fopencookie/funclose is a non-standard API and should not be used
in portable software. Additionally, the way FILE's fd is used in
non-blocking mode is undefined behaviour and cannot be relied on.

This introduces internal abstraction for io streams, that allows
implementing the desired virtualization of read/write operations
with necessary timeout handling.

ASTERISK-24515 #close
ASTERISK-24517 #close

Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85

This reverts commit 93332cb1.

Unfortunately, the aforementioned commit caused a regression (incoming calls
would eventually disconnect). Thus it is being removed.

ASTERISK-26523 #close
ASTERISK-25270

Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d

Added missing account to AMI event of sip show peers

ASTERISK-26176 #close

Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482

Given the scenario where multiple channels are dialed using Dial()
but the caller is picked up using PickupChan() all outgoing channels
except the channel specified to PickupChan() would be marked
as ringing until the call had been hung up.

When using the PickupChan application the channel executing the
application is swapped into place of another channel. As part
of this process the channel is answered. The Dial application
has explicit logic which checks if the channel is answered,
cancels all other outgoing channels, and bridges. This logic is
different than the normal logic that is executed when an outgoing
channel is answered. This different logic failed to publish dial
events stating that the other outgoing channels had been canceled.
As a result references to the outgoing channels were held onto by
the dial masquerade process until the call had been ended and
the channels had gone away. This would result in the channels
appearing in the "core show channels" list despite not being present
anymore and would also result in incorrect device state.

This change makes it so that this logic also publishes
dial events stating that the other outgoing channels have been
canceled.

ASTERISK-26549

Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f

Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.

ASTERISK-26311 #close

Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964

PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
patch.

The patch below fixes a write to freed memory under cartain DNS lookup
conditions.

0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch

ASTERISK-26516
Reported by:  Richard Mudgett

Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5

The res_pjsip_sdp_rtp module did not restrict the number of
formats added to a media stream in the SDP to the defined
limit. If allow=all was used with additional loaded codecs this
could result in the next media stream being overwritten some.

This change restricts the module to limit it to the defined
maximum and also increases the maximum in our bundled pjproject.

ASTERISK-26541 #close

Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8

codecs.conf.sample was missing codec opus's configuration options, descriptions,
and examples. This patch adds the configuration options and examples to
codecs.conf.sample that can be used with codec_opus.

ASTERISK-26538 #close

Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b

This patch adds three new CLI commands:
 - ari show apps: list the registered ARI applications
 - ari show app: show detailed information about an ARI application
 - ari set debug: dump events being sent to an ARI application

Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.

ASTERISK-26488 #close

Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5

If in sip.conf (general section) set option register_retry_403=no,
the command "sip show settings" return value:
Outbound reg. retry 403:0
If in sip.conf (general section) set option register_retry_403=yes,
the command "sip show settings" return value:
Outbound reg. retry 403:-1

* In static char "sip show settings" for "Outbound.reg. retry 403"
option use AST_CLI_YESNO

ASTERISK-26476 #close

Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9

ASTERISK-25070

Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814

PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
define it to a constant. It is indeed not safe to assume there won't be
longer paths and Asterisk generally does err safely on such cases.

So even for HURD we'll just pretend PATH_MAX is 4096.

ASTERISK-25070 #close

Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3

In order for pjsua and its python binding to actually negotiate
audio for the testsuite tests, it needs g711 and resample.  The
pj* libraries themselves do not.  Unfortunately, pjproject relies
on a brand new libresample that most distros don't ship so we need
to use the libresample already bundled with pjproject.  Only the pjsua
executable and the _pjsua.so python library are linked with it so it
shouldn't interfere with asterisk itself.

Also it was pointed out that apply_patches couldn't handle multiple
patches that depended on each other during the dry-run, so the
dry-run was removed.

Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098

The NewConnectedLine event has been added by commit fe7671fe, but the
documentation was missing.

ASTERISK-26537 #close

Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6

Headers declare that memcpy does not accept NULL argument for the first
two parameters.  Add a conditional block to prevent memcpy and ast_free
from running on vectors with NULL element array.

ASTERISK-26526 #close

Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71

Every ao2 object contains storage for a private variable data_size,
though the value is never read if AO2_DEBUG is disabled.  This change
makes the variable conditional, reducing memory usage.

ASTERISK-26524 #close

Change-Id: If859929e507676ebc58b0f84247a4231e11da07f

PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
patch.

The patches below fix the DNS lookup race condition crash caused by
attempting to send the same message twice for the single DNS lookup.

0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch
0006-r5473-svn-backport-Fix-pending-query.patch

The patch below removes a cached DNS response from the hash table when
another thread is referencing the old entry.  The table still contained
the entry when it was destroyed which can result in inexplicable crashes.

0006-r5475-svn-backport-Remove-DNS-cache-entry.patch

ASTERISK-26344 #close
Reported by: Ian Gilmour

ASTERISK-26387 #close
Reported by: Harley Peters

Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4

main/Makefile includes third-party/pjproject/build.mak but
doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak"
evaluates to "/version.mak".  Fix is to set PJDIR in main/Makefile
before the include.

Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604

While publishing device state between multiple instances of Asterisk,
a crash will sporadically occur under high CPS which looks to be a
race condition operating on the publisher queue.

ASTERISK-26506

Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6

It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded.  Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.

ASTERISK-26513 #close

Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21

Removing explicit transport definition for endpoints and registrations. It
isn't necessary and isn't generally advised.

ASTERISK-26514 #close

Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb