Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.

As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.

Much thanks to Torrey as well for providing a scenario that reproduces this
issue.

(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
  always-init-t38.patch uploaded by awinters (License 6477)
  A_PARTY.xml uploaded by tsearle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the CDR unregistration fails due to an inflight CDR, the
res_config_sqlite module needs to bail on unloading itself. Otherwise,
the config could be unloaded (including the CDR table name) while the
CDR engine posts a CDR to the still registered backend, resulting in
a crash.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using this terminator when active results in ${RECORD_STATUS} being set to
'OPERATOR' instead of 'DTMF'

(closes issue AFS-7)

Review: https://reviewboard.asterisk.org/r/3041/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403403 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r403304 | dlee | 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line
  
  Fixed the filename for the ari.conf docs
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  r403310 | file | 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines
  
  Revert revision 403304: Fixed the filename for the ari.conf docs
  
  The changed value refers to the name of the module. The name of the
  configuration file is specified in the configFile section.
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........
remove unwanted property svn:mergeinfo


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403401 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Used a static wrapper around the offending function to alleviate the issue.

Reported by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This crept up during gateway testing where the gateway would receive
the request to negotiate and assume it came from the remote side, causing
the gateway state machine to go a little, to a use a technical term,
"wonky".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Passing a non-zero value causes PJLIB to use the given input as the
hash value. Passing zero causes the parameter to become an output parameter
that receives the hash value that was computed based on the given key.

This change essentially makes ast_sip_dict_get() properly retrieve the
desired value.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Newer versions of PJSIP have changed to using a flag for the
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a
configure check to detect the presence of the flag and use it if found.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Make ast_sorcery_observer_remove() accept a const callbacks struct.

* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL.  Now it can be called within a module unload routine if the
sorcery initialization fails.

* Fix ast_sorcery_observer_add() to fail if the container link fails.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403327 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Media indexing will now skip over files and directories that stat
will not return information about. This can occur under normal
conditions when a symbolic link points to a location that no longer
exists.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Regenerate e164 endpoint list on reload ooh323
(issue ASTERISK-22901)
Reported by: Cyril CONSTANTIN
Patches:
	ASTERISK-22901.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The check for determining whether the T.38 framehook should be added to
the channel or not has now been changed to guarantee adding only occurs
on the first incoming or outgoing INVITE.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Transport type determination for security events has been simplified to use
the type present on the message itself instead of searching through configured
transports to find the transport used.

The actual WebSocket transport has also been simplified. It now leverages the
existing PJSIP transport manager for finding the active WebSocket transport
for outgoing messages. This removes the need for res_pjsip_transport_websocket
to store a mapping itself.

(closes issue ASTERISK-22897)
Reported by: Max E. Reyes Vera J.

Review: https://reviewboard.asterisk.org/r/3036/ 
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Depending on configuration it was possible for a media stream to be
created without any media formats. The produced SDP would fail internal
validation and cause a crash.

The code will now no longer add media streams with no formats to the SDP,
allowing it to pass validation and work.

(closes issue ASTERISK-22858)
Reported by: Anthony Messina
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When sending a re-INVITE to an endpoint it was possible for received
headers to be added as well (since they are stored for retrieval using
the PJSIP_HEADER dialplan function). This caused a broken (and
potentially large) SIP INVITE to be produced and sent.

This changes the module so it will no longer add headers to
re-INVITEs.

(closes issue ASTERISK-22882)
Reported by: David M. Lee
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403222 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds new URI scheme implementations for playing numbers, digits,
and characters. This is done as part of the normal playback mechanism and can
be used with queueing to create a combined sentence.

Review: https://reviewboard.asterisk.org/r/3028/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403210 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.

(closes issue ASTERISK-21710)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2963/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Certain options available that specify a SIP URI perform validation
on the provided URI using the PJSIP URI parser. This operation
requires that the thread executing it be registered with the PJLIB
library. During reloads this was done on a thread which was NOT
registered with it.

This fixes the problem by creating a task which reloads the
configuration on a PJSIP thread.

(closes issue ASTERISK-22923)
Reported by: Anthony Messina
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403180 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).

For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.

The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.

(closes issue ASTERISK-22685)
Review: https://reviewboard.asterisk.org/r/2994/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Some options (such as call_group and pickup_group) share the same configuration
handler and decide what logic to use based on the name of the option. These
handlers were not updated to check for the new option names and were treating
the options as invalid.

