* Use ast_cli_completion_add() to improve performance when large number of
cached items are present.
* Only complete one URI for commands that only accept a single URI.
* Change command documentation to wrap at 80 characters to improve
readability.
Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902
When testing for the existance of a file, the media cache is searched even if
the file has no chance of being in it. This can cause performance issues
as the media cache size increases.
As a result, calls to applications like Read and Playback using local files
must scan through the media cache before playing. Under load and with a
large cache, this can delay the playback of those files.
This patch updates the function that checks for the existance of a file to
only consult the media cache database if the requested file is a remote path.
It introduces a new is_remote_path() function in main/file.c.
ASTERISK-28625 #close
Reported-by: kevin@phoneburner.com
Change-Id: If91137493732d9034dafa381c081c69274a7dcc9
If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.
Action: Originate
Channel: Local/1111
Application: Originate
Data: Local/2222,app,System,touch /tmp/owned
If the "system" authorization isn't set, we now block the
Originate app as well as the System, Exec, etc. apps.
ASTERISK-28580
Reported by: Eliel Sardañons
Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
The current code sets alert levels starting from index 1.
Need to set on whole pool starting from index 0.
Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d
Here's the basic scenario that occurred when executing an AMI fast originate
while at the same time something else locks the channels container, and also
wants a lock on the dialed channel:
1. pbx_outgoing_attempt obtains a lock on a dialed channel
2. concurrently another thread obtains a lock on the channels container, and
subsequently requests a lock on the dialed channel. It waits on #1. For
instance, "core show channel <dialed channel"
3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt
function exits
4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and
attempts to hang up the dialed channel
5. hang up tries to obtain the channels container lock, but can't due to #2.
6. Asterisk is deadlocked.
The solution was to allow the pbx_outgoing_exec function to "steal" ownership
of the dialed channel, and handle hanging it up. The channel now is either hung
up prior to it being potentially locked by the initiating thread, or if locked
the hang up takes place in a different thread, thus alleviating the deadlock.
ASTERISK-28561
patches:
iliketrains.diff submitted by Joshua Colp (license 5000)
Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a
Serializer pools have previously existed in Asterisk. However, for the most
part the code has been duplicated across modules. This patch abstracts the
code into an 'ast_serializer_pool' object. As well the code is now centralized
in serializer.c/h.
In addition serializer pools can now optionally be monitored by a shutdown
group. This will prevent the pool from being destroyed until all serializers
have completed.
Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971
This avoids use of the global variable and ensures topic_all remains
active until all topics are freed.
ASTERISK-28553
patches:
ASTERISK-28553.patch by coreyfarrell (license 5909)
Change-Id: I9a8cd8977f3c3a6aa00783f8336d2cfb9c2820f1
pbx_extension_helper takes two 'context' arguments. One (con) is a
pointer directly to a 'struct ast_context' and the other (context) is
the name of the context. In all cases, one of these arguments is NULL
and the other is non-NULL.
Functions that are ultimately called by pbx_extension_helper expect that
'context' will be non-NULL, so we set it unconditionally on entry into
this function.
ASTERISK-28534 #close
Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d
Added "like" support for 'core show taskprocessors'. Now you
can specify a specific set of taskprocessors (or just one) by
adding the keyword "like" to the above command, followed by
your search criteria.
Change-Id: I021e740201e9ba487204b5451e46feb0e3222464
astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug
threads tracking information in the internal structures of astobj2.
Unfortunately this means that ao2_global_obj contains the statically
allocated debug threads tracking fields which are used by initialization
and cleanup but main/astobj2.c believed those fields and associated
space did not exist.
Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527
Added two new CLI commands to reset stats for taskprocessors. You can
reset stats for a single, specific taskprocessor ('core reset
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
('core reset taskprocessors'). These commands will reset the counter for
the number of tasks processed as well as the max queue size.
Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d
Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based
functions would result in the reference not being logged under
REF_DEBUG. This could sometimes cause inaccurate logging if NULL was
accidentally passed to a reference action. Now reference logging is
only disabled by option passed to the allocation method.
Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54
You can currently capture backtraces of memory allocations but they
only get displayed when you stop asterisk and the atexit hooks
are enabled. Now, if memory backtrace is on and you issue a
"memory show allocations" CLI command for a specific file, then
a backtrace will show for each allocation that occurred after
you turned "memory backtrace on". The backtrace display is shown
only when a specific file's allocations are displayed to prevent
a massive CLI dump of every file's allocations.
Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f
It is possible for topic->name to be NULL, this causes the allocation
reference to not be logged. Use the name variable instead which has
been verified to be a non-empty.
Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883
ast_mwi_topic() returns a borrowed reference which should not be
unreferenced, doing so leads to a FRACK. This was hidden by the fact
that stasis_cache.c leaked the result of cache_remove in
caching_topic_exec.
Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7
This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.
ASTERISK-28533
Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.
Introduce ast_variable_list_replace() and use it where appropriate.
ASTERISK-23756 #close
Patches:
setvar-multiplie.patch submitted by Michael Goryainov
Change-Id: Ie1897a96c82b8945e752733612ee963686f32839
This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.
Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.
ASTERISK-28512
Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2
Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.
