There were several problems using variadic argument macros in res_config_mysql.
 * Improper use of va_end. Multiple calls to va_end were possible resulting in
   an unbalanced matching of va_start/va_end.
 * Calls to va_arg after a possible encounter of a SENTINEL value.

This patch corrects those errors.

(closes issue ASTERISK-19451)
Reported by: wdoekes
patches:
  ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Basically sets the callerid and callername to the first device talked to for the
purposes of putting the the calls made log on the device. Does not affect the device
displaying who the device is currently talking to.

Also some minor changes to use sub->exten in lieu of l->lastnumberdialed.

(closes issue ASTERISK-21095)
Reported by: wedhorn
Tested by: snuffy, myself
Patches: 
    skinny-calllogsoutbound03.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds both fixed and variable prinotify messages and clearprinotify messages to skinny.
Also adds cli function for pushing messages to devices. i

Initial code by snuffy, expanded by myself to include fixed messages.

(closes issue ASTERISK-21091)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    skinny-prinotify02.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Somehow, chan_jingle has managed to operate for years without setting the
sin_family on its bindaddr socket. This patch properly sets the field during
initial module load to AF_INET.

Note that the patch on the issue was modified slightly to change the
initialization of the socket from allocation of a chan_jingle private to the
module initialization, as the bindaddr object (which is static) only needs to
have the address set once.

(closes issue ASTERISK-19341)
Reported by: andre valentin
patches:
  0105-chan_jingle.patch uploaded by avalentin (License 6064)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)

Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.

Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.

(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381949 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:

return_context
    The goto-style label to jump the call back into after timeout. Default
    'priority+1'.

Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.

(closes issue ASTERISK-20113)
Reported by: serginuez
patches:
  app_parkandannounce.diff uploaded by serginuez (License 6405)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381918 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When IPv6 support was added to FastAGI, the intent was to have the ability to
check all addresses resolved for a host since we might receive an IPv4 address
and an IPv6 address.  The problem with the current code, is that, since we are
doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring
this instead of handling it.  We break out of the loop and continue on.  When we
later call ast_poll(), it succeeds but we never check if we have a connection or
not on the socket level.  We then attempt to send data to the host address that
we think is setup and it fails.  We then check the errno and see that we have
"connection refused" and then return with agi failed.

This patch does the following:

* Handles EINPROGRESS by creating the function handle_connection()
  - ast_poll() was moved into this function
  - This function checks the results of the connection on the socket level after
    calling ast_poll()
* Continues to the next address if the above fails to create a connection
* Once all addresses resolved are tried and we still are unable to establish a
  connection, then we return that the FastAGI call failed

(closes issue ASTERISK-21065)
Reported by: Jeremy Kister
Tested by: Jeremy Kister, Michael L. Young
Patches:
  asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2330/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Apparently this feature became broken in 11, probably as a result
of the Hangup Cause project.

(closes issue ASTERISK-21113)
Reprted by: Heiko Wundram
Patches:
	app_dial.patch uploaded by Heiko Wundram (license 5822)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381881 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk was a little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in certain items
in Asterisk that conflict with launchd to not be selectable, such as
res_timing_kqueue.

(closes issue ASTERISK-20749)
Reported by: Oleg Baranov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381869 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The incorrect callid was being written to the "data1" field in queue_log table
for transfer events.  The callid of the queue was being written instead of the
transfer target's callid.  This now gets the correct "transfer to" number and
places that in the "data1" field of the queue_log table when a transfer event
is triggered.

(closes issue ASTERISK-19960)
Reported by: vladimir shmagin
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a module's configuration is not loadable, we still load the module but it
is not in a running state.  When trying to troubleshoot, let's say, why
chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a
loaded module is not currently running.