This change simply updates the handlers with the proper names of the options.

(closes issue ASTERISK-22922)
Reported by: Anthony Messina
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The configure check did not use the provided paths for pjproject
if provided when looking for transaction group lock support.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Created a data model and implemented functionality for an ARI device state
resource.  The following operations have been added that allow a user to
manipulate an ARI controlled device:

Create/Change the state of an ARI controlled device
PUT    /deviceStates/{deviceName}&{deviceState}

Retrieve all ARI controlled devices
GET    /deviceStates

Retrieve the current state of a device
GET    /deviceStates/{deviceName}

Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}

The ARI controlled device must begin with 'Stasis:'.  An example controlled
device name would be Stasis:Example.  A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events.  Any device state, ARI controlled or not, can be subscribed to.

While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same.  Each event resource must now register itself in order to be able
to properly handle [un]subscribes.

(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Created the following AMI commands and corresponding events for res_pjsip:

PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
                     select attributes on each.
  Events:
    EndpointList - for each endpoint a few attributes.
    EndpointlistComplete - after all endpoints have been listed.

PJSIPShowEndpoint - Provides a detail list of attributes for a specified
                    endpoint.
  Events:
    EndpointDetail - attributes on an endpoint.
    AorDetail - raised for each AOR on an endpoint.
    AuthDetail - raised for each associated inbound and outbound auth
    TransportDetail - transport attributes.
    IdentifyDetail - attributes for the identify object associated with
                     the endpoint.
    EndpointDetailComplete - last event raised after all detail events.

PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
                                registrations.
  Events:
    InboundRegistrationDetail - inbound registration attributes for each
                                registration.
    InboundRegistrationDetailComplete - raised after all detail records have
                                been listed.

PJSIPShowRegistrationsOutbound  - Provides a detail listing of all outbound
                                  registrations.
  Events:
    OutboundRegistrationDetail - outbound registration attributes for each
                                 registration.
    OutboundRegistrationDetailComplete - raised after all detail records
                                 have been listed.

PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
                                and their attributes.
  Events:
    SubscriptionDetail - on each subscription detailed attributes
    SubscriptionDetailComplete - raised after all detail records have
                                 been listed.

PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
                                subscriptions and their attributes.
  Events:
    SubscriptionDetail - on each subscription detailed attributes
    SubscriptionDetailComplete - raised after all detail records have
                                 been listed.

(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While there were events defined for playback and recording
these were not actually sent. This change implements the
to_json handlers which produces them.

(closes issue ASTERISK-22710)
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/3026/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Snoop operation can be invoked on a channel to spy or
whisper on it. It returns a channel that any channel operations
can then be invoked on (such as record to do monitoring).

(closes issue ASTERISK-22780)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3003/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the last release of sounds, 1.4.25 we added a vm-msgforwarded prompt for various core languages. Now we use that prompt.

(issue ASTERISK-21413)
(closes issue ASTERISK-21413)
Reported by: netwrkr
Tested by: newtonr


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403106 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes the unit tests that were broken by r403069 and several
functions requiring a new parameter for sanitization of JSON messages
generated from object snapshots.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updated the alembic script for pjsip.  Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403083 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.

This prevents unhelpful error messages from being generated by
ast_json_pack.

This also corrects a bug where BridgeCreated events would not be
created.

(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore).  For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...

Review: https://reviewboard.asterisk.org/r/3002/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.

This change moves code around a bit so that the frame is now
freed after it has been completely used.

(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
	translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
	translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made PickupChan() search by channel uniqueids if the search could not
find a channel by name.

* Ensured PickupChan() never considers the picking channel for pickup.

* Made PickupChan() option p use a common search by name routine.  The
original search was erroneously case sensitive.

(issue AFS-42)

Review: https://reviewboard.asterisk.org/r/3017/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

By the time the directory application exits, a channel variable
DIRECTORY_RESULT will be set for the channel that invoked it which
can be used to determine the reason for exit. The changes log and
the app_directory documentation contain specific details about
each of the possible values for DIRECTORY_RESULT.

Review: https://reviewboard.asterisk.org/r/3016/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402995 65c4cc65-6c06-0410-ace0-fbb531ad65f3