git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389343 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This may alleviate some of the CDR woes with originated channels, as CDRs
do like to know when a channel was originated. Eventually this will get
converted to be a channel flag, so its location is still good to know
post the great CDR shakeup of 2013.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389306 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389251 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389247 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389246 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This exposes stasis_app_control_answer and allows
res_stasis_http_channels to load properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389204 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When this option was added, it was noted in CHANGES, but was missing
the XML documentation that this patch adds.

(closes issue ASTERISK-21780)
Patch-by: Brad Latus (snuffy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389180 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21327)
Reported by: wedhorn
Tested by: myself
Patches: 
    skinny-blindxfer01.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks to Brad Latus, this patch adds a significant amount much-needed
documentation to res_sip. It should cover all existing configuration
options currently in Asterisk trunk.

Patch-by: Brad Latus (snuffy)
Review: https://reviewboard.asterisk.org/r/2471/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CallforwardNoAnswer uses a sched to determine when to forward the call. 
Defaults to 20secs but configurable in skinny.conf.

Adds dialType to each subchannel structure to be used to differentiate
between normal dials that result in a call being placed (default) and
other uses for the skinny_dialer (such as cfwd digit collection).
Restructured all cfwd handling to use this new arrangement.

(closes issue ASTERISK-21292)
Reported by: wedhorn
Tested by: myself
Patches: 
    skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a bug where synchronous origination (oddly enough triggered by doing an async manager Originate) would not work properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389085 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21549)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2512/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either before
it was initialized or after it had been cleaned up. It turns out that
this assertion fires during shutdown.

This actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message types
used by the subscription could be freed before the final message of
the subscription was processed.

This patch adds stasis_subscription_join(), which blocks until the
last message has been processed by the subscription. Since joining was
most commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also added.

Similar functions were also added to the stasis_caching_topic and
stasis_message_router, since they wrap subscriptions and have similar
problems.

Other code in trunk was refactored to join() where appropriate, or at
least verify that the subscription was complete before being
destroyed.

Review: https://reviewboard.asterisk.org/r/2540


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, the buffer for processing "inkeys" is limited to 256 characters.  If
the user has many keys and the names of those key files are long, the 256
character limit is not enough.

* Change inkeys buffer to be dynamic

(closes issue ASTERISK-21398)
Reported by: Pavel Kopchyk
Tested by: Pavel Kopchyk, Michael L. Young
Patches:
    asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
					by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2501/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389009 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does two things:
* It fixes a bug where the outbound channel's application/data set by the
  dialing API/app_dial is not communicated until the channel is hung up.
  If that happens, AMI would incorrectly send a NewExten event immediately
  after a Hangup. This isn't really AMI's fault, as the dialing APIs never
  communicated the 'helpful' app/data on the outbound channel until it was
  hungup.
* It makes public sending a stasis message about a change in channel state.
  This is useful enough that - for now at least - it should be public. If
  operations on a channel go to being more coarse-grained, this function
  could be made private again.

Review: https://reviewboard.asterisk.org/r/2548

Note that this problem was found and reported by Matt DiMeo.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also moves ACL messages to the security topic and gets rid of the
ACL topic

(closes issue ASTERISK-21103)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2496/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also added some missing doc comments for stasis/app.h.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying
to access a possible NULL t->track object.  A NULL check has been added before
trying to access the memory.

(closes issue ASTERISK-21724)
Reported by: Corey Farrell
Fixed by: Corey Farrell
Patches:
	ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388840 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The snapshot API contains an option that allow for combining of new 
and old messages within a single snapshot. New messages, however, 
include options beyond just 'INBOX' - it also includes the Urgent 
folder. A previous patch that combined INBOX and Urgent accidentally 
impacted snapshots that attempted to gain messages from just the Old 
folder. This patch fixes the snapshot gathering such that the API 
returns the appropriate messages for the folder selected, with and 
without the combine option.

This should make it more clear about what's happening.

Review: https://reviewboard.asterisk.org/r/2539/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This allows the SRTP library to be shut down properly when the
functionality is offered by libsrtp.

