If a frame handling routine returns a list of frames (vs. a single frame)
those frames are never passed to a tech's write_stream handler even if one is
available. For instance, if a codec translation occurred and that codec
returned multiple frames then those particular frames were always only sent
to the tech's "write" handler. If that tech (pjsip for example) was stream
capable then those frames were essentially ignored. Thus resulting in bad
audio.

This patch makes it so the "write_stream" handler is appropriately called
for all cases, and for all frames if available.

ASTERISK-28795 #close

Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d

The dial application had 80 characters of destination length
limitation. But this limitation causes unexpected dial string
cut if the dial string is long.

Removed unnecessary limited buffer to support longer dial
destination.

ASTERISK-27946

Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330

named_acl.c (which is really a named_ha) now uses ast_ha_output.

I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set.  If this works then we can add
similar to other modules as required.

Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f

binutils 2.34 merged this commit:

https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\
	h=fd3619828e94a24a92cddec42cbc0ab33352eeb4

Which effectively does things like:

-#define bfd_section_size(bfd, ptr) ((ptr)->size)
-#define bfd_get_section_size(ptr) ((ptr)->size)

+#define bfd_section_size(sec) ((sec)->size)

So in order to remain backwards compatible we need to detect this API
change, and adjust accordingly.  The simplest is to notice that the
bfd_get_section_size and bfd_get_section_vma MACROs are no longer
defined, and define then onto the new API.  The alternative is to litter
the code with a number of #ifdef #else #endif splatters right through
the code.

Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f

Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5

Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.

This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.

ASTERISK-28780

Change-Id: I8044c06daa06f0f16607788c596f55623be26f58

A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.

Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.

ASTERISK-26711 #close
Reported by: Vitold

Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0

The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in
65b83815 such that we would never
actually make a DNS TXT query as described.

This patch restores the documented behavior.

ASTERISK-19460 #close
Reported by: George Joseph

Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376

When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.

Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.

ASTERISK-28755 #close

Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f

There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.

Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e

There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.

Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113

This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.

This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.

ASTERISK-28697

Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b

Dump OpenSSL's error stack to the error log when things fail.

ASTERISK-28750 #close
Reported by: Martin Zeh

Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a

This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.

The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.

Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.

ASTERISK-28733

Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5

When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.

* If the user wants to write to a wav49 file, make sure that it is
  reflected properly in MIXMONITOR_FILENAME.

* Add a note to the documentation describing this behavior.

* Add a note in main/file.c indicating that app_mixmonitor needs to be
  changed if the logic in build_filename was changed.

ASTERISK-24798 #close
Reported by: xrobau

Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c

Change-Id: Ic9b4afcc5398e7f46314419fc3c90433d818e35c

In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use.  To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes"  In Asterisk versions <18, the default
is "no" preserving existing behavior.  Beginning with
Asterisk 18, the option will default to "yes".

NOTE:  This change does not affect UserEvents or the ARI
TextMessageReceived events.

* Added the "hide_messaging_ami_events" option to asterisk.conf.

* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
  the "Message/ast_msg_queue" channel if the option is set in
  asterisk.conf.  This suppresses the reporting of the events.

Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b

When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.

* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.

NOTE:  This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.

ASTERISK-28677
Reported by: Maciej Michno

Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca

Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.

We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.

Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.

Additionally:

* Change 'enablestatic' to 'enable_static' but keep the former for
  backwards compatibility.
* Improve some internal variable names

ASTERISK-28710 #close

Change-Id: I647510f796473793b1d3ce1beb32659813be69e1

SILK @ 24kHz is not shown in the 'core show translation' output because of an
off-by-one-error. Discovered while looking into ASTERISK~19871.

ASTERISK-28706
Reported by: Sean Bright

Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510

In af90afd9, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.

Whitespace only, no functional change.

ASTERISK~23324
Reported by: Kevin McCoy

Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87

ASTERISK-28695 #close
Reported by: Kevin Flyn

Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8

Adds source port matching support when IP matching is used:

  [example]
  type = identify
  match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444

If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.

ASTERISK-28639 #close
Reported by: Mitch Claborn

Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92

When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020

When a topic is added to a pool, its name has the pool's topic
name prepended.  For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020

The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.

When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.

Fix:

* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.

ASTERISK-28633
Reported by: Joeran Vinzens

Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6

The Bridge application was inconsistent if the channel to bridge with is
not specified.  If no parameters are given then a warning is issued and
the current channel is hung up.  If options are given but no channel is
specified then a warning is issued and the current channel is not hung up.

