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

The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out
the ContactStatus AMI event when a contact is updated.
Thist change broke things which rely on old behavior.

This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to preserve old
ContactStatus behavior.
By default new behavior, i.e. the ContactStatus event will not be sent when a
device refreshes its registration.

Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46

In order to get a dump of the running process, we need to find the
pid of the main asterisk process.  This can be tricky if there are
also instances of "asterisk -r" running or if an alternate location
for asterisk.conf was specified on the command line with the -C
option that also specified an alternation location for the pid file.

So now...

1. We find the asterisk executable with "which" or the --asterisk-bin
   command line option.
2. If there's only 1 process with an executable path that matches,
   we use that pid.  If not...
3. We try "<asterisk-bin> -rx 'core show settings'" and parse the
   output to find the pidfile, then read that for the pid.  If that
   didn't work...
4. We get a list of all the pids matching <asterisk-bin> and look
   in /proc/<pid>/cmdline for a -C argument and retry the "core show
   settings" using the same -C option.  We can't parse the output
   of "ps" to get the -C path because it may contain spaces.  The
   contents of /proc/<pid>/cmdline are delimited by NULLs.  For BSDs
   we may have to mount /proc first. :(

ASTERISK-28221
Reported by: Andrew Nagy

Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c

Change-Id: I413f1583c797fb79651786cd8d0b003599f8ed10

Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.

Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.

ASTERISK-28136

Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d

ASTERISK-28087

Change-Id: I046d018015427d0916fab571b5a4f5367476f729

This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.

The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.

The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.

The default value is 'yes' for both options.

Change-Id: I16af967815efd904597ec2f033337e4333d097cd

Given a sip.conf with the following content:

setvar FOO=1
setvar BAR=42

I want my generated pjsip.conf to containt the following set_vars

set_var FOO=1
set_var BAR=42

in the matching endpoint section.

Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26

Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header.  This allows chan_pjsip to have
the same behavour as chan_sip

ASTERISK-28087 #close

Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95

This change implements a few different generic things which were brought
on by Google Voice SIP.

1.  The concept of flow transports have been introduced.  These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target.  These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity).  When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.

2.  Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.

3.  P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module.  If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.

4.  Configurable outbound extension support has been added to the outbound
registration module.  When set the extension will be placed in the
Supported header.

5.  Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.

6.  Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.

All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.

ASTERISK-27971 #close

Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58

This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.

* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.

Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.

Add comments to explain thread safety of the 'struct __priv_data'
bitfields.

Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a

Change-Id: I1297de78628773ca368e687c6f148bf74857cae9

Change-Id: I7ddfed4cd6549a0cd458e4d5cf9ac95d784de6cb

When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock
was ever used, so it can be recorded in the REF_DEBUG destructor entry.

Create contrib/scripts/reflocks.py to process locking used by
allocator.  This can be used to identify places where
AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage.

Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4

In the original commit introducing the feature the column in the alembic
script was called 'suppress_q850_reason_header'.
In the code however the option is called 'suppress_q850_reason_headers'
(trailing 's'). This leads to errors when ARI push configuration is used.

Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f

This is removed in favor of ./configure --with-jansson-bundled.  The
install-unpackaged command would only install jansson once, so once
installed it would never update, where the bundled copy will be kept up
to date.

Change-Id: Ideab1f65419608d3795aa608e9da873823cc42d3

ASTERISK-27988

Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843

When mobile SIP clients register with Asterisk that use some sort of
push notifications, the URI can get quite lengthy due to the
additional push-service annotations (things like tokens, pn-type, etc.)
contained in it.

ASTERISK-28022 #close

Change-Id: I4c7ceadc3bb405f3daf722641c8cd5ca4188cc37

Change-Id: I11641d65592536dea9cbca5aa94a24c25d24dd5f

Mention need to compile Asterisk with systemd development package
installed.

ASTERISK-27968

Change-Id: Ib3a973be403c61cbe09572b0f912fb1aa1bff026

A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.

ASTERISK-27949
Reported-by: Ross Beer

Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1

pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response.  We handle this correctly.  There have
been reported cases where the To tag is the same but we still need to
follow the media.  The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime.  The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.

So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.

The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.

Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer

The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set
to "/tmp" instead of "/some/directory".

Variables set on the command line or that are already in the
environment now take predecence over variables set in the config files.

ASTERISK-27846
Reported by: Ted G

Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387

The script remains compatible with Python 2.7 but now also works with
Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip.

