In ast_rtp_ice_start if the ice session create check list failed, start check
was never initiated and ice_started was never set to true.  Upon re-entering
the function (for instance, [un]hold) it would try to create the check list
again with duplicate remote candidates.

Fixed so that if the create check list fails the necessary data structures
are properly re-initialized for any subsequent retries.

Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a
check list failure because it possible things might still work.  However, a
debug message was added to help with any future troubleshooting.

(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Patches:
     works_on_my_machine.patch uploaded by xytis (license 6558)
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When waitmarked users join a ConfBridge, the conference state is transitioned
from EMPTY -> INACTIVE. In this state, the users are maintined in a waiting
users list. When a marked user joins, the ConfBridge conference transitions
from INACTIVE -> MULTI_MARKED, and all users are put onto the active list of
users. This process works correctly.

When the marked user leaves, if they are the last marked user, the MULTI_MARKED
state does the following:
(1) It plays back a message to the bridge stating that the leader has left the
    conference. This requires an unlocking of the bridge.
(2) It moves waitmarked users back to the waiting list
(3) It transitions to the appropriate state: in this case, INACTIVE

However, because it plays the prompt back to the bridge before moving the users
and before finishing the state transition, this creates a race condition: with
the bridge unlocked, waitmarked users who leave the conference (or are kicked
from it) can cause a state transition of the bridge to another state before
the conference is transitioned to the INACTIVE state. This causes the state
machine to get a bit wonky, often leading to a crash when the MULTI_MARKED state
attempts to conclude its processing.

This patch fixes this problem:
(1) It prevents kicked users from being kicked again. That's just a nicety.
(2) More importantly, it fixes the race condition by only playing the prompt
    once the state has transitioned correctly to INACTIVE. If waitmarked users
    sneak out during the prompt being played, no harm no foul.

Review: https://reviewboard.asterisk.org/r/3108/

Note that the patch committed here is essentially the same as uploaded by
Simon Moxon on ASTERISK-22740, with the addition of the double kick prevention.

(closes issue AST-1258)
Reported by: Steve Pitts

(closes issue ASTERISK-22740)
Reported by: Simon Moxon
patches:
  ASTERISK-22740.diff uploaded by Simon Moxon (license 6546)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds support for Lua 5.2 in pbx_lua which is available on newer
operating systems.

(closes issue ASTERISK-23011)
Review: https://reviewboard.asterisk.org/r/3075/
Reported by: George Joseph
Patch by: George Joseph
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When the patch to add retry-on-forbidden-response was committed, part
of the patch for chan_sip was not committed which caused the feature to
be entirely nonfunctional. This corrects the code in question.

(closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405035 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the 'rewrite_contact' option was enabled and a Contact header was received
which contained a '*' a crash would occur.

This change makes the res_pjsip_nat module ignore the Contact header if it
contains only a '*'.

(closes issue ASTERISK-23101)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The external MWI AMI interface provides a thin wrapper around the core
external MWI resource.

The resource adds the following AMI actions:
MWIGet,
MWIDelete, and
MWIUpdate.

(closes issue AFS-46)

Review: https://reviewboard.asterisk.org/r/3061/
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* The core external MWI resource provides for MWI message counts
persistence using sorcery.  With sorcery, the user is able to configure
which sorcery wizzard backend to use if the default astdb is not desired.

* The core external MWI resoruce provides some debugging CLI commands
enabled by defining MWI_DEBUG_CLI.

The debugging CLI commands are:
"mwi delete all",
"mwi delete like <regex>",
"mwi delete mailbox <mailbox>",
"mwi list all",
"mwi list like <regex>",
"mwi show mailbox <mailbox>", and
"mwi update mailbox <mailbox> [<new> [<old>]]".

(closes issue AFS-43)

Review: https://reviewboard.asterisk.org/r/3061/
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Depending on which threading was loading the outbound registration it was
possible for the registration client to be allocated outside of a pj thread.
This change moves the creation inside the synchronous task where it is
guaranteed it will occur in a pj thread.

Reported by: Rob Thomas
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Even since the fixes of AST-2013-007, Asterisk prints the following
warning on startup if the user decided to live dangerously:

  Privilege escalation protection disabled!
  See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.

This message is intended for the logs and interactive startup. No need
for it to appear on a remote console. This commit removes it from there.

(closes issue ASTERISK-23084)
Review: https://reviewboard.asterisk.org/r/3101/
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Upon reload the module unconditionally "unloaded" the module (freeing memory
and setting pointers to NULL) and then when attempting a "load" if the config
file had not changed then nothing would be reinitialized.

By moving the "unload" to occur conditionally (reload only) after an attempted
configuration load, but before module "loading" alleviates the issue. The module
now loads/unloads/reloads correctly.

(closes issue ASTERISK-22871)
Reported by: Matteo
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Registering yourself with the Asterisk core is the nice thing to do, even
when you're a logging module.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

An md5 hash is 32 bytes long. The char buffer must be at least 33 bytes to
avoid clobbering of the stack. This patch also fixes a potential clobbering
in test_utils.c.

Thanks to Andrew Nagy for reporting and testing this out in #asterisk-dev

Reported by: Andrew Nagy
Tested by: Andrew Nagy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AMI action UserEvent event response would include the action header in its
keyvalue pairs list. Adjusted the start of the header loop to skip over the
action part.

(closes issue ASTERISK-22899)
Reported by: outtolunc
Patches:
     svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license 5198)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:

 'DAHDI/i1/1408409XXXX-6'

then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.

