In order to use nested functions on some versions of GCC (e.g. GCC on OS X),
the -fnested-functions flag must be passed to the compiler. This patch adds
detection logic to ./configure to add the flag if necessary. It also adds
a comment to utils.h as to why the nested function needs a prototype.

(closes issue ASTERISK-20399)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2102/
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For SS7, the companding law for a call was chosen inconsistently depending
upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1).
For incoming calls, the companding law was determined by ss7type.  For
outgoing calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts.  An
A-law/u-law conflict sounds like bad static on the line.

SS7 ITU  signaling with E1 line: ok
SS7 ITU  signaling with T1 line: noise
SS7 ANSI signaling with E1 line: noise
SS7 ANSI signaling with T1 line: ok

* Fix the companding law used to be determined by the SS7 signaling type
only.
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This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
   portions of the SSL library.  Asterisk calls SSL_library_init and
   SSL_load_error_strings during SSL initialization; collectively this
   obviates the need for calling any of the following during initialization
   or client connection handling:
   * ERR_load_crypto_strings (handled by SSL_load_error_strings)
   * OpenSSL_add_all_algorithms (synonym for SSL_library_init)
   * SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
   the SSL library for TLS clients.  This included not freeing the SSL_CTX
   object in the SIP channel driver, as well as not clearing the error
   stack when the TLS client exited.

Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.

(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
  (bugAST-889.patch) by Thomas Arimont (license 5525)

Review: https://reviewboard.asterisk.org/r/2105
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ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!

This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.

(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
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The new API allows for sorted containers, insertion options, duplicate
handling options, and traversal order options.

* Adds the ability for containers to be sorted when they are created.

* Adds container creation options to handle duplicates when they are
inserted.

* Adds container creation option to insert objects at the beginning or end
of the container traversal order.

* Adds OBJ_PARTIAL_KEY to allow searching with a partial key.  The partial
key works similarly to the OBJ_KEY flag.  (The real search speed
improvement with this flag will come when red-black trees are added.)

* Adds container traversal and iteration order options: Ascending and
Descending.

* Adds an AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats <name>" and
"astobj2 container check <name>".  The channels container is normally
registered since it is one of the most important containers in the system.

* Adds ao2_iterator_restart() to allow iteration to be restarted from the
beginning.

* Changes the generic container object to have a v_method table pointer to
support other types of containers.

* Changes the container nodes holding objects to be ref counted.

The ref counted nodes and v_method table pointer changes pave the way to
allow other types of containers.

* Includes a large astobj2 unit test enhancement that tests the new
features.

(closes issue ASTERISK-19969)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2078/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a bug with Jitsi and conference calling. Jitsi implements XEP-0298
which places some conference-info information in the session-initiate request
which chan_motif did not expect to occur.
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(closes issue ASTERISK-20361)
Reported by: Noah Engelberth
Review: https://reviewboard.asterisk.org/r/2108/
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With this option in use, it may be necessary to regulate your log files
externally.

(closes issue ASTERISK-20189)
Reported by: Jaco Kroon
Patches:
    asterisk-logger-norotate-trunk.patch uploaded by Jaco Kroon (license 5671)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "autodestruct with owner in place" message is typically
indicative of a channel reference leak. Printing out the name
of the channel in the message may be helpful when trying to
debug the issue.
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I didn't realize that libasteriskssl.c was still compiled, even when you
disable asteriskssl; it simple gets statically linked into asterisk.
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Changes chan_local channels to use an 8 digit hex identifier generated
atomically and sequentially in order to eliminate the chance of having
multiple channels with the same name during high call volume situations.

(issue ASTERISK-20318)
Reported by: Dan Cropp
Review: https://reviewboard.asterisk.org/r/2104/
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message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.

This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.

(closes issue AST-937)
Reported by Jason Parker
Patches:
	AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
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When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.

