The OUTPUTDIR environment variable can now be set either in the
environment itself or in ast_debug_tools.conf.  If set, it's used
for all work products instead of /tmp.

Also added the --tarball-config option that includes the contents
of /etc/asterisk when either --tarball-coredumps or --tarball-results
are used.

Change-Id: I66b2553319df61caea5b313d084f51978f730b4c

This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.

Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.

ASTERISK-27395

Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4

Since Asterisk 13.17, libSRTP 2.x is supported. Therefore, its latest version
is installed again via the script install_prereq.

ASTERISK-27356

Change-Id: I13125839a79052356469e41edacbebff0a937d39

Adds an extra option, --asterisk-bin=<path> to ast_coredumper. If
provided, the binary given to gdb will be the parameter, rather than
asterisk from the PATH.

ASTERISK-27380 #close

Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3

The ps_endpoints table was missing the bundle column
introduced with the bundle feature in
commit 065c3005.

ASTERISK-27374 #close

Change-Id: Ic900f4f2c20f64b99ea898d50f5c0a7117472d46

When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.

ASTERISK-27206

Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd

The --tarball-coredump option now creates a gzipped tarball of
coredumps processed, their results txt files and copies of
/etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and
/usr/lib(64)/asterisk as those files are needed to properly examine
the coredump.  The file will be named
/tmp/asterisk.<timestamp>.coredumps.tar.gz or
/tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was
specified.

Added dumps of *_siginfo to the top of the txt files so you can
tell what signal was invoked.

Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9

when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa

ASTERISK-27333 #close

Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6

add mappings for udptl redundancy, rtptimeout, and debug flags

Change-Id: Ie73cf5c83c05dee01eb9624ede76c1a30225d73a

A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.

res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.

Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c

The downgrade function was missing "_v2" at the end of the
alter column type.

Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e

MS-SQL has no native Enum-type support and therefore
needs to work with constraints.
Since these constraints need unique names the suggested approach
referenced in the following alembic documentation has been applied:
http://bit.ly/2x9r8pb

ASTERISK-27255 #close

Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95

Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db

Added include for postgresql ENUM type and
redefined values in the same way as in the
other migration scripts.

ASTERISK-27254 #close

Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa

The ps_endpoints table was missing the dtls_fingerprint column
introduced with commit adba2a8d.

ASTERISK-27168 #close

Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd

The fix for the issue is broken up into three parts.

This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled.  Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.

* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown.  If it is shutdown then the contact must be removed because it
is no longer valid.  Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there.  Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request.  The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.

* Prune any rewrite_contact's registered reliable transport contacts on
boot.  The reliable transport no longer exists so the contact is invalid.

* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.

* Made the websocket transport set a unique name since that is what we use
as the ao2 container key.  Otherwise, we would not know which transport we
find when one of them shuts down.  The names are also used for PJPROJECT
debug logging.

* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event.  Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.

* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.

ASTERISK-27147

Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4

When the "webrtc" option was added in res_pjsip it was not added to the alembic
scripts. This patch adds the option for alembic.

Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of
an OPT_BOOL_T so if this field is ever written to a database it will write out
the correct value.

ASTERISK-27119 #close

Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b

Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5

AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0

Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago

Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7

Also add new corosync packages to install_prereq.

Reported by Travis Ryan in #asterisk-dev

Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db

Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187

The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated.  This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.

ASTERISK-27066 #close

Change-Id: Id185b11e84afd9191a2f269e8443019047765e91

The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.

Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.

The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.

Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.

Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.

If a stream has been removed or declined we will now mark it as such
within the resulting SDP.

Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.

Two new configuration options have also been added to PJSIP endpoints:

max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.

max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.

ASTERISK-27076

Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7

This change adds support for socket activation of certain SOCK_STREAM
listeners in Asterisk:
* AMI / AMI over TLS
* CLI
* HTTP / HTTPS

Example systemd units are provided.  This support extends to any socket
which is initialized using ast_tcptls_server_start, so any unknown
modules using this function will support socket activation.

Asterisk continues to function as normal if socket activation is not
enabled or if systemd development headers are not available during
build.

ASTERISK-27063 #close

Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d

This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.

ASTERISK-26919 #close

Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711

This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".

Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".

ASTERISK-26333 #close

Change-Id: Id606fbff2e02e967c02138457badc399144720f2

This change adds an Alembic migration which adds a
ps_resource_list table that can contain resource_list
RLS configuration objects.

