Previously make -j <N> would cause a race between doing cleanup of
certain files (defaults.h, menuselect, ...) and creating them anew.
Add a new target that depends on cleanup only and has a submake doing
the rest as command string. This way the cleanup goes first.

(closes issue ASTERISK-18751)
Tested by: Jeremy Kister
Reviewed by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/1660


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In order to better handle RTP sources with strictrtp enabled (which is now default in 10)
using the learning mode to figure out new sources when they change is handled by checking
for a number of consecutive (by sequence number) packets received to an rtp struct
based on a new configurable value called 'probation'. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source
has been determined.

Review: https://reviewboard.asterisk.org/r/1663/
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If a Contact or a Record-Route header had a quoted string with an
item in angle brackets, then we would mis-parse it. For instance,
"Bob <1234>" <1234@example.org>
would be misparsed as having the URI "1234"
The fix for this is to use parsing functions from reqresp_parser.h
since they are heavily tested and are awesome.

(issue ASTERISK-18990)
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When an inital INVITE occurs that contains image media, a channel
is not yet associated with the SIP dialog.  The file descriptor
associated with the udptl session needs to be set in
initialize_udptl or in sip_new to account for this scenario.
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https://origsvn.digium.com/svn/asterisk/branches/10

................
  r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines
  
  Merged revisions 351182 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines
    
    Add some missing locking in chan_sip.
    
    This patch adds some missing locking to the function 
    send_provisional_keepalive_full().  This function is called from the scheduler,
    which is processed in the SIP monitor thread.  The associated channel (or pbx)
    thread will also be using the same sip_pvt and ast_channel so locking must be
    used.  The sip_pvt_lock_full() function is used to ensure proper locking order
    in a safe manner.
    
    In passing, document a suspected reference counting error in this function.
    The "fix" is left commented out because when the "fix" is present, crashes
    occur.  My theory is that fixing it is exposing a reference counting error
    elsewhere, but I don't know where.  (Or my analysis of this being a problem
    could have been completely wrong in the first place).  Leave the comment in
    the code for so that someone may investigate it again in the future.
    
    Also add a bit of doxygen to transmit_provisional_response().
    
    (closes issue ASTERISK-18979)
    
    Review: https://reviewboard.asterisk.org/r/1648
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When handling a non-2xx final response on an INVITE transaction, we have to
keep the transaction around after we send an ACK in case we receive a
retransmission of the response so we can re-transmit the ACK, but also tear
down the ast_channel as soon as we transmit the ACK. Before this patch, we
could fail at both of these things. Calling sip_alreadygone/needdestroy
prevented us from keeping the transaction up and retransmitting the ACK, and
queueing CONGESTION was not sufficient to cause the channel to be torn down
when originating calls via the CLI, for example.

This patch queues a hangup with CONGESTION instead of just queueing CONGESTION
for these responses and removes the sip_alreadygone and sip_needdestroy calls
from handle_response_invite on non-2xx responses. It relies on the hangup
calling sip_scheddestroy.

For more information, see section 17.1.1.1 of RFC 3261.

(closes issue ASTERISK-17717)
Review: https://reviewboard.asterisk.org/r/1672/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.

(closes issue ASTERISK-18996)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches: session_timer_fix.diff by Terry Wilson (License #5357)
  based on session_timer.patch by Thomas Arimont (License #5525)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, the udptl struct was allocated and initialized when a
dialog was associated with a peer that supported T.38, when a new SIP
channel was allocated, or what an INVITE request was received.  This resulted
in any dialog associated with a peer that supported T.38 having udptl support
assigned to it, including the UDP ports needed for communication.  This
occurred even in non-INVITE dialogs that would never send image media.

This patch creates and initializes the udptl structure only when the SDP
for a dialog specifies that image media is supported, or when Asterisk
indicates through the appropriate control frame that a dialog is to support
T.38.

(closes issue ASTERISK-16698)
Reported by: under
Tested by: Stefan Schmidt
Patches: udptl_20120113.diff uploaded by mjordan (License #6283)

(closes issue ASTERISK-16794)
Reported by: Elazar Broad
Tested by: Stefan Schmidt

review: https://reviewboard.asterisk.org/r/1668/
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You can pass wildcards (%) to the database CLI commands, so this will sort the
returned list of matches.
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This more closely mimics the behavior of 'database show' before the conversion
to sqlite3.
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Add locking around the really-really-quit part of the core stop/restart
part. Previously more than one thread could be called to do cleanup,
causing atexit handlers to be run multiple times, in turn causing
segfaults.

(issue ASTERISK-18883)
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1662/
Review: https://reviewboard.asterisk.org/r/1658/
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Note that I'm not confirming legitimacy of having that file in tree at
all. Is anyone using aelparse/conf2ael?

(issue ASTERISK-15350)
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Recent versions of autoconf (2.68 on my system) won't properly process the configure
script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script
were, but many were not. This patch corrects the unquoted calls.
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  r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
  
  Ensure that two prerequisites are properly installed on Debian-style distributions.
  
  * Don't specify a specific version of libgmime; newer versions are available
    now and acceptable.
  
