Currently if a channel is transferred out of a bridge, the BRIDGEPEER
variable (also BRIDGEPVTCALLID) remain set even once the channel is
out of the bridge. This patch removes these variables when leaving
the bridge.

ASTERISK-25600 #close
Reported by: Mark Michelson

Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da

chan_sip.c:
* Initialize mwi subscription scheduler ids earlier because of ASTOBJ to
ao2 conversion.

* Initialize register scheduler ids earlier because of ASTOBJ to ao2
conversion.

chan_skinny.c:
* Fix more scheduler usage for the valid 0 id value.

ASTERISK-25476

Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95

channels/chan_iax2.c:
* Initialize struct chan_iax2_pvt scheduler ids earlier because of
iax2_destroy_helper().

channels/chan_sip.c:
channels/sip/config_parser.c:
* Fix initialization of scheduler id struct members.  Some off nominal
paths had 0 as a scheduler id to be destroyed when it was never started.

chan_skinny.c:
* Fix some scheduler id comparisons that excluded the valid 0 id.

channel.c:
* Fix channel initialization of the video stream scheduler id.

pbx_dundi.c:
* Fix channel initialization of the packet retransmission scheduler id.

ASTERISK-25476

Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8

dns.c and dns_system_resolver.c were spitting out errors for lookup
failures for things like not finding a SRV record even though
there was an A record.  Those have been changed to debug messages.
Logging not finding ANY record is left to the higher level caller.

Also, dns_system_resolver was using Windows line endings so I
converted them to Unix style.  The actual log changes are on lines
156 and 159.

Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094

Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7

The fastagi record-file testsuite test sometimes fails reporting an empty
recorded file. This was happening because Asterisk was sending the agi result
notification prior to actually closing the file and the data, being buffered,
had not been written to the file yet when the test attempts to check the file
size.

This patch makes it so the record file stream is closed prior to sending the
agi result notification.

ASTERISK-25593 #close

Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde

Several issues are addressed here:
- main() is large, and half of it is only used if we're not rasterisk;
  fixed by spliting up the daemon part into a separate function.
- Call ast_term_init from rasterisk as well.
- Remove duplicate code reading/writing asterisk history file.
- Attempt to tackle background color issues and color changes that
  occur. Tested by starting asterisk -c until the colors stopped
  changing at odd locations.
- Remove unused term_prep() and term_prompt() functions.

ASTERISK-25585 #close

Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f

Fixes some minor typos in the CHANGES file, plus an embarrasing typo in
the StatsD API.

Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7

The usage info for 'pjsip send notify' previously referenced the
chan_sip configuration sip_notify.conf.  Fix this to reference
the correct configuration pjsip_notify.conf.

ASTERISK-25590 #close

Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea

This patch adds a module that emits StatsD statistics about Asterisk
endpoints. This includes:
 * A GAUGE statistic for endpoint states, tracking how many endpoints are in
   a particular state.
 * A GAUGE statistic for each endpoint, counting the number of channels
   currently associated with an endpoint.

ASTERISK-25572

Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305

If the sorcery object type is not found a NULL is returned.
Unfortunately, sorcery_realtime_filter_objectset() will crash after
complaining about not finding the object type and saying to expect errors.

* Use ao2_cleanup() instead of ao2_ref() to prevent the crash.

ASTERISK-25165
Reported by Corey Farrell

Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97

This patch adds the ability to send StatsD statistics related to the
state of PJSIP contacts. This includes:
 * A GUAGE statistic measuring the count of contacts in a particular state.
   This measures how many contacts are reachable, unreachable, etc.
 * The RTT time for each contact, if those contacts are qualified. This
   provides StatsD engines useful time-based data about each contact.

ASTERISK-25571

Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c

This patch adds outbound registration statistics for StatsD. This includes
the following:
 * A GUAGE metric for the overall count of outbound registrations.
 * A GUAGE metric for each state an outbound registration can be in. As the
   outbound registrations change state, the overall count of how many
   outbound registrations are in the particular state is changed.

These statistics are particularly useful for systems with a large number of
SIP trunks, and where measuring the change in state of the trunks is useful
for monitoring.

ASTERISK-25571

Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37

Often, the metric names of statistics we are generating for StatsD have some
dynamic component to them. This can be the name of a particular resource, or
some internal status label in Asterisk. With the current set of functions,
callers of the statsd API must first build the metric name themselves, then
pass this to the API functions. This results in a large amount of boilerplate
code and usage of either fixed length static buffers or dynamic memory
allocation, neither of which is desireable.

