When the ast_data_buffer_put rejects to add a packet, for example because
the buffer already contains a packet with the same sequence number, the
payload will never be freed, resulting in a memory leak.
The data buffer will now return an error if this situation occurs
allowing the caller to free the payload. The res_rtp_asterisk module
has also been updated to do this.
ASTERISK-28826
Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1
If a frame handling routine returns a list of frames (vs. a single frame)
those frames are never passed to a tech's write_stream handler even if one is
available. For instance, if a codec translation occurred and that codec
returned multiple frames then those particular frames were always only sent
to the tech's "write" handler. If that tech (pjsip for example) was stream
capable then those frames were essentially ignored. Thus resulting in bad
audio.
This patch makes it so the "write_stream" handler is appropriately called
for all cases, and for all frames if available.
ASTERISK-28795 #close
Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d
The dial application had 80 characters of destination length
limitation. But this limitation causes unexpected dial string
cut if the dial string is long.
Removed unnecessary limited buffer to support longer dial
destination.
ASTERISK-27946
Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330
named_acl.c (which is really a named_ha) now uses ast_ha_output.
I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set. If this works then we can add
similar to other modules as required.
Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f
binutils 2.34 merged this commit:
https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\
h=fd3619828e94a24a92cddec42cbc0ab33352eeb4
Which effectively does things like:
-#define bfd_section_size(bfd, ptr) ((ptr)->size)
-#define bfd_get_section_size(ptr) ((ptr)->size)
+#define bfd_section_size(sec) ((sec)->size)
So in order to remain backwards compatible we need to detect this API
change, and adjust accordingly. The simplest is to notice that the
bfd_get_section_size and bfd_get_section_vma MACROs are no longer
defined, and define then onto the new API. The alternative is to litter
the code with a number of #ifdef #else #endif splatters right through
the code.
Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f
Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.
This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.
ASTERISK-28780
Change-Id: I8044c06daa06f0f16607788c596f55623be26f58
A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.
Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.
ASTERISK-26711 #close
Reported by: Vitold
Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0
The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in
65b8381550 such that we would never
actually make a DNS TXT query as described.
This patch restores the documented behavior.
ASTERISK-19460 #close
Reported by: George Joseph
Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376
When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.
Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.
That being the case, and since this is technically an API breaking change (no
one should really be affected since things never really worked) the ARI version
was updated to reflect that.
ASTERISK-28755 #close
Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f
There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.
Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113
There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.
Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e
This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.
This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.
ASTERISK-28697
Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b
Dump OpenSSL's error stack to the error log when things fail.
ASTERISK-28750 #close
Reported by: Martin Zeh
Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
ASTERISK-28733
Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.
* If the user wants to write to a wav49 file, make sure that it is
reflected properly in MIXMONITOR_FILENAME.
* Add a note to the documentation describing this behavior.
* Add a note in main/file.c indicating that app_mixmonitor needs to be
changed if the logic in build_filename was changed.
ASTERISK-24798 #close
Reported by: xrobau
Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c
In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use. To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes" In Asterisk versions <18, the default
is "no" preserving existing behavior. Beginning with
Asterisk 18, the option will default to "yes".
NOTE: This change does not affect UserEvents or the ARI
TextMessageReceived events.
* Added the "hide_messaging_ami_events" option to asterisk.conf.
* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
the "Message/ast_msg_queue" channel if the option is set in
asterisk.conf. This suppresses the reporting of the events.
Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b
When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.
* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.
NOTE: This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.
ASTERISK-28677
Reported by: Maciej Michno
Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca
Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.
We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.
Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.
Additionally:
* Change 'enablestatic' to 'enable_static' but keep the former for
backwards compatibility.
* Improve some internal variable names
ASTERISK-28710 #close
Change-Id: I647510f796473793b1d3ce1beb32659813be69e1
SILK @ 24kHz is not shown in the 'core show translation' output because of an
off-by-one-error. Discovered while looking into ASTERISK~19871.
ASTERISK-28706
Reported by: Sean Bright
Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510
In af90afd90c, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.
Whitespace only, no functional change.
ASTERISK~23324
Reported by: Kevin McCoy
Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87
Adds source port matching support when IP matching is used:
[example]
type = identify
match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444
If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.
ASTERISK-28639 #close
Reported by: Mitch Claborn
Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92
When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
When a topic is added to a pool, its name has the pool's topic
name prepended. For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.
When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.
Fix:
* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.
ASTERISK-28633
Reported by: Joeran Vinzens
Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6
The Bridge application was inconsistent if the channel to bridge with is
not specified. If no parameters are given then a warning is issued and
the current channel is hung up. If options are given but no channel is
specified then a warning is issued and the current channel is not hung up.
* Made the Bridge application give a verbose message instead of a warning
if the channel to bridge with is not specified and made not hang up the
current channel. As a result dialplan no longer needs to check if a
channel name is passed before calling Bridge and simply needs to check the
BRIDGERESULT channel variable instead. This is something you likely want
your dialplan to do anyway.
* Fixed up L() option warning message. It is up to the caller to
determine if the channel is hung up because of the warning. Dial() hangs
up the current channel while Bridge() does not.
Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73
When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.
ASTERISK-28562 #close
Reported by: Robert Sutton
Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d
The db_init() function ultimately calls db_sync() which signals the
condition before it is initialized.
Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies. It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.
Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
The problem is essentially the same as in ASTERISK~28245. Besides
the direct media scenario we have an additional scenario where a
special client is involved. This device mutes audio by default in
transmit direction (no rtp frames) and activates audio only by a
foot switch. In this situation dtmf input (pin for conferences,
transfer features codes , etc) using SIP INFO mode is not
understood properly especially when SIP INFO messages are sent
quickly.
This patch ensures that SIP INFO frames are properly queued and
processed in the above scenario. The patch also corrects situations
where successive dtmf events are received quicker than the
signalled event duration (plus minimum gap/pause) allows, i.e. DTMF
events have to be buffered in the ast channel read queue and
emulation has to be processed asynchronously at slower speed.
Reported by: Thomas Arimont
patches:
trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525)
Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2
* 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
Pirmin Walthert
Jean Aunis
Jaco Kroon
Kevin Harwell
sungtae kim
Sean Bright
Joshua C. Colp
George Joseph
Walter Doekes
Friendly Automation
Richard Mudgett
Kevin Reeves
Alexei Gradinari
Joshua Colp
Ben Ford
Corey Farrell
Florian Floimair
Torrey Searle
Walter Doekes
Holger Hans Peter Freyther
Antoni Goldstein
Lucas Mendes
Chris-Savinovich
Sebastian Kemper
Matthew Fredrickson
Sungtae Kim
Giuseppe Sucameli
Gerald Schnabel
Valentin Vidic
mohitdhiman