Adds presence state value to output of core show
hints.  Also reformats the output slightly so it
doesn't use as much space as it would otherwise.

Was:
                   1000@demo                : SIP/1000              State:Unavailable     Watchers  0

Now: 
1000@demo           : SIP/1000              State:Unavailable     Presence:Idle            Watchers  0

AFS-53 #close
Review: https://reviewboard.asterisk.org/r/3604/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415698 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Documentation for how to add custom headers/content to notifies created
with the PJSIPNotify manager action was a little sparse and it also
wasn't vetting application of Content-length headers like its chan_sip
equivalent was (so two Content-length headers could be applied... and
PJSIP determines the content length anyway, so it just opens people up
for error). This patch also flips the variable order so that the
variables are interpreted in the same order as they are put in the AMI
action.

Review: https://reviewboard.asterisk.org/r/3587/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415659 65c4cc65-6c06-0410-ace0-fbb531ad65f3

change return 1 to return AST_MODULE_LOAD_FAILURE on module load routine
few cosmetic changes

ASTERISK-23814 #close

(closes issue ASTERISK-23814)

Reported by: Igor Goncharovsky
Patches:
	ASTERISK-23814-ast11.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415630 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using PJSIP_HEADER() to add custom headers to outgoing INVITE requests, certain
situations could result in the headers being duplicated. For instance, if the request
were retransmitted, or if the INVITE were re-sent with authentication credentials,
the custom headers would be re-added to the request.

The fix here is to, after adding the custom headers to the outbound INVITE, remove
the datastore that holds the custom headers to add. This way, there is no risk in
accidentally adding them if the session supplement is called into a second or third
time.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Replaced a stray echo that should've been a message call in
  safe_asterisk. This replaces a conditional log message by a slightly
  different message. Please update your log parsing scripts.
* Made the $NOTIFY mail Subject more verbose by adding the machine name
  and exitstatus.

(Note that a 'make install' still won't overwrite your old safe_asterisk
if it exists. See ASTERISK-21965.)

ASTERISK-23492 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds thread shutdown to autoservice for graceful shutdowns only.
ast_register_cleanup is backported to 1.8 to allow this.  The logger callid
is also released on shutdown in 11+.

ASTERISK-23827 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3594/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is a re-do of r414122.

When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft
hangup flags have a catastrophic effect on the pbx core if they leak out from
the bridge layer: the channel gets hung up. With the number of threads
involved in a blind transfer, and with the initial patch, it was likely that
this would occur. This caused a large number of test failures

This patch is nearly identical with the one proposed in r414122, save for the
following changes:
 - We explicitly clear the UNBRIDGE flag when setting an after goto on a
   channel in a bridge
 - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it

https://reviewboard.asterisk.org/r/3585/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415428 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, sequential variables would be ordered in reverse
from the order specified in the manager action.

Review: https://reviewboard.asterisk.org/r/3588/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If using the custom URI parsing code (not external uriparser lib) and there
was no query parameters the resulting pointer would be NULL and then an
attempt was made to subtract from it.  The pointer is now set to a valid
value if there is no query parameter(s).

Also, in the 'ast_uri_make_host_with_port' function when setting the terminator
on the resulting string it was writing it one past the end of allocated memory.
It now writes the string terminator appropriately.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, there are situations that can occur when using chan_pjsip
and certain dialplan applications (notably ChanSpy()) that can cause
the channel to get no audio with scrolling warnings about format
mismatches. This is caused by a failure to update translation paths on
a mid-call native format update since the raw formats have already
been updated by res_pjsip_sdp_rtp.c in set_caps(). Removing the
premature raw format updates allows the translation paths to be setup
correctly and the raw read and write formats with them.

