Adding information about including "language" in the standard channel snapshot
output to the CHANGES file. Note the actual source changes have already been
previously committed.

ASTERISK-24553
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429327 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Frames with a payload length of 0 were incorrectly handled in res_http_websocket.
Provided a frame with a payload had been received prior it was possible for a double
free to occur. The realloc operation would succeed (thus freeing the payload) but be
treated as an error. When the session was then torn down the payload would be
freed again causing a crash. The read function now takes this into account.

This change also fixes assumptions made by users of res_http_websocket. There is no
guarantee that a frame received from it will be NULL terminated.

ASTERISK-24472 #close
Reported by: Badalian Vyacheslav

Review: https://reviewboard.asterisk.org/r/4220/
Review: https://reviewboard.asterisk.org/r/4219/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes the MWI test regressions caused by r429127 and ensures that
contacts have non-zero qualify_frequency before attempting scheduling.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429247 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When connecting to the remote console, an id string
is first provided that consts of the hostname, pid,
and version.  This is parsed by the remote instance
using a buffer that may be too short, and can allow
a buffer overrun because it is not terminated. This
patch adds termination and a larger buffer.

Review: https://reviewboard.asterisk.org/r/4182/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The channel "language" was already part of a channel snapshot, however is was
not sent out over AMI or ARI. This patch makes it so the channel "language" is
included in the appropriate AMI or ARI events.

ASTERISK-24553 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4245/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's).  This patch ensures that Asterisk uses the original device
address when using direct media.

ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using a non-default sorcery wizard (in this instance realtime) for outbound
publishes Asterisk will crash after a stack overflow occurs due to the code
infinitely recursing.  The fix entails removing the outbound publish state
dependency from the outbound publish sorcery object and instead keeping an in
memory container that can be used to lookup the state when needed.

ASTERISK-24514 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4178/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429176 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If an originator channel is specified when originating a channel the linked ID
of it will be applied to the newly originated outgoing channel. This allows
an association to be made between the two so it is known that the originator
has dialed the originated channel.

ASTERISK-24552 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4243/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change staggers initiation of outbound qualify (OPTIONS) attempts
to reduce instantaneous server load and prevent network congestion.

Review: https://reviewboard.asterisk.org/r/4246/
ASTERISK-24342 #close
Reported by: Richard Mudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429129 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per
semantic versioning, that warrants a bump in the minor version number, as it
reflects a backwards compatible change. Hence, this commit.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The reviewboard description does a fine job of summarizing this, so here it is:

A reporter discovered that Asterisk would crash when attempting to retransmit
a reinvite that had previously received a 491 response. The crash occurred
because a pjsip_tx_data structure was being saved for reuse, but its reference
count was not being increased. The result was that the pjsip_tx_data was being
freed before we were actually done with it. When we attempted to re-use the
structure when re-sending the reinvite, Asterisk would crash.

The fix implemented here is not to try holding onto the pjsip_tx_data at all.
Instead, when we reschedule sending the reinvite, we create a brand new
pjsip_tx_data and send that instead. Because of this change, there is no need
for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on
it any more. So any code referencing its use has been removed.

When this initial fix was introduced, I encountered a second crash when
processing a subsequent 200 OK on a rescheduled reinvite. The reason was
that when rescheduling the reinvite, we gave the wrong location for a
response callback. This has been fixed in this patch as well.

ASTERISK-24556 #close
Reported by Abhay Gupta

Review: https://reviewboard.asterisk.org/r/4233
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The AMI event is called NewConnectedLine and the ARI event is called
ChannelConnectedLine.

ASTERISK-24554 #close
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/4231
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This corrects several bugs that currently exist in the stasis
application code.

* After a masquerade, the resulting channels have channel topics that
  do not match their uniqueids
** Masquerades now swap channel topics appropriately
* StasisStart and StasisEnd messages are leaked to observer
  applications due to being published on channel topics
** StasisStart and StasisEnd publishing is now properly restricted
   to controlling apps via app topics
* Race conditions exist where StasisStart and StasisEnd messages due to
  a masquerade may be received out of order due to being published on
  different topics
** These messages are now published directly on the app topic so this
   is now a non-issue
* StasisEnds are sometimes missing when sent due to masquerades and
  bridge swaps into and out of Stasis()
** This was due to StasisEnd processing adjusting message-sent flags
   after Stasis() had already exited and Stasis() had been re-entered
** This was corrected by adjusting these flags prior to sending the
   message while the initial Stasis() application was still shutting
   down

Review: https://reviewboard.asterisk.org/r/4213/
ASTERISK-24537 #close
Reported by: Matt DiMeo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When repeatedly starting/stopping a Monitor on a channel, the accumulated
in/out sample counts are never reset to 0. This can cause inadvertent jumps
in the recordings, as the code in the channel core will determine incorrectly
that a jump in the recorded file position should occur. Setting the sample
counts to 0 simply reflects the initial state a Monitor should be in when it
is started, as this is the initial count that would be on the channels at that
time.

