We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0

This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.

A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.

The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.

ASTERISK-26732 #close
Reported by Dan Jenkins

Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5

When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered

ASTERISK-26869 #close

Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525

This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.

ASTERISK-26863

Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a

This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.

Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.

ASTERISK-26863 #close

Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79)

* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.

Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48

Tabs > spaces. Always.

Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1

* res_musiconhold.c: Ensure the general section is not treated as
a moh class.

ASTERISK-26353 #close

Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d

This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.

Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.

ASTERISK-26685

Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647

When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.

Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.

This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.

ASTERISK-26842 #close
Reported by Mark Michelson

Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12

res_config_pgsql should match the behavior of other realtime backend
drivers so that queue_log can disable adaptive logging.

ASTERISK-25628 #close
Reported by: Dmitry Wagin

Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372

The find_table() functions NULL or a locked table pointer. We are
not consistently calling release_table() in failure paths.

Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544

When a subscription was being recreated and the endpoint wasn't
found, we were trying to unref the endpoint.  This was causing
FRACKs.  Removed the unref.

ASTERISK-26823 #close

Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164

This change fixes an assumption in res_pjsip that a contact will
always have a status. There is a race condition where this is
not true and would crash. The status will now be unknown when
this situation occurs.

ASTERISK-26623 #close

Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5

Outbound registration now subscribes to network change events
published by res_stun_monitor and refreshes all registrations
when an event happens.

The 'pjsip send (un)register' CLI commands were updated to accept
'*all' as an argument to operate on all registrations.

The 'PJSIP(Un)Register' AMI commands were also updated to
accept '*all'.

ASTERISK-26808 #close

Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25

This change updates the documentation for the outbound_proxy option
to ensure it is consistently stated that a full SIP URI must be
provided for the option.

The res_pjsip_outbound_registration module has also been changed so
that the provided outbound_proxy value is checked to ensure it is a
URI and if not an error is output stating so.

ASTERISK-26782

Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593

 * Removed all 2.5.5 functional patches.
 * Updated usages of pj_release_pool to be "safe".
 * Updated configure options to disable webrtc.
 * Updated config_site.h to disable webrtc in pjmedia.
 * Added Richard Mudgett's recent resolver patches.

Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7

* A missing AST_LIST_UNLOCK() in find_table()

* The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were
  not consistently locking before calling it.

* There were a handful of other places where pgsqlConn was accessed
  directly without appropriate locking.

Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed

The initial motivation for this patch was to properly handle memory
allocation failures - we weren't checking the return values from the
various LDAP library allocation functions.

In the process, because update_ldap() and update2_ldap() were
substantially the same code, they've been consolidated.

Change-Id: Iebcfe404177cc6860ee5087976fe97812221b822

All of the realtime backends create artificial ast_categorys to pass
back into the core as query results. These categories have no filename
or line number information associated with them and the backends differ
slightly on how they create them. So create a couple helper macros to
help make things more consistent.

Also updated the call sites to remove redundant error messages about
memory allocation failure.

Note that res_config_ldap sets the category filename to the 'table name'
but that is not read by anything in the core, so I've dropped it.

Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897

The inbound authentication object is supposed to be immutable when it is
stored in sorcery.  However, the immutable property is violated if the
authentication object does not have a realm set.

The immutable contract violation has a different effect depending upon
what sorcery back end is used.  If it is the config file back end you
would get the same object back until res_pjsip is reloaded.  If it is the
real-time or AstDB back end you would get a new object on each query.  If
it is cached you would get the same object back until it is refreshed from
the database.

Once an inbound authentication object has its realm set it may or may not
get updated again if the default_realm changes.

If the same authentication object is used for inbound and outbound
authentication then the immutable violation can make it very hard to
determine why the outbound authentication now fails.  The only diagnostic
message is a complaint about no realms matching when it had worked
earlier.  It fails because of the difference in behaviour for an empty
realm setting between inbound and outbound authentication objects.

* Fixed the sorcery object immutable violation by creating a new object
and setting the default_realm on it instead.  The new object is a shallow
copy for speed.

* The auth_store thread storage no longer holds an auth ref.  It
interferes with the shallow copy and never needed a ref anyway.