AFS-63 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415343 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Split astobj2.c into the following files to improve maintainability.

astobj2.c - object primitives, object primitive misc and initialization code.
astobj2_private.h - internal object declarations needed by the containers.
astobj2_container.c - generic conainer and container misc code.
astobj2_container_hash.c - hash container specific code.
astobj2_container_rbtree.c - rbtree container specific code.
astobj2_container_private.h - generic container definitions and rtti prototypes.

https://reviewboard.asterisk.org/r/3576/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Changed naming of included alias templates to avoid confusion between version names. For example, asterisk12 was for asterisk 1.2, so I changed it to asterisk_1dot2, so that later we can use asterisk_12 for Asterisk 12.

Added alias for "features reload" to the template for Asterisk 11 style syntax template, as features reload was removed in 12, but you can still do "module reload features"

Added alias for "pjsip reload" to the friendly template. It is shorter than "module reload res_pjsip.so" and if some are like me; I constantly forget that reloading chan_pjsip doesn't parse config. Remembering "pjsip reload" is just easier.

ASTERISK-23654 #close
ASTERISK-23654 #comment Fixed by adding two new aliases and enhancements for context names.

Review: https://reviewboard.asterisk.org/r/3572/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415302 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415288 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The twisted logic determining if a config file should be reloaded was
mostly broken and disabled.  The incorrect test that ASTERISK-23383 fixed
actually reenabled the broken logic.  The incorrect test was causing the
timestamp to always be cleared which caused config files with includes to
always be reloaded.

* Made wildcard includes always cause a reload.  Determining if a file was
deleted cannot be determined without restructuring the cache to determine
if any files are missing from the last files actually loaded.  Also
without refactoring config_text_file_load(), the glob loop couldn't check
more than one file for changes anyway.

* Made remove the cache entry if the file no longer exists when trying to
get its timestamp or it is no longer a regular file.  This fixes the
corner case where the file was loaded, then deleted, then the config
reloaded, then the file restored with the same timestamp, and then the
config reloaded again.

* Made remove the cache entry include list when actually loading the file.
This gets rid of any stale includes the file had from the last time the
file was loaded.

ASTERISK-23683 #close
Reported by: tootai

Review: https://reviewboard.asterisk.org/r/3575/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a websocket server client in Asterisk. Asterisk has a websocket server,
but not a client. The ability to have Asterisk be able to connect to a websocket
server can potentially be useful for future work (for instance this could allow
ARI to connect back to some external system, although more work would be needed
in order to incorporate that).

Also a couple of things to note - proxy connection support has not been
implemented and there is limited http response code handling (basically, it is
connect or not).

Also added an initial new URI handling mechanism to core.  Internet type URI's
are parsed into a data structure that contains pointers to the various parts of
the URI.

(closes issue ASTERISK-23742)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/3541/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, users waiting to enter a ConfBridge were not considered
when muted via the CLI or via AMI. Instead, a confusing message would be
emitted stating that the channel did not exist.

This patch allows a user to be muted when waiting to enter a ConfBridge
conference. This is equivalent to start when muted, only toggled via the CLI
or AMI.

Review: https://reviewboard.asterisk.org/r/3582

#ASTERISK-23824 #close
patches:
  rb3582.patch uploaded by tm1000 (License 6524)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415208 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This makes chan_pjsip send connected line information when it is called
so that connected line information is available on the connected
channel.

(closes issue DPMA-442)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3584/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND
option that allows the debian asterisk init script to capture the
right pid.

* Drop the vim #modeline which wasn't used. Use test consistently
  without the odd configure xno syntax. Double quote all paths.
  General cleanup.
* Don't output message()s to the console but only to TTY if set.
* Allow TTY to be "no" as well as empty (debian compatibility with
  debian/patches/safe_asterisk-config).
* Add option to export ASTSAFE_FOREGROUND=1 from the init script
  that calls this to disable backgrounding. Debian uses a similar
  method in debian/patches/safe_asterisk-nobg).

ASTERISK-23492 #close
Review: https://reviewboard.asterisk.org/r/3574/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415173 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds some debug statements that aid with determining why a direct
media request may or may not be initiated.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This small patch adds a debug level 3 statement indicating how a session
refresh is being sent - either as a re-INVITE or as an UPDATE - and where
the session refresh is going.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Conference names were not checked for maximum length, allowing unexpected
behaviour.  This change adds checking to ensure the maximum length is not
exceeded.  The maximum length is also changed from 32 to AST_MAX_EXTENSION.

