CEL wrongly assumed that a channel would only have a single dial
event on it. This is incorrect. Particularly in a queue each
call attempt to a member will result in a dial event, adding
a new dial status in CEL without removing the old one. This
would cause the container to grow with only one dial status
being removed when the channel went away. The other dial status
entries would remain leaking memory.

This change fixes the memory leak by ensuring that only one dial
status will only ever exist for each channel.

The behavior during the scenario where multiple events are received
has also been improved. For failure cases the first failure will
be the dial status. If an answer dial status is received, though,
it will take priority and the dial status for the channel will be
answer.

Memory usage has also been decreased by storing the minimal
amount of information and the code has been cleaned up slightly.

ASTERISK-25262 #close

Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe

Scenario: Caller blonde transfer
Bob calls Charlie who answers.
Bob puts Charlie on hold and calls Alice.
Before Alice answers, Bob transfers Charlie to Alice.

Charlie's channel triggers an assert because he gets an "ANSWERED"
event even though he never dialed anything. With recent changes to dial
events, this is now a valid scenario so the assert needed to be removed.

ASTERISK-26103 #close

Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2

bridge_native_rtp can call into an RTP-capable channel driver in order
for the driver to update information about who the channel is
communicating with. For SIP channel drivers, this means deactivating
RTCP and sending a reinvite so that the endpoints can communicate
directly.

bridge_native_rtp does the right thing and has the channel locked when
calling into the channel driver. chan_pjsip can't alter session
properties in this thread, though. chan_pjsip queues a task on the
session serializer in order to update properties there.

The problem is that this queued task was not locking the channel. This
meant that the queued task could attempt to deactivate RTCP at the same
time that the channel thread was attempting to process an incoming RTCP
packet. This could lead to a crash.

This patch fixes the issue by locking the channel in the queued task
when altering RTP properties.

ASTERISK-26092 #close
Reported by Niklas Larsson

Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159

A change to glibc 2.22 changed the order of the sockadddr_storage
members which caused the places where we do an initialization of
ast_sockaddr with '{ { 0, 0, } }' to fail compilation.  Those
initializers (which we shouldn't have been using anyway) have been
replaced with memsets.

Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4

Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb

POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.

Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc

A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not
loaded and does not have a configuration file. Previously when this
occurred, checks were put in to see if the configuration was loaded
successfully. While this is a good idea - and has been added to the
offending function in res_hep - the reality is res_hep_pjsip and
res_hep_rtcp have no business running if res_hep isn't also running.

As such, this patch also adds a function to res_hep that returns whether
or not it successfully loaded. Oddly enough, ast_module_check returns
"everything is peachy" even if a module declined its load - so it cannot
be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this
function to see if they should continue to load; if it fails, they
decline their load as well.

ASTERISK-26096 #close

Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea

This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compactheaders=yes via the file sip.conf.

ASTERISK-25578 #close

Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044

Testing has shown that our usage of UnixODBC is problematic
due to bugs within UnixODBC itself as well as the heavy weight
cost of connecting and disconnecting database connections, even
when pooling is enabled.

For users of UnixODBC 2.3.1 and earlier crashes would occur due
to insufficient protection of the disconnect operation. This was
fixed in UnixODBC 2.3.2 and above.

For users of UnixODBC 2.3.3 and higher a slow-down would occur
under heavy database use due to repeated connection establishment.
A regression is present where on each connection the database
configuration is cached again, with the cache growing out of
control.

The connection pool implementation present in this change helps
to mitigate these issues by reducing how much we connect and
disconnect database connections. We also solve the issue of
crashes under UnixODBC 2.3.1 by defaulting the maximum number of
connections to 1, returning us to the previous working behavior.
For users who may have a fixed version the maximum concurrent
connection limit can be increased helping with performance.

The connection pool works by keeping a list of active connections.
If the connection limit has not been reached a new connection is
established. If the connection limit has been reached then the
request waits until a connection becomes available before
continuing.

ASTERISK-26074 #close
ASTERISK-26054 #close

Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff

Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.

This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.

The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.

ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson

Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665

Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore.
Therefore, the symbol RAND_bytes is used instead of crypto_get_random.

ASTERISK-24436 #close

Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96

In several internal library projects, the files are archived with the help of
'ar cr'. Only the projects editline and the Objective Open H.323 stack
implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms
changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier
ignored since `D' is the default (see `U')". For consistency and to avoid this
message all projects use 'ar cr' now.

ASTERISK-26091 #close

Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40

Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])

Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.

Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9

ASTERISK-25629 #close

Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006

Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.

Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.

ASTERISK-26059 #close

Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2

If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192.  While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.

In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.

To facilitate determination of format names, the format name has been
added to "core show codecs".

ASTERISK-26070 #close

Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b

These flags are non-portable GNU extensions. Make their use
optional. This fixes complication error on e.g. musl c-library
based systems.

Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3

Resolver state is not part of res_search API. This fixes
compilation error:

dns.c:261:8: error: too many arguments to function 'res_search'
  ret = res_search(&dns_state,

Change-Id: Ia600a58557040df83f744da3dde23225293845a5

POSIX defines poll.h, sys/poll.h should not be used at is c-library
internal header which may or may not exist. Notable in musl it
generates warning of being incorrect. And add explict include of
sys/cdefs.h where needed.

Change-Id: I142930df53fe7585a06b854b6faddc5301e024be

Add Uptime and LastReload to event FullyBooted.

ASTERISK-26058 #close
Reported by: Niklas Larsson

Change-Id: I909b330801c0990d78df9b272ab0adc95aecb15e

The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting
to use UniqueConstraint and failing. It was not imported and after
importing it also continued to fail.

I've changed the script to use the explicit name of the constraint
instead.

Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9

The stringfields refactor to allow adding stringfields to the end of a
structure (f6f4cf45) exposed some
incomplete cleanup code by some stringfield users.

The most noticeable leaker is the logging system where there is a leak for
every log message generated.

ASTERISK-26078 #close
Reported by:  Etienne Lessard
Patches:
      jira_asterisk_26078_v13.patch (license #5621) patch uploaded
      by Richard Mudgett

Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782