ASTERISK-24573 #close
Reported by: Nuno Borges
patches:
  24573.patch uploaded by Nuno Borges (License 6116)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the app_meetme module is loaded without its configuration file, the
module settings aren't initialized. In particular, this impacts the use
of logging realtime members. This patch guarantees that we always set the
default module settings on initial load.

Review: https://reviewboard.asterisk.org/r/4242/

ASTERISK-24572 #close
Reported by: Nuno Borges
patches:
  24572.patch uploaded by Nuno Borges (License 6116)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add new global, instance and wizard observers.
instance_created
wizard_registered
wizard_unregistered
instance_destroying
instance_loading
instance_loaded
wizard_mapped
object_type_registered
object_type_loading
object_type_loaded
wizard_loading
wizard_loaded

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4215/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429001 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On a 32-bit system, a type of intmax_t will result in a compilation warning
when formatted as a 'long int'. Use the format specifier of %jd (which was
what was used originally in manager.c) to format the JSON extracted integer
on both 32-/64-bit systems.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a race condition between the raising of test AMI events (which
drive many tests in the Asterisk Test Suite) and other AMI events. Prior to
this patch, the Stasis messages published to the test topic were not forwarded
to the AMI topic. Instead, the code in manager had a dedicated handler for test
messages that was independent of the topics forwarded to the AMI topic. This
results in no synchronization between the test messages and the rest of the
Stasis messages published out over AMI. In some test with very tight timing
constraints, this can result in out of order messages and spurious test
failures. Properly forwarding the Test Suite topic to the AMI topic ensures
that the messages are synchronized properly.

This patch does that, and moves the message handling to the Stasis definition
of the Test Suite message in test.c as well.

Review: https://reviewboard.asterisk.org/r/4221/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Despite failing less often, the ordering of the ATTENDEDTRANSFER event and the
BRIDGE_EXIT event for the Alice and David channels is not defined. This makes
the test still fail.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the publication of attended transfer messages were pushed to another
thread, some subtle race conditions were introduced with the CEL unit tests.
This patch fixes one of them, and pushes the other to ASTERISK-22367, which
already exists to fix another bouncy CEL unit test.

In particular, this patch fixes the test_cel_attended_transfer_bridges_link
test, and defers the test_cel_attended_transfer_bridges_swap test to the
aforementioned JIRA issue.

ASTERISK-22367
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The UW IMAP library is instrinsically not thread-safe, and relies upon higher
level applications to guarantee thread safety. For the most part, this is
provided by the vms object, which provides locking for individual streams.
Unfortunately, this is not sufficient for calls to mail_open which create the
IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
function for determining the address of a system based on a hostname,
ip_nametoaddr.

In the ip6_unix implementation of this function, static variables are used
to hold parsing buffers. This can cause a crash if multiple threads attempt
to convert a hostname to an address at the same time. Locking on a single
mail stream is not sufficient to prevent simultaneous access to these static
variables.

In the IMAP library, this function can be called from the mail_open and
imap_status functions. As the imap_status function is not used by
app_voicemail, locking on access to mail_open is sufficient to prevent
any mangling of the buffers.

Review: https://reviewboard.asterisk.org/r/4188/

ASTERISK-24516 #close
Reported by: David Duncan Ross Palmer
Tested by: David Duncan Ross Palmer
patches:
  ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Tested-by: George Joseph
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When merging the changes up stream in r428687, I missed the fact that the
signature for stasis_message_type_create was changed. This patch fixes
the compilation issues introduced by that merge.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch makes a small rearrangement to only do dialplan lookups during
loopback switches if the pattern matches. Prior to this patch, the dialplan
lookups were always performed, even when the result would be discarded.
Dialplan lookups can be very costly if remote switches - like DUNDi - are
present. In those cases extension matching is sped up considerably, making
the issue of lost digits more manageable.

As collateral damage, 6 trailing spaces were killed.

Review: https://reviewboard.asterisk.org/r/4211

ASTERISK-24577 #close
Reported by: Birger Harzenetter
patches:
  ast-loopback.patch uploaded by Birger Harzenetter (License 5870)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are two methods within res_pjsip_refer for keeping track of the state of a transfer.
The first is a framehook which looks at frames passing by to determine the state. The second
subscribes to know when the channel joins a bridge. In the case when the channel joins the
bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology
from getting used.

