With ptlib 2.10.9, the configure script fails due to grep returning multiple
matches for the pattern it searches for. This patch updates the pattern
matching to return only the actual version for the symbol searched for,
PTLIB_VERSION.

(closes issue ASTERISK-20980)
Reported by: Stefan Reuter
patches:
  ASTERISK-20980-1.patch uploaded by Stefan Reuter (license 5339)
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There is currently an edge case where call number 32768 might be allocated for
a call, even though the IAX2 protocol requires call numbers be only 15 bits.
This resulted in some unpredictable behavior when call number 32678 is chosen.

This patch was mostly written by Richard Mudgett via ReviewBoard.  I'm just
committing it.

Review: https://reviewboard.asterisk.org/r/2293/
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This patch came about due to a problem observed where wav files had an
empty header.  The header is supposed to be updated in wav_close().  It
turns out that this was broken when the cache_record_files option from
asterisk.conf was enabled.  The cleanup code was moving the file to its
final destination *before* running the close() method of the file
destructor, so the header didn't get updated.

Another problem here is that the move was being done before actually
closing the FILE *.

Finally, the last bug fixed here is that I noticed that wav_close()
checks for stream->filename to be non-NULL.  In the previous cleanup
order, it's checking a pointer to freed memory.  This doesn't actually
cause anything to break, but it's treading on dangerous waters.  Now the
free() of stream->filename is happening after the format module's
close() method gets called, so it's safer.

Review: https://reviewboard.asterisk.org/r/2286/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an option that lets you specify that the timestamps going into the realtime
queue log should be in GMT instead of local time.

Review: https://reviewboard.asterisk.org/r/2287/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "sound_only_one" sound was not being set even though it was configured.  In
looking into this, I found that the "join" and "leave" prompts were not being
set either.

(closes issue ASTERISK-20898)
Reported by: Stephan
Tested by: Stephan
Patches:
    asterisk-20898-custom-sounds-ignored.diff uploaded by 
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2289/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380178 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380165 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Rename multiplexed_thread variables to muxed_thread.  It is shorter and
my editer tagging works much better.  Struct names and variable names have
different purposes and therefore should have different names.

* Renamed the multiplexed_threads container to muxed_threads for
consistency.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380142 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made multiplexed_bridge_destroy() check if anything to destroy and
cleared bridge_pvt pointer after destruction.

* Made multiplexed_add_or_remove() handling of the chans array simpler.

* Extracted bridge_channel_poke().

* Simplified bridge_array_remove() handling of the bridge->array[].  The
array does not have a NULL sentinel pointer.

* Made ast_bridge_new() not create a temporary bridge just to see if it
can be done.  Only need to check if there is an appropriate bridge tech
available.

* Made ast_bridge_new() clean up on allocation failures.

* Made destroy_bridge() free resources in the opposite order of creation.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Breaking long lines
* Word wrapping comment blocks.
* Removing redundant initializers.
* Debug message wording.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380108 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Sorcery is a unifying data access layer which provides a pluggable mechanism to allow
object creation, retrieval, updating, and deletion using different backends (or wizards).

This is a fancy way of saying "one interface to rule them all" where them is configuration,
realtime, and anything else that comes along.

Review: https://reviewboard.asterisk.org/r/2259/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds a dial softkey when the device is in DAFD. The softkey is greyed (unusable) 
until a possible dialplan match is entered. Code includes updating 
transmit_selectsoftkeys to allow the use of a button mask. Also add option
to use # or * as a dial now button. Original patch by snuffy cleaned up by myself.

Review: https://reviewboard.asterisk.org/r/2277/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380057 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk responds with an SDP ANSWER for SRTP, it had the code to
correctly fill in the crypto data, which was overwritten by a call to
sdp_crypto_offer. Corrected the situation by changing sdp_crypto_offer
to not replacing crypto data if it already exists.

(closes issue ASTERISK-20849)
Reported by: José Luis Millán
Tested by: Iñaki Baz Castillo
Patches:
	fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380044 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The documentation for ConfbridgeList states that the Conference field is
optional. That's not really the case: if you fail to provide a Conference
number, the command will kick back an error.

(closes issue AST-1090)
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the ability to get the DPMA version, a listing of the local
firmware directory, and indexes of configured remote directories.

(closes issue AST-1070)
Reported By: Malcolm Davenport
Tested By: Kinsey Moore <kmoore@digium.com>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Put the external obj pointer in the message instead of the internal version.
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This patch fixes the problem, but the issue includes a test which is still
being considered for the automated test suite.

