git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the kqueue timing module to conform to current timer API.

This fixes issues with using the kqueue timing source on Asterisk 13 on
FreeBSD 10. These issues include:

- Remove support for kevent64().  The values used to support Asterisk timers
  fit within 32bits and so can be handled on all platforms via kevent().

- Provide debug logging for, but do not track, unacked events.  This matches
  the behavior of all other timer implementations.

- Implement continuous mode by triggering and leaving active, a user event.
  This ensures that the file descriptor for the timer returns immediately from
  poll(), without placing the load of a high speed timer on the kernel.

- In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
  On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
  type kqueue supports for timers.

- In kqueue_timer_get_event(), assume the caller woke up from poll() and just
  return the mode the timer is currently in. This matches all other timer
  implementations.

- Adjust the test code now that unacked events are not tracked.

Review: https://reviewboard.asterisk.org/r/4465/

ASTERISK-24857 #close
Reported by: scsiguy
Tested by: Ed Hynan
patches:
  rb4465.patch submitted by scsiguy (License 6692)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use the standard ast_log instead of ast_log_safe for STANDALONE programs.

Review: https://reviewboard.asterisk.org/r/4538/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This introduces a new logger routine ast_log_safe.  This routine should be
used for all error messages in code that can be run as a result of ast_log.
ast_log_safe does nothing if run recursively.  All error logging in
astobj2.c, strings.c and utils.h have been switched to ast_log_safe.

This required adding support for raw threadstorage.  This provides direct
access to the void* pointer in threadstorage.  In ast_log_safe, NULL is used
to signify that this thread is not already running ast_log_safe, (void*)1 when
it is already running.  This was done since it's critical that ast_log_safe
do nothing that could log during recursion checking.

ASTERISK-24155 #close
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/4502/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add ast_register_cleanup to utils/clicompat.c to deal with
  any utils that copy sources from main.
* Asterisk 13+: remove unused variables from core_local.c.

Review: https://reviewboard.asterisk.org/r/4534/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433501 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups.  Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe.  ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.

Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.

ASTERISK-24142 #close
Reported by: David Brillert

ASTERISK-24683 #close
Reported by: Peter Katzmann

ASTERISK-24805 #close
Reported by: Badalian Vyacheslav

ASTERISK-24881 #close
Reported by: Corey Farrell

Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
  caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
  crashes in some cases.  In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
  shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.

ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.

ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* In res/res_sorcery_realtime.c: Broke long line.

* In main/bucket.c: Eliminated unnecessary NULL check as
ast_sorcery_unref() is NULL tolerant and set the global object to NULL
after unref in the system shutdown bucket_cleanup().
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
setting was never actually processed. This was due to not properly copying
over the global settings to the client settings when applying the
configuration to the run-time object.

Review: https://reviewboard.asterisk.org/r/4496/

ASTERISK-14233
ASTERISK-24780 #close
Reported by: Simon Arlott
patches:
  asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.

This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.

Unit tests have also been written for all of the above to confirm the API and
functionality.

ASTERISK-24834 #close
Reported by: Matt Jordan

ASTERISK-24836 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens.  If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.

Consequences of these unnecessary messages:

* The caller can start hearing ringback before the far end even gets the
call.

* Many phones tend to grab the first connected line information and refuse
to update the display if it changes.  The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.

When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled.  When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.

* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages.  The default is "no" to disable sending the
unnecessary messages.

ASTERISK-24781 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4473/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to
(long) when printing members of certain time structs.

Review: https://reviewboard.asterisk.org/r/4507

ASTERISK-24879 #close
Reported by: snuffy
Tested by: snuffy
patches:
  openbsd-time64.diff uploaded by snuffy (License 5024)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch addresses compilation issues for OpenBSD. Specifically, it
addresses:
 * It allows including <sys/vmmeter.h> in asterisk.c
 * Provides a needed (size_t) cast in xmldoc.c

In 13+, it also addresses a conditional inclusion in loader.c.