This change gets the channel and if it still exists remove the framehook.

Review: https://reviewboard.asterisk.org/r/4218/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add
const char *ast_variable_find_in_list(const struct ast_variable *list,
   const char *variable);

ast_variable_find() requires a config category to search whereas
ast_variable_find_in_list() just needs the root list element which is
useful if you don't have a category.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4217/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While troubleshooting other things I realized there were no pjsip cli
commands for identify.  This patch adds them.  It also also fixes a
reference leak when a 'show endpoint' displayed identifies and properly
sets the return code if load_module can't allocate a cli formatter structure.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4212/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428732 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rtp_engine: Add support for transporting signed linear at 12kHz, 24kHz, 32kHz, 44kHz, 48kHz, 96kHz, and 192kHz over RTP.

This change adds mappings in the RTP engine layer for the remaining signed linear formats.

ASTERISK-24274 #close
Reported by: Frankie Chin

Review: https://reviewboard.asterisk.org/r/4093/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.

For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
  a single message - the subscription is created, a message is published, the
  delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.

This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.

Review: https://reviewboard.asterisk.org/r/4193

ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording.

The Record dialplan function trims 1/4 of a second from the end of recordings in case
they are terminated because of DTMF. When hanging up, however, you don't want this to happen.
This change makes it so on hangup this does not occur.

ASTERISK-24530 #close
Reported by: Ben Smithurst
patches:
 app_record_v2.diff submitted by Ben Smithurst (license 6529)

Review: https://reviewboard.asterisk.org/r/4201/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The static buffer for codecs when invoking the "core show channeltype" CLI command
did not have enough room for all codecs. This has been extended so it does.

ASTERISK-24542 #close
Reported by: snuffy
patches:
 channeltype-tech.diff submitted by snuffy (license 5024)

Review: https://reviewboard.asterisk.org/r/4204/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix the failing /channels/features/test_features_channel_dtmf unit test.

DTMF emulation does not work without a stream of packets to prod the
emulation code.

Review: https://reviewboard.asterisk.org/r/4199/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Any partially collected DTMF digits for a DTMF hook need to be pushed into
the bridge when a channel leaves the bridging system as if there were a
timeout.

Review: https://reviewboard.asterisk.org/r/4199/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When shutting down Asterisk that has an active AMI connection, you get
several "failed to extend from %d to %d" messages because use of the
EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the
event.

* Created MAX_AUTH_PERM_STRING to use when creating stack based struct
ast_str variables used with the authority_to_str() and
user_authority_to_str() functions instead of a variety of magic numbers
that could be too small.

* Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so
it will not attempt to add all permission level strings.

Review: https://reviewboard.asterisk.org/r/4200/
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As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime
was tossing database fields that didn't have an exact match to a sorcery
registered field.  This broke the ability to use regexes as field names which
manifested itself as a failure of res_pjsip_phoneprov_provider which uses
this capability.  It also broke handling of fields that start with '@' in
realtime but I don't think anyone noticed.

This patch does the following...
* Modifies ast_sorcery_fields_register to pre-compile the name regex.
* Modifies ast_sorcery_is_object_field_registered to test the regex if it
  exists instead of doing an exact strcmp.
* Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work
  with realtime.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4185/
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In r428165, two bugs were introduced:

* Prior to entering the features retry loop, the buffer that holds the
  collected digits is wiped. However, this inadvertently wipes out the
  first collected digit on the first pass through, which is obtained
  in ast_stream_and_wait. This caused all of the features tests to fail.
* If ast_app_dtget returns a hangup (-1), the loop would retry incorrectly.
  If we detect a hangup, we have to stop trying the feature.

This patch fixes both issues.

Review: https://reviewboard.asterisk.org/r/4196/
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Prior to this commit, the address family of the first item in an ACL
was used to compare all incoming traffic. This could lead to traffic
of other IP address families bypassing ACLs.

ASTERISK-24469 #close

Reported by Matt Jordan
Patches:
	ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283)

AST-2014-012
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The DB dialplan function when executed from an external protocol (for instance
AMI), could result in a privilege escalation.

Asterisk now inhibits the DB function from being executed from an external
interface if the live_dangerously option is set to no.

ASTERISK-24534
Reported by: Gareth Palmer
patches: submitted by Gareth Palmer (license 5169)
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The biggest problem this patch fixes is that ACLs weren't previously being
loaded when the res_pjsip_acl module was loaded. Yikes. In addition, the
ACL options contact_permit and contact_acl were effectively interpreted as
contact_deny and this patch fixes that as well.

AST-1418 #close
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4120/

ASTERISK-24531 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4171/
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