* Added CHANNEL(callid) to retrieve the call identifier log tag associated
with the channel.  Dialplan now has access to the call log search key
associated with the channel so it can be saved in case there is a problem
with the call.

ASTERISK-26878

Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f

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)

Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.

ASTERISK-26568 #close
Reported by: Roman Shubovich

Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89

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

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

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

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

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

When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
	app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957

The 'ari set debug' command has been enhanced to accept 'all' as an
application name.  This allows dumping of all apps even if an app
hasn't registered yet.  To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.

'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.

* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
  to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
  failed, the consumption of the body was moved from the ari stubs
  to ast_ari_callback in res_ari.c and the moustache templates were
  then regenerated.  The body is now passed to ast_ari_invoke and then
  on to the handlers.  This results in code savings since that template
  was inserted multiple times into all the stubs.

An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function.  The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.

Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874)

Add an application that allows tracking outbound calls
using app_queue.

ASTERISK-19862

Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e

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

Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.

This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.

ASTERISK-26658 #close
Reported by: Jonathan R. Rose

Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30

The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.

PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead.  Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.

For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.

ASTERISK-26644 #close

Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a

Increasing the testsuite shutdown timeout before forcibly killing
Asterisk allowed more events to be sent out.  Some tests failed as
a result.  The tests/channels/pjsip/statsd/registrations failed
because we now get the statsd events that a comment in the test
configuration stated couldn't be intercepted.  Unfortunately, we
get a variable number of events because of internal status state
transition races generating redundant statsd events.

We were reporting redundant statsd PJSIP.registrations.state changes
for internal state changes that equated to the same thing publicly.

* Made update_client_state_status() filter out redundant statsd
updates.

ASTERISK-26527

Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646

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

Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call.  The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works.  Have also tested both 'exten'
and 'app' versions of app_originate.

Opened by: dkerr
Patch by: dkerr

Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57

In order to not have version number overlap between different versions
of Asterisk, each new major version of Asterisk will mean we also bump
the ARI major version number.

This particular change does NOT introduce any known breaking changes to
ARI.

For discussion relating to this topice, see:
http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html

Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665

In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

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

Radcli is yet another RADIUS client library, generally compatible with
freeradius and radiusclient-ng.

This commit adds autoconf option for detecting it as well and changes
cdr_radius and cel_radius to use its header file in that case.

ASTERISK-26540 #close

Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f

ASTERISK-26558

Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e

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

This patch adds three new CLI commands:
 - ari show apps: list the registered ARI applications
 - ari show app: show detailed information about an ARI application
 - ari set debug: dump events being sent to an ARI application

Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.

ASTERISK-26488 #close

Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5

ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966

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

- Call "rc_openlog" optional. If you do not call,
you will simply NULL instead of a name.

- On the one PID can be only one syslog channel.
And it can already be run in logger.c

- Calling rc_openlog we assigns a new name for
the channel syslog. This unexpected behavior for logger.c.

Most lesser evil, is to agree on a NULL name syslog
if the channel was not launched in logger.c.

It also solves the problem of memory leaks.

ASTERISK-26455 #close

Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd

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

This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.

ASTERISK-26470 #close

Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d

The "Q" option will set the cause on the unanswered channels when
another channel answers.  It overrides the default of
ANSWERED_ELSEWHERE.

NOTE:  chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.

ASTERISK-26446 #close

Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47

This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.

ASTERISK-26398 #close

Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d

sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).

To use this, use a systemd unit with 'Type=notify' for Asterisk.

This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.

Also adds support for libsystemd detection in the configure script.

Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811

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

Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.

This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.

Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
  conference (if the channel and conference use the same language)

ASTERISK-26289 #close
Reported by Mark Michelson

Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a

Following the Encrypt-all-the-things paradigm:

The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone
determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts
the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile
(sRTP is preferred aka optional; not mandatory). If the VoIP server does not
support sRTP and TLS, the phone shows an open padlock icon.

This paradigm is supported by several VoIP/SIP clients on default. Some
implementations even cannot be changed to RTP/sAVP. Therefore here, this
change allows Preferred sRTP for ingress. For egress, please, create a dial
plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP.

ASTERISK-20234 #close
Reported by: tootai
Tested by: tootai, Alexander Traud
patches:
 srtp_patches.diff submitted by Matt Jordan

Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd

The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect.  Any that are selected will automatically be
downloaded and installed when "make install" is run.  Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.

Example use with codecs:

The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included.  Their support levels are 'external', which
triggers the download and install, and defaultenabled is no.  Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name.  You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory.  In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.

A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.

To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball.  The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.

bash and xmlstarlet are required for downloader operation.  If they're
not installed, the external items in menuselect will be unavailable.

Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a

We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals.  Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object.  With a
memory cache for realtime, there is about the same amount of overhead as
for config files.  Either way, it is still fairly expensive to access the
sorcery object that much.

* Cache the global config options so we can access them faster.  You must
now always perform a res_pjsip reload to change the global options.

Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7