Details:
- The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2"
- As a result, the memcmp() in dahdi_r2_get_link() always fails
- This cause dahdi_r2_get_link() to create new link for every channel
(instead of a new link for every ~30 channels)
- With the fix, far less links are generated -- so we use far less threads
Walter Doekes [Wed, 17 Jul 2019 13:06:12 +0000 (15:06 +0200)]
sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread
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.
George Joseph [Tue, 16 Jul 2019 12:55:49 +0000 (06:55 -0600)]
Build: Separate header install/uninstall
Asterisk headers are no longer installed and uninstalled
automatically when performing a "make install" or a
"make uninstall". To install/uninstall the headers, use
"make install-headers" and "make uninstall-headers".
The headers also continue to be uninstalled when performing a
"make uninstall-all".
Also corrects an issue where /usr/include/asterisk.h was never
being removed at all.
chan_sip: Handle invalid SDP answer to T.38 re-invite
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
Kevin Harwell [Wed, 12 Jun 2019 18:49:30 +0000 (13:49 -0500)]
app_voicemail: Remove dependency on the stasis cache
app_voicemail utilized the stasis cache when polling mailboxes for MWI. This
caused a memory leak (items were not being appropriately removed from the
cache), and subsequent slowdown in system processing. This patch removes the
stasis cache dependency, thus alleviating the memory leak. It does this by
utilizing the new MWI API that better manages state lifetime.
Kevin Harwell [Wed, 12 Jun 2019 18:11:42 +0000 (13:11 -0500)]
MWI: Update modules that subscribe to MWI to use new API calls
The MWI core recently got some new API calls that make tracking MWI state
lifetime more reliable. This patch updates those modules that subscribe to
specific MWI topics to use the new API. Specifically, these modules now
subscribe to both MWI topics and MWI state.
Kevin Harwell [Tue, 11 Jun 2019 19:12:12 +0000 (14:12 -0500)]
mwi: Update the MWI core to use stasis_state API
** Note **
This patch is meant to be the minimum needed in order for the MWI core to use
the now underlying stasis_state module. As such it does not completely remove
its reliance on the stasis_cache. Doing so has allowed current consumers to
not have to change, and update those code paths for this patch. When time
allows, subsequent patches can/will be made to those consumers to take advantage
of some of the new MWI API included here. Thus, eventually and ultimately
removing MWI dependency on the stasis_cache.
** End Note **
This patch makes it so the MWI core now takes advantage of the new stasis_state
API. Consumers of MWI should no longer need to depend upon stasis topic pooling,
and the stasis cache directly. Similar functionality and implementation details
have now been pushed into the stasis_state module. However, all MWI state should
be accessed via the MWI API itself.
As such a few new methods, and constructs have been added to the MWI core that
facilitate consumer publishing, subscribing, and iterating over MWI state data.
* ast_mwi_subscriber *
Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox
in order to receive updates about the given mailbox. Adding a subscriber will
create the underlying topic, and associated state data if those do not already
exist for it. The topic, and last known state data is guaranteed to exist for
the lifetime of the subscriber.
* ast_mwi_publisher *
Before publishing to a particular topic a publisher should be created. This can
be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then
be done using one of the MWI publish functions. This ensures the message is
published to the appropriate topic, and the last known state is maintained.
* ast_mwi_observer *
Add an observer in order to watch for particular MWI module related events. For
instance if a submodule needs to know when a subscription is added to any
mailbox an observer can be added to watch for that.
* other *
Urgent message count is now part of the published MWI state object. Also state
can be iterated over using defined callbacks.
Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous
patch introduced a variable of type unassigned long long which is 64-bits.
Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work
with 32-bit systems.
Kevin Harwell [Tue, 11 Jun 2019 17:30:27 +0000 (12:30 -0500)]
stasis_state: Add new stasis_state module
This new module describes an API that can be thought of as a combination of
stasis topic pools, and caching. Except, hopefully done in a more efficient
and less memory "leaky" manner.
