A long time ago, in a land far far away, we added "asterisk/ast_version.h",
which provides the ast_get_version() and ast_get_version_num() functions. These
were added so that modules that needed the version information for the Asterisk
instance they were loaded in could actually get it (as opposed the version that
they were compiled against). We changed everything in the tree to use the
new mechanism (although later main/test.c was added using the old method).
However, the old mechanism was never removed, and as a result, new code is
still trying to use it.

This commit removes asterisk/version.h and replaces it with a header that
will generate a compile-time error if you try to use it (the error message
tells you which header you should use instead). It also removes the Makefile
and build_tools bits that generated the file, and it updates main/test.c to
use the 'proper' method of getting the Asterisk version information.

This is an API change and thus is being committed for trunk only, but it's
a fairly minor one and definitely improves the situation for out-of-tree
modules.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352626 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Avoid unnecessary rebuilds of main/test.c.

main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352565 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.

* Pass up better From header contents for SIP to use.  Now is in the
"display-name" <URI> format expected by MessageSend.  (Note that this is a
behavior change that could concievably affect some people.)

* Block user from adding standard headers that are added automatically.
(To, From,...)

* Allow the user to override the Content-Type header contents sent by
MessageSend.

* Decrement Max-Forwards header if the user transferred it from an
incoming message.

* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.

* Documents what SIP expects in the MessageSend(from) parameter.

(closes issue ASTERISK-18992)
Reported by: Yuri

(closes issue ASTERISK-18917)
Reported by: Shaun Clark

Review: https://reviewboard.asterisk.org/r/1683/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1) Be sure and free at unload the epa_backend we allocate at startup
2) Do the same sip_registry cleanup at unload we do at reload

Review: https://reviewboard.asterisk.org/r/1689/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These files have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between 'modified'
and 'unmodified' states.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.

(closes issue ASTERISK-15537)
Reported by: yarique
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-16550)
reported by: Olle Johansson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!

(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
	confbridge.tar.001 uploaded by Cameron Twomey
    confbridge.tar.002 uploaded by Cameron Twomey
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed a potential memory leak when an existing datastore is manually
destroyed by inline code instead of calling ast_datastore_free().

(closes issue ASTERISK-17948)
Reported by: Archie Cobbs

Review: https://reviewboard.asterisk.org/r/1687/
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Move RTP timeout check to before bridged channel check so it is actually executed.

(issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury

(closes issue ASTERISK-14534)
Reported by: kriborgen
Patches:
	chan_sip.patch uploaded by kriborgen (license 6138)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents the 'h' extension from being run on the transferee
channel when it is transferred via a native transfer mechanism such
as SIP REFER.

(closes ASTERISK-19173)
Reported by: Ross Beer
Tested by: Kristjan Vrban
Patches:
	ASTERISK-19173 by Mark Michelson (license 5049)

Review: https://reviewboard.asterisk.org/r/1685
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While the FAXOPT function could be used to set the modem capabilities, the
input to that function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting the spandsp
layer.

(closes issue: ASTERISK-16409)
Reported by: Kristijan Vrban
Tested by: Matt Jordan, Matthew Nicholson
Patches:
  spandsp-modems-1.8.diff uploaded by mnicholson (license 5081)
  spandsp-modems-10.diff uploaded by mnicholson (license 5081)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is a feature patch which allows an 'announcement' option to be specified in
musiconhold.conf which should be set to the name of a sound. If a valid sound is
specified for this option, then it will be played on that music on hold class whenever
a channel bound to that class is put on hold as well as when Asterisk is able to detect
that a song has ended before starting the next song (excludes external players).

(closes ASTERISK-18977)
Reported by: Timo Teräs
Patches:
	asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.

(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The invalid value used when notifycid was enabled was benign.  As far as
the code was concerned -1 and 1 are equivalent.

(closes issue ASTERISK-19232)
Reported by: Eike Kuiper
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Note: Noone calls ast_app_dtget() with the timeout parameter of zero so
the bad code normally will never get executed.

* Fix unnecessary floating point division in func_timeout.c
timeout_write() when all other values are integers.

(closes issue ASTERISK-16817)
Reported by: Dmitry Andrianov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.

This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.

(issue ASTERISK-19192)

Review: https://reviewboard.asterisk.org/r/1681/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-16877)
Reported by: klaus3000
Patches:
      show-complete-routeset-patch.txt (license #5054) patch uploaded by klaus3000 (modified)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.

Event description:

Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id

`source` can be either RTPTimeout or SIPSessionTimer

(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled

(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.

Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't be alarmed. This only affected trunk, and it would have
required manager access to your system.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These changes are in a file that is not compiled by default, and so were
missed on earlier checks.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351862 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These functions are not noops and modify the array that is passed in. Thanks
for the catch Richard.
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GCC 4.6.3 caught these in dev mode as well.
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(closes ASTERISK-19057)
Reported By: Yuri
Patches: 348360chan_sip.diff uploaded by Yuri (license 5242)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

gcc version 4.6.2 caught an unused variable in the ilbc codec
library.  This would prevent compilation with --enable-dev-mode;
variable removed.
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* Fix corner cases in get_calleridname() parsing and ensure that the
output buffer is nul terminated.

* Make get_calleridname() truncate the name it parses if the given buffer
is too small rather than abandoning the parse and not returning anything
for the name.  Adjusted get_calleridname_test() unit test to handle the
truncation change.

* Fix get_in_brackets_test() unit test to check the results of
get_in_brackets() correctly.

* Fix parse_name_andor_addr() to not return the address of a local buffer.
This function is currently not used.

* Fix potential NULL pointer dereference in sip_sendtext().

* No need to memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it is nul
terminated.

* Reply with an accurate response if get_msg_text() fails in
receive_message().  This is academic in v1.8 because get_msg_text() can
never fail.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351667 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Change the RTCP RR and SR generation code to convert Asterisk's internal jitter
statistics to be represented in RTP timestamp units based on the rate of the
codec in use instead of in seconds.

(closes issue ASTERISK-14530)
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This patch prevents the domain string from getting mangled during the initreqprep
step by moving the initialization to before its immediate use.  It also documents
this pitfall for the ast_sockaddr_stringify functions.

(issue ASTERISK-19057)
Reported by: Yuri
Review: https://reviewboard.asterisk.org/r/1678/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351561 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.

(closes issue ASTERISK-19202)
Reported by: Catalin Sanda
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351506 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch includes the iLBC source code for distribution with Asterisk.
Clarification regarding the iLBC source code was provided by Google, and
the appropriate licenses have been included in the codecs/ilbc folder.

Review: https://reviewboard.asterisk.org/r/1675
Review: https://reviewboard.asterisk.org/r/1649

(closes issue: ASTERISK-18943)
Reporter: Leif Madsen
Tested by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 number from a P-Asserted-Identity cause the header parsing logic was wrong. 
Changing the parsing functions to the sip header parsing APIs in 
reqresp_parser.h solves this problem.

Review: https://reviewboard.asterisk.org/r/1673
Reviewed by: wdoekes2 and Mark Michelson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351409 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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