* Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h.
* Fix incorrect function parameters in utils/extconf.c.

(closes issue ASTERISK-23141)
Reported by: Maxim
Review: https://reviewboard.asterisk.org/r/3241/
........

Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408786 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408787 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk didn't support the dynamic payload change in rtp mapping in the 200
OK response.

Scenario:
Asterisk sends the INVITE proposing alaw and telephone-event, it proposes
rtpmap:101 for telephone-event.  Peer responds with 2xx, it answers with
alaw and telephone-event also, but it proposes a different rtpmap number
(rtpmap:103) for telephone-event.

Expected Behaviour:
Asterisk should honour the rtpmapping in the response and send DTMF packets
using 103 as payload type for DTMF.

Actual Behaviour: Asterisk sends DTMF packets using payload type 101.

With this patch asterisk now supports changes that can occur in the rtp mapping
in the response.

(closes issue ASTERISK-23279)
Reported by: NITESH BANSAL
Review: https://reviewboard.asterisk.org/r/3225/
Patches:
     dynamic_payload_change.patch uploaded by nbansal (license 6418)
........

Merged revisions 408729 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408730 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed use of uninitialized ao2 container iterator in an off-nominal
condition.  Either a memory allocation error or the requested channel is
an internal channel not exposed to the outside.
........

Merged revisions 408715 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408723 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed off-nominal json ref counting issue with using the following API
calls: ast_json_object_set() and ast_json_array_append().

* Fixed off-nominal error reporting in ast_ari_endpoints_list().

* Fixed some miscellaneous off-nominal json ref counting issues in
report_receive_fax_status() and dial_to_json().
........

Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed json ref counting issue with json API wrapper code for
ast_json_object_update_existing() and ast_json_object_update_missing()
when the json library is earlier than version 2.3.0.
........

Merged revisions 408711 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix regression caused by ASTERISK-22582.  Empty Route
headers were added when the route had a single strict
hop.

(closes issue ASTERISK-23306)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3236/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed the output of CHANNEL(rtpqos,audio,all) to use txjitter instead
of rxjitter.

(closes issue ASTERISK-23261)
Reported by: rsw686
Patches:
     rtpqos.patch uploaded by rsw686 (license 5887)
........

Merged revisions 408646 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408647 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408649 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.

Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.

The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.

Credit to Olle Johansson for pointing me in the right direction on this issue.

(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
........

Merged revisions 408642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408643 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408644 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408645 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 408589 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408590 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408591 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408592 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using the "x" option (specify a DTMF digit to exit the application), it is
not obvious in the documentation that this only works when spying on a channel.
If a channel being used to spy on other channels is waiting to connect to a
channel or is no longer attached to a channel, the DTMF is ignored.

As noted on the issue tracker, since there are workarounds available and this is
a rarely used option we are opting for a documentation change here.

(closes issue ASTERISK-22661)
Reported by: Chris Hillman
Patches:
    asterisk-22661-doc-clarify-chan_spy.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2990/
........

Merged revisions 408536 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408537 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408538 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408539 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added 'show registrations' and 'show contacts' to pjsip cli to make things
a little more consistent.  The output is exactly the same as the list command.

Just needed to add entries to their respective ast_cli_entry structures.

(closes issue ASTERISK-23275)
Review: http://reviewboard.asterisk.org/r/3210/
........

Merged revisions 408522 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408523 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed memory leaks in ast_sip_cli_print_sorcery_objectset and
ast_variable_list_sort.  

(closes issue ASTERISK-23266)
Review: http://reviewboard.asterisk.org/r/3200/
........

Merged revisions 408520 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408521 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.

ast_sorcery_init now creates a hashtab as a registry.

ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name.  If it finds one, it bumps the 
refcount and returns it.  If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.

ast_sorcery_retrieve_by_module_name is a new function that does a hashtab 
lookup by module name.  It can be called by the future dialplan function.

res_pjsip/config_system needed a small change to share the main res_pjsip 
sorcery instance.

tests/test_sorcery was updated to include a test for the registry.