(issue ASTERISK-20919)
Reported by: NITESH BANSAL
Patches:
	patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During the conversion to the newer CLI command structure the old definitions were
commented out.  I think it's safe to remove them completely now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The old prompts for the administrator menu were inadequate. They didn't mention
that the menu had additional options through the 8 key and pressing the 8 key
wouldn't reveal what those options were. This patch fixes all of that while
also organizing code pertaining to each individual menu type which was
previously all stored in one gigantic function along with many of the basic
conference functions.

(closes issue AST-996)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/360/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379864 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.

Review: https://reviewboard.asterisk.org/r/2265/

(closes issue ASTERISK-20882)
Reported by: mjordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two bugs:
 * If an outbound call is made from a SLA phone using SLAStation, then there is
   no ringtone audible to the phone that originates the call. The indication of
   the ringing was not being passed to the SLA station; this patch fixes that
   by passing through the progress indications.
 * If an SLA station hangs up before the called party answers, then the channel
   to the called party continues to ring until a timeout occurs. If the called
   party manages to answer, Asterisk attempts to connect the called party to
   a non-existant MeetMe room. This patch corrects the behavior by abandoning
   the call attempt if it detects that the SLA station is no longer in use
   while attempting to call the called party.

Review: https://reviewboard.asterisk.org/r/2275/

(closes issue ASTERISK-20462)
Reported by: dkerr
patches:
  asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
  asterisk-11-bugid20462.patch uploaded by dkerr (license 5558)

(closes issue ASTERISK-20440)
Reported by: dkerr
patches:
  asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
  asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
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* Generate a warning message if sound files do not exist when trying to
play the user count to the conference.  Use the new helper routine
sound_file_exists() for consistency.

* Put the new user into autoservice when playing user counts to the
conference.

* Check the return value of ast_bridge_impart().
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When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
 * Not displaying the splash screen on a remote console.
 * Displaying an error message on stderr when a remote console cannot connect
   to a running instance of Asterisk.

In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.

Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.

(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
  asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
  ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379753 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When iLBC is being used with a jitter buffer and the jb has to
interpolate frames, it generates frames with a null pointer and a
non-zero datalen. This is now handled properly.

(closes issue ASTERISK-20914)
Reported By: John McEleney
Patches:
  ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379720 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Skinny device call logging (ie missed, place and received calls) has issues 
because the incorrect sequence of callstates is/can be sent to the device.
This patch removes some extra callstate updates driven by forces external
to skinny and ensures the needed intermediary callstate messages are sent.

(closes issue ASTERISK-20964)
Reported by: wedhorn
Tested by: snuffy, myself
Patches: 
    ast11-skinny-calllog01.diff uploaded by wedhorn (license 5019)
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Add LDAP dev package to Debian/Ubuntu install list.  Existed in Redhat already.

(issue ASTERISK-20886)
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An incorrect string initializations was left in ast_str_encode_mime from the
patch that converted string manipulations to use ast_str strings (r191140).
The string initialization causes a crash when ast_str_set is called on
the string later on in the function.

(closes issue ASTERISK-18697)
Reported by: Chris Boot
patches:
  minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309)

(issue ASTERISK-20854)
Reported by: Chris Warr
Tested by: Chris Warr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes a couple of issues with the way skinny handles sessions by ensuring
sessions aren't used after being freed. Some other minor changes.

Review: https://reviewboard.asterisk.org/r/2272/
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(closes issue ASTERISK-16854)
Review: https://reviewboard.asterisk.org/r/2276
Reported-by: Ovidiu Sas
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379549 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk forks itself into the background via a call to daemon, it must
re-set the pid value of the new process. Otherwise, astcanary gets the pid
value of the process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer communicate
with the proverbial canary in the coal mine.

This patch ensures that the correct process identifier is used by astcanary.

Note that this is getting committed to 10 as a regression fix.

(closes issue ASTERISK-20947)
Reported by: Jakob Hirsch
Tested by: mjordan
patches:
  asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113)
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 * This allows us to remove some special-case build logic.
 * 10.5 is down to less that 8% of the OS X market share. 10.4 is down to 
   under 2%.
 * Apple is no longer releasing security updates for 10.5 and earlier.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379495 65c4cc65-6c06-0410-ace0-fbb531ad65f3