Switch logger callid's from AO2 objects to simple integers.
This helps in two ways.  Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead.  This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.

ASTERISK-24833 #comment Committed callid conversion to trunk. 
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The distinctive ring feature interferes with detecting Caller ID and
appears to have been broken for years.  What happens is if you have a
ring-ring cadence as used in the UK you get too many DAHDI events for the
distinctive ring pattern array and Caller ID detection is aborted.  I
think when Zapata/DAHDI added the ring begin event it broke distinctive
ring.  More events happen than before and the code does no filtering of
which event times are recorded in the pattern array.

* Made distinctive ring only record the ringt count when the ring ends
instead of on just any DAHDI event.  Distinctive ring can be ring,
ring-ring, ring-ring-ring, or different ring durations for the up to three
rings.

* Fixed the distinctive ring detection enable (chan_dahdi.conf option
usedistinctiveringdetection) to be per port instead of somewhat per port
and somewhat global.  This has been broken since v1.8.

* Fixed using the default distinctive ring context when the detected
pattern does not match any configured dringX patterns.  The default
context did not get set when the previous call was a matched distinctive
ring pattern and the current call is not matched.  This has been broken
since v1.8.

* Made distinctive ring have no effect on Caller ID detection when it is
disabled.  Caller ID detection just monitors for 10 seconds before giving
up.

* Fixed leak of struct callerid_state memory when a polarity reversal
during Caller ID detection causes the incoming call to be aborted.

DAHDI-1143
AST-1545
ASTERISK-24825 #close
Reported by: Richard Mudgett

ASTERISK-17588
Reported by: Daniel Flounders

Review: https://reviewboard.asterisk.org/r/4444/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

With the log messages on one line, you can search for the log message seen
in the log and expect to find it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the featdmf signaling mode the incoming MF Caller-ID information is
formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#

Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
with what is received instead.

AST-1368 #close
Reported by: Denis Martinez
Patches:
      extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Remove functions:
	ast_smdi_interface_unref
	ast_smdi_md_message_putback
	ast_smdi_mwi_message_putback
	ast_smdi_md_message destructor
	ast_smdi_mwi_message destructor

Includes for astobj.h are removed everywhere it's possible.

ASTERISK-24066 #close
Review: https://reviewboard.asterisk.org/r/3758/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

PJSIP would never send the final 200 Notify for a blind transfer
when transferring to parking. This patch fixes that. In addition,
it fixes a reference leak when performing blind transfers to
non-bridging extensions.

Review: https://reviewboard.asterisk.org/r/3485/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Original commit message by mmichelson (asterisk 12 r403311):

"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."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 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

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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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

Transferring an analog call using a flash-hook to parking would fail to
park the call and result in an invalid ao2 object unref.

* Park the correct bridged channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.

This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.

Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It adds support for externally initiated parking requests. In particular,
   chan_skinny has a protocol level message that initiates a call park.
   This patch now supports that option, as well as the protocol specific
   mechanisms in chan_dahdi/sig_analog and chan_mgcp.
 * A parking bridge features virtual table has been added that provides
   access to the parking functionality that the Bridging API needs. This
   includes requests to park an entire 'call' (with little or no additional
   information, thank you chan_skinny), perform a blind transfer to a parking
   extension, determine if an extension is a parking extension, as well as the
   actual "do the parking" request from the Bridging API.
 * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
   functions
 * The removal of some - but not all - dead parking code from features.c

This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)

Review: https://reviewboard.asterisk.org/r/2710

(closes issue ASTERISK-22134)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.

A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.

(closes issue ASTERISK-22130)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.

This adds tests for blind transfers, several types of attended
transfers, and call pickup.

The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.

Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change removes JitterBufStats, ChannelReload, and ChannelUpdate
and refactors the following events to travel over Stasis-Core:
* LocalBridge
* DAHDIChannel
* AlarmClear
* SpanAlarmClear
* Alarm
* SpanAlarm
* DNDState
* MCID
* SIPQualifyPeerDone
* SessionTimeout

Review: https://reviewboard.asterisk.org/r/2627/
(closes issue ASTERISK-21476)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21523)
Reported by: Matt Jordan

(closes issue ASTERISK-21524)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2600/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.

In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.

Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.

Review: https://reviewboard.asterisk.org/r/2578/

(issue ASTERISK-21542)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387181 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
  
  Fix misuses of timeouts throughout the code.
  
  Prior to this change, a common method for determining if a timeout
  was reached was to call a function such as ast_waitfor_n() and inspect
  the out parameter that told how many milliseconds were left, then use
  that as the input to ast_waitfor_n() on the next go-around.
  
  The problem with this is that in some cases, submillisecond timeouts
  can occur, resulting in the out parameter not decreasing any. When this
  happens thousands of times, the result is that the timeout takes much
  longer than intended to be reached. As an example, I had a situation where
  a 3 second timeout took multiple days to finally end since most wakeups
  from ast_waitfor_n() were under a millisecond.
  
  This patch seeks to fix this pattern throughout the code. Now we log the
  time when an operation began and find the difference in wall clock time
  between now and when the event started. This means that sub-millisecond timeouts
  now cannot play havoc when trying to determine if something has timed out.
  
  Part of this fix also includes changing the function ast_waitfor() so that it
  is possible for it to return less than zero when a negative timeout is given
  to it. This makes it actually possible to detect errors in ast_waitfor() when
  there is no timeout.
  
  (closes issue ASTERISK-20414)
  reported by David M. Lee
  
  Review: https://reviewboard.asterisk.org/r/2135/
........
  r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
  
  Remove some debugging that accidentally made it in the last commit.
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Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second.  Dial, ExternalIVR, and SendDTMF.

* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'.  The 'w' pauses dialing for half a
second.  The 'W' pauses dialing for one second.

* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.

(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
      jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
      Expanded patch to add support in chan_dahdi.
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds HANGUPCAUSE information to called channels so that hangup
handlers can, in conjunction with predial dialplan execution, access
the hangupcause information when the dialed channel hangs up on a
one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE
usage on the caller channel.

Review: https://reviewboard.asterisk.org/r/2069/
(closes issue ASTERISK-20198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port.  It now also
flash-hooks the correct channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370902 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.

(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make starting analog_ss_thread() or __analog_ss_thread() failure paths
hangup the channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file.  If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file.  Without the python scripts, these XML fragments will not validate.

This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds some basic documentation for a number of modules.  This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri.  The DTD
has also been updated to allow referencing of AMI commands.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It is not necessary to generate information cause code frames on every
protocol event that occurs.  This removes all the instances where the
frame was not conveying a cause code and was instead just conveying a
protocol-specific message.  This also corrects the generation of the
message associated with disconnects for MFC/R2 to use the MFC/R2
specific text for the disconnect cause.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.

review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?"
project and covers the implementation for the technologies implemented
in sig_analog.c and sig_pri.c. Tested on a local machine to verify
protocol and cause information is available.

Review: https://reviewboard.asterisk.org/r/1953/
(issue SWP-4222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369185 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.

(issue ASTERISK-19537)

(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter

(closes issue ASTERISK-19801)
Reported by: Alec Davis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer.  If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.

* Release the channel and private locks when creating a new channel as a
result of a flash hook.

(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
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Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3