(issue ASTERISK-20335)
Reported by: aragon
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(closes issue ASTERISK-20406)
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(issue AST-969)
Reported by John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, The acknowledgement wasn't produced until after
executing the sip_poke_peer action actually responsible for
qualifying the peer. Now the response is given immediately once it is
known that a peer will be qualified and a SIPqualifypeerdone event
is issued when the process is finished. Thanks to OEJ for identifying
the problem and helping to come up with a solution.

(issue AST-969)
Reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/2098/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.

(closes issue ASTERISK-20338)
Reported-by: John Covert
Patch-by: John Covert
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chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of
using chan_motif and res_xmpp. They are a feature-equivalent
replacement and are written to be more easily maintainable.

(closes issue ASTERISK-20298)
Review: https://reviewboard.asterisk.org/r/2082/
Reported-by: Leif Madsen
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Removes "res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer
will break strict-aliasing rules" warning from the build on 32-bit platforms.

The problem is that 'size' was referenced aliased to both (pj_size_t *) and
(pj_ssize_t *). Now just make a copy of size that is the right type so there
isn't any pointer aliasing happening.

It also adds comments and asserts regarding what looks like an inappropriate
use of pj_sock_sendto, but is actually totally fine.

(closes issue ASTERISK-20368)
Reported by: Shaun Ruffell
Tested by: Michael L. Young
Patches:
  0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch uploaded by Shaun Ruffell (license 5417)
    slightly modified by David M. Lee.
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(closes issue AST-991)
Reported by John Bigelow
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Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.

(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
    masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
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In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an
SDP offer and the ability to re-create an SRTP session when the crypto keys
changed.  In certain circumstances - most notably when a phone is put on
hold after having been bridged for a significant amount of time - the act
of re-creating the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session regardless
of whether or not the cryptographic keys changed.  Since this is technically
not necessary, this patch modifies the behavior to only re-create the SRTP
session if Asterisk detects that the remote key has changed.  This allows
models of phones that do not handle the SRTP session changing to continue
to work, while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys.

(issue ASTERISK-20194)
Reported by: Nicolo Mazzon
Tested by: Nicolo Mazzon

Review: https://reviewboard.asterisk.org/r/2099
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Without this flag, those files will compile with the system installed
OpenSSL headers (if they exist). This is a real bummer if a different
path was specified using --with-ssl=

(closes issue ASTERISK-20392)
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(closes issue ASTERISK-20349)
Reported by: Brent Eagles
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(closes issue AST-1001)
Reported by: Guenther Kelleter
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(closes issue ASTERISK-20380)
Reported by: Jeremy Pepper
Patches:
      fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper
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The AMI action VoicemailUsersList VoicemailUserEntry event headers
ServerEmail and MailCommand did not report the global values if they were
not overridden.  The VoicemailUserEntry event header ServerEmail was not
populated with the global value if the voicemail user did not override it.
The VoicemailUserEntry event header MailCommand was never populated with a
value.

* Removed unused struct ast_vm_user member mailcmd[].

(closes issue AST-973)
Reported by: John Bigelow
Tested by: rmudgett
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* pjproject bin and lib directories should pretty much ignore everything
* Ignore *.o in codecs/ilbc
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Fixes a build regression introduced in r369517 "Add support for ICE/STUN/TURN
in res_rtp_asterisk and chan_sip." [1].

[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=369517

When compiling asterisk in parallel like:
    $ make -j 10

It's possible to get errors like the following:

    .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing separator.  Stop.
    make[4]: *** [depend] Error 2
    make[3]: *** [dep] Error 1
    make[2]: *** [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2
    make[3]: warning: jobserver unavailable: using -j1.  Add `+' to parent make rule.

This is because the build system is trying to build each of the libraries in
pjproject in parallel. Now the build will build pjproject in a single job and
link the results into res_asterisk_rtp.

