app_voicemail: Add blank argument to externnotify if no context argument

At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.

(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
	modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
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Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix parallel make problems.

Occasionally, make -j would fail due to missing includes, or other
unusual errors.

This was due to the 'cleantest' target, which was designed to force a
make clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile did
not depend upon cleantest, hence would run in parallel to it. By
adding the dependency, make -j runs happily now.

Review: https://reviewboard.asterisk.org/r/2418/
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Merged revisions 384410 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Remove silly use of strncmp.
Fixed spurious rebuilds of func_version.

func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that was
added in r384410 to fix parallel make bugs.

Now build.h will only be created if it does not exist, which was the
original behavior of the Makefile.
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Merged revisions 384544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Update documentation for CHANNEL function

Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
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Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
chan_dahdi: Add inband_on_proceeding compatibility option.

The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.

Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio.  However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.

ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.

(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Blocked revisions 384779

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Backport Appropiate NAT Setting Cleanup

In ASTERISK-20904, the focus was around the changes to NAT that took place in
Asterisk 11.  Since the report stated that 1.8 was fine, we didn't take a look
at 1.8 at the time.

While working on ASTERISK-21225, I could see that 1.8 would benefit from having
some of those changes applied to it.

This patch does the following:

* The important part of this patch is that it sets the peer's flags earlier in
  build_peer so that the code properly uses the peer's flags based on the peer's
  configuration.
* constify req parameter in check_via()
* update realtime schemas under the contrib directory to handle properly the NAT
  settings available in 1.8 as well as to handle the changes made in 11 to make
  upgrading easier when installing newer versions of Asterisk

(closes issue ASTERISK-21243)
Reported by: Michael L. Young
Patches:
    asterisk-20904-changes_for_1.8.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2422/
Fix For Not Overriding The Default Settings In chan_sip

The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting.  Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.

This patch works similar to what occurs in build_peer().  We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.

In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.

This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.

(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
  asterisk-21225-handle-options-default-prob_v4.diff
						Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2385/
Blocked revisions 385008

........
Fix For Not Overriding The Default Settings In chan_sip

The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting.  Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.

This patch works similar to what occurs in build_peer().  We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.

In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.

(closes issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
  asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
						Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2386/
Modified the list of keys for the driver backends for sake of sample clarity

Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
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Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE

When a BYE request is processed in chan_sip, the current SIP dialog is detached
from its associated Asterisk channel structure. The tech_pvt pointer in the
channel object is set to NULL, and the dialog persists for an RFC mandated
period of time to handle re-transmits.

While this process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no way of knowing
that the channel they've just obtained (which is still valid) and that they are
attempting to lock is about to have its tech_pvt pointer removed. By the time
they obtain the channel lock and call the channel technology callback, the
tech_pvt is NULL.

This patch adds a few checks to some channel callbacks that make sure the
tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
callbacks, which would crash if AMI initiated a DTMF on the channel at the
same time as a BYE was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this function), as well
as sip_indicate (as lots of things can queue an indication onto a channel).

Review: https://reviewboard.asterisk.org/r/2434/

(closes issue ASTERISK-20225)
Reported by: Jeff Hoppe
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Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Use LDAP memory management functions instead of Asterisk's

When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.

This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.

Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.

(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
  issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
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Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix 'pri intense debug span' alias.
Add dependency on libuuid, for res_rtp_asterisk

pjproject is what actually requires libuuid.

(closes issue ASTERISK-21125)
reported by Private Name

(Ed. note: Really?  Private Name?  I am rolling my eyes so hard right now.)
IAX2, prevent network thread starting before all helper threads are ready

On startup, it's possible for a frame to arrive before the processing threads were ready.

In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive
before we are at ast_cond_wait, the signal will be ignored.
The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued.  

Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled
after each thread is started.
 
(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2427/
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Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8
IAX2 defer_full_frames fail to get sent

Ensure iax2_process_thread is signalled when a deferred frame is queued to it.

(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2426/
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Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX

When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
turned on and off when using the auto_force_rport and auto_comedia nat settings
go back to the default setting off.  These flags are turned on when needed or
off when not needed at the time that a peer registers, re-registers or initiates
a call.  This would apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would only affect the
force_rport flag.

Everything is good except for the following:  The nat setting is set to
auto_force_rport and auto_comedia.  We reload Asterisk and the peer's
registration has not expired.  We load in the settings for the peer which turns
force_rport and comedia back to off.  Since the peer has not re-registered or
placed a call yet, those flags remain off.  We then initiate a call to the peer
from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved,
we end up with one-way audio since we never checked to see if the peer is behind
NAT or not.

This patch does the following:

* Moves the checking of whether a peer is behind NAT into its own function

* Create a function to set the peer's NAT flags if they are using the auto_* NAT
  settings

* Adds calls in sip_request_call() to these new functions in order to setup the
  dialog according to the peer's settings

(closes issue ASTERISK-21374)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2421/
Fix app_voicemail Segfault And A Few Memory Leaks

The original report was that app_voicemail would crash.  This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status.  After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.

During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.

This patch does the following:

* Creates a helper function to check if the configuration is valid

* Adds calls to the new helper function where appropiate

* Fixes memory leaks where the code returned without running
  ast_config_destroy() on the configuration that was loaded

(closes issue ASTERISK-21302)
Reported by: Jaco Kroon
Tested by: Jaco Kroon, Michael L. Young
Patches:
    asterisk-11.3.0-app_voicemail-ast_config-fixes.patch
                                                       Jaco Kroon (license 5671)
    asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2443/
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Merged revisions 385551 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Allow codec_resample to be unloaded

Ensure that trans_size is correct to prevent uninitialized entries from
preventing reload.

(closes issue ASTERISK-21401)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
    codec_resample-unload.patch uploaded by Corey Farrell
Fix Manager Segfault When app_queue Is Unloaded

When app_queue is unloaded, some manager commands are not being unregistered
which result in a segfault.  This patch corrects this.

(closes issue ASTERISK-21397)
Reported by: Peter Katzmann, Corey Farrell
Tested by: Corey Farrell
Patches:
    asterisk-21397-missing-unreg-manager-cmd_1.8.diff
                                                 Michael L. Young (license 5026)
    asterisk-21397-missing-unreg-manager-cmd_11.diff
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2444/
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Merged revisions 385593 from http://svn.asterisk.org/svn/asterisk/branches/1.8