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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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)


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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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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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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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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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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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(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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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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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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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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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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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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 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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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

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  r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines
  
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    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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