Review: https://reviewboard.asterisk.org/r/2538/
(closes issue ASTERISK-21719)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

macros.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.

This patch breaks the major components of res_stasis.c into individual
files.

 * res/stasis/app.c - Stasis application tracking
 * res/stasis/control.c - Channel control objects
 * res/stasis/command.c - Channel command object

This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.

The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.

Review: https://reviewboard.asterisk.org/r/2530/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.

(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
      jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This moves the JSON event generators out of the Stasis-HTTP modules and
into standalone JSON-related counterparts so that Stasis-HTTP and
res_stasis can depend on them without creating dependency cycles. This
also provides a future location for Swagger Model validator functions
once the generators for that code are written.

Review: https://reviewboard.asterisk.org/r/2534/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388668 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CALL-ID (ie [C-00000074]) is missing when logging to syslog.  This was just
an oversight when this feature was added.

* Add CALL-IDs when using syslog

(closes issue ASTERISK-21430)
Reported by: Nikola Ciprich
Tested by: Nikola Ciprich, Michael L. Young
Patches:
    asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2526/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The prior code committed, r385473, failed to take into consideration that not
all outgoing calls will be to a peer.  My fault.

This patch does the following:

* Check if there is a related peer involved.  If there is, check and set NAT 
  settings according to the peer's settings.

* Fix a problem with realtime peers.  If the global setting has auto_force_rport
  set and we issued a "sip reload" while a peer is still registered, the peer's
  flags for NAT are reset to off.  When this happens, we were always setting the
  contact address of the peer to that of the full contact info that we had.

(closes issue ASTERISK-21374)
Reported by: jmls
Tested by: Michael L. Young
Patches:
   asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2524/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388602 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding the cleanup function needs some deeper thought since it
apparently doesn't exist for all variants of libsrtp.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21723)
Reported by: Corey Farrell
Patches:
    core-pbx-cleanup.patch uploaded by Correy Farrell (license 5909)
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Merged revisions 388532 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388579 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure that libsrtp is shutdown properly when res_srtp is unloaded.

(closes issue ASTERISK-21719)
Reported by: Corey Farrell
Patches:
    res_srtp-library-shutdown.patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21670)
Reported by: Snuffy
Review: https://reviewboard.asterisk.org/r/2473/
Patches:
    gulp-coding-guide.diff uploaded by snuffy (license 5024)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AMI actions must never return non-zero unless they intend to close the AMI
connection.  (Which is almost never.)

(closes issue ASTERISK-21779)
Reported by: Paul Goldbaum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made isdn_msg_parser.c build a progress message with the mandatory
progress indicator IE.  (The mISDNuser NT state machine rejected sending
the incomplete message.)

Note: The associated mISDN and mISDNuser patches respectively are viewable
here:
http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
http://svnview.digium.com/svn/thirdparty?view=rev&rev=201

(closes issue AST-1153)
Reported by: Guenther Kelleter
Patches:
      progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter
      progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter
      progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

pbx_dundi added an io context without removing
it. This caused a memory leak when the module was
unloaded.

(closes ASTERISK-21718)
Reported by Corey Farrell
Patches:
	pbx_dundi-ast_io_remove.patch uploaded by Corey Farrell (License #5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After the merge of support for the realtime sorcery module, extensions that
contained a pattern were not being found through odbc realtime.  It was tracked
down to this one line that was advancing to the next variable list before it
should have been.  The removal of this one line fixes this.

Tested this fix on my machine.

Received confirmation that this is the right fix from file on IRC.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I've noticed when doing a graceful shutdown that the res_stasis_http.so
module gets unloaded before the modules that use it, which causes some
asserts during their unload.

While r386928 was a quick hack to get it to not assert and die, this
patch increases the use counts on res_stasis.so and res_stasis_http.so
properly. It's a bigger change than I expected, hence the review instead
of just committing it.

Review: https://reviewboard.asterisk.org/r/2489/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

STASIS_MESSAGE_TYPE_*() macros.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388318 65c4cc65-6c06-0410-ace0-fbb531ad65f3