Before this change, make failed with the error
Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH
when CFLAGS were supplied to the configure script. This was introduced with
<https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when
CFLAGS were supplied. Those who need different -march= values, please, go for
./configure
make menuselect.makeopts or make menuselect
./menuselect/menuselect --disable BUILD_NATIVE

ASTERISK-25289 #close

Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc

Thanks to ibercom for pointing out a memory leak that was missed
in the earlier patch for the issue.

ASTERISK-26119
Reported by: Alexei Gradinari

Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71

Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being
recorded to the refs log for the node being replaced.  This prevents
logging of those unrefs since they would produce errors in
refcounter.py.

ASTERISK-26181 #close

Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4

Some T.38 implementations may send another re-invite after the initial
one which adds additional negotiation details (such as the max bitrate).
Currently this will fail when passthrough is being done in chan_sip as we
do nothing if T.38 is already active.

Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
scenario so this change adds support for it to chan_sip and res_pjsip_t38.
If a request to negotiate is received while T.38 is already enabled a
new re-INVITE is sent and negotiation is done again.

ASTERISK-26179 #close

Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c

When using TCP transport with chan_pjsip, the TCP_NODELAY
option value was allocated on the stack, then passed as a
pointer to the tcp transport configuration structure, and
later re-used on subsequently created sockets when it was
no longer valid.  This patch changes the allocation to be
a static.

ASTERISK-26180 #close
Reported by: Scott Griepentrog

Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0

When an answer SDP is invalid we were disconnecting the outgoing call and
sending two BYE requests.  The first BYE was sent by PJPROJECT because of
the invalid SDP answer.  The second BYE was sent by Asterisk because it
thought the canceled call was the result of the RFC5407 section 3.1.2 race
condition.

* Made not send the BYE on a canceled session if the SDP negotiation is
incomplete because PJPROJECT has already sent a BYE for the failed
negotiation.

ASTERISK-25772 #close
Reported by:  Dmitriy Serov

Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836

When an incoming call defers SDP negotiation and then sends us an invalid
SDP in the ACK, we need to send a BYE to disconnect the call.  In this
case SDP negotiation has failed and we don't have valid media streams
negotiated.

ASTERISK-25772

Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8

Registering the PJMEDIA error codes allows errors found when parsing an
incoming SDP to be easier to figure out.

"Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
is much easier to understand than "Unknown error 220030".

ASTERISK-25772

Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0

Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa

pjsip_inv_end_session() is documented as being able to return the
passed in tdata parameter set to NULL on success.

Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047

Found as a result of the testsuite tests/callparking test crashing.

Several calls to ast_get_chan_featuremap_config() and
ast_get_chan_features_xfer_config() did not lock the channel before
calling so the channel's datastore list was accessed without the lock's
protection.  Apparently another thread deleted a datastore on the
channel's list while the crashing thread was walking the list.  Crash at
0xdeaddead due to MALLOC_DEBUG's memory filler value as a result.

* Add missing channel locks to calls that were not already protected
as the doxygen for those calls indicates.

Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1

There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK
from getting set when using an external pjproject.

ASTERISK-26099 #close
Reported-by: Ross Beer

Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae

Following the principle of least surprise, we should not be sending
massive numbers of PJSIP and RTCP HEP packets out into the ether to some
only-slightly-random IP address. Having 'enabled' set to 'no' in the
sample configuration file should prevent this from happening for those
who run 'make samples'.

ASTERISK-26159 #close

Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1

When negotiating ICE candidates with WebRTC capable endpoints, many
networks will result in a browser offering ICE candidates that exceeds
the default number of max candidates, 16. This patch bumps the max
candidates to 32, with the max checks at twice the number of candidates.
In practice, this has shown to be sufficient for browser/WebRTC
negotiation.

Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5

Adding format_name even to the end of ast_codec caused issued with
binary codec modules because the pointer would be garbage in asterisk
when they registered.  So, the ast_codec structure was reverted and an
internal_ast_codec structure was created just for use in codec.c.  A new
internal-only API was also added (__ast_codec_register_with_format) so
that codec_builtin could register codecs with the format_name in a
separate parameter rather than in the ast_codec structure.

ASTERISK-26144 #close
Reported-by: Alexei Gradinari

Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba

gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.

ASTERISK-26157 #close

Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e

This patch removes the following modules:
 - pbx_functions: It never existed.
 - res_pjsip_log_forwarder: It no longer exists.
 - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
                  aren't going to be installing HOMER
 - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
                  loaded, and we aren't configured to make use of the
                  module

Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5

This change removes hardcoded SDP parsing and generation for
Siren7 and Siren14 from chan_sip and moves it to format attribute
modules so it can also be used by chan_pjsip.

With this the fmtp lines for both are added with the bitrate
information.

ASTERISK-26021

Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037

Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C
but requires ANSI C anyway.

ASTERISK-26046

Change-Id: I914c014385e1862102d90fe7650621def78db02e

fax_v21_session_new created a session details object but only released
the allocation reference during error conditions.  fax_session_new adds
it's own reference to details if needed so the caller is always
responsible for cleaning it's own reference.

ASTERISK-26141 #close

Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88