Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.

Review: https://reviewboard.asterisk.org/r/1794/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I was getting confused during some testing why Asterisk was saying that
a subscription was being added when it was clearly being removed. This
fixes that confusion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Global ao2 objects must always exist after initialization because there is
no access control to obtain another reference to the global object.

It is expected that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object.

With these new API calls, the global object could be replaced, removed, or
referenced without the risk of someone using a stale global object
pointer.

Review: https://reviewboard.asterisk.org/r/1824/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360626 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-17842)
Reported by: Bryon Clark
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Update CHANGES for r360471
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Rather then flood the CLI with verbose messages, we've changed the level to
debug. This will help keep the CLI clean.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes ASTERISK-17842)
Reported by: Bryon Clark
Patches:
	20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003)
	configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is just a minor code cleanup change.  These uses of ao2_callback() would
never return anything since the callbacks always returned 0.  However, be more
explicit that no returned results are wanted by specifying OBJ_NODATA.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360369 65c4cc65-6c06-0410-ace0-fbb531ad65f3

dial_list is a dynamically allocated array that is allocated at the beginning
of Page() based on how many devices will be dialed.  This was never being freed.
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  r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines
  
  expression parser: Fix (theoretical) memory leak.
  
  Fix a memory leak that is very unlikely to actually happen.  If a malloc()
  succeeded, but the following strdup() failed, the memory from the original
  malloc() would be leaked.
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  r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
  
  Rebuild parsers.
  
  This is needed to include the last fix to main/ast_expr2.y.  The changes look
  much bigger as this regeneration of the code was done with newer versions of
  flex and bison.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".

* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking".  This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.

* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c.  This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360264 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Similar to dial and queue F option.

(Closes issue ASTERISK-19282)
Reported by: To
Patches:
	bridge_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1825/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Everything still compiled after making these changes, so I assume these
whitespace-only changes didn't break anything (and shouldn't have).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19367)
Reported by: Andrew Latham
Patches:
	debian_install_prereq.diff uploaded by Andrew Latham (license 5985)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360140 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also detect gmime version 2.6 (Michael Biebl)

Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360137 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk detects a hangup and cannot send a BYE due to a pending
INVITE, it sets the pendingbye flag and waits for the final response to that
INVITE.  When the response is received, it transmits the BYE.  If, however,
that INVITE request is a pending re-INVITE, it needs to first send a CANCEL
request to terminate the pending re-INVITE.  In that circumstance, Asterisk
was, in some scenarios, clearing the pendingbye flag after processing the
CANCEL request and not checking for a pending BYE when receiving the final
487 response to the INVITE.

This patch ensures that if the pendingbye flag is set, it is honored
regardless of the nature of the INVITE request currently in flight.

(closes issue ASTERISK-19365)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
  bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283)

Review: https://reviewboard.asterisk.org/r/1807
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Echo()'s description states that it echoes audio, video, and DTMF except for #
while it actually echoes any frame that it receives other than DTMF #.  This
was causing frame storms in the test suite in some circumstances where Echo()
was attached to both ends of a pair of local channels and control frames
were being periodically generated.  Echo()'s behavior and description have
been modifed so that it only echoes media and non-# DTMF frames.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The PeerStatus event for IAX2 channels currently includes a header named Post
which should have been Port.  Post was removed and the AMI version has been
updated to 1.3.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix AMI module reload deadlock regression from ASTERISK-18479 when it
tried to fix the race between calling an AMI action callback and
unregistering that action.  Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.

Locking the ao2 object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called.  Unfortunately, this
causes the deadlock situation.  The patch stops locking the ao2 object to
allow multiple threads to invoke the callback re-entrantly.  There is no
way to guarantee a module unload will not crash because of an active
callback.  The code attempts to minimize the chance with the registered
flag and the maximum 5 second delay before ast_manager_unregister()
returns.

The trunk version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory while an
action callback is active.

* Don't hold the lock while calling the AMI action callback.

(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer

Review: https://reviewboard.asterisk.org/r/1818/
Review: https://reviewboard.asterisk.org/r/1820/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359981 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added missing error exits with cause in manager_mutestream().

* Cleaned up manager_mutestream() and func_mute_write().

* Some whitespace and comment cleanup.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359942 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch addresses a bug with chanspy on local channels which roughly 50% of the time
would create a situation where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never be able to hang up.

(closes issue ASTERISK-19493)
Reported by: lvl
Review: https://reviewboard.asterisk.org/r/1819/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Remove unnnecessary const from const char * const var declaration in the
ast_app_run_macro() and ast_app_run_sub() prototypes.  The second const is
unnecessary.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The code may be just fine, but it had not received a "ship it!" on
review board yet.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

from Review: https://reviewboard.asterisk.org/r/1699/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359811 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There exists a remotely exploitable stack buffer overflow in HTTP digest
authentication handling in Asterisk.  The particular method in question
is only utilized by HTTP AMI.  When parsing the digest information, the
length of the string is not checked when it is copied into temporary buffers
allocated on the stack.

This patch fixes this behavior by parsing out pre-defined key/value pairs
and avoiding unnecessary copies to the stack.

(closes issue ASTERISK-19542)
Reported by: Russell Bryant
Tested by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added 'b' and 'B' options to Dial.  These options will allow you to run
  last-minute dialplan on the caller and callee channels while the Dial
  application is executing, but before the call is started.  For example you
  can use the 'b' option to run dialplan on the callee channel to get the name
  of the newly created channel right away.

Review: https://reviewboard.asterisk.org/r/1229/

(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Milliwatt is vulnerable to a remotely exploitable stack overrun when using
the 'o' option.  This occurs due to the milliwatt_generate function not
accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer.

This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET
when determining the maximum number of samples allowed.  Note that at no
point is remote code execution possible.  The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.

(closes issue ASTERISK-19541)
Reported by: Russell Bryant
Tested by: Matt Jordan
Patches:
  milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283)
  Note that this patch was written by Russell, even though Matt uploaded it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Missed on the previous commit


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359651 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's caller-id is
implicitly imported into the incoming channel's connected line data.  If
you are using the interception macros, you would expect that they get run
for every change to a channel's connected line information outside of
normal dialplan execution.

Review: https://reviewboard.asterisk.org/r/1817/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359644 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1696/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359607 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Initialize a struct sockaddr_in in try_transfer() so that the code isn't
(potentially) trying to read from it while uninitialized.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Avoid potential use of idroster in gtalk_alloc() before it has been
initialized.
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Ensure that status is set before it is used by resetting it during each loop
iteration.  This could have resulted in incorrect results from this app.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359495 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Scan results indicated that this array could be used uninitialized.  At a quick
look, it looks correct.  In any case, initializing it is a Good Thing (tm).
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