Change-Id: I050ceffe5a133d5add2dab46687209813d58f597

* The bridging core no longer uses the stasis cache for bridge
  snapshots.  The latest bridge snapshot is now stored on the
  ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache
  is no longer used:
    ast_bridge_topic_cached()
    ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no
  longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the
  container of all bridges.  Users formerly calling
  ast_bridge_cache() can use the new function to iterate over
  bridges and retrieve the latest snapshot directly from the
  bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to
  ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve
  the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic
  not a cached one so you can't use stasis cache functions on it
  either.

* The ast_bridge_snapshot_type() stasis message now has the
  ast_bridge_snapshot_update structure as it's data.  It contains
  the last snapshot and the new one.

* cdr, cel, manager and ari have been updated to use the new
  arrangement.

Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369

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

Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.

As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()

The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.

The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.

The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.

ast_channel_snapshot_get_latest() still returns the latest snapshot.

The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.

ASTERISK-28102

Change-Id: I9334febff60a82d7c39703e49059fa3a68825786

The marker bit set on the voice packet indicates the start
of a new stream and a new time stamp.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet.

If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.

This bug was caught using Cisco ip-phone SPA50X and codec g722.
On SIP session update the SPA50X resets stream indicating it with market bit
and a new timestamp is twice smaller then the previous.

ASTERISK-28162 #close

Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620

Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.  Remove ao2_container_alloc macro.

Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088

This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.

This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.

There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.

ASTERISK-28103

Change-Id: I99bee23895baa0a117985d51683f7963b77aa190

The res_ari(POST /channels/create handler) deos not check the endpoint
parameter length. And it causes core
dump.
Fixed it to check the parameter length. Also fixed memory leak.

ASTERISK-28169

Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993

PJSIP assumes that these header names are not allocated, does not clone
the name strings when reusing headers.

Block unload of res_pjsip_caller_id until shutdown to ensure static
memory stays valid.  It was previously unsafe to unload while any
sessions are active.

Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f

The presence of Record-Route in re-invites is optional, thus it is
important to make sure the dialog doesn't have a routset before
rewriting the contact header.

ASTERISK-28129 #close

Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc

When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.

Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836

The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.

This patch replaces the pipe with a comma.

ASTERISK-28150

Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719

The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests.  Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.

Longer running tasks with the round-robin method can delay processing
tasks.

* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.

Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd

ASTERISK-28087

Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4

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

The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.

For example if there are 10k endpoints the module makes 10k requests
of these 10k records.

Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.

This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.

ASTERISK-28137 #close

Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b

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

Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b

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

Change-Id: I1a00ca22d59d6b6d2166aa56f0e9338a33e5ac60

These macros have been documented as legacy for a long time but are
still used in new code because they exist.  Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc

These macro's are also removed.  Only ao2_container_alloc remains due to
it's use in over 100 places.

Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a

Most were in comments.  A couple were in warning messages.

Pointed out by Jonathan H on the Asterisk users mailing list.

Change-Id: I6286939dff5d0a27a2758140570106f1cb351855

* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.

Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377

This patch sets the callerid_tag to empty string by default.

If the callerid_tag is set to NULL then the tag does not
become part of a connected line update.
For example:
Alice's tag is "Alice".
Bob's tag is empty.
Charlie's tag is "Charlie".
Alice calls Bob and then does attended transfer to Charlie.
When Alice hangs up the CONNECTEDLINE(tag) is "Alice"
on the interception routine on the Charlie's channel, but should be empty.

Ths patch also fix memory leaks if there are more then one options
"callerid", "callerid_tag", "voicemail_extension" and "contact_user"
in the pjsip.conf endpoint definition.

Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4

The return status when there was no change in statsd.conf was incorrect.
This resulted in the wrong status message on the CLI when reloading the
module.

* Fixed cleanup on initial load if initializing statsd failed.

Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481

I think this module is so screwed up that it doesn't work anymore.  Even
with these attempts to fix things it still won't gracefully shut down.
The module refs will not go to zero to allow unloading the module.

* Fix module ref counting dealing with the SMDI interface object.  There
were several off-nominal paths that unbalanced the module ref count.  Also
the destructor freed the ao2 object itself which is bad.  Made the
smdi_read thread not hold its own ref to the SMDI interface object so when
all refs go away the destructor will stop the listener thread.

* Fixed the smdi_load() return code of 1 concerning the number of
listeners.  The test was inverted.

Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784

This module is an optional dependency of a couple of other modules.  If it
declines to load, it then forces other modules that can optionally use
this module to also decline.

* Made use the default configuration if the config file does not exist and
simplified some of the logic.

