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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408590 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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

(closes issue ASTERISK-23069)
Reported By: Bryan Anderson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408448 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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408388 65c4cc65-6c06-0410-ace0-fbb531ad65f3

send receiveAndTransmit user input our caps instead of receive only
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408330 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by: Gabriele Odone
Patches:
	ASTERISK-22738-1.patch
Tested by: Gabriele Odone


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408312 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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408201 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In pbx.c ast_custom_function_unregister(), a list
of escalations being removed from the list wasn't
being free'd creating a leak. This patch corrects
that by freeing the records.

Review: https://reviewboard.asterisk.org/r/3213/
Reported by: Corey Farrell
Patches:
     acf_escalating_leak.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408143 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)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408137 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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "goodbye" parameter is not implemented in the source code, it does nothing.

(closes issue SWP-6518)
Reported By: Steve Pitts
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Merged revisions 408020 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408021 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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407874 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.

This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.

This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).

It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.

Backported from trunk/12.

Review: https://reviewboard.asterisk.org/r/726/
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Merged revisions 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407765 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect.  The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.

For example:
1) v1.4 calls v1.8 (or later) using IAX2

2) v1.8 answers and sends a connected line update control frame.  (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)

3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)

4) v1.4 disconnects the call once the receive queue becomes empty.

Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:

* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.

* Made block sending and receiving control frames that have no reason to
go over the wire.

* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.

* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.

(closes issue AST-1302)

Review: https://reviewboard.asterisk.org/r/3174/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407727 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* If the "stutter" (voicemail indication) tone is indeed a stutter tone,
  and it ends with a constant tone, make sure that it is the dial tone.
  This was done for India (in), Mexico (mx) and the Philippines (ph).
* If no "stutter" tone exists for a country, provide one. This was done for
  Spain (es), Malaysia (my) and Venezuela (ve).

Review: https://reviewboard.asterisk.org/r/3158/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension.

(closes issues ASTERISK-22310)
Reported by: Jim Credland
Review: https://reviewboard.asterisk.org/r/3188/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-23232)
Reported by: Leon Roy


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407457 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes path handling for log files so that an extra / is not
appended to the file path when the path is absolute (begins with /).
This would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
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Merged revisions 407455 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407456 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Nothing actually cares about the value anyway.

(closes issue ASTERISK-23178)
Reported by: Jonathan Rose
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Merged revisions 407337 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks to Guillaume Martres for doing the necessary research to validate
the change.

(closes issue ASTERISK-17727)
Reported by: LN
Patches:
      use_certificate_chain.patch (license #5864) patch uploaded by st
      documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres
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Merged revisions 407272 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.

The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.

(closes issue ASTERISK-19773)
Reported by: Joel Vandal

(closes issue ASTERISK-22757)
Reported by: Gareth Blades
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407210 65c4cc65-6c06-0410-ace0-fbb531ad65f3

STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter.  This
protects against situations where those parameters are blank or missing by
logging an error and returning.

(closes issue ASTERISK-23220)
Reported by: James Sharp
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Merged revisions 407100 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407103 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The parsing for the destination of the macro/gosub uses the '^' character to
separate out context, extension, and priority. However, the logic for the
macro/gosub execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply when the
macro/gosub jump occurs in a priority/priority label. This patch changes
the logic so that the parsing still occurs, but the jump will occur even
for priorities/priority labels.

(issue ASTERISK-23164)

Review: https://reviewboard.asterisk.org/r/3154
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Merged revisions 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@407074 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_bind to a port reserved for another program by SELinux causes
errno == EACCES.  This caused random failures when binding rtp or
udptl sockets.  Treat EACCES as a non-fatal error, try next port.

(closes issue ASTERISK-23134)
Reported by: Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406934 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406918 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Closes issue ASTERISK-22662
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406861 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.

(closes issue ASTERISK-22980)
Reported by: Jeremy Lainé
Patches:
     freeradius-client.patch uploaded by sharky (license 6561)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406802 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not.  This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.

(issue AST-1276)
Review: https://reviewboard.asterisk.org/r/3156/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").

I can't figure out any good reason why this should be limited.  This patch just
removes the limit and uses the default nesting depth limit.

Closes issue ASTERISK-17837

Review: https://reviewboard.asterisk.org/r/3159/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406644 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ast_filestream object gets tacked on to a channel via
chan->timingdata.  It's a reference counted object, but the reference
count isn't used when putting it on a channel.  It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.

Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.

Bug reported by Leif Madsen.

Review: https://reviewboard.asterisk.org/r/3135/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CEL data structures need to be protected during a configuration reload
and shutdown.  Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.

* Protected the appset and linkedids ao2 containers using the reload_lock.
As a result appset, linkedids, and held objects don't need a lock.

* Added NULL checks before use of the appset and linkedids ao2 containers
in case the CEL module is already shutdown.

* Fixed overloading of the linkedids held objects reference count.  During
shutdown any held objects would be leaked.

* Fixed memory leak of linkedids held objects if the LINKEDID_END is not
being tracked.  The objects in the linkedids container were not removed if
the LINKEDID_END event is not used.

* Added access protection to the appset container during the CLI "cel show
status" command.

* Made CEL config reload not set defaults if the cel.conf file is invalid.

(closes issue AST-1253)
Reported by: Guenther Kelleter

Review: https://reviewboard.asterisk.org/r/3127/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made register atexit shutdown routine only once in __init_manager().

* Fixed some initial load failure conditions in __init_manager().

* Made reset options to defaults on reload when the reload will actually
happen.

* Removed unnecessary container traversals of the white/black filters
during manager_free_user().

* ast_free() does not need a NULL check before calling.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3141/
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Merged revisions 406360 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406361 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.

* Added ao2_global_obj protection to the sessions global container.

* Fixed the order of unreferencing a session object in session_destroy().

* Removed unnecessary container traversals of the white/black filters
during session_destructor().

(closes issue AST-1242)
Reported by: Guenther Kelleter

Review: https://reviewboard.asterisk.org/r/3144/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing.  The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.

(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
     ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406245 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Confbridge AMI and CLI commands for mute, unmute, and setting the
single video source can accept channel prefixes in lieu of a full
channel name, but documentation states only that it is required and is
a channel name. This corrects the documentation.

(closes issue PQ-1397)
Reported by: Steve Pitts


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406217 65c4cc65-6c06-0410-ace0-fbb531ad65f3