This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the barge audiohook was attached prior to the spyee and its peer
actually being bridged, the audiohook would not be applied and the
connected peer would not be able to hear audio from the spy when the
spy is in barge mode.

(closes issue ASTERISK-23381)
Reported by: Robert Moss
Review: https://reviewboard.asterisk.org/r/3505/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413486 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed ref leak in the CLI "confbridge kick" command when the channel to be
kicked was not in the conference.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patches:
	res_config_odbc-take2.patch by John Hardin (License #6512)
........

Merged revisions 413396 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patches:
	res_config_odbc.patch by John Hardin (License #6512)
........

Merged revisions 413304 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413305 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patches:
    odbc-crash.patch by John Hardin (License #6512)
........

Merged revisions 413241 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413251 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The realtime API specifies that the store callback is supposed to return the number
of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which
during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were
inserted causes problems to the function's callers.

To give an idea of how strange code can be, this is the necessary code change to fix
a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that
the registrar would attempt to insert contacts into the database. Because of the 0
return from res_config_pgsql, the registrar would think that the contact was not successfully
inserted, even though it actually was. As such, even though the contact was query-able
and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered,
meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE.

The necessary fix applies to all versions of Asterisk, so even though the bug reported
only applies to Asterisk 12+, the code correction is being inserted into 1.8+.

Closes issue ASTERISK-23707
Reported by Mark Michelson
........

Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed early exit in sip_msg_send() not destroying the message iterator.

* Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
tolerant of a NULL iter parameter in case ast_msg_var_iterator_init()
fails.

* Made ast_msg_var_iterator_destroy() clean up any current message data
ref.

* Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(),
ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and
ast_msg_var_iterator_destroy() use iter instead of i.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This resolves a race condition where data could be written to a NULL
FILE pointer causing a crash as a websocket connection was in the
process of shutting down by adding locking to websocket session writes
and by deferring session teardown until session destruction.

(closes issue ASTERISK-23605)
Review: https://reviewboard.asterisk.org/r/3481/
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413123 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On congested networks, it is possible for the DTLS handshake messages to get
lost. This patch adds a timer to res_rtp_asterisk that will periodically
check to see if the handshake has succeeded. If not, it will retransmit the
DTLS handshake.

Review: https://reviewboard.asterisk.org/r/3337

ASTERISK-23649 #close
Reported by: Nitesh Bansal
patches:
  dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Backport -r411687 and fix the fix because content_length is the length of
out plus the length of the file controlled by fd.

When a response has an out content length of 0, fwrite would be called to
write a buffer with no data in it.  This resulted in the following classic
error message:

  [Apr  3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success

This patch makes it so that we only attempt to write the content of out if
the out string is non-zero.
........

Merged revisions 412922 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412923 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-1301)

(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
........

Merged revisions 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 412764 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the TCP_NODELAY option to accepted connections on the HTTP
server built into Asterisk. This option disables the Nagle algorithm
which controls queueing of outbound data and in some cases can cause
delays on receipt of response by the client due to how the Nagle
algorithm interacts with TCP delayed ACK. This option is already set on
all non-HTTP AMI connections and this change would cover standard HTTP
requests, manager HTTP connections, and ARI HTTP requests and
websockets in Asterisk 12+ along with any future use of the HTTP
server.

Review: https://reviewboard.asterisk.org/r/3466/
........

Merged revisions 412745 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.

(closes issue AST-1301)

(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski

Review: https://reviewboard.asterisk.org/r/3447/
........

Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix wrong dialtone. The "modulation" should not be referenced for tone+tone as it refers to the on-off characteristic - this often resulted in a single tone rather than the multitone as in the UK.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two issues in app_sms:
(1) Firstly, the 'flags' field on the stack in sms_exec() is uninitialised,
    causing it to use the wrong protocol in some cases. This patch correctly
    initializes the flags fields.

(2) Secondly, when disconnect supervision is not working or
    inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was failing to
    terminate the call after it sent the REL(ease) message and the peer stopped
    talking to it. This patch fixes the code to handle the 'bad stop bit'
    message more gracefully in that case, and hang up the call.

Review: https://reviewboard.asterisk.org/r/1392/

ASTERISK-18331 #close
Reported by: David Woodhouse
patches:
  asterisk-fix-sms.patch uploaded by David Woodhouse (License 5754)
........

Merged revisions 412655 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In sounds/Makefile

 1 Adds and moves some lines necessary for the en_GB core set. I'm just following how the other sets are defined here.
 2 removes the ES extra sounds related lines as we don't have ES extra sound sets. 

In sounds/sounds.xml

 3 Adds member definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in extra sound sets

ASTERISK-23550 #close
Review: https://reviewboard.asterisk.org/r/3464/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
channels/chan_oss: Fix compilation problem on SmartOS/Illumos/SunOS

THis patch fixes an issue in chan_oss when building on certain platforms. It
ensures that soundcard.h is found.

Review: https://reviewboard.asterisk.org/r/3426

Note that this patch is a part of the patch on ASTERISK-23576; the Makefile
portion only applies to Asterisk 11+.