The API defines methods, and data structures for managing, and tracking
published message state through stasis. By adding a subscriber or publisher,
consumers can more easily track the lifetime of the contained state. For
instance, when no more publishers and/or subscribers have need of the topic,
and associated state its data is removed from the managed container.
* stasis_state_manager *
The manager stores and well, manages state data. Each state is an association
of a unique stasis topic, and the last known published stasis message on that
topic. There is only ever one managed state object per topic. For each topic
all messages are forwarded to an "all" topic also maintained by the manager.
* stasis_state_subscriber *
Topic and state can be created, or referenced within the manager by adding a
stasis_state_subscriber. When adding a subscriber if no state currently exists
new managed state is immediately created. If managed state already exists then
a new subscriber is created referencing that state. The managed state is
guaranteed to live throughout the subscriber's lifetime. State is only removed
from the manager when no other entities require it.
* stasis_state_publisher *
Topic and state can be created, or referenced within the manager by also adding
a stasis_state_publisher. When adding a publisher if no state currently exists
new managed state is created. If managed state already exists then a new
publisher is created referencing that state. The managed state is guaranteed to
live throughout the publisher's lifetime. State is only removed from the
manager when no other entities require it.
* stasis_state_observer *
Some modules may wish to watch for, and react to managed state events. By
registering a state observer, and implementing handlers for the desired
callbacks those modules can do so.
* other *
Callbacks also exist that allow consumers to iterate over all, or some of the
managed state.
Chris-Savinovich [Thu, 27 Jun 2019 18:50:57 +0000 (13:50 -0500)]
app_voicemail.c: Build all three variants for app_voicemail at the same time
Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.
George Joseph [Thu, 27 Jun 2019 17:46:44 +0000 (11:46 -0600)]
pjproject_bundled: Add peer information to most SSL/TLS errors
Most SSL/TLS error messages coming from pjproject now have either
the peer address:port or peer hostname, depending on what was
available at the time and code location where the error was
generated.
sungtae kim [Mon, 15 Apr 2019 23:26:46 +0000 (01:26 +0200)]
res/ari/resource_channels.c: Added hangup reason code for channels
Currently, DELETE /ari/channels/<channelID> supports only few hangup reasons.
It's good enough for simple use, but when it needs to set the detail reason,
it comes challenges.
Added reason_code query parameter for that.
Alexei Gradinari [Wed, 29 May 2019 22:54:16 +0000 (18:54 -0400)]
res_fax: gateway sends T.38 request to both endpoints if V.21 detected
According T.38 Gateway 'Use case 3'
https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway
T.38 Gateway should send T.38 negotiation request to called endpoint
if FAX preamble (using V.21 detector) generated by called endpoint.
But it does not, because fax_gateway_detect_v21 constructs T.38
negotiation request, but forwards it only to other channel,
not to the channel on which FAX preamble is detected.
Some SIP endpoints could be improperly configured to rely on the other side
to initiate T.38 re-INVITEs.
With this patch the T.38 Gateway tries to negotiate with both sides
by sending T.38 negotiation request to both endpoints supported T.38.
George Joseph [Wed, 19 Jun 2019 16:58:39 +0000 (10:58 -0600)]
CI: New way to determnine libdir
We were using the presence of /usr/lib64 to determine where
shared libraries should be installed. This only existed on
Redhat based systems and was safe. If it existed, use it,
otherwise use /usr/lib.
Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT
NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is
installed there, it won't work.
The new method, just looks for $ID in /etc/os-release and if it's
centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib.
NOTE: This applies only to the CI scripts. Normal asterisk
build and install is not affected.
Alexei Gradinari [Fri, 14 Jun 2019 20:45:39 +0000 (16:45 -0400)]
translate.c do not log WARNING on empty audio frame
There is WARNING "no samples for ..." on each Playtones.