ASTERISK-26799 #close

Change-Id: I2328a52f61b78ed5fbba38180b7f183ee7e08956

There was code attempting to update the artificial authentication object
whenever the default_realm changed.  However, once the artificial
authentication object was created it would never get updated.  The
artificial authentication object would require a system restart for a
change to the default_realm to take effect.

ASTERISK-26799

Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802

Using the same auth section for inbound and outbound authentication is not
recommended.  There is a difference in meaning for an empty realm setting
between inbound and outbound authentication uses.

An empty inbound auth realm represents the global section's default_realm
value when the authentication object is used to challenge an incoming
request.  An empty outgoing auth realm is treated as a don't care wildcard
when the authentication object is used to respond to an incoming
authentication challenge.

ASTERISK-26799

Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce

* Removed overloaded unmatched response ignore.  We obviously sent the
request so we shouldn't ignore it because it isn't new work.

ASTERISK-26669
ASTERISK-26738

Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37

When listing a container, we now print the number of objects
in the container at the end of the list.

Change-Id: I791cbc3ee9da9a2af9adc655164b5d32953df812

OpenLDAP will raise an error when we try to delete an LDAP attribute
that doesn't exist. We need to filter out LDAP_MOD_DELETE requests
based on which attributes the current LDAP entry actually has. There
is of course a small window of opportunity for this to still fail,
but it is much less likely now.

Change-Id: I3fe1b04472733e43151563aaf9f8b49980273e6b

Extraneous line numbers were being output in many log messages. These
have been removed.

Change-Id: Ice9efa3d252ee87f37fa8f5ea852fda482675431

The code in update_ldap() and update2_ldap() was using both Asterisk's
memory allocation routines as well as OpenLDAP's. I've changed it so
that everything that is passed to OpenLDAP's functions are allocated
with their routines.

Change-Id: Iafec9c1fd8ea49ccc496d6316769a6a426daa804

The "_general" configuration section allows administrators to provide
both general configuration options (host, port, url, etc.) as well as a
global realtime-to-LDAP-attribute mapping that is a fallback if one of
the later sections do not override it. This neglected to exclude the
general configuration options from the mapping. As an example, during
my testing, chan_sip requested 'port' from realtime, and because I did
not have it defined, it pulled in the 'port' configuration option from
"_general." We now filter those out explicitly.

Change-Id: I1fc61560bf96b8ba623063cfb7e0a49c4690d778

We always treat the first change of our modification batch as a
replacement when it sometimes is actually a delete. So we have to pass
the correct arguments to the OpenLDAP library.

ASTERISK-26580 #close
Reported by: Nicholas John Koch
Patches:
	res_config_ldap.c-11.24.1.patch (license #6833) patch uploaded
	by Nicholas John Koch

Change-Id: I0741d25de07c9539f1edc6eff3696165dfb64fbe

When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has
already destroyed the ast_config struct for us. Trying to do it again
results in a crash.

Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6

This creates the following:
* Asterisk's internal representation of an SDP
* An API for translating SDPs from one format to another
* An implementation of a translator for PJMEDIA

Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b

The realtime framework allows for components to look up values using a
LIKE clause with similar syntax to SQL's. pbx_realtime uses this
functionality to search for pattern matching extensions that start with
an underscore (_).

When passing an underscore to SQL's LIKE clause, it will be interpreted
as a wildcard matching a single character and therefore needs to be
escaped. It is (for better or for worse) the responsibility of the
component that is querying realtime to escape it with a backslash before
passing it in. Some RDBMs support escape characters by default, but the
SQL92 standard explicitly says that there are no escape characters
unless they are specified with an ESCAPE clause, e.g.

	SELECT * FROM table WHERE column LIKE '\_%' ESCAPE '\'

This patch instructs 3 backends - res_config_mysql, res_config_pgsql,
and res_config_sqlite3 - to use the ESCAPE clause where appropriate.