* Made the Bridge application give a verbose message instead of a warning
if the channel to bridge with is not specified and made not hang up the
current channel.  As a result dialplan no longer needs to check if a
channel name is passed before calling Bridge and simply needs to check the
BRIDGERESULT channel variable instead.  This is something you likely want
your dialplan to do anyway.

* Fixed up L() option warning message.  It is up to the caller to
determine if the channel is hung up because of the warning.  Dial() hangs
up the current channel while Bridge() does not.

Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73

When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.

ASTERISK-28562 #close
Reported by: Robert Sutton

Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d

The db_init() function ultimately calls db_sync() which signals the
condition before it is initialized.

Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab

ASTERISK-28667 #close

Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2

When testing for the existance of a file, the media cache is searched even if
the file has no chance of being in it.  This can cause performance issues
as the media cache size increases.

As a result, calls to applications like Read and Playback using local files
must scan through the media cache before playing.  Under load and with a
large cache, this can delay the playback of those files.

This patch updates the function that checks for the existance of a file to
only consult the media cache database if the requested file is a remote path.
It introduces a new is_remote_path() function in main/file.c.

ASTERISK-28625  #close
Reported-by:  <kevin@phoneburner.com>

Change-Id: If91137493732d9034dafa381c081c69274a7dcc9

ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee

Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies.  It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.

Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

The problem is essentially the same as in ASTERISK~28245. Besides
the direct media scenario we have an additional scenario where a
special client is involved. This device mutes audio by default in
transmit direction (no rtp frames) and activates audio only by a
foot switch. In this situation dtmf input (pin for conferences,
transfer features codes , etc) using SIP INFO mode is not
understood properly especially when SIP INFO messages are sent
quickly.

This patch ensures that SIP INFO frames are properly queued and
processed in the above scenario. The patch also corrects situations
where successive dtmf events are received quicker than the
signalled event duration (plus minimum gap/pause) allows, i.e. DTMF
events have to be buffered in the ast channel read queue and
emulation has to be processed asynchronously at slower speed.

Reported by: Thomas Arimont
patches:
  trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525)

Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2

* Use ast_cli_completion_add() to improve performance when large number of
  cached items are present.

* Only complete one URI for commands that only accept a single URI.

* Change command documentation to wrap at 80 characters to improve
  readability.

Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902

If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.

Action: Originate
Channel: Local/1111
Application: Originate
Data: Local/2222,app,System,touch /tmp/owned

If the "system" authorization isn't set, we now block the
Originate app as well as the System, Exec, etc. apps.

ASTERISK-28580
Reported by: Eliel Sardañons

Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa

The current code sets alert levels starting from index 1.
Need to set on whole pool starting from index 0.

Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d

This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

* Pass caller information to frame allocation functions.
* Disable caching as it interfers with MALLOC_DEBUG reporting.
* Stop using ast_calloc_cache.

Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260

Here's the basic scenario that occurred when executing an AMI fast originate
while at the same time something else locks the channels container, and also
wants a lock on the dialed channel:

1. pbx_outgoing_attempt obtains a lock on a dialed channel
2. concurrently another thread obtains a lock on the channels container, and
   subsequently requests a lock on the dialed channel. It waits on #1. For
   instance, "core show channel <dialed channel"
3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt
   function exits
4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and
   attempts to hang up the dialed channel
5. hang up tries to obtain the channels container lock, but can't due to #2.
6. Asterisk is deadlocked.

The solution was to allow the pbx_outgoing_exec function to "steal" ownership
of the dialed channel, and handle hanging it up. The channel now is either hung
up prior to it being potentially locked by the initiating thread, or if locked
the hang up takes place in a different thread, thus alleviating the deadlock.

ASTERISK-28561
patches:
  iliketrains.diff submitted by Joshua Colp (license 5000)

Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a

Serializer pools have previously existed in Asterisk. However, for the most
part the code has been duplicated across modules. This patch abstracts the
code into an 'ast_serializer_pool' object. As well the code is now centralized
in serializer.c/h.

In addition serializer pools can now optionally be monitored by a shutdown
group. This will prevent the pool from being destroyed until all serializers
have completed.

Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971

This avoids use of the global variable and ensures topic_all remains
active until all topics are freed.

ASTERISK-28553
patches:
  ASTERISK-28553.patch by coreyfarrell (license 5909)

Change-Id: I9a8cd8977f3c3a6aa00783f8336d2cfb9c2820f1