One of the code paths in __ast_file_read_dirs will only get executed if
the OS doesn't support dirent->d_type OR if the filesystem the
particular file is on doesn't support it.  So, while standard Linux
systems support the field, some filesystems like XFS do not.  In this
case, we need to call stat() to determine whether the directory entry
is a file or directory so we append the filename to the supplied
directory path and call stat.  We forgot to truncate path back to just
the directory afterwards though so we were passing a complete file name
to the callback in the dir_name parameter instead of just the directory
name.

The logic has been re-written to only create a full_path if we need to
call stat() or if we need to descend into another directory.

Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba

During the development of Asterisk 14 the behavior of
the Command AMI action was altered such that the result
was returned on lines with a prefix of "Output: ". While
this was documented in the UPGRADE.txt file it is also
reasonable that this should bump the AMI version number.

ASTERISK-26556

Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42

The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how
many pairs of local/remote candidates will be made. If for some reason
we reach this upper bound, ICE will generally fail and no media will
flow between the browser and Asterisk.

This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of
pairs of candidates we'd theoretically allow, which is
PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied
PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame
Docker), this is far too low to allow WebRTC calls to succeed.

Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed
even when the system Asterisk was running on had quite a few virtual
interfaces.

Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55

In 9785e8d0, app_echo was updated to relay video source updates to the
channel for the purposes of displaying video in WebRTC tests.
Unfortunately, this can cause a Kafkaesque nightmare if two or more
Local channels are in a bridge together where their ends are in
app_echo. When this situation occurs, a video update sent into app_echo
will cause the video update to be relayed to the other Local channels,
causing another round of video updates, etc. In not much time at all,
the channel length queues will be overwhelmed, channel alert pipes will
fail, and all hell will break loose as Asterisk merrily continues to
throw more video update requests onto the channels.

This patch updates app_echo to *only* relay a single video update. Once
a video update has been made, all further video updates are dropped.
This meets the intended purpose of the original patch: if we get a video
update and we're in app_echo, go ahead and ask the sender to update
themselves. However, once we've got that video stream sync'd up, don't
keep spamming the world.

Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74

In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

This works the same as for AMI manager variables. Set
"channelvars=foo,bar" in your ari.conf general section, and then the
channel variables "foo" and "bar" (along with their values), will
appear in every Stasis websocket channel event.

ASTERISK-26492 #close
patches:
  ari_vars.diff submitted by Mark Michelson

Change-Id: I5609ba239259577c0948645df776d7f3bc864229

Libedit 3.1 is not build with unicode on as a default and so the
prototype for the el_gets callback changed from expecting a char buffer
to accepting a wchar buffer.  If ast_el_read_char isn't changed,
the cli reads garbage from teh terminal.

Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
updated ast_el_read_char to use the HAVE_ define to detemrine whether
to use char or wchar.

ASTERISK-26592 #close

Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a

When optimistic SRTP was on it was possible for us to still
set up a call without an audio stream if an offer was received
with required SRTP.

This change makes it so this scenario will now fail with a 488
response.

ASTERISK-26575

Change-Id: I7d14187037681f48879bd20319ac79d0877318f3

Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.

Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc

* Don't hold the req_wrapper lock too long in endpt_send_request().  We
could block the PJSIP monitor thread if the timeout timer expires.
sip_get_tpselector_from_endpoint() does a sorcery access that could take
awhile accessing a database.  pjsip_endpt_send_request() might take awhile
if selecting a transport.

* Shorten the time that the req_wrapper lock is held in the callback
functions.

* Simplify endpt_send_request() req_wrapper->timeout code.

* Removed some redundant req_wrapper->timeout_timer->id assignments.

Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9

Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b

Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94

Correct typo of end-pints to end-points
Re-wrap session timer parameter docs to max 80 chars wide; this
eases reading on terminals with lower resolution, commonly the case
for those with visual impairments.

ASTERISK-26573

Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b
Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>

This change fixes the SIP resolver such that if an IPv6 transport
is explicitly used it will resolve NAPTR, SRV, and AAAA records.

You can explicitly use one by specifying it on an endpoint.

ASTERISK-26571

Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36

ASTERISK-26558

Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e

This reverts commit f073f648.
Multiple testsuite failures were detected after the fact.

Change-Id: I968c380418bf65c7166f6ecff30fe8e247ea6682

This reverts commit 28926d1c.
Multiple testsuite failures were detected after the fact.

Change-Id: I8d4f5ccbb421a351d616254844ae7e5a31053edb

This reverts commit afef1b8e.
Multiple testsuite failures were detected after the fact.

Change-Id: Ib4cb0c0a6475681ce817f71b4050be25640ab67f

This reverts commit 39220230.

Multiple testsuite issues were discovered after the fact.

Change-Id: I848c4196dca2994b1a368087004326ea354cff95

2 of the sanitizers didn't have default values so in systems that
don't support sanitizers menuselect would spit out warnings.  They
were harmless but confusing.  They've now been set to "0".

Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58

sets the variable ABANDONED to TRUE if the call was not answered.

ASTERISK-26558

Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3