While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.
Also made a minor fix to the data buffer unit test.
Change-Id: I56107c7411003a247589bbb6086d25c54719901b
This change removes the assumption that a frame will always have
a src set on it. This assumption is incorrect.
Given a scenario where an RTP packet is received with no payload
the resulting audio frame will have no samples. If this frame goes
through a signed linear translation path an interpolated frame can
be created (if generic packet loss concealment is enabled) that has
minimal data on it, including no src. If this frame is given to a
translation path a crash will occur due to the lack of src.
ASTERISK-28499
Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8
The new function takes in a pointer to an ast_sockaddr structure,
a hostname and an optional port and then dispatches parallel
"AAAA" and "A" record queries. If an "AAAA" record is returned,
it's parsed into the ast_sockaddr structure along with the port
if it was supplied. If no "AAAA" record was returned, the
first "A" record returned (if any) is parsed instead.
This is a synchronous call. If you need asynchronous lookups,
use ast_dns_query_set_resolve_async and roll your own.
Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
Somehow it's possible for the srtp session object to be NULL even though the
Asterisk srtp object itself is valid. When this happened it would cause a
crash down in the srtp code when attempting to protect or unprotect data.
After looking at the code there is at least one spot that makes this situation
possible. If Asterisk fails to unprotect the data, and after several retries
it still can't then the srtp->session gets freed, and set to NULL while still
leaving the Asterisk srtp object around. However, according to the original
issue reporter this does not appear to be their situation since they found
no errors logged stating the above happened (which Asterisk does for that
situation).
An issue was found however, where a possible race condition could occur between
the pjsip incoming negotiation, and the receiving of RTP packets. Both places
could attempt to create/setup srtp for the same rtp instance at the same time.
This potentially could be the cause of the problem as well.
Given the above this patch adds locking around srtp setup for a given rtp, or
rtcp instance. NULL checks for the session have also been added within the
protect and unprotect functions as a precaution. These checks should at least
stop Asterisk from crashing if it gets in this situation again.
This patch also fixes one other issue noticed during investigation. When doing
a replace the old object was freed before creating the replacement. If the new
replacement object failed to create then the rtp/rtcp instance would now point
to freed srtp data which could potentially cause a crash as well when the next
attempt to reference it was made. This is now fixed so the old srtp object is
kept upon replacement failure.
Lastly, more logging has been added to help diagnose future issues.
ASTERISK-28472
Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc
When updating times on CDR or CEL records using the time at which
it is done can result in times being incorrect if the system is
heavily loaded and stasis message processing is delayed.
This change instead makes it so CDR and CEL use the time at which
the stasis messages that drive the systems are created. This allows
them to be backed up while still producing correct records.
ASTERISK-28498
Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:
unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);
would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.
ASTERISK-28480
Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1
incorrect handling of UDPTL squence number wrap arounds causes
loss of packets every time the wrap around occurs
ASTERISK-28483 #close
Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234
The functions that build manager message headers do so in a way that
results in a single messages being split across multiple packets. While
this doesn't matter to the remote end, it makes network captures noisier
and harder to follow, and also means additional system calls.
With this patch, we build up more of the message content into the TLS
buffer before flushing to the network. This change is completely
internal to the manager code and does not affect any of the existing
API's consumers.
Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9
When a module fails to register itself (usually a coding error
in the module), dlerror() can return NULL. We weren't checking
for that in load_dlopen() before trying to strdup the error message
so a SEGV was thrown. dlerror() is now surrounded with an S_OR
so we don't SEGV.
Change-Id: Ie0fb9316f08a321434f3f85aecf3c7d2ede8b956
When fixing ASTERISK~24212, a change was done so a scheduled callback could not
be removed while it was running. The caller of ast_sched_del would have to wait.
However, when the caller of ast_sched_del is the callback itself (however wrong
this might be), this new check would cause a deadlock: it would wait forever
for itself.
This changeset introduces an additional check: if ast_sched_del is called
by the callback itself, it is immediately rejected (along with an ERROR log and
a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the
after-ast_sched_del-refcall function is only run if ast_sched_del returned
success.
This should fix the following spurious race condition found in chan_sip:
- thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE)
- thread 2: run sip_poke_peer_now
- thread 2: blank out sched-ID (too soon!)
- thread 1: set sched-ID (too late!)
- thread 2: try to delete the currently running sched-ID
After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor
excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and
other madness) should occur.
(Thanks Richard Mudgett for reviewing/improving this "scary" change.)
Note that this change does not fix the observed race condition: unlocked
access to peer->pokeexpire (and potentially other scheduled items in chan_sip),
causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the
deadlock go away. And in the observed case, it will not have adverse affects
(like memory leaks) because the scheduled item is removed through a different
path.
ASTERISK-28282
Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856
When manager debugging is turned on, this patch makes it so incoming AMI actions
are now also logged.
Change-Id: I8047524510e7ac97d99482b2448f8e368f29cd47
Where possble, hostname and port has been added to error
messages, mostly on the server side.
ASTERISK-26006
Reported by: Oleksandr Natalenko
Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0
There is WARNING "no samples for ..." on each Playtones.
