When a configuration file in Asterisk is loaded
information about it is stored such that on a
reload it is not reloaded if nothing has changed.
This can be problematic when an error exists in
a configuration file in PJSIP since the error
will be output at start and not subsequently on
reload if the file is unchanged.

This change makes it so that if an error is
encountered when res_sorcery_config is loading
a configuration file a reload will always read
in the configuration file, allowing the error
to be seen easier.

Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed

The flags of a previous selection could have been set within the
object 'srtp', for example, when the previous selection returned
failure after setting just 'some' flags. Now, not to clutter the
code, all possible flags are cleared first, and then the selected
flags are set as before.

ASTERISK-28903

Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee

Added two unit tests, one for signing and another for verifying.
stir_shaken_sign checks to make sure that all the required parameters
are passed in and then signs the actual payload. If a signature is
produced and a payload returned as a result, the test passes.
stir_shaken_verify takes the signature from a signed payload to verify.
This unit test also verifies that all the required information is passed
in, and then attempts to verify the signature. If verification is
successful and a payload is returned, the test passes.

Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13

The PJSIP packet logger now has the following CLI commands:

pjsip set logger pcap <filename>

When used this will create a pcap file containing the incoming
and outgoing SIP packets, in unencrypted form.

pjsip set logger verbose <on / off>

This allows you to toggle logging to verbose on and off.

pjsip set logger host <IP/subnet mask> add

This allows you to add an additional IP address or subnet
mask to logging, allowing you to log multiple instead of
just a single IP address or all traffic.

The normal "pjsip set logger host" CLI command has also been
expanded to allow subnet masks as well.

ASTERISK-28895

Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e

A warning was triggered that there may be a problem regarding file
extension (which is correct and should not be set anyway). The warning
also appeared if there was dot within the path itself.

E.g.
[sales-queue-hold]
mode=playlist
entry=/var/www/domain.tld/moh/funky_music

The music played correctly but you get a warning message.

Now there will be a check if the position of a potential dot character
is after the last position of a slash character. This dot charachter
will be treated as a extension naming. Dots within the path then ignored.

ASTERISK-28892
Reported-By: Nicholas John Koch

Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e

Added freeifaddrs() for memory releasing.

ASTERISK-28904

Change-Id: I109403866e85a30659351946903a679de9727a8f

This change adds the same variable functionality that
is available for originating a channel to the create
call. Now when creating a channel you can specify
dialplan variables to set instead of having to do another
API call.

ASTERISK-28896

Change-Id: If13997ba818136d7c070585504fc4164378aa992

1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups
   when an IPV6 transport type has been requested.

2. Rename all occurrences of pjsip_transport_get_type_name to
   pjsip_transport_get_type_desc. This ensures that the log/debug info
   shows whether the transport is IPv6 or IPv4.

3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport
   types. This results in invalid values. Use a bitwise or instead.

ASTERISK-26780
Patches:
    pjsip_resolver.c uploaded by Peter Sokolov (License #7070)

Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58

Adds the "STIR_SHAKEN" dialplan function and an API call to add a
STIR_SHAKEN verification result to a channel. This information will be
held in a datastore on the channel that can later be queried through the
"STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results
including identity, attestation, and verify_result. Here are some
examples:

STIR_SHAKEN(count)
STIR_SHAKEN(0, identity)
STIR_SHAKEN(1, attestation)
STIR_SHAKEN(2, verify_result)

Getting the count can be used to iterate through the results and pull
information by specifying the index and the field you want to retrieve.

Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82

Changed source and destination address fields in struct
pjsip_history_entry so that they are long enough to hold an IPv6
address.

ASTERISK-28854

Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e

Change-Id: Ice5d92ecea2f1101c80487484f48ef98be2f1824

There are a lot of moving parts in this patch, but the focus of it is on
the verification of the signature using a public key located at the
public key URL provided in the JSON payload. First, we check the
database to see if we have already downloaded the key. If so, check to
see if it has expired. If it has, redownload from the URL. If we don't
have an entry in the database, just go ahead and download the public
key. The expiration is tested each time we download the file. After
that, read the public key from the file and use it to verify the
signature. All sanity checking is done when the payload is first
received, so the verification is complete once this point is reached.

The XML has also been added since a new config option was added to
general (curl_timeout). The maximum amount of time to wait for a
download can be configured through this option, with a low value by
default.

Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c

Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined.

ASTERISK-28852 #close

Change-Id: I381718893b80599ab8635f2b594a10c1000d595e

Some places in Asterisk did not treat the formats on a stream
as immutable when they are.

The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.

An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.