  * Install libsrtp so that res_srtp can be built.
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  r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
  
  Correct some 'set-but-not-used' variable warnings.
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configure and autoconfig.h.in were not regenerated when the fix was committed.
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OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not
'struct ucred', which causes compilation of main/asterisk.c to fail in
read_credentials().  This allows configure to check for sockpeercred and
asterisk to deal with it properly.

(closes issue ASTERISK-18929)
Reported-by: Barry Miller
Patch-by: Barry Miller
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None of the parameters to ast_bridge_call() can be NULL for the bridge to
work so no need to check for it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350644 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Multiple revisions 350555,350571

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  r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines
  
  Add missing CEL logging fields to various CEL backends.
  
  * Add missing eventextra to cel_psql.c and cel_odbc.c.
  
  * Add missing PeerAccount and EventExtra to cel_manager.c.
  
  * Add missing userdeftype support for cel_custom.conf.sample and
  cel_sqlite3_custom.conf.sample.
  
  (closes issue ASTERISK-17190)
  Reported by: Bryant Zimmerman
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  r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines
  
  Use compatible names for event extra data for various CEL backends.
  
  * Change eventextra to extra in cel_psql.c and cel_odbc.c.
  
  * Change EventExtra to Extra in cel_manager.c.
  
  (issue ASTERISK-17190)
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Previously, realtime queues could be loaded without defining the queue member
table.  This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage.  Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned.  Previously, an empty ast_config object was
expected.

(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches: 
  rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
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This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
   bridge_pvt an ao2 ref counted object

Patch by David Vossel (mjordan was merely the commit monkey)

(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)

(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1654/
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(closes issue ASTERISK-17940)
Reporter: Nic Colledge
Patches:
	features_18.patch uploaded by Nic Colledge (license 6245)
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(closes issue ASTERISK-19180)
Reported by: Corey Farrell
Patches:
      asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell
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Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.

* Added the 'I' option just like the app_dial and app_queue 'I' option.

* Made 'N' option ignored if the call is already answered.

(closes issue ASTERISK-18969)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1656/
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Initially, support was left in for the old style of searching, even
though it wasn't actually used. In the case of name_len != 0, the
OBJ_KEY flag isn't passed because we aren't matching on a full key
and therefor can't use the hash function to optimize. The code left
in to support the old way of searching unfortunately treated a prefix
search like this as though an ast_channel struct was passed as an arg
and caused a crash.

This patch also adds needed parentheses around some matching conditions.

(closes issue ASTERISK-19182)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The time passed by the LOCK function to an internal function was relative
time when the function expected absolute time.

* Don't use C++ keywords in get_lock().

(closes issue ASTERISK-16868)
Reported by: Andrey Solovyev
Patches:
      20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The iax2_process_thread() can exit without anyone waiting to join the
thread.  If noone is waiting to join the thread then a large memory leak
occurs.

* Made iax2_process_thread() deatach itself if nobody is waiting to join
the thread.

(closes issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
      asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified)

(closes issue ASTERISK-17825)
Reported by: wangjin
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If a db_sync was scheduled just before shutdown, the atexit code calling
db_sync would have no effect, causing the astdb commit thread to stay
alive. This caused the SIP/realtime_sipregs test to fail. (The fallback
kill would run the atexit code again and that would wreak havoc.) This
fixes that the atexit kill condition is picked up properly.

(closes issue ASTERISK-18883)
Reviewed by: Terry Wilson

Review: https://reviewboard.asterisk.org/r/1659
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........
  r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines
  
  Update contrib script live_ast to invoke Asterisk with valgrind and suppression file.
  
  * Added valgrind_compare script to compare two valgrind log files for
  differences.
  
  (issue ASTERISK-17339)
  Reported by: Tzafrir Cohen
  Patches:
        valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
        live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen
        live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger
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  r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines
  
  live_ast: valgrind: run asterisk under valgrind
  
  Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
  valgrind. The extra command-line parameters are passed to Asterisk as
  usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
  in live.conf .
  
  Review: https://reviewboard.asterisk.org/r/1109/
  
  Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10
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The Asterisk -x command line parameter is documented inconsistently.

* Made the -x documentation and behavior consistent.

* Since this is also a new year, updated the copyright notices while here.

(closes issue ASTERISK-19094)
Reported by: Eugene
Patches:
      issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified)
Tested by: Eugene
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350077 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed.  Moving the settings reset later in the reload
process fixes this.

(closes issue AST-744)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350025 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When someone does Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows
"Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From
header will still display the callerid info, even though we supply an rpid
header with the anonymous info. It seems like we shouldn't leak that info in
any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems
to indicate that one shouldn't send identifying info in the From in this case.

This patch anonymizes the From header as well even when sendrpid=yes/pai.

(closes issue ASTERISK-16538)

Review: https://reviewboard.asterisk.org/r/1649/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A bug in the pbx_lua goto detection was causing the dialplan to hangup
unexpectedly after confbridge exited if it had called lua dialplan code during
execution.

Patch-by: Timo Teras
Acked-by: Matt Nicholson
(closes issue ASTERISK-18976)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349929 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19055)
Reported by: Matt Jordan
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