The function ast_playtones_start calls ast_activate_generator,
which calls ast_prod.
The function ast_prod calls ast_write with empty audio frame.
In this case it's spam log.
Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660
When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.
Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1
Various fixes for issues caught by gcc 9. Mostly snprintf
trying to copy to a buffer potentially too small.
ASTERISK-28412
Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e
Added a conversion for umax (largest maximum sized integer allowed). Adjusted
the other current conversion functions (uint and ulong) to be derivatives of
the umax conversion since they are simply subsets of umax.
Also made the negative check move the pointer on spaces since strtoumax does it
anyways.
Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08
After a bridge has been deleted the stasis control will depart
the channel and might attempt to re-add it to the dial bridge.
The later can fail and this can lead to a situation that the stasis
control is unlinked but the after_bridge_cb_failed cb is executed trying
to access a dangling control object.
Fix it by calling the after_cb's before bridge_channel_impart_signal.
ASTERISK-26718
Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496
The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.
For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.
The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.
ASTERISK-28400
Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc
When compiling in dev mode, stasis statistics are enabled and can cause
a crash at shutdown due to the following:
- Containers are freed
- Topics and subscriptions remain
- When those topics and subscriptions are deallocated, they go to do
things with the container
This changes the containers to global ao2 objects, and whenever needed
in the code, a reference must be obtained and checked before any
operations can be done.
ASTERISK-28353 #close
Change-Id: Ie7d5e907fcfcb4d65bd36d5e4eb923126fde8d33
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
ASTERISK-28363
Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
We changed the validation of autocomplete parameter in the "indications
remove" command to avoid continue the execution of the command after
asking for autocomplete out of range parameters.
ASTERISK-28391
Reported by: lmendes86
Change-Id: I92b24131fd02f2e3c7fec966eea6f7a663310d40
Because hyphens are not matched literally in Asterisk dialplan, we need
to ignore them in our candidate extensions as well.
ASTERISK-17695 #close
Reported by: test011
Change-Id: I227f02301577b1633e8a55b9fe9dc149935c03f0
If the target of a Goto is a label that starts with a number, we
erroneously treat the leading digits as a priority.
ASTERISK-20182 #close
Reported by: Janu
Change-Id: Ia78408c0805a729103917247ecfc802f6fafc94b
When extconfig.conf file is parsed, the code previously searched for
character comma without verifying if error (null or blank). This caused
a segmentation error.
Change-Id: Id76b452d8f330d11c2742c37232761ad71472a8b
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.
ASTERISK-28343
Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c
Asterisk assumes that dlopen() will always run the constructor of a
shared library and every dlclose() will run its destructor. But dlopen()
may be permanent, meaning the constructor will only be run once, as is
the case with musl libc.
With a permanent dlopen() the Asterisk module loader does not work
correctly, because it's expectations regarding when the constructors and
destructors are run are not met. In fact a segmentation fault will occur
when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS
set (the dlopen() does not call the constructor, resource_being_loaded
is not set to NULL, then strlen is called with NULL instead of a string,
see issue ASTERISK-28319).
This commit adds code to the loader that will manually run the
constructors/destructors of the (non-builtin) modules where needed. To
achieve this a new ao2 container (linked list) is started and filled
with objects that contain the names of the modules and the pointers to
their respective info structs.
This behavior can be activated when configuring Asterisk
(--enable-permanent-dlopen). By default this is disabled, of course.
ASTERISK-28319 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875
Added topic_all container for centralizing the topic. This makes more
easier to managing the topics.
Added cli commands.
stasis show topics : It shows all registered topics.
stasis show topic <name> : It shows speicifed topic's detail info.
ASTERISK-28264
Change-Id: Ie86d125d2966f93de74ee00f47ae6fbc8c081c5f
Added ARI resource for channel statistics.
GET /ari/channels/{channelId}/rtp_statistics : It returns given
channel's rtp statistics detail.
ASTERISK-28320
Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376
When notifying a manager session that new events were available
the same lock was used that was also held when doing things within
the session (such as sending events out). If the manager session
blocked for a period of time this would cause a back up of messages
in Stasis and would also block any other sessions from receiving
events.
This change adds a separate lock to the manager session which is
strictly used for notifying it that new events are available.
ASTERISK-28350
Change-Id: Ifbcac007faca9ad0231640f5e82a6ca9228f261b
Since the new names went in, the maximum taskprocessor name is too
short. This patch increases the name field to a length to better
handle the new names.
Change-Id: I32f32d6926f25c8ef5a91303fd2988d2c2858877
Added ability to specifiy a wizard is read-only when applying
it to a specific object type. This allows you to specify
create, update and delete callbacks for the wizard but limit
which object types can use them.
Added the ability to allow an object type to have multiple
wizards of the same type. This is indicated when a wizard
is added to a specific object type.
Added 3 new sorcery wizard functions:
* ast_sorcery_object_type_insert_wizard which does the same thing
as the existing ast_sorcery_insert_wizard_mapping function but
accepts the new read-only and allot-duplicates flags and also
returns the ast_sorcery_wizard structure used and it's internal
data structure. This allows immediate use of the wizard's
callbacks without having to register a "wizard mapped" observer.
