Original commit: cfbf5fbe

Change-Id: I34a841d73c429ca8d944481f8dccb756ee231c9c

Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.

Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa

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.

Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7

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.

ASTERISK-28018

Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06

ASTERISK-28234
Reported-by: Kirsty Tyerman

Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1

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.

ASTERISK-28403

Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59

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.

ASTERISK-28403

Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42

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:

[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address

This causes both 192.168.1.1 and 1.2.3.4 to be advertized.

Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db

When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.

This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.

ASTERISK-28401

Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c

There a long history here:

In commit dd1e62c0 has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf

After that, the commit 4b50e3f1 fix the
sample file.

We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.

Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c

Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.

res_pjsip_refer:  Configures PJSIP norefersub capability accordingly.

Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.

This is useful for Cisco switches that do not follow RFC4488.

ASTERISK-28375 #close
Reported-by: Dan Cropp

Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9

chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP.  This new flag allows chan_pjsip to have the same
behavior as chan_sip.

ASTERISK-28322 #close

Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a

Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857

The res_pjsip_websocket module requires the res_http_websocket
module so ensure it is loaded. As well the res_pjsip_notify
module needs the pjsip_notify.conf configuration file so
ensure it is installed.

ASTERISK-28272

Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd

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

When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.

This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.

This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.

ASTERISK-28277

Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6

The option value "sdp" for some of the settings was removed a while back,
however the sample conf was not updated.

This patch removes any wording with regards to the old "sdp" option value,
and adjusts the defaults to what they are now.

ASTERISK-28263

Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445

You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes.  These can be used anywhere
the mailbox is specified.

Example:
[general]
aliasescontext = myaliases

[default]
1234 = yadayada

[myaliases]
4321@devices = 1234@default

Now you can use 4321@devices to refer to the 1234@default mailbox.

This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.

Change-Id: I395b9205c91523a334fe971be0d1de4522067b04

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

The module has been removed, so it shouldn't be in the default config any more.

Change-Id: Ie7e09f00f9c9a885574e29478250de4c2cefd9f1

This reverts commit 29115e23.

That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf.  This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.

ASTERISK-28151
Reported by: Ronald Raikes

Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15

When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.

This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.

The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.

Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.

ASTERISK-28119

Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423

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

This officially deprecates chan_sip in Asterisk 17+.  A warning is
printed upon startup or module load to tell users that they should
consider migrating.  chan_sip is still built by default but the default
modules.conf skips loading it at startup.

Very important to note we are not scheduling a time where chan_sip will
be removed.  The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.

Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93

Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90

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

PARKINGSLOT was deprecated in Asterisk 12 but the sample config was not
updated to reflect that.

Change-Id: I3e087c19d9ee587094fa5304102d8084a79c2b3c

When networks experience disruptions, there can be large gaps of time
between receiving packets. When strictrtp is enabled, this created
issues where a flood of packets could come in and be seen as an attack.
Another option - seqno - has been added to the strictrtp option that
ignores the time interval and goes strictly by sequence number for
validity.

Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71

app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers.  It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled.  For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.

Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.

This paves the way for disabling the caching of stasis subscription
change messages.

Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.

ASTERISK-27121

Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee

Change disables loading of res_hep.so in default installation.  Loading
res_hep has a performance impact whether it's used or not.  This disables
loading of it in sample config files.

Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0

Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849

It is valid for a config file to be empty or contain only comments, but
not valid for a config value to be set when no uncommented context
exists.  This caused an error to be loged numerous times during start
when loading the default pjsip.conf.

Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6

Remove the note that SRV records are not supported as that is no longer
true.

ASTERISK-27993

Change-Id: Id0dd6ef40e52702be9727a2b6122216cb00bb4ca

Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52

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

The Websocket transport uses the built-in HTTP server. As a result
the TLS configuration is done in http.conf and not in pjsip.conf.

This change adds a warning if this is configured in pjsip.conf and
also clarifies in the sample configuration file.

Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9

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

ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc.  In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.

For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.

To control this behavior, the following options have been added to
confbridge.conf:

bridge_profile/enable_events:  This must be enabled on any bridge where
events are desired.

user_profile/send_events:  This must be set for a user profile to send
events.  Different user profiles connected to the same bridge can have
different settings.  This allows admins to get events but not normal
users for instance.

user_profile/echo_events:  In some cases, you might not want the user
triggering the event to get the event sent back to them.  To prevent it,
set this to false.

A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used.  This allows participant A's video
stream to appear as the same label to all other participants.

Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e

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

Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers.  It has three timeout constants that this commit converts to
channel-level configuration options:

* firstdigit_timeout: Default time (ms) to detect first digit

* interdigit_timeout: Default time (ms) to detect following digits

* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match.  This happens when the dialed digits match a number in the current
context but are also the prefix of another number.

Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213