You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri,
without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S).

ASTERISK-28936

Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a

Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an
INVITE, the Identity header is retrieved, parsing the message to verify
the signature. If any of the parsing fails,
AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this
caller ID. If verification itself fails,
AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in
the payload does not line up with the SIP signaling,
AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps
pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the
verification process.

A new config option has been added to the general section for
stir_shaken.conf. "signature_timeout" is the amount of time a signature
will be considered valid. If an INVITE is received and the amount of
time between when it was received and when it was signed is greater than
signature_timeout, verification will fail.

Some changes were also made to signing and verification. There was an
error where the whole JSON string was being signed rather than the
header combined with the payload. This has been changed to sign the
correct thing. Verification has been changed to do this as well, and the
unit tests have been updated to reflect these changes.

A couple of utility functions have also been added. One decodes a BASE64
string and returns the decoded string, doing all the length calculations
for you. The other retrieves a string value from a header in a rdata
object.

Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913

In a particular fax gateway scenario whereby it would
have to translate using the read translation path on a
channel the frame being translated would be consumed.
When the frame is in the write path it is not permitted
to free the frame as the caller expects it to continue
to exist.

This change makes it so that the frame is only consumed
on the read path where it is acceptable to free it.

ASTERISK-28900

Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d

When writing tx messages to pcap files, Asterisk is using the wrong
pointer resulting in lots of wasted space. This patch fixes it to use
the correct pointer.

ASTERISK-28932 #close

Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23

files

fwrite() does return the number of elements written and not the
number of bytes. However asterisk is currently comparing the return
value to the size of the written element what means that asterisk logs
five WARNING messages on every packet written to the pcap file.

This patch changes the code to check for the correct value, which will
always be 1.

ASTERISK-28921 #close

Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2

When replacing the user portion of the Contact URI the code
was using the ephemeral pool instead of the tdata pool. This
could cause the Contact user value to become invalid after a
period of time.

The code will now use the tdata pool which persists for the
lifetime of the message instead.

ASTERISK-28794

Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5

While asterisk is filtering out the x-ast-orig-host parameter from the
contact on response messages, it is not filtering it out from the
request URI and the to header on SIP requests (for example INVITE).

ASTERISK-28884 #close

Change-Id: Id032b33098a1befea9b243ca994184baecccc59e

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