Added:
 * outbound-publish
 * resource_list
 * inbound-publication
 * asterisk-publication

Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a

Two new parameters have been added to the pjsip config wizard.

 * Setting 'sends_line_with_registrations' to true will cause the wizard
   to skip the creation of an identify object to match incoming request
   to the endpoint and instead add the line and endpoint parameters to
   the outbound registration object.

 * Setting 'outbound_proxy' is a shortcut for adding individual
   endpoint/outbound_proxy, aor/outbound_proxy and
   registration/outbound_proxy parameters.

Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0
(cherry picked from commit a827892f)
(cherry picked from commit 27344675)

ASTERISK-26086 #close
Reported by: Jens Bürger

Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e

Dynamic payload types were statically defined in Asterisk. This unfortunately
limited the number of dynamic payloads that could be registered. With this patch
dynamic payload type numbers are now assigned dynamically and per RTP instance.
However, in order to limit any issues where some clients expect the old
statically defined value this patch makes it so the value Asterisk used to pre-
designate is used for the dynamic assignment if available.

An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf)
that turns the new dynamic behavior on or off. When off it reverts back to using
statically defined payload values. This option defaults to "yes" in Asterisk 15.

ASTERISK-26515 #close
patches:
  ASTERISK-26515.diff submitted by jcolp (license 5000

Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc

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

ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

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 patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

Use the description of useragent from sip.conf here.

ASTERISK-26825 #close

Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

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

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

Original patch by John Covert, slight modifications by me.

ASTERISK-17428 #close
Reported by: John Covert
Patches:
	app_voicemail.c.patch (license #5512) patch uploaded by
        John Covert

Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6

The entry for 'identify' was incorrectly placed in the
res_pjsip section when it should be in
res_pjsip_endpoint_identifier_ip.

ASTERISK-26785 #close

Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a

ASTERISK-26704 #close
Reported by: Anthony Messina

Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f

This change adds experimental support for providing RTCP
feedback information to codec modules so they can dynamically
change themselves based on conditions.

ASTERISK-26584

Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857

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)

Adds a new formula for SL2 and documentation

ASTERISK-26559

Change-Id: I0970c620460507cd9d45b0d43600779c8915e770

Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.

Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages.  Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible.  Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.

* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.

* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.

* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.

* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject.  Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.

* In log_forwarder(), made always log enabled and mapped pjproject log
messages.  DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.

* Removed RAII_VAR() from res_pjproject.c:get_log_level().

ASTERISK-26630 #close

Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389

This works the same as for AMI manager variables. Set
"channelvars=foo,bar" in your ari.conf general section, and then the
channel variables "foo" and "bar" (along with their values), will
appear in every Stasis websocket channel event.

ASTERISK-26492 #close
patches:
  ari_vars.diff submitted by Mark Michelson

Change-Id: I5609ba239259577c0948645df776d7f3bc864229

Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.

ASTERISK-26311 #close

Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964

codecs.conf.sample was missing codec opus's configuration options, descriptions,
and examples. This patch adds the configuration options and examples to
codecs.conf.sample that can be used with codec_opus.

ASTERISK-26538 #close

Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b

Support for referring to DAHDI channels by logical names was added in
(FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support
of refering to channels by name.

While technically usable, it has never been properly supported in
dahdi-tools, as using it would require many changes at the Asterisk
level. Instead logical mapping was added at the kernel level.

Thus it seems that refering to DAHDI channels by name is not really used
by anyone, and therefore should probably be removed.

Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485

When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.

The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.

The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.

ASTERISK-26423 #close

Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc

This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.

ASTERISK-26309 #close

Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d

Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the
form ice_blacklist = <subnet spec>, e.g. ice_blacklist =
192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay
discovery. This is useful for optimizing the ICE process where a system
has multiple host address ranges and/or physical interfaces and certain
of them are not expected to be used for RTP. Multiple ice_blacklist
configuration lines may be used. If left unconfigured, all discovered
host addresses are used, as per previous behavior.

Documention in rtp.conf.sample.

ASTERISK-26418 #close

Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9

We use a lot res_calendar, we are very happy with that, especially
because you use libical, the almost alone opensource library that
supports really ical format with all types of recurrency.

Nevertheless, some features are missed for our business use cases.

This first patch adds a new option in calendar.conf:
fetch_again_at_reload. Be my guest for a better name.

If it's true, when you'll launch "module reload res_calendar.so",
Asterisk will download again the calendar.

The business use case is that we have a WebUI with a scheduler planner,
we know when the calendars are modified.

For now, we need to define 1 minute of timeout to have a chance that
our user doesn't wait too long between the modification and the real
test.  But it generates a lot of useless HTTP traffic.


ASTERISK-26422 #close

Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077

The pooling, shared_connection, limit, and idlecheck options
are no longer used in res_odbc.

ASTERISK-26389

Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6

* Make 'cdrzone=UTC' work properly.
* Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone

ASTERISK-26359 #close

Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778

This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn

Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762

Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6

When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL
SSLv23_method. This was documented incorrectly in the file sip.conf.sample.

SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method
enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that
function should have been called 'secure_method' or 'automatic_method' back in
the 90s.

Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if
you face a server which has problems like not falling back to TLSv1.0
automatically.

ASTERISK-24425

Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3

contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.

Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4

The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.

This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.

With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
    res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
    pbx_dundi, res_xmpp

Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".

ASTERISK-26164 #close

Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6

ASTERISK-26246 #close

Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657

Make it clear that we're talking about device state hints and add
an entry to the sample config.

Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433

The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.

This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.

This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.

ASTERISK-26230 #close

Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a