George Joseph [Wed, 18 Dec 2019 14:30:50 +0000 (07:30 -0700)]
Revert "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up"
There are reports that this commit causes deadlocks when issuing
a "pri show" CLI command and a channel is in the process of being
hung up. Further work is in progress to determine the cause of
the deadlock and provide a permanent fix.
ASTERISK~28605 #keeping open
Reported by: Dirk Wendland
Joshua C. Colp [Tue, 17 Dec 2019 01:18:37 +0000 (21:18 -0400)]
json: Support older jansson versions.
The use of '?' is a fairly new addition to jansson and is not
supported in the version of jansson that can be used by 13.
This change returns to previous supported behavior and removes
usage of '?'.
Joshua C. Colp [Mon, 16 Dec 2019 12:35:31 +0000 (08:35 -0400)]
configure: Add check for MySQL client bool and my_bool type usage.
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
Sean Bright [Wed, 4 Dec 2019 21:26:46 +0000 (16:26 -0500)]
res_pjsip_registrar.c: Prevent potential double free if AOR is not found
The simple fix here is simply to NULL out username and password after we call
ast_free on them. Unfortunately, I noticed that we weren't checking for
allocation failures for username and password, and adding those checks made
things noisy and cumbersome.
So instead we partially rollback the recent LGTM patch, and move the alloca
calls into find_aor_name().
Sean Bright [Wed, 4 Dec 2019 21:12:39 +0000 (16:12 -0500)]
res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases
We're appropriately sizing the id_domain_alias buffer, but then copying the data
into the id_domain one. We were then using the uninitialized id_domain_alias
buffer we just allocated.
This is ASTERISK~28641 adjacent, but significant enough to warrant its own
patch.
Frederic LE FOLL [Wed, 27 Nov 2019 18:11:33 +0000 (19:11 +0100)]
chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.
During capabilities selection (joint capabilities of us and peer,
configured capability for this peer, or general configured
capabilities), if sip_new() does not keep framing information,
then directmedia activation will fail for any framing different
from default framing.
George Joseph [Fri, 22 Nov 2019 16:39:36 +0000 (09:39 -0700)]
channel.c: Resolve issue with receiving SIP INFO packets for DTMF
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)
Ben Ford [Mon, 21 Oct 2019 19:55:06 +0000 (14:55 -0500)]
chan_sip.c: Prevent address change on unauthenticated SIP request.
If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.
This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.
George Joseph [Thu, 24 Oct 2019 17:41:23 +0000 (11:41 -0600)]
manager.c: Prevent the Originate action from running the Originate app
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.
Ben Ford [Fri, 8 Nov 2019 19:21:15 +0000 (13:21 -0600)]
res_pjsip_session.c: Check for port of zero on incoming SDP.
If a re-invite comes in initiating T.38, but there is no c line in the
SDP and the port is also 0, a crash can occur. A check is now done on
the port to see if the steam is already declined, preventing the crash.
The logic was moved to res_pjsip_session.c because it is handled in a
similar manner in later versions of Asterisk.
Salah Ahmed [Fri, 15 Nov 2019 17:34:26 +0000 (11:34 -0600)]
res_pjsip_t38: T.38 error correction mode selection at 200 ok received
if asterisk offer T38 SDP with none error correction scheme and
the endpoint respond with redundancy EC scheme, asterisk switch
to that mode. Since we configure the endpoint as none EC mode
we should not switch to any other mode except none.
following logic implemented in code.
1. If asterisk offer none, and anything except none in answer
will be ignored.
2. If asterisk offer fec, answer with fec, redundancy and none will
be accepted.
3. If asterisk offer redundancy, answer with redundancy and none
will be accepted.
George Joseph [Tue, 19 Nov 2019 14:51:56 +0000 (07:51 -0700)]
CI: Increase clone depth and do better cleanup
The original clone depth of 10 was causing the need to rebase
changes whose parent was older than the 10 commits. The clone
depth has been increased to 100.
Workspace cleanup was only happening for successful builds which
wasn't enough to keep the 8G workspace in-memory drives on the
docker slaves from filling up. Now the workspaces are cleaned up
after every build regardless of success/failure. If you need to
preserve builds temporarily, you can log into Jenkins/Manage
Jenkins/Configure System and change the CLEANUP_WS_* environment
variable for the job type you're troubleshooting to "FALSE".
chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and
may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events.
If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection
and no Q.921 reconnection sequence occurs, chan_dahdi will keep
seeing span status "Down" at the end of alarm.
This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit
unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects
a disconnection of Q.921 layer and this will clear DCHAN_UP if required.
Martin Tomec [Thu, 7 Nov 2019 17:54:06 +0000 (17:54 +0000)]
func_curl.c: Support custom http headers
When user wants to send json data, the default Content-Type header
is incorect (application/x-www-form-urlencoded). This patch allows
to set any custom headers so the Content-Type header can be
overriden. User can set multiple headers by multiple calls of
curlopt(). This approach is not consistent with other parameters,
but is more readable in dialplan than one call with multiple
headers.
George Joseph [Wed, 6 Nov 2019 11:47:17 +0000 (04:47 -0700)]
stasis: Don't hold app_registry and session locks unnecessarily
resource_events:app_handler() was locking the session, then
attempting to determine if the app had debug enabled which
locked the app_registry container. res_stasis:__stasis_app_register
was locking the app_registry container then calling app_update
which caused app_handler (which locks the session) to run.