(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
     svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed a compiler warning (errors in 'dev-mode') given by gcc version 4.8.1.
The one in app_meetme involved the 'sizeof-pointer-memaccess'
(see: http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so
it would no longer issue a warning and can compile again in 'dev-mode'.

Review: https://reviewboard.asterisk.org/r/3098/
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This change moves outbound registration URI validation into the task executed
within a pjlib thread.

Reported by: Andrew Nagy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When calling REPLACE() with an empty replace-char argument, strcpy
is used to overwrite the the matching <find-char>.  However as the
src and dest arguments to strcpy must not overlap, it causes other
parts of the string to be overwritten with adjacent characters and
the result is mangled.  Patch replaces call to strcpy with memmove
and adds a test suite case for REPLACE.

(closes issue ASTERISK-22910)
Reported by: Gareth Palmer
Review: https://reviewboard.asterisk.org/r/3083/
Patches:
    func_strings.patch uploaded by Gareth Palmer (license 5169)
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Added a new 'set_var' option for ast_sip_endpoint(s).  For each variable
specified that variable gets set upon creation of a pjsip channel involving
the endpoint.

(closes issue ASTERISK-22868)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3095/
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Channel creation in Asterisk is broken up into two steps: requesting and calling.
In some cases a channel may be requested but never called. This happens in the
ChanIsAvail dialplan application for determining if something is reachable or
not. The PJSIP channel driver did not take this situation into account and
attempted to end a session that was never called out on.

The code now checks the session state to determine if the session has been
called out on and if not terminates it instead of ending it.

(closes issue ASTERISK-23074)
Reported by: Kilburn
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Hostnames specified in the 'match' field will be resolved and all addresses
returned. Each address will be added to the endpoint identifier for the
matching process.

Reported by: Rob Thomas
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A deadlock can happen between a thread unloading or reloading the cel_pgsql
module and the core_event_dispatcher taskprocessor thread. Description of
what is happening:

Thread 1 (for example, a netconsole thread):

    a "module reload cel_pgsql" is launched
    the thread enter the "my_unload_module" function (cel_pgsql.c)
    the thread acquire the write lock on psql_columns
    the thread enter the "ast_event_unsubscribe" function (event.c)
    the thread try to acquire the write lock on ast_event_subs[sub->type]

Thread 2 (core_event_dispatcher taskprocessor thread):

    the taskprocessor pop a CEL event
    the thread enter the "handle_event" function (event.c)
    the thread acquire the read lock on ast_event_subs[sub->type]
    the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events
    the thread try to acquire a read lock on psql_columns

(closes issue ASTERISK-22854)
Reported by: Etienne Lessard
Patches:
     cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license 6394)
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When applying configuration for outbound registrations the 'server_uri' and
'client_uri' fields were not validated. The code will now confirm that they
exist and that they contain parseable SIP URIs.

Reported by: Andrew Nagy
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'core show channeltypes' type column is being sliced, resulting in incomplete
type names.

(closes issue ASTERISK-22919)
Reported by: outtolunc
Patches:
     svn_channel.c.format_15.diff.txt uploaded by outtolunc (license 5198)
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Asterisk does not support any of the transfer encodings specified in
HTTP/1.1, other than the default "identity" encoding.

According to RFC 2616:

   A server which receives an entity-body with a transfer-coding it does
   not understand SHOULD return 501 (Unimplemented), and close the
   connection. A server MUST NOT send transfer-codings to an HTTP/1.0
   client.

This patch adds the 501 Unimplemented response, instead of the hard work
of actually implementing other recordings.

This behavior is especially problematic for Node.js clients, which use
chunked encoding by default.

(closes issue ASTERISK-22486)
Review: https://reviewboard.asterisk.org/r/3092/
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When destroying a subscription we remove the serializer from its dialog
and decrease its reference count. Depending on which thread dropped the
subscription reference count to 0 it was possible for this to occur in
a thread where it is not possible.

(closes issue ASTERISK-22952)
Reported by: Matt Jordan
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If ignore_failed_channels is set to "true" for a channel, the channel
will continue to be configured even if configuring it has failed.

This allows Asterisk to start before all the DAHDI initialization is
done and thus not force the starting order dahdi -> asterisk.

Review: https://reviewboard.asterisk.org/r/3063/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When wanting to pass *free as a function pointer, ast_free_ptr has to be used
instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled.
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When we added support for specifying channel variables for an
origination, we didn't consider how that would interact with another
feature, namely specifying request parameters in a JSON request body.

The method of specifying channel variables (as a flat JSON object passed
in the JSON body) interferes with parsing parameters out of the request
body.

Unfortunately, fixing this would be a backward incompatible change. In
the interest of keeping the API sane and keeping our release schedule,
we're dropping the feature for specifying channel variables in the
origination request.

We will bring the feature back soon, as a backward compatible addition
to the API.

(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3088
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Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)

Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.

New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.

(issue ASTERISK-22610)
patches:
  pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_say_date_with_format_pl(), change ast_say_number() to
use tm_sec instead of tm_mn.

(closes issue ASTERISK-22856)
Reported by: Robert Mordec
Review: https://reviewboard.asterisk.org/r/3082/
Patches:
     say.c.patch uploaded by veilen (license 6555)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404461 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When transferring to a dialplan extension that will not place any outbound
calls, the only control frames that the PJSIP REFER framehook will receive
are inconsequential (such as unhold or srcchange). As such, we shouldn't
allow for the reception of those types of frames prevent us from signaling
to the transferring party that the transfer has completed successfully once
voice frames are read.

Thanks to Jonathan Rose for pointing this out.
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