* ast_sorcery_object_type_apply_wizard which does the same
thing as the existing ast_sorcery_apply_wizard_mapping function
but has the added capabilities of
ast_sorcery_object_type_insert_wizard.
* ast_sorcery_object_type_remove_wizard which removes a wizard
matching both its name and its original argument string.
* The original logic in __ast_sorcery_insert_wizard_mapping was moved
to __ast_sorcery_object_type_insert_wizard and enhanced for the
new capabilities, then __ast_sorcery_insert_wizard_mapping was
refactored to just call __ast_sorcery_insert_wizard_mapping.
* Added a unit test to test_sorcery.c to test the read-only
capability.
Change-Id: I40f35840252e4313d99e11dbd80e270a3aa10605
A size_t is not always an unsigned long.
* Use the %zu format specifier in the ast_cli() printf format string since
AST_VECTOR_SIZE() returns a size_t value.
Change-Id: Ib102dd36bbe6c2a7a4ce6870ae9110d978dd7e98
As part of an earlier voicemail refactor, ast_delete_mwi_state_full
was modified to remove the pool topic for a mailbox when the state
was deleted. This was an attempt to prevent stale topics from
accumulating when app_voicemail was reloaded and a mailbox went
away. Unfortunately because of the fact that when app_voicemail
reloads, ALL mailboxes are deleted then only current ones recreated,
topics were being removed from the pool that still had subscribers
on them, then recreated as new topics of the same name. So now
modules like res_pjsip_mwi are listening on a topic that will
never receive any messages because app_voicemail is publishing on
a different topic that happens to have the same name. The solutiuon
to this is not easy and given that accumulating topics for
deleted mailboxes is less evil that not sending NOTIFYs...
* Removed the call to stasis_topic_pool_delete_topic in
ast_delete_mwi_state_full.
Also:
* Fixed a topic reference leak in res_pjsip_mwi
mwi_stasis_subscription_alloc.
* Added some debugging to mwi_stasis_subscription_alloc,
stasis_topic_create, and topic_dtor.
* Fixed a topic reference leak in an error path in
internal_stasis_subscribe.
ASTERISK-28306
Reported-by: Jared Hull
Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27
Topic names now follow: <subsystem>:<functionality>[/<object>]
This ensures that they are all unique, and also provides better
insight in to what each topic is for.
Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.
Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.
Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.
ASTERISK-28335
Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d
Add a json_pack at startup that will fail if runtime links against a
library older than jansson-2.11.
Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a
This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.
ASTERISK-28279
Change-Id: I460238c488eca4d216b9176576211cb03286e040
Currently, the Asterisk does not support seperated HTTP request.
This patch make the Asterisk enables to wait lest part of HTTP request.
Also increases acceptable HTTP body length to 40k to support more
larger request.
ASTERISK-28236
Change-Id: I48a401aa64a21c3b37bf3cb4e0486d64b7dd8aa1
This change provides an easier mechanism to determine which
subscribers are subscribed to a topic. Using this you can
inspect the specific subscribers for further details.
Change-Id: I8deea21703cd5c5357b85593b46c3eaf24e18c0c
To prevent one subsystem's taskprocessors from causing others
to stall, new capabilities have been added to taskprocessors.
* Any taskprocessor name that has a '/' will have the part
before the '/' saved as its "subsystem".
Examples:
"sorcery/acl-0000006a" and "sorcery/aor-00000019"
will be grouped to subsystem "sorcery".
"pjsip/distributor-00000025" and "pjsip/distributor-00000026"
will bn grouped to subsystem "pjsip".
Taskprocessors with no '/' have an empty subsystem.
* When a taskprocessor enters high-water alert status and it
has a non-empty subsystem, the subsystem alert count will
be incremented.
* When a taskprocessor leaves high-water alert status and it
has a non-empty subsystem, the subsystem alert count will be
decremented.
* A new api ast_taskprocessor_get_subsystem_alert() has been
added that returns the number of taskprocessors in alert for
the subsystem.
* A new CLI command "core show taskprocessor alerted subsystems"
has been added.
* A new unit test was addded.
REMINDER: The taskprocessor code itself doesn't take any action
based on high-water alerts or overloading. It's up to taskprocessor
users to check and take action themselves. Currently only the pjsip
distributor does this.
* A new pjsip/global option "taskprocessor_overload_trigger"
has been added that allows the user to select the trigger
mechanism the distributor uses to pause accepting new requests.
"none": Don't pause on any overload condition.
"global": Pause on ANY taskprocessor overload (the default and
current behavior)
"pjsip_only": Pause only on pjsip taskprocessor overloads.
* The core pjsip pool was renamed from "SIP" to "pjsip" so it can
be properly grouped into the "pjsip" subsystem.
* stasis taskprocessor names were changed to "stasis" as the
subsystem.
* Sorcery core taskprocessor names were changed to "sorcery" to
match the object taskprocessors.
Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56
Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a
little easier to retrieve a string field from the JSON object.
Also added an 'ast_strings_equal' function that safely checks (checks for NULLs)
for equality between two strings.
Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b
The CDR was overwriting the start time when the call continued the
dialplan from the ARI stasis or a Local channel was originated.