Looking through documentation and source tarballs, I was able to
determine that the ESCAPE clause is supported in:

MySQL 5.0.15   (released 2005-10-22 - earliest version available from
                archives)
PostgreSQL 7.1 (released 2001-04-13)
SQLite 3.1.0   (released 2005-01-21)

The versions of the relevant libraries that we depend on to access MySQL
and PostgreSQL will not work on versions that old, and I've added an
explicit check in res_config_sqlite3 to only use the ESCAPE clause when
we have a sufficiently new version of SQLite3.

res_config_odbc already handles the escape characters appropriately, so
no changes were required there.

ASTERISK-15858 #close
Reported by: Humberto Figuera

ASTERISK-26057 #close
Reported by: Stepan

Change-Id: I93117fbb874189ae819f4a31222df7c82cd20efa

There were two specific issues resolved here:

1) The code that iterated over the required fields
   (via ast_realtime_require) was broken for the RQ_INTEGER1 field
   type. Iteration would stop when the first RQ_INTEGER1 (0) field
   was encountered.

2) sqlite3_changes() was used to try and count the number of rows
   returned by a SELECT statement. sqlite3_changes() only counts
   affected rows, so this was always returning the value from the
   most recent data modification statement. We now separate read-only
   queries from data modification queries and count rows appropriately
   in both cases.

ASTERISK-23457 #close
Reported by: Scott Griepentrog

Change-Id: I91ed20494efc3fcfbc2a96ac7646999a49814884

This patch fixes 2 original issues and more that those 2 exposed.

* When we send a NOTIFY, and the client either doesn't respond or
  responds with a non OK, pjproject only calls our
  pubsub_on_evsub_state callback, no others.  Since
  pubsub_on_evsub_state (which does the sub_tree cleanup) does not
  expect to be called back without the other callbacks being called
  first, it just returns leaving the sub_tree orphaned.  Now
  pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
  which is what pjproject will set to tell us that it was the
  transaction that timed out or failed and not the subscription
  itself timing our or being terminated by the client. If is
  TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
  regardless of the state of the subscription.

* When a client renews a subscription, we don't update the
  persisted subscription with the new expires timestamp.  This causes
  subscription_persistence_recreate to prune the subscription if/when
  asterisk restarts.  Now, pubsub_on_rx_refresh calls
  subscription_persistence_update to apply the new expires timestamp.
  This exposed other issues however...

* When creating a dialog from rdata (which sub_persistence_recreate
  does from the packet buffer) there must NOT be a tag on the To
  header (which there will be when a client refreshes a
  subscription).  If there is one, pjsip_dlg_create_uas will fail.
  To address this, subscription_persistence_update now accepts a flag
  that indicates that the original packet buffer must not be updated.
  New subscribes don't set the flag and renews do.  This makes sure
  that when the rdata is recreated on asterisk startup, it's done
  from the original subscribe packet which won't have the tag on To.

* When creating a dialog from rdata, we were setting the dialog's
  remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
  When the client tried to resubscribe after a restart with the
  correct cseq, we'd reject the request with an Invalid CSeq error.

* The acts of creating a dialog and evsub by themselves when
  recreating a subscription does NOT restart pjproject's subscription
  timer.  The result was that even if we did correctly recreate the
  subscription, we never removed it if the client happened to go away
  or send a non-OK response to a NOTIFY.  However, there is no
  pjproject function exposed to just set the timer on an evsub that
  wasn't created by an incoming subscribe request.  To address this,
  we create our own timer using ast_sip_schedule_task.  This timer is
  used only for re-establishing subscriptions after a restart.

  An earlier approach was to add support for setting pjproject's
  timer (via a pjproject patch) and while that patch is still included
  here, we don't use that call at the moment.

While addressing these issues, additional debugging was added and
some existing messages made more useful.  A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.

ASTERISK-26696
ASTERISK-26756

Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e

pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND,
which is a compile-time constant. Instead of hard-coding 16 when we
enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can
potentially collect more interfaces if the compile time options are
changed.

Tangentially related to ASTERISK~24464

Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd

* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca

This change adds the media stream topology definition and API for
accessing and using it.

Some refactoring of the stream was also done.

ASTERISK-26786

Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568

The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE
but we have no authenticator registered to create the challenge.

Change-Id: I62368180d774b497411b80fbaabd0c80841f8512