With newer version of linux /var/run/ is a symlink to /run/ that has
been turned into tmpfs.

Added note that if asterisk has to bind to a specific IP that
systemd has to wait until the network is up.

Added note on how to make sure that the environment variable
HOSTNAME is included.

ASTERISK-29216
Reported by: Mark Petersen
Tested by: Mark Petersen

Change-Id: Ib3e560655befd3e99eec743687144f5569533379

This reverts commit 2fe76dd8.

Reason for revert: Too many issues reported.  Need to research and correct.

ASTERISK-29230
ASTERISK-29231
Reported by: Michael Maier

Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb

Under contention it becomes possible that multiple channels will be told
they successfully obtained the lock, which is a bug.  Please refer

ASTERISK-29217

This introduces a couple of changes.

1.  Replaces requesters ao2 container with simple counter (we don't
    really care who is waiting for the lock, only how many).  This is
    updated undex ->mutex to prevent memory access races.
2.  Correct semantics for ast_cond_timedwait() as described in
    pthread_cond_broadcast(3P) is used (multiple threads can be released
    on a single _signal()).
3.  Module unload races are taken care of and memory properly cleaned
    up.

Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

On Gentoo it's possible to have multiple lua versions installed, all
with a path of /usr, so it's not possible to use the current --with-lua
option to determisticly pin to a specific version as is required by the
Gentoo PMS standards.

This environment variable allows to lock to specific versions,
unversioned check will be skipped if this variable is supplied.

Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

launch_monitor_thread is responsible for creating and initializing
the mixmonitor, and dependent data structures. There was one off
nominal path after the datastore gets created that triggers when
the channel being monitored is hung up prior to monitor starting
itself.

If this happened the monitor thread would not "launch", and the
mixmonitor object and associated objects are freed, including the
underlying datastore data object. However, the datastore itself was
not removed from the channel, so when the channel eventually gets
destroyed it tries to access the previously freed datastore data
and crashes.

This patch removes and frees datastore object itself from the channel
before freeing the mixmonitor object thus ensuring the channel does
not call it when destroyed.

ASTERISK-28947 #close

Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8

ASTERISK-28992 #close

Change-Id: Ia7d608924036139ee2520b840d077762d02668d0

Add channel reference count for PJSIP REFER. The call could be terminated
prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
occurred several minutes later, it would attempt to access a session which was
no longer valid.  Terminate event subscription if pjsip_xfer_initiate() or
pjsip_xfer_send_request() fails in transfer_refer().

ASTERISK-29201 #close
Reported-by: Dan Cropp

Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435

A prior patch segmented channel snapshots, and changed the underlying
data object type associated with ast_channel_snapshot_type stasis
messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but
now it type ast_channel_snapshot_update.

When publishing ast_channel_snapshot_type in pbx_realtime the
ast_channel_snapshot was being passed in as the message data
object. When a handler, expecting a data object type of
ast_channel_snapshot_update, dereferenced this value a crash
would occur.

This patch makes it so pbx_realtime now uses the expected type, and
channel snapshot publish method when publishing.

ASTERISK-29168 #close

Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896

Rename check_manager_enabled() and check_webmanager_enabled() to begin
with ast_ so that the symbols are automatically exported by the
linker.

ASTERISK~29184

Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9

Segfault occurs during outbound UDP registration when all
transport states are being iterated over. The transport object
in the transport is accessed, but flow transports have a NULL
transport object.

Modify to not iterate over any flow transport

ASTERISK-29210 #close

Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a

This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.

Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940

Add a check to see if the URI is a Tel URI and prevent crashing on
trying to retrieve the reason parameter.

ASTERISK-29191
ASTERISK-29219

Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
(cherry picked from commit a7aea71e)

Fix the usual compile problem when someone adds a new callback to struct
ast_channel_tech.

Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32

AST_VECTOR_SIZE() returns a size_t.  This is not always equivalent to an
unsigned long on all machines.

Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338

Added NULL pointer check to prevent Asterisk crash.

ASTERISK-29215

Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95

The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.

    https://wiki.asterisk.org/wiki/x/Xc5uAg

However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.

For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.

ASTERISK-28883 #close

Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f

Fixed wrong null object handle in
/channels/<channel_id>/play request handler.

ASTERISK-29188

Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe

Scope tracing allows you to not specify a format string or variable,
in which case it just prints the indent, file, function, and line
number.  The trace output automatically adds a newline to the end
in this case.  If you also have debugging turned on for the module,
a debug message is also printed but the standard log functionality
which prints it doesn't add the newline so you have messages
that don't break correctly.

 * format_log_message_ap(), which is the common log
   message formatter for all channels, now adds a
   newline to the end of format strings that don't
   already have a newline.

ASTERISK-29209
Reported by: Alexander Traud

Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da

In rewrite_uri asterisk was not making deep copies of strings when
changing the uri. This was in some cases causing garbage in the route
header and in other cases even crashing asterisk when receiving a
message with a record-route header set. Thanks to Ralf Kubis for
pointing out why this happens. A similar problem was found in
res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
to avoid garbage in CANCEL messages.