The function ast_playtones_start calls ast_activate_generator,
which calls ast_prod.
The function ast_prod calls ast_write with empty audio frame.
In this case it's spam log.
George Joseph [Mon, 17 Jun 2019 17:11:49 +0000 (11:11 -0600)]
chan_dahdi: Address gcc9 issues
Fixed format-truncation issues in chan_dahdi.c and
sig_analog.c. Since they're related to fields provided
by dahdi-tools we can't change the buffer sizes so we're just
checking the return from snprintf and printing an errior if we
overflow.
George Joseph [Mon, 10 Jun 2019 21:58:59 +0000 (15:58 -0600)]
app_confbridge: Attended transfer event fixup
When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.
Joshua Colp [Tue, 11 Jun 2019 12:26:42 +0000 (09:26 -0300)]
res_rtp_asterisk: Add support for DTLS packet fragmentation.
This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.
This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.
[custom_atxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,AttendedTransfer(${dest})
same => n,Return()
agupta [Thu, 6 Jun 2019 12:48:18 +0000 (18:18 +0530)]
chan_pjsip.c: Check for channel and session to not be NULL in hangup
We have seen some rare case of segmentation fault in hangup function
and we could notice that channel pointer was NULL. Debug log shows
that there is a 200 OK answer and SIP timeout at the same time. It
looks that while the SIP session was being destroyed due to timeout
call hangup due to answer event lead to race condition and channel
is being destroyed from two different places. The check ensures we
check it not to be NULL before freeing it.
[custom_blindxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,BlindTransfer(${dest},default)
same => n,Return()
;;;
Fixes an error occurring in function pgsql_reconnect() caused when value of
hostname is blank. Which in turn will cause the connection string to look
like this: "host= port=xx", which creates a sintax error. This fix now checks
if the corresponding values for host, port, dbname, and user are blank. Note
that since this is a reconnect function the database library will replace any
missing value pairs with default ones.
Alexei Gradinari [Tue, 28 May 2019 20:35:17 +0000 (16:35 -0400)]
res_fax: fix segfault on inactive "reserved" fax session
The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.
The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.
The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.
Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.
This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.
This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.
Alexei Gradinari [Tue, 28 May 2019 22:15:40 +0000 (18:15 -0400)]
res_fax: add channel name to CLI 'fax show session'
This patch adds a channel name to output of CLI 'fax show session'
and also expands the channel name field up to 30 characters on
CLI 'fax show sessions'
Matt Jordan [Fri, 10 May 2019 14:36:01 +0000 (09:36 -0500)]
res_prometheus: Add metrics for PJSIP outbound registrations
When monitoring Asterisk instances, it's often useful to know when an
outbound registration fails, as this often maps to the notion of a trunk
and having a trunk fail is usually a "bad thing". As such, this patch
adds monitoring metrics that track the state of PJSIP outbound registrations.
It does this by looking for the Registry events coming across the Stasis
system topic, and publishing those as metrics to Prometheus. Note that
while this may support other outbound registration types (IAX2, SIP, etc.)
those haven't been tested. Your mileage may vary.
(And why are you still using IAX2 and SIP? It's 2019 folks. Get with the
program.)
This patch also adds Sorcery observers to handle modifications to the
underlying PJSIP outbound registration objects. This is useful when a
reload is triggered that modifies the properties of an outbound registration,
or when ARI push configuration is used and an object is updated or
deleted. Because we rely on properties of the registration object to
define the metric (label key/value pairs), we delete the relevant metric when
we notice that something has changed and wait for a new Stasis message to
arrive to re-create the metric.
Matt Jordan [Thu, 3 Jan 2019 16:28:28 +0000 (10:28 -0600)]
res_prometheus: Add CLI commands
This patch adds a few CLI commands to the res_prometheus module to aid
system administrators setting up and configuring the module. This includes:
* prometheus show status: Display basic statistics about the Prometheus
module, including its essential configuration, when it was last scraped,
and how long the scrape took. The last two bits of information are useful
when Prometheus isn't generating metrics appropriately, as it will at
least tell you if Asterisk has had its HTTP route hit by the remote
server.