(issue ASTERISK-23576)
Reported by: Sebastian Wiedenroth
patches:
  fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412482 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two issues when building on SmartOS:

- channels/chan_oss.c: it makes sure soundcard.h is found
- main/Makefile: only use "-Wl,--version-script" when GNU LD is used as the Sun
  Linker doesn't support that. Similar checks are already used elswhere in the
  Makefile

Review: https://reviewboard.asterisk.org/r/3426

ASTERISK-23576 #close
Reported by: Sebastian Wiedenroth
patches:
  fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end
of the function.  The unref needs to happen after the last use of the
pointer.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412348 65c4cc65-6c06-0410-ace0-fbb531ad65f3

```yaml:frontmatter
  r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines

  chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

  Prior to this patch, the P-Asserted-Identity header would include anonymous
  caller id information which seems to go against the point of the
  P-Asserted-Identity header. Now the real caller ID information will be
  included in this header. Also, no privacy header would be included.
  This patch adds 'Privacy: id' to outgoing SIP messages that include the
  P-Asserted-Identity header.

  (closes issue AST-1301)
```
........

Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412329 65c4cc65-6c06-0410-ace0-fbb531ad65f3

autoservice acquires a local reference to the logger callid of each channel
in a loop.  This local reference was not released, causing the callid of
every channel in autoservice to leak.  This change moves the callid unref
inside the loop.

ASTERISK-23616 #close
Reported by: ibercom


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412305 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23620 #close
Reported by: Bradley Watkins
Patches:
      ASTERISK-23620_unlock_oldlist.patch (license #5021) patch uploaded by Bradley Watkins
........

Merged revisions 412225 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412226 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
    REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
    Every run will now blow away the previous run (as large ref files
    sometimes caused issues). We now also no longer open/close the file
    on each write, instead relying on fflush to make sure data gets written
    to the file (in case the ao2 call being performed is about to cause a
    crash)
(3) It goes with a comma delineated format for the ref debug file. This
    makes parsing much easier. This also now includes the thread ID of the
    thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
    contrib/scripts folder.

Review: https://reviewboard.asterisk.org/r/3377/
........

Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412115 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add notice messages during execution that the -I command line option and
the astersik.conf internal_timing option are no longer needed.  The
internal timing functionality is now always enabled if there is a timing
module loaded.

NOTE: Since the command line options and the asterisk.conf config file are
processed before the logging system is initialized, the messages are
output to stderr.

Change requested as a result of asterisk-dev list comments about the
commit for ASTERISK-22846 that removed the -I and internal_timing options.

Review: https://reviewboard.asterisk.org/r/3423/
........

Merged revisions 411964 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a long standing bug in CB_ADD_LEN() behaving like CB_ADD().

ASTERISK-23546 #close
Reported by: Walter Doekes
........

Merged revisions 411960 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411961 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixed copy pasta error.

ASTERISK-23545 #close
Reported by: John Knott


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411944 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 411807 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross.  Local channel optimization requires frames
flowing to trigger when optimization can happen.  When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing.  If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received.  With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.

* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed.  Asterisk now always uses internal
timing when needed if any timing module is loaded.  The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used.  The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.

* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().

ASTERISK-22846 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3414/
........

Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411716 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23391 caused a regression where the symbol 'defaultlanguage'
was used by app_voicemail but not exported by main/asterisk.  This
change renames the variable to ast_defaultlanguage.  The variable was
already renamed in Asterisk 12+.

(closes issue ASTERISK-23559)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3408/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.

This patch causes realtime queue members to remain in queues during the reload process. Previously these
members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY".

ASTERISK-23547 #close
ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409)

Review: https://reviewboard.asterisk.org/r/3404/
........

Merged revisions 411584 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created

(closes issue ASTERISK-23460)

Reported by: Dmitry Melekhov
Patches:
	ASTERISK-23460-2.patch

Tested by: Dmitry Melekhov



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:

a) Client request comes from node.js user agent
   "Shred" via use of swagger-client library.

b) Asterisk and Client are *not* on the same
   host or TCP/IP stack

In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function.  The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission.  See review for more details.


ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
........

Merged revisions 411462 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 411457 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411458 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.

This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It should be disabled
for database backends (such as PostgreSQL) that require NULL instead of an
empty string for Integer columns.

Review: https://reviewboard.asterisk.org/r/3375

(issue ASTERISK-23459)
Reported by: zvision
patches:
  res_config_odbc.diff uploaded by zvision (License 5755)
........

Merged revisions 411399 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.

ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)

Review: https://reviewboard.asterisk.org/r/3396/
........

Merged revisions 411372 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411373 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
........

Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Backport ast_register_cleanup from Asterisk 12.
* Use ast_register_cleanup for format_attr_shutdown.

ast_register_cleanup was originally commited in r390122 by dlee.

(closes issue ASTERISK-23103)
Reported by: JoshE


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411310 65c4cc65-6c06-0410-ace0-fbb531ad65f3