This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.

Review: https://reviewboard.asterisk.org/r/3035
........

Merged revisions 403616 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When switching to using a vector for authentication, I initialized
the vector for the artificial endpoint to be of size 1. However, this
does not result in AST_VECTOR_SIZE() returning 1 since there isn't
actually anything in the vector.

Rather than trifle with the vector by putting unnecessary elements in,
I simply changed the callback in res_pjsip_authenticator_digest.c to
explicitly report that the artificial endpoint requires authentication.

Thanks to Joshua Colp for pointing this out.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22936)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3042/
........

Merged revisions 403587 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403588 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AFS-14)
Review: https://reviewboard.asterisk.org/r/3045/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_sorcery_astdb.c: Fix get multiple records by regex.

* Fix sorcery_astdb_retrieve_regex() pattern matching.  Let the regexec()
function match the stored key values instead of having astdb prefilter
them.  Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
........

Merged revisions 403559 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix sorcery_astdb_retrieve_regex() pattern matching.  Let the regexec()
function match the stored key values instead of having astdb prefilter
them.  Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.

* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
........

Merged revisions 403545 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.

* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.

* Remove redundant test in __ast_sorcery_apply_config().  !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
........

Merged revisions 403541 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403544 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.

A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.

(issue ASTERISK-22801)
Reported by: Matt Jordan
........

Merged revisions 403542 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

You would think that a new file would start off without any whitespace
oddities.
........

Merged revisions 403527 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3009



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the time, this is only useful for retrieving the filename.

The purpose of this function is to better facilitate multiple
mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel
variable is not conducive to such behavior, so allowing finer
grained access to individual mixmonitor properties improves
the situation. The MIXMONITOR_FILENAME channel variable is still
set, though, so there is no worry about backwards compatibility.

Review: https://reviewboard.asterisk.org/r/3023



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to the way pjproject internally works it was possible for the
NAT module to not be invoked on messages with-in a session dialog.
This means that the various parts of the message would not get rewritten
with the source IP address and port.

This change uses a session supplement to add the NAT module
to the dialog on the first incoming or outgoing INVITE.

(closes issue ASTERISK-22941)
Reported by: Leif Madsen
........

Merged revisions 403510 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.

Review: https://reviewboard.asterisk.org/r/3044



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)
........

Merged revisions 403449 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 403450 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 403458 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403435 from http://svn.asterisk.org/svn/asterisk/branches/12


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

........

Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403342 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........
  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.
........

Merged revisions 403304,403310 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403402 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
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
........

Merged revisions 403377 from http://svn.asterisk.org/svn/asterisk/branches/12


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".
........

Merged revisions 403364 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403349 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403312 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

Merged revisions 403288 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 403290 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12


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/ 
........

Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12


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

........

Merged revisions 403240 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

Merged revisions 403223 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

Merged revisions 403221 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 403209 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

Merged revisions 403179 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

Merged revisions 403173 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403174 65c4cc65-6c06-0410-ace0-fbb531ad65f3