ASTERISK-26929

Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05

This change adds database tables for the PUBLISH support so it
can be configured using realtime. A minor fix to the
res_pjsip_publish_asterisk module was done so that it read the
sorcery configuration from the correct section. Finally the
sample configuration files have been updated.

ASTERISK-26928

Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952

Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c

Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.

ASTERISK-26864

Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6

A new transport parameter 'symmetric_transport' has been added.

When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output.  On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.

* config_transport was modified to accept and store the new parameter.

* config_transport/transport_apply was updated to store the transport
  name in the pjsip_transport->info field using the pjsip_transport->pool
  on UDP transports.

* A 'multihomed_on_rx_message' function was added to
  pjsip_message_ip_updater that, for incoming requests, retrieves the
  transport name from pjsip_transport->info and retrieves the transport.
  If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
  containing the transport name is added to the incoming Contact header.

* An 'ast_sip_get_transport_name' function was added to res_pjsip.
  It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
  transport name if endpoint->transport is set or if there's an
  'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
  ipv6 address.  Otherwise it returns NULL.

* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
  which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
  pjsip_tpselector.  It calls ast_sip_get_transport_name() and if
  a non-NULL is returned, sets the selector and sets the transport
  on the dialog.  If a selector was passed in, it's updated.

* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
  were modified to call ast_sip_dlg_set_transport() instead of their
  original logic.

* res_pjsip/create_out_of_dialog_request was modified to call
  ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
  instead of its original logic.

* Existing transport logic was removed from endpt_send_request
  since that can only be called after a create_out_of_dialog_request.

* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
  a new 'ast_sip_create_rdata_with_contact' function which allows
  a contact_uri to be specified in addition to the existing
  parameters.  (See below)

* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
  since all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.

* 'contact_uri' was added to subscription_persistence.  This was
  necessary because although the parsed rdata contact header has the
  x-ast-txp parameter added (if appropriate),
  subscription_persistence_update stores the raw packet which
  doesn't have it.  subscription_persistence_recreate was then
  updated to call ast_sip_create_rdata_with_contact with the
  persisted contact_uri so the recreated subscription has the
  correct transport info to send the NOTIFYs.

* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
  all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac.

* pjsip_message_ip_updater/multihomed_on_tx_message was updated
  to remove all traces of the x-ast-txp parameter from the
  outgoing headers.

NOTE:  This change does NOT modify the behavior of permanent
contacts specified on an aor.  To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated.  If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.

You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.

Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f

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

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)

Forgot to install it with the original patch

Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c

The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified.  If asterisk is running when it is executed,
the same commands will be issued to the running instance.  The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.

The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid

Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.

A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.

Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543

ast_loggrabber gathers log files from customizable search patterns,
optionally converts POSIX timestamps to a readable format and
tarballs the results.

Also a few tweaks were made to ast_coredumper.

Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495
(cherry picked from commit c70915287837704090d75f181525765de7a17221)

This utility allows easy manipulation of asterisk coredumps.

* Configurable search paths and patterns for existing coredumps
* Can generate a consistent coredump from the running instance
* Can dump the lock_infos table from a coredump
* Dumps backtraces to separate files...
  - thread apply 1 bt full -> <coredump>.thread1.txt
  - thread apply all bt -> <coredump>.brief.txt
  - thread apply all bt full -> <coredump>.full.txt
  - lock_infos table -> <coredump>.locks.txt
* Can tarball corefiles and optionally delete them after processing
* Can tarball results files and optionally delete them after processing
* Converts ':' in coredump and results file names '-' to facilitate
  uploading.  Jira for instance, won't accept file names with colons
  in them.

Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1].

[1] For *BSDs, the "devel/gdb" package might have to be installed to
get a recent gdb.  The utility will check all instances of gdb
it finds in $PATH and if one isn't found that can run python, it
prints a friendly error.

Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd
(cherry picked from commit cb47b4556053cd50d9102eef913671ad0306062d)

This change implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.

This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".

ASTERISK-26693

Change-Id: I6b641e275bf96629320efa8b479737062aed82ac

Pyflake is a python (2) source checker. This patch fixes various
(mostly trivial) errors and warnings it reports.

Change-Id: Ia35c5ac61751b927814cf693994c632c412386ea

Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7