(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3184/
........

Merged revisions 408518 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When 'use_avpf' is set to True, inbound offers must use the AVPF/SAVPF RTP
profile. However, when 'use_avpf' is set to False, Asterisk will accept
both AVP/SAVP or AVPF/SAVPF RTP profiles in inbound offers. The documentation
previously implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was
set to False and a UA offered said profile in an INVITE request.
........

Merged revisions 408502 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Macro is executed on the called channel, not the calling channel.

(closes issue ASTERISK-23069)
Reported By: Bryan Anderson
........

Merged revisions 408447 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408448 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408449 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408450 65c4cc65-6c06-0410-ace0-fbb531ad65f3

config: Add file size and nanosecond resolution fields to the cached modified config file information.

Repeatedly modifying config files and reloading too fast sometimes fails
to reload the configuration because the cached modification timestamp has
one second resolution.

* Added file size and nanosecond resolution fields to the cached config
file modification timestamp information.  Now if the file size changes or
the file system supports nanosecond resolution the modified file has a
better chance of being detected for reload.

* Added a missing unlock in an off-nominal code path.

(closes issue AST-1303)

Review: https://reviewboard.asterisk.org/r/3235/
........

Merged revisions 408387 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408388 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408389 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408390 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The sorcery astDB wizzard does not handle regex correctly if the pattern
begins with an anchor character.

This patch attempts to convert the anchored regex pattern to a prefix
pattern supported by astDB for performance reasons.  If it is not able to
convert the pattern it falls back to getting all astDB members of the
family and doing a normal regex pattern matching on the retrieved records.

Review: https://reviewboard.asterisk.org/r/3161/
........

Merged revisions 408385 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

send receiveAndTransmit user input our caps instead of receive only
........

Merged revisions 408328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408330 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408331 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408332 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by: Gabriele Odone
Patches:
	ASTERISK-22738-1.patch
Tested by: Gabriele Odone
(closes issue ASTERISK-22738)
........

Merged revisions 408312 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408314 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408315 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 408297 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When an endpoint sends a REGISTER request to Asterisk, we now will
associate the User-Agent header with all contacts that were bound in
that REGISTER request.
........

Merged revisions 408270 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.

This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.

(closes issue ASTERISK-23297)
Reported by: CJ Oster

Review: https://reviewboard.asterisk.org/r/3222
........

Merged revisions 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408201 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408220 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/ ) broke the
build. This patch fixes it by ignoring the .lastclean dependencies if the
MENUSELECT_EMBED variable is not defined.

patches:
  tmp.diff uploaded by wdoekes (License 5674)

Review: https://reviewboard.asterisk.org/r/3228/
........

Merged revisions 408193 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408194 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408199 65c4cc65-6c06-0410-ace0-fbb531ad65f3

URI's are supposed to be case sensitive and all
lower case.  In practice some portions of URI's
in ARI are case insensitive and others are not,
such as TECH, which in one instance would match
a lower case name and in another would not.  In
this patch, the ast_endpoint_lastest_snapshot()
function is modified to change the TECH portion
to full upper case before lookup. This resolves
the discrepancy noted by the reporter.  However
I chose to avoid forcing the /ari prefix of the
URI's to be lower case for now.  Except for the
two cases here, all URI's should be lower case,
unless they are part of a resource name or id.

Review: https://reviewboard.asterisk.org/r/3211/
Reported by: Zane Conkle
(closes issue ASTERISK-23125)
........

Merged revisions 408140 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408141 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later.  This patch clears up and corrects the test.

Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
     main_format.patch uploaded by marcelloceschia (license 6036)
	 ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 408137 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408138 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Directory 'main' only needs to depend on embedded modules. If no
module embedding is selected, the dependency is dropped.

Review: https://reviewboard.asterisk.org/r/3212/
........

Merged revisions 408083 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 408084 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 408085 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.

This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.

This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.

(closes issue ASTERISK-23290)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3214/
........

Merged revisions 408069 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A helper script to copy a source file substituting any
__ASTERISK_<foo>_DIR__ with the content of $AST<foo>DIR.