Parallel builds, on one test system, saves ~1.5 minutes from a default Asterisk
build:

Single job:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make >/dev/null 2>&1 )

    real    2m34.529s
    user    1m41.810s
    sys     0m15.970s

Parallel make:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 )

    real    1m2.353s
    user    2m39.120s
    sys     0m18.850s

(closes issue ASTERISK-20362)
Reported by: Shaun Ruffel
Patches:
    0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch uploaded by Shaun Ruffel (License #5417)
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The previous commit (r372554) was from a patch that was written before
r366880, which ensured that ast_str objects allocated in the sendmail
routine were free'd in off nominal paths.  This commit frees the
string objects in the off nominal path introduced in r372554.

(issue ASTERISK-17133)
Reported by: Tzafrir Cohen
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When MiniVM sends an e-mail and it has the volgain option set, it will spawn
sox in a separate process to handle the manipulation of the sound file.  In
doing so, it creates a temporary file.  There are two problems here:
  1) The file descriptor returned from mkstemp is leaked
  2) The finalfilename character pointer points to a buffer that loses scope
     once volgain processing is finished.

Note that in r316265, Russell fixed some gcc warnings by using the return
value of the mkstemp call.  A warning was placed in minivm that the file
descriptor was going to be leaked.  This patch reverts that change, as it
handles the leak and 'uses' the file descriptor returned from mkstemp.

(closes issue ASTERISK-17133)
Reported by: Tzafrir Cohen
patches:
  minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035)
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The Status: header in a QueueMemberStatus event (and other QueueMember* events)
is the numeric value of the device state corresponding to that Queue Member.
As those values are not exactly obvious, listing them in the documentation is
useful.

Matt Riddell reported this indirectly through the wiki page.

(closes issue ASTERISK-20243)
Reported by: Matt Riddell
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Using the AMI redirect action to take an ISDN call out of a parking lot
causes the MOH state to get confused.  The redirect action does not take
the call off of hold.  When the call is subsequently parked again, the
call no longer hears MOH.

* Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames
if it is already in a state where it is supposed to be sending MOH.  The
MOH may have been stopped by other means.  (Such as killing the generator.)

This simple fix is done rather than making the AMI redirect action post an
AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus
potentially breaking something with an unexpected AST_CONTROL_UNHOLD.

(closes issue ABE-2873)
Patches:
      jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by rmudgett
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When using tab-completion for the list of queues on "queue reset stats"
or "queue reload {all|members|parameters|rules}", the tab-completion
listing for further queues erroneously listed queues that had already
been added to the list. The tab-completion listing now only displays
queues that are not already in the list.

(closes issue AST-963)
Reported-by: John Bigelow
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r366547 introduced a change to the directmedia ACL for chan_sip which
modified the behavior significantly. Prior to the patch, this option would
bridge peers with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the bridged peer's
ACL instead. This change has been present since 1.8.14.0. That patched failed
to document the change in Upgrade.txt, so this patch adds mention of that
change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches)

(issue AST-876)
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Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.

(closes issue AST-958)
Reported-by: John Bigelow
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Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors,
PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors.  If the connection
is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in
the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send
messages to PBX3.  If it does, PBX2 will assume that PBX3 already received the
message and fail to forward the message on to PBX3 itself.  This patch fixes
this by only including peers in a DPDISCOVER message that are reachable by the
sending node.  This includes all peers with an empty address
(00:00:00:00:00:00) and that are have been reached by a qualify message.

This patch also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers.

The patch uploaded by Peter was modified slightly for this commit.

(closes issue ASTERISK-19309)
Reported by: Peter Racz
patches:
  dundi_routing.patch uploaded by Peter Racz (license 6290)

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When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters.  This can artificially limit some parallel dial scenarios.  This
patch allows for numbers of any length to be defined in the configuration
file.

Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue.  The patch originally expanded the buffer to 256
characters.  Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.

(closes issue ASTERISK-16879)
Reported by: Clod Patry
Tested by: mjordan
patches:
  followme_no_limit.diff uploaded by Clod Patry (license #5138)

Slightly modified for this commit.
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Fix compile error.
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