(closes issue ASTERISK-21108)
Reported by: Rusty Newton
Tested by: Michael L. Young
Patches:
  asterisk-21108_add_status-v2.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2331/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If you started/stopped recording of a conference multiple times channels
would remain active even when all participants left the conference.  This
was due to the fact that a reference to the confbridge was being added
every time a start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the conference alive.
Made sure only a single reference is added for the record thread no matter how
many times recording is started/stopped.  A de-reference is issued upon thread
ending.

Note, this issue is being fixed under AST-1088 since it relates to it and
should have been corrected along with those modifications.

(issue AST-1088)
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, if one starts, stops, and then starts a recording again for a
conference the recorded data is appended to the file originally created
on the first record start.  An option record_file_append has been added
that defaults to "yes", but when set to "no" will force creation of a new
file between every record start/stop.

(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patch adds all the packet and structure stuff to skinny to enable setting 
service URLs in skinny, such as corporate directories.

This stuff is only relevant during load/unload as when activated. Also 
some minor changes removing duplicated counting of addons and speedials in 
handle_skinny_show_devices.

Review: https://reviewboard.asterisk.org/r/2321/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Auto complete for skinny debug allows multiple options and negation, also add 
debug all option. Usage example: 'skinny debug all -packets' (each can be 
autocompleted including -packet).

Change show device to use device name. Remove the duplicate ast_strdup's from 
place calling device complete return immediately from complete devicename and 
complete linename so that multiple options are displayed on the CLI if more 
than one option available.

Review: https://reviewboard.asterisk.org/r/2333/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A deadlock occurred after starting/stopping and then restarting a confbridge
recording.  Upon starting a recording a record thread is created that holds a
lock until just before exiting.  Stopping the recording does not stop/exit the
thread or release the lock.  The thread waits until recording begins again.
Starting a stopped recording signals the thread to continue and start recording
again.  However restarting the recording also created another record thread
resulting in a deadlock.  The fix was to make sure the record thread was only
created once.

Also it was noted that filenames for the recordings were being concatenated for
each start/stop.  This was fixed by creating a new file for each conference
session and appending the actual recorded data within the file (e.g. passing
the 'a' option to MixMonitor).

(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "registertrying" option was removed in r343220. The "rtp_engine"
option was added in r186078 but erroneously named "engine" in the sample.
Note that there is no global sip setting for a different engine.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381670 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Notes that the 'e' option actually decodes data when used as a write function
such as with the SET application while it encodes data when used to read.

Review: https://reviewboard.asterisk.org/r/2335/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added the following flags to the CLI "confbridge list <conference>" output:
A - The user is an admin
M - The user is a marked user
W - The user must wait for a marked user to join
E - The user will be kicked after the last marked user leaves the conference
w - The user is waiting for a marked user to join

* Added the following header to the AMI ConfbridgeList events:
WaitMarked, EndMarked, and Waiting.

(closes issue AST-1101)
Reported by: John Bigelow
Patches:
      confbridge-show-admin3.txt (license #5091) patch uploaded by John Bigelow
      Modified


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381644 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously, presencestate information was sent whenever the state was not
NOT_SET. When r381594 actually returned INVALID presence state in all the
places it was supposed to, it caused chan_sip to start adding presence
state information to NOTIFY requests that it previously would not have
added. chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to invalid and
an invalid state always implies that the provider is in an error condition.

(issue AST-1084)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2329/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.

(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reference counting for the channel and its tech_pvt got messed up at
some point between 1.8 and 11. The result was that if a BYE for a dialog
that had been replaced (via an INVITE with Replaces) was received, Asterisk
would crash due to trying to access data on a channel that was no longer there.

The fix I introduced is to remove code that both unrefs the sip_pvt and sets
the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This
way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can
be processed and not cause a crash.

(closes issue ASTERISK-20929)
reported by Kristopher Lalletti
patches:
	ASTERISK-20929.patch uploaded by Mark Michelson (License #5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does two things:
 1. It disables (temporarily) strict XML documentation checking for module
    configurations. We should re-enable it before making any release from
    trunk.
 2. Pass the module flag AST_MODULE through sorcery. This means several of the
    API calls are now macros and will do this automatically for you. The config
    framework needs the module that objects are registering to so it can
    properly construct the documentation. (This was already a required field,
    but sorcery was getting by without it)




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.

Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything).  If it is different then
store/replace and log, otherwise just leave as is.  Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.

(issue ASTERISK-21014)
(closes issue ASTERISK-21014)
Report by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2324/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Some bad copy/pasting resulted in using the audio crypto attribute for both
text and video RTP. Also the audio crypto isn't set until after these, so it
was really just bad all around.

(closes ASTERISK-20905)
Reported by: Kristopher Lalletti
patches:
	rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:

1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
   options that use the configuration framework

This patch include configuration documentation for the following modules:
 * chan_motif
 * res_xmpp
 * app_confbridge
 * app_skel
 * udptl

Two new CLI commands have been added:
 * config show help - show configuration help by module, category, and item
 * xmldoc dump - dump the in-memory representation of the XML documentation to
   a new XML file.

Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058

patches:
  on review 2058 uploaded by twilson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't know why it seemed to work during testing, but it really is needed 
for protocol v17 (and probably above).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either, but it should not
be set outside of a bridge.

(issue ASTERISK-20492)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix chan_skinny so that it respects callerID presentation of inbound calls to 
device and a couple of other minor fixes: 145 packet (add OCTAL_FROM amd callerid),
and dont send dialednumber message if protocol >= 17. 

(closes issue ASTERISK-21066)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    skinny-respect-clid-restrictions-v2.diff uploaded by snuffy (license 5024)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381465 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The core module related to coloring terminal output was old and needed
some love.  The main thing here was an attempt to get rid of the
obscene number of stack-local buffers that were allocated for no other
reason than to colorize some output.  Instead, this uses a simple trick
to allocate several buffers within threadlocal storage, then
automatically rotates between them, so that you can make multiple calls
to the colorization routine within one function and not need to
allocate multiple buffers.

Review: https://reviewboard.asterisk.org/r/2241/
Patches:
    bug.patch uploaded by Tilghman Lesher


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While adding red-black tree containers to astobj2 in r376575, Richard pointed
out the way chan_iax2 finds unused call numbers will prevent ao2_container
integrity checks at runtime.

This patch removes the ao2_container and instead uses fixed sized arrays and a
modified Fisher-Yates-Durstenfeld shuffle to maintain the call number list.

While the locking semantics are similar to the ao2_container implementation,
this implementation should be faster and more memory efficient.

Review: https://reviewboard.asterisk.org/r/2288/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381398 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The function call ast_db_deltree returns the number of row deleted, or a
negative number if it failed. DBdeltree was treating any non-zero return
as an error, causing a spurious verbose error message to be displayed.

This patch handles the return code of ast_db_deltree correctly.

(closes issue ASTERISK-21070)
Reported by: ianc
patches:
  dbdeltree.diff uploaded by ianc (License #5955)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381366 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the ability to create a serializer from a thread pool. A
serializer is a ast_taskprocessor with the same contract as a default
taskprocessor (tasks execute serially) except instead of executing out
of a dedicated thread, execution occurs in a thread from a
ast_threadpool. Think of it as a lightweight thread.

While it guarantees that each task will complete before executing the
next, there is no guarantee as to which thread from the pool individual
tasks will execute. This normally only matters if your code relys on
thread specific information, such as thread locals.

This patch also fixes a bug in how the 'was_empty' parameter is computed
for the push callback, and gets rid of the unused 'shutting_down' field.

Review: https://reviewboard.asterisk.org/r/2323/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381326 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The RTP engine will no longer allow for local and remote native RTP bridges
if packetization of streams differs. Allowing native bridging in this scenario
has been known to cause FAX failures.

(closes ASTERISK-20650)
Reported by: Maciej Krajewski
Patches:
	ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)

Review: https://reviewboard.asterisk.org/r/2319
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381285 65c4cc65-6c06-0410-ace0-fbb531ad65f3