The result was a deadlock.
* Updated resource_events:app_handler() to determine if debug was
set (which locks the app_registry) before obtaining the session lock.
* Updated res_stasis:__stasis_app_register to release the app_registry
container lock before calling app_update (which locks the sesison).
Michael Cargile [Tue, 5 Nov 2019 18:16:48 +0000 (13:16 -0500)]
app_amd: Fixed timeout issue
ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.
* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began
OpenSSL can not tolerate if the packet sent out does not
match the length that it provided to the sender. This change
lies and says that each time the full packet was sent. If
a problem does occur then a retransmission will occur as
appropriate.
Kevin Harwell [Wed, 13 Nov 2019 23:24:48 +0000 (17:24 -0600)]
bridge_softmix: clear hold when joining a softmix bridge
MOH continues to play to a channel if that channel was on hold prior to
entering a softmix bridge. MOH will not stop even if the original "holder"
attempts an unhold.
For the most part a softmix bridge ignores holds, so a participating channel
shouldn't join while on hold. This patch checks to see if the channel joining
the softmix bridge is currently on hold. If so then it indicates an unhold.
Joshua Colp [Tue, 12 Nov 2019 11:00:44 +0000 (07:00 -0400)]
parking: Use channel snapshot instead of channel.
There exists a scenario where a thread can hold a lock on the
channels container while trying to lock a bridge. At the same
time another thread can hold the lock for said bridge while
attempting to retrieve a channel. This causes a deadlock.
This change fixes this scenario by retrieving a channel snapshot
instead of a channel, as information present in the snapshot
is all that is needed.
Kevin Harwell [Wed, 23 Oct 2019 21:34:27 +0000 (16:34 -0500)]
various files - fix some alerts raised by lgtm code analysis
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
Joshua Colp [Thu, 24 Oct 2019 10:21:31 +0000 (07:21 -0300)]
res_ari_events: Add module reference when a WebSocket is open.
This change ensures that the module isn't unloaded when a
WebSocket is open. Previously it was possible to unload the
module manually or during shutdown which could cause a crash
when any active WebSockets were terminated.
Salah Ahmed [Thu, 10 Oct 2019 11:56:34 +0000 (13:56 +0200)]
Crash during "pjsip show channelstats" execution
During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.
app_voicemail.c: Support multiple file formats for forwarded messages.
If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.
cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12
PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef
(column default values), which has been deprecated since version 8.0.
Since then, the official/correct/supported way to retrieve the column
default value from the catalog is function pg_catalog.pg_get_expr().
This change breaks compatibility with pre-8.0 PostgreSQL servers,
but has reached end-of-support more than a decade ago.
cdr_pgsql and res_config_pgsql still have support for pre-7.3
servers, but cleaning that up is perhaps a topic for a major release,
not this bugfix.
Kevin Harwell [Thu, 10 Oct 2019 20:30:06 +0000 (15:30 -0500)]
res_pjsip_mwi: potential double unref, and potential unwanted double link
When creating an unsolicited MWI aggregate subscription it was possible for
the subscription object to be double unref'ed. This patch removes the explicit
unref as it is not needed since the RAII_VAR will handle it at function end.
Less concerning there was also a bug that could potentially allow the aggregate
subscription object to be added to the unsolicited container twice. This patch
ensures it is added only once.
George Joseph [Wed, 9 Oct 2019 14:32:45 +0000 (08:32 -0600)]
pjproject_bundled: Replace earlier reverts with official fixes.
Issues in pjproject 2.9 caused us to revert some of their changes
as a work around. This introduced another issue where pjproject
wouldn't build with older gcc versions such as that found on
CentOS 6. This commit replaces the reverts with the official
fixes for the original issues and allows pjproject to be built
on CentOS 6 again.
Kevin Harwell [Wed, 9 Oct 2019 20:17:59 +0000 (15:17 -0500)]
pbx: deadlock when outgoing dialed channel hangs up too quickly
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)
Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.
Kevin Harwell [Wed, 2 Oct 2019 16:08:05 +0000 (11:08 -0500)]
res_pjsip_mwi: use an ao2_global object for mwi containers
On shutdown it's possible for the unsolicited mwi container to be freed before
other dependent threads are done using it. This patch ensures this can no
longer happen by wrapping the container in an ao2_global object. The solicited
container was also changed too.
Kevin Harwell [Tue, 1 Oct 2019 20:54:59 +0000 (15:54 -0500)]
res_pjsip/res_pjsip_mwi: use centralized serializer pools
Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they
both implemented their own serializer pool functionality that was pretty much
identical in each of the source files. This patch removes the duplicated code,
and uses the new 'ast_serializer_pool' object instead.
Additionally res_pjsip_mwi enables a shutdown group on the pool since if the
timing was right the module could be unloaded while taskprocessor threads still
needed to execute, thus causing a crash.
Kevin Harwell [Tue, 1 Oct 2019 20:43:47 +0000 (15:43 -0500)]
serializer: move/add asterisk serializer pool functionality
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.
There are some warning messages which are not informative without endpoint:
"No registered subscribe handler for event presence.winfo"
"No registered publish handler for event presence"
This patch adds an endpoint name to these messages.
projects / thirdparty / asterisk.git / log