ASTERISK-29024 #close

Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b

ASTERISK-29211 #close

Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877

This adds support for both Digium and Sangoma user agent strings
for the Sangoma specific body supplement.

Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482

In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.

This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.

ASTERISK-29022

Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9

By default libcurl does not follow redirects, so we explicitly enable
it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl
will follow up to CURLOPT_MAXREDIRS redirects, which by default is
configured to be unlimited.

This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If
we determine at some point that this needs to be increased on
configurable it is a trivial change.

ASTERISK-29173 #close

Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30

As described in the issue, /tmp is not a suitable location for a
large amount of cached media files, since most distributions make
/tmp a RAM-based tmpfs mount with limited capacity.

I opted for a location that can be configured separately, as opposed
to using a subdirectory of spooldir, given the different storage
profile (transient files vs files that might stay there indefinitely).

This commit just makes the cache directory configurable, and changes
the default location from /tmp to /var/cache/asterisk.

ASTERISK-29143

Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d

Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6

the 'J' is missing in module description.
"PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk"

ASTERISK-29175 #close

Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a

When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.

ASTERISK-29118 #close

Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb

In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO.

ASTERISK-29148

Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2

Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a

RFC 3261 says that the Accept-Encoding header should be present
in an options response. Permitted values according to RFC 2616
are only compression algorithms like gzip or the default identity
encoding. Therefore "text/plain" is not a correct value here.
As long as the header is hard coded, it should be set to "identity".

Without this fix an Alcatel OmniPCX periodically logs warnings like
"[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed"
on a SIP Trunk.

ASTERISK-29165 #close

Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840

12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
vanished. However, the struct member itself and all seven set/uses
remained as dead code.

ASTERISK-28798

Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59

Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641.
common_recall_channel_setup() setups common things on the recalled transfer
target, but used same target as source instead trasfered.

ASTERISK-29161 #close

Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f

Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306

Operations that update queues when shared_lastcall is set lock the
queue in question, then have to lock the queues container to find the
other queues with the same member. On the other hand, __queues_show
(which is called by both the CLI and AMI) does the reverse. It locks
the queues container, then iterates over the queues locking each in
turn to display them.  This creates a deadlock.

* Moved queue print logic from __queues_show to a separate function
  that can be called for a single queue.

* Updated __queues_show so it doesn't need to lock or traverse
  the queues container to show a single queue.

* Updated __queues_show to snap a copy of the queues container and iterate
  over that instead of locking the queues container and iterating over
  it while locked.  This prevents us from having to hold both the
  container lock and the queue locks at the same time.  This also
  allows us to sort the queue entries.

ASTERISK-29155

Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41

* Instead of using the pjproject timer heap, we now use our own
  pjsip_scheduler.  This allows us to more easily debug and allows us to
  see times in "pjsip show/list registrations" as well as being able to
  see the registrations in "pjsip show scheduled_tasks".

* Added the last registration time, registration interval, and the next
  registration time to the CLI output.

* Removed calls to pjsip_regc_info() except where absolutely necessary.
  Most of the calls were just to get the server and client URIs for log
  messages so we now just save them on the client_state object when we
  create it.

* Added log messages where needed and updated most of the existong ones
  to include the registration object name at the start of the message.

Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2

* Added a ONESHOT type that never reschedules.

* Added "like" capability to "pjsip show scheduled_tasks" so you can do
  the following:

  CLI> pjsip show scheduled_tasks like outreg
  PJSIP Scheduled Tasks:

  Task Name                                     Interval  Times Run ...
  ============================================= ========= ========= ...
  pjsip/outreg/testtrunk-reg-0-00000074            50.000   oneshot ...
  pjsip/outreg/voipms-reg-0-00000073              110.000   oneshot ...

* Fixed incorrect display of "Next Start".

* Compacted the displays of times in the CLI.

* Added two new functions (ast_sip_sched_task_get_times2,
  ast_sip_sched_task_get_times_by_name2) that retrieve the interval,
  next start time, and next run time in addition to the times already
  returned by ast_sip_sched_task_get_times().

Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3

The data can be freed if the old object '_data' is the same object as
new 'data'. Because at first the object is unreferenced which can lead
to destroying it.

This could happened in res_pjsip_pubsub when the publication is updated
which could lead to segfault in function publish_expire.

Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da

ASTERISK-28933
Reported-by: Walter Doekes

Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f

This avoids the inclusion of the OpenSSL headers in the public header,
which avoids one external library dependency in res_pjsip_stir_shaken.

Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4

Currently any response from res_curl where we get an answer from the
web server, regardless of what the response is (404, 403 etc.) Asterisk
currently treats it as a success. This patch allows you to set which
codes should be considered as a failure by Asterisk. If say we set
failurecodes=404,403 then when using curl in realtime if a server gives
a 404 error Asterisk will try to failover to the next option set in
extconfig.conf

ASTERISK-28825

Reported by: Dovid Bender
Code by: Gobinda Paul

Change-Id: I94443e508343e0a3e535e51ea6e0562767639987