This patch adds two new functions to the StatsD API that support a printf
style format specifier for constructing the metric name. A dynamic string,
allocated in threadstorage, is used to build the metric name. This eases
the burden on users of the StatsD API.

Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea

When a channel is in a direct media bridge, a re-INVITE may arrive that forces
Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
must change its technology to a simple bridge, and re-INVITE the media back
to Asterisk.

Generally, this logic mostly already exists in Asterisk. However, prior to this
patch, there were a few bugs:
(1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
    ever entering into a direct media bridge. This applies even when the only
    media being passed over the channel is audio. This patch fixes this bug
    by having the framehook specify that it defers caring about any frame type.
    This allows the channels to enter into a direct media bridge, which will
    be broken when a re-INVITE is received.
(2) When a re-INVITE is received, nothing instructed the bridging layer to
    re-inspect the allowed bridging technology. This now occurs when either
    a re-INVITE is received from a peer, or when a response is received from
    the far end (that is, when the T.38 state changes to either
    T38_PEER_REINVITE or T38_LOCAL_REINVITE).
(3) chan_pjsip needs to do a small amount of work to prevent a direct media
    bridge from being chosen when a T.38 session is in progress. When a T.38
    session supplement has a t38 datastore - which is added when we detect
    we should start thinking about T.38 on a channel - we now refuse a native
    RTP bridge.
(4) When a BYE request is received, we don't terminate the T.38 session. If
    the other side of a T.38 fax survives the hangup (due to the 'g' flag
    in Dial, for example), we don't currently re-INVITE the media on the
    other channel back to audio. This patch now has res_pjsip_t38 intercept
    BYE requests and inform the far side that the T.38 session is terminated.
    This naturally causes the correct re-INVITEs to be sent.

ASTERISK-25582

Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb

Because the context, extension, and application are stored in stringfields,
checking for them being NULL doesn't work so well. This patch uses the
appropriate string library call, ast_strlen_zero, to see if there is a value
in the context/exten/app values.

Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23

This patch adds some debug statements to res_pjsip_t38. These statements help
to determine which SDP negotiation callbacks are being executed, and, when
a particular callback exits, why a callback may not have applied its logic
to the local or remote SDP.

Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77

When Asterisk is configured to use a dynamic sorcery backend (such as
res_sorcery_astdb) with 'registration' objects, it will fail to create the
internal state objects associated with the registration objects on module
load. This is due to nothing actually querying for the specific objects
and calling their sorcery apply handler during module load.

This patch fixes that by calling get_registrations in the sorcery observer's
object_type_loaded handler. Doing this causes the sorcery backends to be
asked for the current state of all registration objects, which causes the
apply handler to be called and the internal run-time state to be created.

ASTERISK-25575 #close

Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23

Previously, a trancoding module did not have access to the joint but cached
format. Therefore, the module did not have access to the attributes negotiated
via SDP (line fmtp). Now, a translation module receives the joint format.

ASTERISK-25545 #close

Change-Id: Id6878a989b50573298dab115d3371ea369e1a718

When no parameter is present, Asterisk does not generate the line fmtp, as
expected. However, because a buffer was reset, even rtpmap and fmtp of previous
media codecs got removed. Now, Asterisk does not reset other codecs in case of
no parameter for H.264.

ASTERISK-25573 #close

Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286

To be able to barge into a call by dialling a prefix+extension that maps
to the extensions device.

Senario is that DECT headset users may be away from their desks and need
to transfer the call, the goal is that from any phone they dial a prefix
then their extension and are added to the bridge that they are in, from
there they can drop the headset call, as it's also on the handset,
and transfer the caller.

The dialplan would look like, where prefix=73, extension = 8512;
exten => _738512,1,BridgeAdd(SIP/cisco0001)

ASTERISK-25551 #close
Reported By: Alec Davis

Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540

Implemented support for the StatsD sample rate parameter,
which is a parameter for determining when to send computed
statistics to a client.

Valid sample rate values are:
Less than or equal to 0.0 will never be sent.
Between 0.0 and 1.0 will randomly be sent.
Greater than or equal to 1.0 will always be sent.

ASTERISK-25419
Reported By: Ashley Sanders

Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52

Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d