ASTERISK-23035 #close
Reported by: Iñaki Cívico
Tested by: Iñaki Cívico
Patches:
    confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909)
    confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415080 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The change that removed the fixed size buffers in odbc-related code --
removing arbitrary column width limits -- was incomplete. This change
adds: no segfault on writesql without insertsql and return value checks
after strdup.

While I was in the vicinity I cleaned up the linefeeds in the odbc
function descriptions, moved some code for clarity, removed some blobs
and noted (but didn't fix) that the 'odbc write ... exec' CLI command
doesn't behave as the dialplan equivalent when insertsql= is used.

ASTERISK-23582 #close
Review: https://reviewboard.asterisk.org/r/3579/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415000 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The bridge_native_rtp module currently uses the bridge result of the first
channel that joins a bridge as the ultimate result. This means that if the
first channel has direct media enabled but the second does not a direct
media bridge will still occur.

This change makes it so that both sides are taken into account. If either
side forbids the bridge or responds with a local bridge result then
either a generic or local bridge occurs.

ASTERISK-23541 #close
Reported by: Justin E

Review: https://reviewboard.asterisk.org/r/3577/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Blind transfers don't go too well with NULL channels which can occur if
the channel has already been transferred away.

(closes issue ASTERISK-23718)
Reported by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414949 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new channel function TALK_DETECT that, when set on a
channel, causes events indicating the start/stop of talking on a channel to be
emitted to both AMI and ARI clients. 

The function allows setting both the silence threshold (the length of silence
after which we decide no one is talking) as well as the talking threshold (the
amount of energy that counts as talking). Parameters can be updated on a channel
after talk detection has been enabled, and talk detection can be removed at
any time.

The events raised by the function use a nomenclature similar to existing AMI/ARI
events.
For AMI: ChannelTalkingStart/ChannelTalkingStop
For ARI: ChannelTalkingStarted/ChannelTalkingFinished

Review: https://reviewboard.asterisk.org/r/3563/

#ASTERISK-23786 #close
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When invoking UpdateConfig AMI action with Action set to EmptyCat, Asterisk
will make all categories empty in the config but the one requested with a
Cat variable. This is due to a bug in ast_category_empty (main/config.c)
that makes an incorrect comparison for a category name.

This patch corrects the comparison such that only the requested category
is cleared.

Review: https://reviewboard.asterisk.org/r/3573/

#ASTERISK-23803 #close
Reported by: zvision
patches:
  manager.c.diff uploaded by zvision (License 5755)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Dynamic and pattern matching hints should not be checked for their last
known state until they are instantiated by subscribers.

(closes issue AFS-56)
Reported by: John Hardin
Patch AFS-56-pbx.diff submitted by Matt Jordan (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
 * A number of chatty verbose messages were removed or demoted to DEBUG
   messages. Verbose messages with a verbosity level of 5 or higher were -
   if kept as verbose messages - demoted to level 4. Several messages
   that were emitted at verbose level 3 were demoted to 4, as announcement
   of dialplan applications being executed occur at level 3 (and so the
   effects of those applications should generally be less).
 * Some verbose messages that only appear when their respective 'debug'
   options are enabled were bumped up to always be displayed.
 * Prefix/timestamping of verbose messages were moved to the verboser
   handlers. This was done to prevent duplication of prefixes when the
   timestamp option (-T) is used with the CLI.
 * Verbose magic is removed from messages before being emitted to
   non-verboser handlers. This prevents the magic in multi-line verbose
   messages (such as SIP debug traces or the output of DumpChan) from
   being written to files.
 * _Slightly_ better support for the "light background" option (-W) was
   added. This includes using ast_term_quit in the output of XML
   documentation help, as well as changing the "Asterisk Ready" prompt to
   bright green on the default background (which stands a better chance of
   being displayed properly than bright white).