* prometheus show metrics: Dump the current metrics to the CLI. Useful for
system administrators to see what metrics are currently available without
having to cURL or go to Prometheus itself.
Matt Jordan [Thu, 9 May 2019 14:41:49 +0000 (09:41 -0500)]
res_prometheus: Add Asterisk bridge metrics
This patch adds basic Asterisk bridge statistics to the res_prometheus
module. This includes:
* asterisk_bridges_count: The current number of bridges active on the
system.
* asterisk_bridges_channels_count: The number of channels active in a
bridge.
In all cases, enough information is provided with each bridge metric
to determine a unique instance of Asterisk that provided the data, along
with the technology, subclass, and creator of the bridge.
Matt Jordan [Thu, 9 May 2019 14:41:02 +0000 (09:41 -0500)]
res_prometheus: Add Asterisk endpoint metrics
This patch adds basic Asterisk endpoint statistics to the res_prometheus
module. This includes:
* asterisk_endpoints_state: The current state (unknown, online, offline)
for each defined endpoint.
* asterisk_endpoints_channels_count: The current number of channels
associated with a given endpoint.
* asterisk_endpoints_count: The current number of defined endpoints.
In all cases, enough information is provided with each endpoint metric
to determine a unique instance of Asterisk that provided the data, as well
as the underlying technology and resource definition.
Morten Tryfoss [Tue, 21 May 2019 16:29:05 +0000 (18:29 +0200)]
res_rtp_asterisk: timestamp should be unsigned instead of signed int
Using timestamp with signed int will cause timestamps exceeding max value
to be negative.
This causes the jitterbuffer to do passthrough of the packet.
Matt Jordan [Fri, 3 May 2019 00:45:27 +0000 (19:45 -0500)]
res_prometheus: Add Asterisk channel metrics
This patch adds basic Asterisk channel statistics to the res_prometheus
module. This includes:
* asterisk_calls_sum: A running sum of the total number of
processed calls
* asterisk_calls_count: The current number of calls
* asterisk_channels_count: The current number of channels
* asterisk_channels_state: The state of any particular channel
* asterisk_channels_duration_seconds: How long a channel has existed,
in seconds
In all cases, enough information is provided with each channel metric
to determine a unique instance of Asterisk that provided the data, as
well as the name, type, unique ID, and - if present - linked ID of each
channel.
Matt Jordan [Mon, 29 Apr 2019 15:10:35 +0000 (10:10 -0500)]
pjproject/Makefile: Updates for Darwin compatible builds
This patch fixes three compatibility issues for Darwin compatible builds:
(1) Use BSD compatible command line option for sed
For some versions of BSD sed, the -r command line option is unknown.
Both GNU and BSD sed support the -E command line option for enabling
extended regular expressions; as such, this patch replaces the -r
option with -E.
(2) Look for '_' in pjproject generated symbols
In Darwin comaptible systems, the symbols generated for pjproject may be
prefixed with an '_'. When exporting these to a symbol file, the invocation
to sed has to optionally look for a prefix of said '_' character.
(3) Use -all_load/-noall_load when linking
The flags -whole-archive/-no-whole-archive are not supported by the
linker, and must instead be replaced with -all_load/-noall_load.
Matt Jordan [Thu, 3 Jan 2019 16:28:28 +0000 (10:28 -0600)]
Add core Prometheus support to Asterisk
Prometheus is the defacto monitoring tool for containerized applications.
This patch adds native support to Asterisk for serving up Prometheus
compatible metrics, such that a Prometheus server can scrape an Asterisk
instance in the same fashion as it does other HTTP services.