Review: https://reviewboard.asterisk.org/r/4506

ASTERISK-24880 #close
Reported by: snuffy
Tested by: snuffy
patches:
  misc-openbsd.diff uploaded by snuffy (License 5024)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Valgrind found some memory leaks associated with
ast_pjsip_rdata_get_endpoint().  The leaks would manifest when sending
responses to OPTIONS requests, processing MESSAGE requests, and
res_pjsip supplements implementing the incoming_request callback.

* Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in
res/res_pjsip.c:supplement_on_rx_request(),
res/res_pjsip/pjsip_options.c:send_options_response(),
res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
res/res_pjsip_messaging.c:send_response().

* Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
res/res_pjsip_nat.c:nat_on_rx_message().

* Fixed inconsistent but benign return value in
res/res_pjsip/pjsip_options.c:options_on_rx_request().

Review: https://reviewboard.asterisk.org/r/4511/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Valgrind found a memory leak and invalid access.

* Fix invalid access by sscanf() being fed a non-nul terminated string of
digits in res/res_pjsip_sdp_rtp.c:get_codecs().

* Fix memory leak in main/sorcery.c:sorcery_object_field_destructor().

* Fix potential NULL pointer dereference in
main/xmldoc.c:xmldoc_get_syntax_config_option().

Review: https://reviewboard.asterisk.org/r/4513/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433200 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When r432935 was merged, it did correctly fix a situation where a FILE read
operation on the middle of a file buffer would not read the requested length
in the parameters passed to the FILE function. Unfortunately, it would also
allow the FILE function to append more bytes than what was available in the
buffer if the length exceeded the end of the buffer length.

This patch takes the minimum of the remaining bytes in the buffer along with
the calculated length to append provided by the original patch, and uses
that as the length to append in the return result. This patch also updates
the unit tests with the scenarios that were originally pointed out in
ASTERISK-21765 that the original implementation treated incorrectly.

ASTERISK-21765
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The script was added in 13, but when committed to trunk it caused a branch to
occur due to some trunk only alemebic changes. This fixes it so that the new
'add_pjsip_endpoint_identifier_order script points to the correct down revision.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When logger.conf is missing or invalid enable console logging and display
an error message.

ASTERISK-24817 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4497/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433130 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Replace functions for ref/undef of dialogs and peers with macro's
  to call ao2_t_bump/ao2_t_cleanup.
* Enable passthough of REF_DEBUG caller information to sip_alloc and
  find_call.

ASTERISK-24882 #close 
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4189/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Release the scheduler reference to the dialog for reinvite timeout during
dialog_unlink_all.

ASTERISK-24876 #close 
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4491/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433114 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using DEBUG_CHAOS several instances of a null
pointer crash, and one uninitialized variable
were uncovered and fixed.  Also added details
on why Asterisk failed to initialize.

Review: https://reviewboard.asterisk.org/r/4468/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Locate potential crashes by exercising seldom
used code paths.  This patch introduces a new
define DEBUG_CHAOS, and mechanism to randomly
return an error condition from functions that
will seldom do so.  Functions that handle the
allocation of memory get the first treatment.

Review: https://reviewboard.asterisk.org/r/4463/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433062 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In this collection of small patches to prevent
Valgrind errors are: fixes for reference leaks
in config hooks, evaluating a parameter beyond
bounds, and accessing a structure after a lock
where it could have been already free'd.

Review: https://reviewboard.asterisk.org/r/4407/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Valgrind found some memory leaks associated with ast_sockaddr_resolve().
Most of the leaks had already been fixed by earlier memory leak hunt
patches.  This patch performs an audit of ast_sockaddr_resolve() and found
one more.

* Fix ast_sockaddr_resolve() memory leak in
apps/app_externalivr.c:app_exec().

* Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs
parameter for safety so the pointer will never be uninitialized on return.
The same goes for res/res_pjsip_acl.c:extract_contact_addr().

* Made functions that call ast_sockaddr_resolve() with RAII_VAR()
controlling the addrs variable use ast_free instead of ast_free_ptr to
provide better MALLOC_DEBUG information.