Review: https://reviewboard.asterisk.org/r/3547/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

pjsip.conf: privkey_file should be priv_key_file, mediaencryption=yes should be mediaencryption=sdes

privkey_file was missed in the snake case update. An example included an invalid value for the mediaencryption option.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414781 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update the semantic versioning of ARI to 1.3.0 and AMI to 2.3.0 to account for
backwards compatible changes going from 12.2.0 to 12.3.0.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When generating SQL files via the repotools alembic_creator.py script, a
configuration object is used programatically with SQLAlechemy, as opposed to
a configuration file. This patch ignores failures to interpret a config file,
as ... there isn't one in this case.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak
video RTP ports if the codec were not negotiated by an incoming call.

* Made add_sdp_streams() associate the handler with the media stream if
the handler handled the media stream.  Otherwise, when the
ast_sip_session_media object was destroyed it didn't know how to clean up
the RTP resources.

* Fixed sdp_requires_deferral() associating the handler with the media
stream when deciding if the SDP processing needs to be deferred for T.38.
Like the leaked video RTP ports, the T.38 handler needs to clean up
allocated resources from deciding if SDP processing needs to be deffered.

* Cleaned up some dead code in handle_incoming_sdp() and
sdp_requires_deferral().

ASTERISK-23721 #close
Reported by: cervajs

Review: https://reviewboard.asterisk.org/r/3571/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Improvements to the agent pool functionality.

* AgentRequest no longer hangs up the caller if the agent fails to connect
with the caller.  It now continues in the dialplan.

* AgentRequest returns AGENT_STATUS set to NOT_CONNECTED if the agent
failed to connect with the call.  Most likely because the agent did not
acknowledge the call in time or got disconnected.

* The agent alerting play file configured by the agent.conf custom_beep
option can now be disabled by setting the option to an empty string.  The
agent is effectively alerted to a call presence when MOH stops.

* Fixed bridge reference leak when the agent connects with a caller.

ASTERISK-23499 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3551/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23582 #close
ASTERISk-23582 #comment Reported by: Walter Doekes

Review: https://reviewboard.asterisk.org/r/3557/
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#ASTERISK-23792 #close
Reported by: Peter Whisker

Review: https://reviewboard.asterisk.org/r/3567/
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Asterisk started counting the session timer at INVITE while the other
end correctly started at 200. This meant that for short session-expiries
(90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk
would wrongly assume that the timer was hit before the other end thought
it was time to send a session refresh. This resulted in prematurely
ended calls.

This changes the session timer to start counting first at 200 like RFC
says it should.

(Also removed a few excess NULL checks that would never hit, because if
they did, asterisk would have crashed already.)

ASTERISK-22551 #close
Reported by: i2045 

Review: https://reviewboard.asterisk.org/r/3562/
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The ODBC realtime driver uses ^NN parameter encoding to cope with the
special meaning of the semi-colon. A semi-colon in a field is
interpreted as if the key was supplied twice, something which isn't
otherwise possible with fixed database columns. E.g. allow=alaw;ulaw
is parsed as allow=alaw and allow=ulaw. A literal semi-colon is
rewritten to ^3B when stored in the database.

The module uses a stringfield to efficiently store the encoded
parameters. However, this stringfield wasn't always freed in some
off-nominal cases.

Commit r413241 fixed initialization so the encoding for INSERT and
DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked
apparently.) But that commit forgot the frees. This change cleans
that up.

Review: https://reviewboard.asterisk.org/r/3555/
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When a channel is destroyed (such as via ast_channel_release in off nominal
paths in core_unreal), it will attempt to free (via ast_free) the channel tech
pvt. This is problematic for a few reasons:
1. The channel tech pvt is an ao2 object in core_unreal. Free'ing the pvt
   directly is no good.
2. The channel tech pvt's reference count is dropped just prior to calling
   ast_channel_release, resulting in the pvt's destruction. Hence, the
   channel destructor is free'ing an invalid pointer.

This patch keeps the dropping of the reference count, but sets the pvt to
NULL on the channel prior to releasing it. This models what would occur if the
channel was hung up directly.
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