This change fixes this by no longer reinitializing the CDR when
transitioning out of the dialed pending state to the single state.
ASTERISK-28181
Change-Id: I921bc04064b6cff1deb2eea56a94d86489561cdc
Split destroy_hint method to separate hint removal and extension hint
state changed callback, the latter now called via stasis.
This avoids deadlock between res_parking reload that is removing the
parking lot and the related hint and subscribe requests coming for the
same parking lot.
ASTERISK-28173
Change-Id: I5b03c3455b3b12b6f83cea4cc34f4b4b20444f7e
Testing revealed that the cache added no benefit but that it could
consume excessive memory.
Two new index related functions were created:
ast_sounds_get_index_for_file() and ast_media_index_update_for_file()
which restrict index updating to specific sound files.
The original ast_sounds_get_index() and ast_media_index_update()
calls are still available but since they no longer cache the results
internally, developers should re-use an index they may already have
instead of calling ast_sounds_get_index() repeatedly. If information
for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().
The media_index directory scan code was elimininated in favor of
using the existing ast_file_read_dirs() function.
Since there's no more cache, ast_sounds_index_init now only
registers the sounds cli commands instead of generating the
initial index and subscribing to stasis format register/unregister
messages.
"sounds" is no longer a valid target for the "module reload"
command.
Both the sounds cli commands and the sounds ari resources were
refactored to only call ast_sounds_get_index() once per invocation
and to use ast_sounds_get_index_for_file() when a specific sound
file is requested.
Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d
The type value extracted from stasis message data in channel_hangup_handler_cb
isn't compared against the valid values "run", "pop" and "push". Thus the
manager events HangupHandlerPush, HangupHandlerPop and HangupHandlerRun are
never thrown.
This regression was introduced by ASTERISK_21462.
ASTERISK-28252
Change-Id: I9956e35e18da1873113644df1ddc3c7cd37bf524
A bug in GCC causes TEST_CEL to return failure under the following
conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline. The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()
Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7
If the Monitor is started with the i option the read_stream will be
NULL. One code path in channel.c checks if write_stream is set but than
uses read_stream instead causing a segfault.
ASTERISK-28249
Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525
Previously both AMI and ARI used a default route on
their stasis message router to handle some of the
messages for publishing out their respective
connection. This caused messages to be given to
their subscription that could not be formatted
into AMI or JSON.
This change adds an API call to the stasis message
router which allows a default route to be set as well
as formatters that the default route is expecting.
This allows both AMI and ARI to specify that their
default route only wants messages of their given
formatter. By doing so stasis can more intelligently
filter at publishing time so that they do not receive
messages which will not be turned into AMI or JSON.
ASTERISK-28244
Change-Id: I65272819a53ce99f869181d1d370da559a7d1703
During Bridging of two channels if masquerade operation is performed on a
channel (clone channel) which was created with endpoint details
(ast_channel_alloc_with_endpoint()) and the original channel which is created
without endpoint details (ast_channel_alloc()) then both the channels must
exchange their endpoint details or else after masquerade when clone channel
is being destroyed the endpoint cleanup callbacks will be destroyed too and
after call completion unique_id of original channel will still be there in
ast_endpoint structure's channel_ids container.
ASTERISK-28197
Change-Id: Ied0451f378a3f2a36acc8c0984959a69895efa17
An int64_t is not likely the same size as a long.
* Changed the int64_t values in the statistics structs to longs so casting
is not necessary when generating the formatted CLI output. The offending
members did not need to be int64_t anyway as they were only set by an int
type variable which was already truncating bits.
* Reordered the statistics structs to reduce potential padding bytes.
Change-Id: Ic090a070e9dc4ca650ebdb9c01ed50a581289962
This commit caused issues with polling when combined with
the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis"
This reverts commit 17d6d9e1e7.
ASTERISK-28222
Reported by: abelbeck
Change-Id: Ib6a16cbe2c0b74c76c64264f525ab647959c1483
The backtrace library bfd.h include file does not get the sizes of
pointers and ints right on some platforms. On my old test box the size
of bfd_vma is 8 while the size of a pointer is 4. gcc on the box
complains of the integer casting to/from pointers size mismatch.
* uintptr_t to the rescue by doing an appropriate two stage cast.
Change-Id: Icb2621583f50c8728de08a3c824d95fe53cc45d0
This change adds statistics gathering to Stasis topics,
subscriptions, and message types. These can be viewed using
CLI commands and provide insight into how Stasis is used
and how long certain operations take to execute.
These are only available when Asterisk is compiled in
developer mode and do not have any impact under normal
operation.
ASTERISK-28117
Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f
Some platforms provide an implementation of socket() and pipe2() that allow the
caller to specify that the resulting file descriptors should be non-blocking.
Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous
calls to fcntl() to set the O_NONBLOCK flag afterwards.
In passing, change ast_alertpipe_init() to use pipe2() directly instead of the
wrapper if it is available.
Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0
A subscriber can now indicate that it only wants messages
that have formatters of a specific type. For instance,
manager can indicate that it only wants messages that have a
"to_ami" formatter. You can combine this with the existing
filter for message type to get only messages with specific
formatters or messages of specific types.