Review: https://reviewboard.asterisk.org/r/4509/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updated some documentation stating that endpoint identifiers registered without
a name are place at the front of the lookup list. Also renamed register method
'ast_sip_register_endpoint_identifier_by_name' to
'ast_sip_register_endpoint_identifier_with_name'

ASTERISK-24840
Reported by: Mark Michelson
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433032 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.

ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In realtime, it is normal to have a database with both 'allow' and 'disallow'
columns in the schema. It is perfectly valid to have an 'allow' value of
'!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files,
you can't *not* provide the disallow value. Thus, the empty disallow value
causes a spurious WARNING message, which is kind of annoying.

This patch makes it so that a 'disallow' value with no ... value ... is
ignored. Granted, you can still screw this up as well, as technically
specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no
'allow' value in your database. But really, why would you do that? WHY?

ASTERISK-16779 #close
Reported by: Atis Lezdins
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432972 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This code originally kept a lock held when performing the HTTP
request to ensure that the options provided to curl remain valid.
This doesn't seem to be necessary these days and holding the lock
caused requests to happen sequentially instead of in parallel.

ASTERISK-18708 #close
Reported by: Dave Cabot
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In some countries, privacy laws specify that SMS content cannot be saved by a
provider. This patch adds a new option to the SMS application, 'n', which
prevents the SMS content from being written to the SMS log.

ASTERISK-22591 #close
Reported by: Jan Juergens
patches:
  DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "core set debug channel" CLI command mistakenly had source filenames
added to its tab completion. This occurred because the CLI generator fell back
to the "core set debug" command which permits setting debug at a source
filename level.

ASTERISK-21038 #close
Reported by: Richard Kenner
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The loop that reads in a file was not correctly using the offset when
determining what bytes to append to the output. This patch corrects
the logic such that the correct portion of the file is extracted when an
offset is specified.

ASTERISK-21765
Reported by: John Zhong
Tested by: Matt Jordan, Di-Shi Sun
patches:
  file_read_390821.patch uploaded by Di-Shi Sun (License 5076)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch corrects the documentation for the AMD application. Specifically:
* It documents the maximum_word_length option, which limits the maximum allowed
  length of a single utterance.
* It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH
  was documented as MAXWORDS, while MAXWORDS was undocumented.

Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues.

ASTERISK-19470 #close
Reported by: Frank DiGennaro
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432921 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also fixed similar problem with AMI action PJSIPShowEndpoints.

ASTERISK-24872 #close
Reported by: Dmitriy Serov

Review: https://reviewboard.asterisk.org/r/4487/
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Merged revisions 432894 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The res_pjsip modules were manually checking both name and number
presentation values when there is a function that determines the combined
presentation for a party ID struct.  The function takes into account if
the name or number components are valid while the manual code rarely
checked if the data was even valid.

* Made use ast_party_id_presentation() rather than manually checking party
ID presentation values.

* Ensure that set_id_from_pai() and set_id_from_rpid() will not return
presentation values other than what is pulled out of the SIP headers.  It
is best if the code doesn't assume that AST_PRES_ALLOWED and
AST_PRES_USER_NUMBER_UNSCREENED are zero.

* Fixed copy paste error in add_privacy_params() dealing with RPID
privacy.

* Pulled the id->number.valid test from add_privacy_header() and
add_privacy_params() up into the parent function add_id_headers() to skip
adding PAI/RPID headers earlier.

* Made update_connected_line_information() not send out connected line
updates if the connected line number is invalid.  Lower level code would
not add the party ID information and thus the sent message would be
unnecessary.

* Eliminated RAII_VAR usage in send_direct_media_request().

Review: https://reviewboard.asterisk.org/r/4472/
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Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.

ASTERISK-24840
Reported by: Mark Michelson
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Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Revision 432834 introduced a build error when MALLOC_DEBUG
is used.  Switch callid threadstorage to simple
AST_THREADSTORAGE since we no longer need custom cleanup.

Reported by: Corey Farrell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432851 65c4cc65-6c06-0410-ace0-fbb531ad65f3