An option has been added, srv_lookups, which controls whether
SRV lookups are performed on the provided match hosts or not.
It was possible for this option to be applied after resolution
had already happened.

This change makes it so hosts are stored away, settings are read
and applied, and then resolution is done. This ensures that no
matter the ordering the srv_lookups option is in effect.

ASTERISK-26735

Change-Id: I750378cb277be0140f8c5539450270afbfc43388

Change-Id: Ie0b69a830385452042fa19e7d267c6790ec6b6be

Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1

ast_loggrabber gathers log files from customizable search patterns,
optionally converts POSIX timestamps to a readable format and
tarballs the results.

Also a few tweaks were made to ast_coredumper.

Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495
(cherry picked from commit c70915287837704090d75f181525765de7a17221)

Change-Id: I573f0343c0c63a785cd4da60d57cc9f8b9ce7f49

It was possible for a frame to be re-inserted into a jitter buffer after it
had been removed from it. A case when this happened was if a frame was read
out of the jitterbuffer, passed to the translation core, and then multiple
frames were returned from said translation core. Upon multiple frames being
returned the first is passed on, but sebsequently "chained" frames are put
back into the read queue. Thus it was possible for a frame to go back into
the jitter buffer where this would cause problems.

This patch adds a flag to frames that are inserted into the channel's read
queue after translation. The abstract jitter buffer code then checks for this
flag and ignores any frames marked as such.

Change-Id: I276c44edc9dcff61e606242f71274265c7779587

Add an application that allows tracking outbound calls
using app_queue.

ASTERISK-19862

Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e

The task processor queue reached X scheduled tasks message was originally
intended to get logged only once per task processor to prevent spamming
the log.  This is no longer necessary since high and low water thresholds
can better control when the message is logged.

It is beneficial to generate the warning each time a task processor
reaches the high water level because PJSIP stops processing new requests
while any high water alert is active.  Without this change you would have
to enable at least debug level 3 logging to know about a repeated alert
trigger.

* Made generate the warning message whenever a task is pushed into the
task processor that triggers the high water alert.

* Appended 'again' to the warning for a repeated high water alert trigger.

Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999

Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter)
always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and
"AST_RTP_STAT_STRCPY".
It should compare "combined" with "stat" not "current_stat".

ASTERISK-26710 #close
Reported-by: Aaron An
Tested-by: AaronAn

Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15

This utility allows easy manipulation of asterisk coredumps.

* Configurable search paths and patterns for existing coredumps
* Can generate a consistent coredump from the running instance
* Can dump the lock_infos table from a coredump
* Dumps backtraces to separate files...
  - thread apply 1 bt full -> <coredump>.thread1.txt
  - thread apply all bt -> <coredump>.brief.txt
  - thread apply all bt full -> <coredump>.full.txt
  - lock_infos table -> <coredump>.locks.txt
* Can tarball corefiles and optionally delete them after processing
* Can tarball results files and optionally delete them after processing
* Converts ':' in coredump and results file names '-' to facilitate
  uploading.  Jira for instance, won't accept file names with colons
  in them.

Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1].

[1] For *BSDs, the "devel/gdb" package might have to be installed to
get a recent gdb.  The utility will check all instances of gdb
it finds in $PATH and if one isn't found that can run python, it
prints a friendly error.

Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd
(cherry picked from commit cb47b4556053cd50d9102eef913671ad0306062d)

* changes:
  res_rtp_asterisk.c: Fix uninitialized memory crash.
  chan_rtp.c: Fix uninitialized memory crash.
  res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().

When MALLOC_DEBUG was specified, make was failing.  Immediately
remaking would work.  The issues was in the ordering of the make
dependencies.

Change-Id: If6030b54fc693f3179f32bfd20c6b5d5f1b3f7cd

This change implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.

This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".

ASTERISK-26693

Change-Id: I6b641e275bf96629320efa8b479737062aed82ac

Adds a new formula for SL2 and documentation

ASTERISK-26559

Change-Id: I0970c620460507cd9d45b0d43600779c8915e770

Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.

This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.

ASTERISK-26658 #close
Reported by: Jonathan R. Rose

Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30

This feature was available in the SIP channel driver chan_sip. For example,
Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not
local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect
and dial remote SIP-URIs. This change here sets the SIP destination domain of
an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well.

ASTERISK-26670 #close

Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243

After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats
but remember the joint format. Cached formats contain default parameters,
often create an empty fmtp line. However, a joint format might have passed
format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and
contain the resulting format parameters from a SDP negotiation.

ASTERISK-26691 #close

Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc

A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6.
This allowed us to control the log level better from inside Asterisk.
An unfortunate side effect of this was that the pjsua binary and
python bindings were also compiled with log level set to 6 so whenever
a testsuite test that uses pjsua runs, it spits out 6795 lines of
debug in an instant even before the test starts.  I believe this
overruns the Jenkins capture buffer and prevents the test from
properly terminating.  In turn, this results in the testsuite just
hanging until the job is killed.  It's more frequent on the higher
end agents because they can spit out the messages faster.

Unfortunately, the messages are all spit out before we have control
of the python pj.Lib instance where we can set logging levels so the
only alternative was to actually compile pjsua and _pjsua.so with an
overridden PJ_LOG_MAX_LEVEL.  Although defining a lower max level was
done in the Makefile, the define in config_site.h had to be wrapped
with "#ifndef" so the change would take effect.

Change-Id: I2af9e7d48dde1927279c586c9c725d868fe6f3ff

The CHANNEL() dialplan function implementation for PJSIP allows
querying of PJSIP specific information. This used the channel
passed in to get the PJSIP session and associated information.
It is possible for this channel to be masqueraded and end
up as a different channel type by the time the information
request is actually acted upon.

This change retrieves the PJSIP session safely and accesses
data from it (including channel). This provides a guarantee
that the session and channel will not be altered when the
request is being acted upon.

ASTERISK-26673

Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6

ASTERISK-26684 #close

Change-Id: Ifd7e401c45015119dd5e8421dbfe3afa6381744a

When "fetch_again_at_reload" is set in config, we create now
new object and thread for each reloaded calendar (with new
configuration). Old calendar should be then unlinked, so the
old thread can exit and free memory.

ASTERISK-26683

Change-Id: Ic17fba9371c5a8b26a6bc54ea4957c13a32a343e

refer_incoming_refer_request needed to look for the "r" header as well
as the "Refer-To" header.

ASTERISK-26655 #close
patches:
	refer_compact_fix.diff	submitted by JoshE (license 6075)

Change-Id: I610410a99b02427ea5db887aeb454d5f12c2259f

In native_rtp_bridge_compatible_check()

* Made one variable declaration per line.

* Extracted if test assignment to make the test easier to see.

* Made long if tests easier to see the combinatorial logic.

* Added bridge id to a couple debug messages.

Change-Id: I65bc5732aa7c9a2537f062f106fbea711cf2daad