ASTERISK-28186
Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c
This adds documentation to handle_cli_malloc_trim() indicating how it
can be useful when debugging OOM conditions.
Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78
We've had multiple opportunities where Richard Mudgett's
malloc_trim patch has been useful. Let's get it
pushed up to gerrit and merged.
Since malloc_trim is only available in libc, an entry is
added to configure.ac to create a definition for
HAVE_MALLOC_TRIM.
Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.
ao2_container_alloc is now restricted to modules only and is being
removed from Asterisk 17.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
Create ao2_container_dup_weakproxy_objs to perform a similar function to
ao2_container_dup. This function expects the source container to have
weakproxy objects, inserts the associated non-weak objects into the
destination container. Orphaned weakproxy objects are ignored.
Create test for this new function and for ao2_weakproxy_find.
Change-Id: I898387f058057e08696fe9070f8cd94ef3a27482
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
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
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 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
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
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.
Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c
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.
ASTERISK-28107
Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499
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 still available for use but only in modules. Only
ao2_container_alloc remains due to it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
__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.
Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
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.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.
* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.
Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.
Add comments to explain thread safety of the 'struct __priv_data'
bitfields.
Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a
thread_worker_pair, set_size_data and task_pushed_data structures are
allocated with AO2 objects, passed to a taskprocessor, then released.
They never have multiple owners or use locking so AO2 only adds
overhead.
Change-Id: I2204d2615d9d952670fcb48e0a9c0dd1a6ba5036
The field used to store call arguments was not large enough to hold the
arguments string that can be constructed for 'open'. Expand it to
prevent this warning/error.
Change-Id: I514927f256481bc84df10a51b19d5b5fb1bc387e
ast_sendtext_data() would create an incorrect T.140 text frame which
length include the null terminator byte. It causes ultimately RTP
packets to be send with this trailing 0. The proposed fix just set the
correct length to the text frame
ASTERISK-28089
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: I7ab1b9ed1e21683b2b667ea0a59d9aba3c77dd96
This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts
startup, but it's possible for this code to be returned on manual load
of a module after startup.
It is an error for a module to not have a load callback but this is not
a fatal system error. In this case flag the module as declined, return
AST_MODULE_LOAD_FAILURE only if a required module is broken.
Expand doxygen documentation for AST_MODULE_LOAD_*.
Change-Id: I3c030bb917f6e5a0dfd9d91491a4661b348cabf8
There is currently no way to indicate to Asterisk that TLS certificates
and/or keys have been updated other than by modifying http.conf or
restarting Asterisk.
There is already code in main/tcptls.c that determines if a reload is
actually necessary based on the hashes of the certicate and dependent
files, so this change merely gives us a way to request a reload without
explicitly modifying http.conf.
Change-Id: Ie795420dcc7eb3d91336820688a29adbcc321276
* ACO options
* Indications
* Module loader ref_debug object
* Media index info and variants
* xmldoc items
These allocation locations were identified using reflocks.py on the
master branch.
Change-Id: Ie999b9941760be3d1946cdb6e30cb85fd97504d8
When a module reload fails we never set AST_MODULE_RELOAD_ERROR. This
caused reload failures to incorrectly report 'No module found'.
Change-Id: I5f3953e0f7d135e53ec797f24c97ee3f73f232e7
* Display list of unavailable dependencies when they cause another
module to fail loading.
* When a module declines to load find all modules which depend on it so
they can be declined and listed together.
* Prevent retry of declined modules during startup.
* When a module fails to dlopen try loading it with RTLD_LAZY so we can
attempt to display the list of missing dependencies.
These changes are meant to reduce logger spam that is caused when a
module has many dependencies and declines to load. This also fixes some
error paths which failed to recognize required modules.
Module load/start errors are delayed until the end of loader startup.
Change-Id: I046052c71331c556c09d39f47a3b92975f3e1758
If a channel creates an AST_TEXT_FRAME with datalen == 0, the ast_frdup()
and ast_frisolate() functions could create a clone frame with an invalid
data.ptr which would cause a crash. The proposed fix is to make sure that
for such empty text frames, ast_frdup() and ast_frisolate() return cloned
text frames with a valid data.ptr.
ASTERISK-28076
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: Ib882dd028598f13c4c233edbfdd7e54ad44a68e9
When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock
was ever used, so it can be recorded in the REF_DEBUG destructor entry.
Create contrib/scripts/reflocks.py to process locking used by
allocator. This can be used to identify places where
AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage.
Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4
Add a volatile flag to lock tracking structures so we only need to use
the global lock when first initializing tracking.
Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by
astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is
not defined.
Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b
Reduce options to 2-bit field, magic to 30 bit field. Move ref_counter
next to options and explicitly use int32_t so the fields will pack.
This reduces memory overhead for every ao2 object by 8 bytes on x86_64.
Change-Id: Idc1baabb35ec3b3d8de463c4fa3011eaf7fcafb5
* In main/config.c, AST_INCLUDE_GLOB is fixed to '1' making the #ifdefs
pointless.
* In utils/extconf.c, AST_INCLUDE_GLOB is never defined so there is a
lot of dead code.