The core module in this patch provides an API that future work can build
on top of. The API manages metrics in one of two ways:
(1) Registered metrics. In this particular case, the API assumes that
the metric (either allocated on the stack or on the heap) will have
its value updated by the module registering it at will, and not
just when Prometheus scrapes Asterisk. When a scrape does occur,
the metrics are locked so that the current value can be retrieved.
(2) Scrape callbacks. In this case, the API allows consumers to be
called via a callback function when a Prometheus initiated scrape
occurs. The consumers of the API are responsible for populating
the response to Prometheus themselves, typically using stack
allocated metrics that are then formatted properly into strings
via this module's convenience functions.
These two mechanisms balance the different ways in which information is
generated within Asterisk: some information is generated in a fashion
that makes it appropriate to update the relevant metrics immediately;
some information is better to defer until a Prometheus server asks for
it.
Note that some care has been taken in how metrics are defined to
minimize the impact on performance. Prometheus's metric definition
and its support for nesting metrics based on labels - which are
effectively key/value pairs - can make storage and managing of metrics
somewhat tricky. While a naive approach, where we allow for any number
of labels and perform a lot of heap allocations to manage the information,
would absolutely have worked, this patch instead opts to try to place
as much information in length limited arrays, stack allocations, and
vectors to minimize the performance impacts of scrapes. The author of
this patch has worked on enough systems that were driven to their knees
by poor monitoring implementations to be a bit cautious.
Additionally, this patch only adds support for gauges and counters.
Additional work to add summaries, histograms, and other Prometheus
metric types may add value in the future. This would be of particular
interest if someone wanted to track SIP response types.
Finally, this patch includes unit tests for the core APIs.
George Joseph [Fri, 17 May 2019 23:44:37 +0000 (17:44 -0600)]
res_rtp_asterisk: Add ability to propose local address in ICE
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
Alexei Gradinari [Mon, 13 May 2019 20:37:50 +0000 (16:37 -0400)]
pjsip: replace 180 by 183 if SDP negotiation has completed
The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.
This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".
In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.
Joshua Colp [Wed, 8 May 2019 15:41:43 +0000 (15:41 +0000)]
res_rtp_asterisk: Fix sequence number cycling and packet loss count.
This change fixes two bugs which both resulted in the packet loss
count exceeding 65,000.
The first issue is that the sequence number check to determine if
cycling had occurred was using the wrong variable resulting in the
check never seeing that cycling has occurred, throwing off the
packet loss calculation. It now uses the correct variable.
The second issue is that the packet loss calculation assumed that
the received number of packets in an interval could never exceed
the expected number. In practice this isn't true due to delayed
or retransmitted packets. The expected will now be updated to
the received number if the received exceeds it.
Ben Ford [Tue, 7 May 2019 16:08:33 +0000 (11:08 -0500)]
pjsip_options.c: Allow immediate qualifies for new contacts.
When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.
Kevin Harwell [Mon, 6 May 2019 21:26:46 +0000 (16:26 -0500)]
conversions.c: Add conversions for largest max sized integer
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.
agupta [Fri, 3 May 2019 15:49:31 +0000 (21:19 +0530)]
stasis: Hangup channel for Local channel No such extension error
When we use early bridge with create and dial from stasis using Local channel
and the dialplan does not any entry the it is returned from core_local.c with
No such extension .
In such case asterisk locks up till the channel is not hangup with the error
Exceptionally long voice queue length
* Found that in such case app_control_dial fails on ast_call method and
return -1
* Since it is called from stasis_app_send_command_async and return -1 does
not cause resources to be freed and since no PBX exist it is not able to
read from channel causing exceptionally long queue
* After putting this code found that the channel was releasing immediately
and resources were freed.
George Joseph [Fri, 3 May 2019 18:31:06 +0000 (12:31 -0600)]
build: Pass --fno-partial-inlining to third-party when appropriate
When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.
projects / thirdparty / asterisk.git / log