Change-Id: Ib93191f1fe28c0dd9ebe3d84c7762b32f83c4eb9

This module is an optional dependency of many modules.  If it declines to
load it then forces other modules that can optionally use this module to
also decline.

* Made use default configuration if there is a config error or the config
file does not exist.

Change-Id: If1068a582ec54ab7fb437265cb5370a97a825737

CLI command 'pjsip show contacts' inefficiently make a lot of DB requests.

For example if there are 10k aors then asterisk requests these 10k records
of aor and then does 10k requests of contact - one request per aor.

Even if use 'like <pattern>' the asterisk requests all aor's and contact's
records and then filters them by itself.

This patch gathers contact's container by
- retrieving all dynamic contacts by regex (filtered by reg_server)
- retrieving all aors with permanent contacts
- finally filters container by regex

ASTERISK-28077 #close

Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618

Fixed an issue that resulted in "Allocation failed" each time an ARI
request was made to start playing MOH on a bridge.

In bridge_moh_create() we were attaching the after bridge callbacks to
chan which is the ;1 channel of the unreal channel pair.  We should have
attached them to the ;2 channel which is pushed into the bridge by
ast_unreal_channel_push_to_bridge().  The callbacks are called when the
specific channel leaves the bridging system.  Since the ;1 channel is
never put into a bridge the callbacks never get called.  The callbacks
then never remove the moh_wrapper from the app_bridges_moh container.  As
a result we cannot find the channel associated with the wrapper to start
MOH because it has hungup.  This is the reason causing the reported issue.

* Rather than using after bridge callbacks to cleanup, we now have
moh_channel_thread() doing the cleanup when the channel hangs up.

* Fixed moh_channel_thread() accumulating control frames on the stasis
bridge MOH channel until MOH is stopped.  Control frames are no longer
accumulated while MOH is playing.

* Fixed channel ref counting issue.  stasis_app_bridge_moh_channel() may
or may not return a channel ref.  As a result ast_ari_bridges_start_moh()
wouldn't know it may have a channel ref to release.
stasis_app_bridge_moh_channel() will now return a ref with the channel it
returns.

* Eliminated RAII_VAR in bridge_moh_create().

ASTERISK-26094 #close

Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c

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

On SQL error there is not diagnostic information about this error.
There is only
WARNING res_odbc.c: SQL Execute error -1!

The function ast_odbc_print_errors calls a SQLGetDiagField to get the number
of available diagnostic records, but the SQLGetDiagField returns 0.
However SQLGetDiagRec could return one diagnostic records in this case.

Looking at many example of getting diagnostics error information
I found out that the best way it's to use only SQLGetDiagRec
while it returns SQL_SUCCESS.

Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR
to res_config_odbc.

ASTERISK-28065 #close

Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1

In order to do this and provide good feedback, a new macro was
created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and
path setups for the library then compiles, links and runs a supplied
code fragment to do the final determination.  In this case, the
final code fragment compares UNBOUND_VERSION_MAJOR
and UNBOUND_VERSION_MINOR to determine if they're greater than or
equal to 1.5.

Since we require version 1.5, some code in res_resolver_unbound
was also simplified.

ASTERISK-28045
Reported by: Samuel Galarneau

Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72

This change raises a testsuite event to provide what port
Asterisk has actually allocated for RTP. This ensures that
testsuite tests can remove any assumption of ports and instead
use the actual port in use.

ASTERISK-28070

Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044

* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
  safe to just use ast_json_ref.

Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped.  This was previously
  avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null().  The
  "?" is a new feature of ast_json_pack starting with Asterisk 16.

Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48

ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call
the destroy callback.  Remove call to ast_free from ast_rtp_new, leave
it to rtp_engine.c to initiate the full cleanup.  Add error detection
for the ssrc_mapping vector initialization.  In rtp_allocate_transport
set rtp->s = -1 in the failure path where we close that FD to ensure we
don't try closing it twice.

ASTERISK-27854 #close

Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451

'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults
if they are not present in the updated configuration file.

Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670

The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.

* No longer allocate memory from the stack in a loop to parse the header
values.  NOTE: There is a slight API change when using the passed in
strings as is.  We now require the passed in strings to no longer have
leading or trailing whitespace.  This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.

ASTERISK-28013 #close

Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a

This adds a module which registers a CLI command that can set the
REMB bitrate value for REMB as it enters or exits Asterisk. This
allows you to ignore what Asterisk or a client produces and is
useful for demonstrations.

This does not generate REMB frames, however, but just modifies
them as they flow to or from a channel.

Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec

This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.

ASTERISK-28059

Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189