Change-Id: I1bad1a46d7466ddf90d52cc724e997195495226c
When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log
before we shutdown astobj2_container. This caused the AO2_DEBUG
container registration container to be reported as a leak.
Change-Id: If9111c4c21c68064b22c546d5d7a41fac430430e
Use json_vsprintf from versions which contain fix for va_copy leak.
Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.
Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539
* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
safe to just use ast_json_ref.
Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped. This was previously
avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null(). The
"?" is a new feature of ast_json_pack starting with Asterisk 16.
Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48
When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.
This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.
Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.
Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0
does_id_conflict() was passing a pointer to an int to a callback
that expected a pointer to a size_t.
Found by the Address Sanitizer.
Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503
app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload. This resulted in leaks in both
areas.
* app_voicemail now calls ast_delete_mwi_state_full when it frees
a user structure and ast_delete_mwi_state_full in turn now calls
the new stasis_topic_pool_delete_topic function to clear the topic
from the pool.
Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8
There's been a long standing leak when using topic pools. The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically. If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.
* Added stasis_topic_pool_delete_topic() so modules can clean up
topics from pools.
* Registered the topic pool containers so it can be examined from
the CLI when AO2_DEBUG is enabled. They'll be named
"<topic_pool_name>-pool".
Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25
Stasis message types are global ao2 objects and we make stasis messages
and cache entries hold references to them. Since there are currently
situations where cache objects are never deleted, the reference count on
the types can exceed 100000 and generate a FRACK assertion message. The
stasis message cache could conceivably also have that many messages
legitimately on large systems.
The only down side to not holding the message type ref in the stasis
message is it only makes a crash either at shutdown or when manually
unloading a busy module slightly more likely. However, this is more
exposing a pre-existing stasis shutdown ordering issue than a problem with
not holding a message type ref in stasis messages.
* Made stasis messages and cache entries no longer hold a ref to the
message type.
Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707
* Create the stasis message object without a lock as it is immutable.
* Create the stasis message type object without a lock as it is immutable.
* Creating the stasis message type could crash if the passed in type name
is NULL and REF_DEBUG is enabled. Added missing NULL check when passing
the ao2 object tag string.
Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32
Since app_voicemail no longer uses the cache to maintain its state
there is no longer a need to cache these messages.
ASTERISK-27121
Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4
The documentation already specified EVENT_FLAG_DIALPLAN for this
event, but the implementation was using EVENT_FLAG_CALL.
Using EVENT_FLAG_DIALPLAN allows AMI clients to opt out of receiving
this highly verbose event.
ASTERISK-28033
Change-Id: I45b3119f30e4dbc17b49831f2b1a4f2c1beadafe
Generators provide such functionality as tone generation or
silence generation. RTCP frames provide RTCP information and
should not stop generators from operating.
ASTERISK-28005
Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9
The stasis cache provides a way to reconstruct the current state
of topic subscribers. Unfortunately, since every subscribe and
unsubscribe is cached, the cache continues to grow unabated while
asterisk is running. This patch removes subscribe messages from
the cache when the corresponding unsubscribe is received.
This patch also registers the cache containers with ao2 so that if
AO2_DEBUG is turned on, you can list the container and get its
stats from the CLI.
ASTERISK-27121
Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56
Added a check when we receive a HTTP request line or header line that is
too long. We now return an error response to the sender because we are
not able to process the request.
Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d
Providing a buffer larger than the internal buffer of ast_iostream_gets()
fails to get lines longer than the internal buffer.
* Made ast_iostream_gets() fill the supplied buffer with read data until
either a '\n' is found or the supplied buffer is filled just like fgets().
Change-Id: If18b3f6ee500e22f0633a68779ed09f7e0f305ed
All HTTP/AMI message headers are being sent to the verbose channel.
There are multiple places this is happening. Consolidate the loop into
a function. Drop the debug/verbose message.
Convert to using ast_asprintf to perform the length calculation, memory
allocation and snprintf all in one step.
Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1
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
When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.
Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37
With the new module loader it was missed that built-in modules never
parsed dependencies from mod->info into vectors of mod. This caused
manager to be initialized before acl (named_acl). If manager.conf
used any named ACL's they would not be found and result in no ACL being
applied to the AMI user.
In addition to the manager ACL fix this adds "extconfig" to all builtin
modules which support realtime configuration. This only matters if one
of the builtin modules is configured with 'preload', depending on
"extconfig" will cause config.c to automatically be initialize during
the preload stage.
Change-Id: I482ed6bca6c1064b05bb538d7861cd7a4f02d9fc
When converting from a json object to an ast variables list the conversion
algorithm was doing a complete traversal of the entire variables list for
every item appended from the json structure.
This patch makes it so the list is no longer traversed for each new ast
variable being appended.
Change-Id: I8bf496a1fc449485150d6db36bfc0354934a3977
When publishing a device state the change can be marked as being
cachable or not. If it is not cached the change is just published
to all interested and not stored away for later query. This was not
fully taken into account when publishing in stasis. The act of
publishing would create a topic for the device even if it may be
ephemeral.
This change makes it so messages which are not cached won't create
a topic for the device. If a topic does already exist it will be
published to but otherwise the change will only be published to
the device state all topic.
