send_request()
deadlocked itself right after the wrapper object got created.
Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited
MWI NOTIFY messages can hit this deadlock.
* Replaced the struct send_request_wrapper pjsip lock with the mutex lock
that can come with an ao2 object since all of Asterisk's mutexes are
recursive. Benefits include removal of code maintaining the pjsip
non-recursive lock since ao2 objects already know how to maintain their
own lock and the lock will show up in the CLI "core show locks" output.
ASTERISK-25435 #close
Reported by: Dmitriy Serov
Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d
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frame->subclass.frame_ending"
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In ast_rtp_read, the value of the variable 'mark' which we try to assign to a
frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate
it to 0 or 1.
ASTERISK-25451 #close
Change-Id: I53bdf5c026041730184a6a809009c028549ce626
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Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not
exist, i.e. when a new CustomPresence is added in the dialplan.
ASTERISK-25400 #close
Reported by: Andrew Nagy
Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a
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When we decide we will no longer schedule an RTCP write, we remove the
reference to the RTP instance, then assign -1 to the stored scheduler ID
in case something else comes along and wants to see if anything is scheduled.
That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to
fix the regression introduced by 3cf0f29310, this improper assignment on a
potentially destroyed object started getting tripped on the build agents.
Frankly, this should have been crashing a lot more often earlier. I can only
assume that the timing was changed just enough by both changes to start
actually hitting this problem.
As it is, simply moving the assignment prior to the ao2 deference is sufficient
to keep the RTP instance from being referenced when it is very, truly,
aboslutely dead.
(Note that it is still good practice to assign -1 to the scheduler ID when we
know we won't be scheduling it again, as the ao2 deref *may* not always destroy
the ao2 object.)
ASTERISK-25449
Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
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If a Via header containes an IPv6 address and a port number is ommitted,
as it is the standard port, we now leave the port empty and to not set it
to the value after the first colon of the IPv6 address.
ASTERISK-25443 #close
Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70
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Apparently some endpoints attempt to send a reINVITE before completing the
initial INVITE transaction. In this case PJSIP responds appropriately to
the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip
is using the initial INVITE transaction state to determine if an INVITE is
the initial INVITE or a reINVITE. Since the initial INVITE transaction
has not been confirmed yet chan_pjsip thinks the reINVITE is an initial
INVITE and starts another PBX thread on the channel. The extra PBX thread
ensures that hilarity ensues.
* Fix checks for a reINVITE on incoming requests to look for the presence
of a to-tag instead of the initial INVITE transaction state.
* Made caller_id_incoming_request() determine what to do if there is a
channel on the session or not. After a channel is created it is too late
to just store the new party id on the session because the session's party
id has already been copied to the channel's caller id.
ASTERISK-25404 #close
Reported by: Chet Stevens
Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be
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When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of
'0' returned. While this was valid per the documentation for the API, it was
apparently never returned previously. As a result, several users of the
scheduler API viewed the result as being invalid, causing them to reschedule
already scheduled items or otherwise fail in interesting ways.
This patch corrects the users such that they view '0' as valid, and a returned
ID of -1 as being invalid.
Note that the failing HEP RTCP tests now pass with this patch. These tests
failed due to a duplicate scheduling of the RTCP transmissions.
ASTERISK-25449 #close
Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39
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Some systems require the REFER packet to include a Referred-By header.
If the channel variable SIPREFERREDBYHDR is set, it passes that value as the
Referred-By header value. Otherwise, it adds the current dialog’s local info.
Reported by: Dan Cropp
Tested by: Dan Cropp
Change-Id: I3d17912ce548667edf53cb549e88a25475eda245
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When GetConfigJSON was introduced back in 1.6, it returned each
section as an array of strings: ["key=value", "key2=value2"].
Afterwards, it was changed a few times and became
["key": "value", "key2": "value2"], which is not a correct JSON.
This patch fixes that by constructing a JSON object {} instead of
an array [].
Also, the keys "istemplate" and "tempates" that are used to
indicate templates and their inherited categories are now wrapped in
quotes.
ASTERISK-25391 #close
Reported by: Bojan Nemčić
Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8
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A deadlock can happen when a sorcery object is being expired from the
memory cache when at the same time another object is being placed into the
memory cache. There are a couple other variations on this theme that
could cause the deadlock. Basically if an object is being expired from
the sorcery memory cache at the same time as another thread tries to
update the next object expiration timer the deadlock can happen.
* Add a deadlock avoidance loop in expire_objects_from_cache() to check if
someone is trying to remove the scheduler callback from the scheduler.
ASTERISK-25441 #close
Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc
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Make sorcery_memory_cache_close() call remove_all_from_cache() instead of
partially inlining it.
ASTERISK-25441
Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c
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Basically you should shutdown in the opposite order of how you setup since
later setup pieces likely depend on earlier setup pieces. e.g.,
Registering your external API with the rest of the system should be the
last thing setup and the first thing unregistered during shutdown.
Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e
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Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160
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Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c
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In practice the set_role API callback can be invoked even
when no ICE is present on an RTP instance. This can occur
if ICE has not been enabled on it.
ASTERISK-25438 #close
Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69
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Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc
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Made use the ao2 sort compare template function and OBJ_SEARCH_xxx
identifiers.
Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c
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* Now conf_alloc() has more off nominal error checking.
* Eliminated RAII_VAR() use in conf_alloc().
* Eliminated a dubius shortcut when destroying cfg->general in
conf_destructor() that would cause a crash if cfg->general failed to get
allocated.
* Add some ACO registration section comments.
Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a
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Need to finish initializing the string fields in the ao2 object before
putting any default strings into them.
ASTERISK-25383 #close
Reported by: yaron nahum
Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84
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When Asterisk is part of a larger distributed system, log files are often
gathered using tools (such as logstash) that prefer to consume information
and have it rendered using other tools (such as Kibana) that prefer a
structured format, e.g., JSON. This patch adds support for JSON formatted
logs by adding support for an optional log format specifier in Asterisk's
logging subsystem. By adding a format specifier of '[json]':
full => [json]debug,verbose,notice,warning,error
Log messages will be output to the 'full' channel in the following
format:
{
"hostname": Hostname or name specified in asterisk.conf
"timestamp": Date/Time
"identifiers": {
"lwp": Thread ID,
"callid": Call Identifier
}
"logmsg": {
"location": {
"filename": Name of the file that generated the log statement
"function": Function that generated the log statement
"line": Line number that called the logging function
}
"level": Log level, e.g., DEBUG, VERBOSE, etc.
"message": Actual text of the log message
}
}
ASTERISK-25425 #close
Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238
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