Review: https://reviewboard.asterisk.org/r/3202/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408059 65c4cc65-6c06-0410-ace0-fbb531ad65f3

PJSIP has built-in MWI code that could be useful to some
degree, but our utilization of the API actually made our
code a bit more cluttered since we had to have special
cases peppered throughout.

With this change, we move to using the pjsip_evsub API
instead, which streamlines the code by removing special
cases.

Review: https://reviewboard.asterisk.org/r/3205
........

Merged revisions 408005 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If an AOR has no permanent contacts, then the
permanent_contacts container is never allocated.
This makes the code safe in the face of NULLs.

I also changed the variable that counts contacts
from "num" to "total_contacts" since there are now
two variables that are indicate numbers of things.
........

Merged revisions 407988 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407990 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the ability to dynamically add and remove logger channels
from Asterisk via the CLI.

(closes issue AST-1150)
Review: https://reviewboard.asterisk.org/r/3185/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The old code depended on undefined va_arg behaviour: calling a function
twice with the same va_list parameter and expecting it to continue where
it left off. The changed code behaves like the manpage says it should.

Also added a bunch of early returns to trap errors (e.g. OOM) instead of
crashing.

The problem was found by Julian Lyndon-Smith. The deviant behaviour on
the raspberry PI also uncovered another bug (fixed in r407875) in the
res_config_pgsql.so driver.

Reported by: jmls
Tested by: jmls
Review: https://reviewboard.asterisk.org/r/3201/
........

Merged revisions 407968 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407970 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is a first stab at tweaking the performance profile of the scheduler. Removing
the hashtab usage removes an extra memory allocation when scheduling something and
makes it so rescheduling does not incur any memory allocation at all.

Review: https://reviewboard.asterisk.org/r/3199/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch tweaks the behaviour of POST /channels with channel variables such
that the variables are passed into the pbx.c routines that perform the
origination. This allows the variables to be assigned to the newly created
channels immediately upon their construction, as opposed to be assigned after
the originate has completed.

The upshot of this is that the variables are available on the channels if
they execute in the dialplan, as opposed to only being available once the
channels are answered.

Review: https://reviewboard.asterisk.org/r/3183/
........

Merged revisions 407937 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Move route code to sip/route.c + sip/include/route.h
* Rename functions to sip_route_*
* Replace ad-hoc list code with macro's from linkedlists.h
* Create sip_route_process_header() to processes Path and Record-Route headers
  (previously done with different code in build_route and build_path)
* Add use of const where possible
* Move struct uriparams, struct contact and contactliststruct from sip.h to
  reqresp_parser.h.  sip/route.c uses reqresp_parser.h but not sip.h, this was
  a problem.  These moved declares are not used outside of reqresp_parser.
* While modifying reqprep() the lack of {} caused me trouble.  I added them.
* Code outside route.c treats sip_route as an opaque structure, using macro's
  or procedures for all access.

(closes issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3173/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407926 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix so multiple updates from a single call works (add missing ',').
Remove bogus ast_free's that weren't supposed to be there.
Moved a few spaces for readability.

Review: https://reviewboard.asterisk.org/r/3194/
........

Merged revisions 407873 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 407874 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 407875 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407876 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.

This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.

(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
........

Merged revisions 407857 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 407858 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407859 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 407764 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 407765 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 407766 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When formatting an optional IE, the value is, of course, optional. As such, it
is entirely appropriate for ast_json_object_get to return NULL. If that occurs,
we now simply skip the IE that was requested, as it was not provided by the
entity that raised the event.

Thanks to George Joseph (gtjoseph) for catching this and reporting it in
#asterisk-dev
........

Merged revisions 407750 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407752 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change allows timing implementation data to be stored directly
on the timer itself thus removing the requirement for many
implementations to do a container lookup for the same information.

This means that API calls into timing implementations can directly
access the information they need instead of having to find it.

Review: https://reviewboard.asterisk.org/r/3175/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407749 65c4cc65-6c06-0410-ace0-fbb531ad65f3