ASTERISK-27591
Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a
The "xmldoc dump" cli command was simply concatenating xml documents
into the output file. The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.
* Added 2 new functions to xml.c:
ast_xml_copy_node_list creates a copy of a list of children.
ast_xml_add_child_list adds a list to an existing list.
* Modified handle_dump_docs to create a new output document and
add to it the children from each input file. It then dumps the
new document to the output file.
Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07
In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.
This assertion is no longer present so the special logic can be
removed to make it the same between them both.
Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8
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
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode. With this
fix if message is disabled
a warning will print.
ASTERISK-25548
Change-Id: Ie0d866d1cbc60c16dbef08bc65e99505c3c1adfa
* Merge the preload and load stages, use load ordering to try preload's
first. This fixes an issue where `preload=res_config_curl` would fail
unless res_curl and func_curl were also preloaded. Now it is only
required that those modules be loaded during startup: autoload or
regular load is good enough.
* The configuration option `require` and `preload-require` were only
effective if the modules failed to load. These options will now abort
Asterisk startup if required modules fail to reach the 'Running'
state.
* Missing or invalid 'module.conf' did not prevent startup. Asterisk
doesn't do anything without modules so this a fatal error.
Change-Id: Ie4176699133f0e3a823b43f90c3348677e43a5f3
Separate "name" into "classname" and "name".
Use '.' for classname separator instead of '/'.
Prefix reserved words with '_'.
Wrap output with a top-level "testsuites" element.
Change-Id: Iec1a985eba1c478e5c1d65d5dfd95cb708442099
There is a rare case (do to the infrequent timing involved) where
CDR submission threads in batch mode can deadlock with a currently
running CDR batch process. This patch should remove the need for
holding the lock in the scheduler and should clean a few code
paths up that inconsistently submitted new work to the CDR batch
processor.
ASTERISK-27909
Change-Id: I6333e865db7c593c102c2fd948cecdb96481974d
Reported-by: Denis Lebedev
The AMI action was directly sending the text to the channel driver.
However, this makes two threads attempt to handle media and runs afowl of
CHECK_BLOCKING.
* Queue a read action to make the channel's media handling thread actually
send the text message. This changes the AMI actions success/fail response
to just mean the text was queued to be sent not that the text actually got
sent. The channel driver may not even support sending text messages.
ASTERISK-27943
Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379
In addition to text/* content types, incoming_in_dialog_request now
accepts application/* content types.
Also fixed a length issue when copying the body text. It was one
character short.
ASTERISK-27942
Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818
* changes:
channel.c: Make CHECK_BLOCKING() save thread LWP id for messages.
channel.c: Fix usage of CHECK_BLOCKING()
autoservice: Don't start channel autoservice if the thread is a user interface.
With ./configure --enable-dev-mode[=noisy], the build fails because every
warning gets an error. Therefore, Asterisk has to be free of warnings and this
variable must go.
Change-Id: I63dd2bc4833b9bdb04602f83422d16caf289d46a
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905
The CHECK_BLOCKING() macro is used to indicate if a channel's handling
thread is about to do a blocking operation (poll, read, or write) of
media. A few operations such as ast_queue_frame(), soft hangup, and
masquerades use the indication to wake up the blocked thread to reevaluate
what is going on.
ASTERISK-27625
Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d
Executing dialplan functions from either AMI or ARI by getting a variable
could place the channel into autoservice. However, these user interface
threads do not handle the channel's media so we wind up with two threads
attempting to handle the media.
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49
In Asterisk there are some dynamic codecs that have
a fixed payload number. This number was being improperly
used to negotiate the codec, instead of using the name
and sample rate. This could result in the wrong payload
number being negotiated for a codec.
This change makes it so that only static payloads
will be negotiated using their payload number.
ASTERISK-27848
Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570
Before Asterisk sends an HTTP response (at least in the case of errors),
it attempts to read & discard the content of the request. If the client
lies about the Content-Length, or the connection is closed from the
client side before "Content-Length" bytes are sent, the request handling
thread will busy loop.
ASTERISK-27807
Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762
ast_indicate is invoked with the bridge locked. As ast_indicate locks the
other end of the bridge as well this can lead to a deadlock in some situations.
(Especially when a different thread does the same in the reverse order).
This patch calls ast_indicate after unlocking the bridge which fixes the
deadlock. Calling ast_indicate with these parameters without locking the
bridge should be safe as this is done at different places without a
bridge lock.
ASTERISK-27094 #close
Reported-by: David Brillert
Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f
Sean Bright
Kevin Reeves
George Joseph
Friendly Automation
Kevin Harwell
Alexei Gradinari
Joshua Colp
Ben Ford
Corey Farrell
Florian Floimair
Torrey Searle
Walter Doekes
Holger Hans Peter Freyther
Antoni Goldstein
Lucas Mendes
Chris-Savinovich
sungtae kim
Sebastian Kemper
Matthew Fredrickson
Richard Mudgett
Sungtae Kim
Giuseppe Sucameli
Gerald Schnabel
Valentin Vidic
mohitdhiman
neutrino88
lvl
Jaco Kroon
Alexander Traud
Pirmin Walthert