ASTERISK-28846

Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe

Fixed it to copy the entire string from the requested endpoint body except tech-prefix.

ASTERISK-28847

Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25

This change fixes a few re-negotiation issues
uncovered with fax.

1. The fax support uses its own mechanism for
re-negotiation by conveying T.38 information in
its own frames. The new support for re-negotiating
when adding/removing/changing streams was also
being triggered for this causing multiple re-INVITEs.
The new support will no longer trigger when
transitioning between fax.

2. In off-nominal re-negotiation cases it was
possible for some state information to be left
over and used by the next re-negotiation. This
is now cleared.

ASTERISK-28811
ASTERISK-28839

Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2

ASTERISK-28835

Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26

ASTERISK-28838

Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25

MODULEINFO is checked while buidling/linking the module.
AST_MODULE_INFO is checked while loading/running the module.

ASTERISK-28838

Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042

ASTERISK-28838

Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52

ASTERISK-28838

Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074

When the receive buffer was flushed by a received packet while it
already contained a packet with the same sequence number, Asterisk
never left the while loop which tried to order the packets.

This change makes it so if the packet is in the receive buffer it
is retrieved and freed allowing the buffer to empty.

ASTERISK-28827

Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3

When the ast_data_buffer_put rejects to add a packet, for example because
the buffer already contains a packet with the same sequence number, the
payload will never be freed, resulting in a memory leak.

The data buffer will now return an error if this situation occurs
allowing the caller to free the payload. The res_rtp_asterisk module
has also been updated to do this.

ASTERISK-28826

Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1

When the first DTMF receive in RF2833 codec have TimeStamp at 0
is not processed.

ASTERISK-28812

Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504

Change-Id: Idba5151a3079f9dcc0076d635422c5df5845114f

Change-Id: I40c014c2cb88e943cf6f1b99a08c7c885e855b3a

By using pjproject to give us a list of candidates, and then filtering,
if the host has >32 addresses configured, then it is possible that we
end up filtering out all 32 of those, and ending up with no candidates
at all.  Instead, get getifaddrs (which pjsip is using underlying
anyway) to retrieve all local addresses, and iterate those, adding the
first 32 addresses not excluded by the ICE ACL.

In our setup at any point in time We've got between 6 and 328 addresses
on any given system.  The lower limit is the lower limit but the upper
limit is growing on a near daily basis currently.

Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b

Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec

ASTERISK-28809

Change-Id: I269731715347c8e5ef7db1b6ffd3f8d15fc04be4

This reverts commit a3a2fbae.

Reason for revert: There is a lot of code that relies on the broken
behavior that this fixes.

Change-Id: I410c395a0168acbdaf89e616e3cb5e1312d190cb

When an AOR is modified endpoints are updated that reference
the AOR so they can start receiving updates and reflect the
correct state. If this is the case then we shouldn't change
the endpoint to be offline if it does not reference the AOR
but instead only when the endpoint is completely updated for
all its AORs.

ASTERISK-28056
patches:
  pjsip_options-aor.diff submitted by jhord (license 6978)

Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164

This unit test runs through combinations of...
	* Local codecs
	* Remote Codecs
	* Codec Preference
	* Incoming/Outgoing

A few new APIs were created to make it easier to test
the functionality but didn't result in any actual
functional change.

ASTERISK_28777

Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164

Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437

This change provides functions that take in a JSON payload, verify that
the contents contain all the mandatory fields and required values (if
any), and signs the payload with the private key. Four fields are added
to the payload: x5u, attest, iat, and origid. As of now, these are just
placeholder values that will be set to actual values once the logic is
implemented for what to do when an actual payload is received, but the
functions to add these values have all been implemented and are ready to
use. Upon successful signing and the addition of those four values, a
ast_stir_shaken_payload is returned, containing other useful information
such as the algorithm and signature.

Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393

RFC5621 requires any content type with a Content-Disposition
with handling=required to be rejected with a 415 response

ASTERISK-28782 #close

Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4

When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.

This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.

ASTERISK-28787

Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f

ASTERISK-28790

Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99

This commit sets up some of the initial framework for the module and
adds a way to read the private key from the specified file, which will
then be appended to the certificate object. This works fine for now, but
eventually some other structure will likely need to be used to store all
this information. Similarly, the caller_id_number is specified on the
certificate config object, but in the end we will want that information
to be tied to the certificate itself and read it from there.

A method has been added that will retrieve the private key associated
with the caller_id_number passed in. Tab completion for certificates and
stores has also been added.

Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f

When examining a stream to determine hold/unhold information we
only care about the default audio stream. Other streams aren't
used for hold/unhold.

ASTERISK-28784

Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4