ASTERISK-27811

Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835

ASTERISK-27820

Change-Id: I718ddb000fe5184b1bdc7759da67a370a7520144

* Consistently use spaces in rest-api-templates/asterisk_processor.py.
* Exclude third-party from docs/full-en_US.xml.
* Add docs/full-en_US.xml to .gitignore.
* Use list() to convert python3 view.
* Use python3 print function.
* Replace cmp() with equivalent equation.
* Replace reference to out of scope subtype variable with name
  parameter.
* Use unescaping triple bracket notation in mustache templates where
  needed.  This causes behavior of Python2 to be maintained when using
  Python3.
* Fix references to has_websocket / is_websocket in
  res_ari_resource.c.mustache.
* Update calculation of has_websocket to use any().
* Use unicode mode for writing output file in transform.py.
* Replace 'from swagger_model import *' with explicit import of required
  symbols.

I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the
print syntax has been fixed.

Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33

ASTERISK-27770

Change-Id: Ib87e0483c785542238cfe34c1e884d5a31edfaab

ASTERISK-27769

Change-Id: Ieb13293cd67481f3a33f58f6f7c8c3ee1e338e7a

Asterisk does not need the development package of libltdl, because it does not
use any symbol of -lltdl directly. Instead, it uses the runtime package via the
shared library -lodbc. On the supported platforms, that shared library declares
its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have
failed.

ASTERISK-27745

Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba

ASTERISK-27738

Change-Id: I7ca620e3c4dfb4b064a19382c4915aeb42a2a09f

ASTERISK-27736

Change-Id: I4cafc8973349d50a7cb7919ddf0bb1aaef4bfc3e

Because the code review system Gerrit creates merge conflicts even when one line
apart another change happened, the previous update to the FreeBSD libraries had
to be rebased via Git. Because of a break for training of the original
contributor, this rebase was done by another contributor and the variant for
Asterisk 13 was cherry-picked to all branches. By this, dependencies for new
features added in newer Asterisk version got lost. This can be seen, when not
the original path set but a previous patch set is compared.

This change here fixes this by adding those (optional) dependencies for
Asterisk 15 and newer (again).

ASTERISK-27686

Change-Id: I6638a3d0dc37ad4ff5f94be15463e3dd8a2bfe74

ASTERISK-27729

Change-Id: I7a706d51375d54cf5e36d32397bfe09a48670804

* Fix --tarball-config so the option doesn't cause an error.

* Allow for missing /etc/os-release.

* Add a sleep between tarballing the coredump and removing the
  output directory to allow the filesystem to settle.

Change-Id: I73e03b13087978bcc7f6bc9f45753990f82d9d77

deleted
 autoconf gcc libsamplerate sqlite

changed
 binutils to libbfd
 freetds-devel to freetds
 gmime2 to gmime26
 mysql55-client to mysql57-client

added
 alsa-lib bison bzip2 cclient corosync doxygen libedit flex graphviz
 libhoard libical libilbc libltdl lua neon newt net-snmp
 openldap-client openssl patch pkgconf portaudio postgresql10-client
 python radcli speexdsp subversion uriparser xmlstarlet libzip

ASTERISK-27686

Change-Id: Ibe88c9b26e59c30d26cdb313a3ef01c9f37ac80d

deleted
 jack sqlite

renamed
 freetds-0.63p1-msdblib to freetds
 mysql-client to mariadb-client

added
 bison bzip2 c-client doxygen e2fsprogs graphviz gsm libical jansson libltdl
 lua neon net-snmp libsrtp portaudio-svn postgresql-client python speexdsp
 subversion uriparser xmlstarlet
 fftw3 libsndfile

ASTERISK-27684

Change-Id: I26bdcb0a1d0e484a8dad1052da97f194aefd3370

Add a new script that can read from legacy realtime peers & generate
an sql file for populating pjsip endpoints, identify, and aor records.

ASTERISK-27348 #close

Change-Id: Idd3d7968a3c9c3ee7936d21acbdaf001b429bf65

ASTERISK-27670

Change-Id: I07a8ef8bbd6001e25711fa1bff152eb6c9efa729

Jansson is thread safe for all read-only functions and reference
counting starting v2.11.  This allows simplification of our code and
removal of locking around reference counting and dumping.

Change-Id: Id985cb3ffa6681f9ac765642e20fcd187bd4aeee

ASTERISK-27555

Change-Id: Idc36e91db30c0163c560d04c5a82bca5d6ce92a8