ao2_replace() bumps the reference count of the object that is doing the
replacing, which is not what we want. We just want to drop the old ref
on the old object and update the pointer to point to the new object.

Pointed out by George Joseph in #asterisk-dev

Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d

For RTCP to work, we update the ssrc to be the one corresponding to
the native bridge while active.  However when the bridge ends we
should generate a new SSRC as the sequence numbers will not continue
from the native bridge left off.

ASTERISK-29300 #close

Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10

Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.

This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.

ASTERISK-29321

Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1

Although the dlg session count was incremented in a pjsip servant
thread, there's no guarantee that the last thread to unref this
progress object was one.  Before we decrement, we need to make
sure that this is either a servant thread or that we push the
decrement to a serializer that is one.

Because pjsip_dlg_dec_session requires the dialog lock, we don't
want to wait on the task to complete if we had to push it to a
serializer.

Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41

When registering it can be useful to see the source IP address and
port in cases where multiple devices are using the same endpoint
or when anonymous is in use.

ASTERISK-29325

Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0

When Asterisk sends a reinvite negotiating T38 faxing, it's possible a
crash can occur if the response contains a m=image and zero port. The
reinvite callback code now checks session_media to see if it is null or
not before trying to access the udptl variable on it.

ASTERISK-29305

Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9

Fixed:
* RFC 4629 does not allow the value "0" for MPI, K, and N.
* Allow value "0" for PAR.
* BPP is printed only when specified because "0" has a meaning.

New:
* Added CPCF and MaxBR.
* Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1
  Although a violation of RFC 3555 section 3, we can support that.

Changed:
* Resorts the CIFs from large to small which partly fixes ASTERISK~29267.

Change-Id: I95a650c715007b8dde11a77cb37d9c6c123a441e

When sending a SIP response to an incoming REGISTER request
we don't want to change the Contact header as it will
contain the Contacts registered to the AOR and not our own
Contact URI.

ASTERISK-29235

Change-Id: I35a0723545281dd01fcd5cae497baab58720478c

This change will check is the remote ICE session got reset or not by
checking the offered ufrag and password with session. If the remote ICE
reset session then Asterisk reset its local ufrag and password to reject
binding request with Old ufrag and Password.

ASTERISK-29266

Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb

If set_outbound_initial_authentication_credentials() fails,
handle_client_registration() bails early without creating or
sending a register message.

[set_outbound_initial_authentication_credentials() failures
can occur during the process of retrieving an oauth access
token.]

The return from handle_client_registration is ignored, so
returning an error doesn't do any good.

This is a real problem when the registration request is a
re-register, because then the registration will still be
marked 'active' despite the re-register never being sent at all.

So instead, log a warning but let the registration be created
and sent (and probably fail) and follow the normal registration
failed retry/abort logic.

ASTERISK-29315 #close

Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa

If the remote Station ID contains invalid UTF-8 characters
the asterisk fails to publish the Stasis and ReceiveFax status messages.

json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28]
1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc]
2: /usr/sbin/asterisk(ast_channel_publish_varset+0x2b) [0x57aa0b]
3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x121) [0x530641]
4: /usr/lib64/asterisk/modules/res_fax.so(+0x44fe) [0x7f27f4bff4fe]
...
stasis_channels.c: Error creating message

json.c: Error building JSON from '{s: s, s: s, s: s, s: s, s: s, s: s, s: o}': Invalid UTF-8 string.
0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28]
1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc]
2: /usr/lib64/asterisk/modules/res_fax.so(+0x5acd) [0x7f27f4c00acd]
...
res_fax.c: Error publishing ReceiveFax status message

This patch replaces the invalid UTF-8 Station IDs with an empty string.

ASTERISK-29312 #close

Change-Id: Ieb00b6ecf67db3bfca787649caa8517f29d987db

Although refer_progress_notify() always runs in the progress
serializer, the pjproject evsub module itself can cause the
subscription to be destroyed which then triggers
refer_progress_on_evsub_state() to clean it up.  In this case,
it's possible that refer_progress_notify() could get the
subscription pulled out from under it while it's trying to use
it.

At one point we tried to have refer_progress_on_evsub_state()
push the cleanup to the serializer and wait for its return before
returning to pjproject but since pjproject calls its state
callbacks with the dialog locked, this required us to unlock the
dialog while waiting for the serialized cleanup, then lock it
again before returning to pjproject. There were also still some
cases where other callers of refer_progress_notify() weren't
using the serializer and crashes were resulting.

Although all callers of refer_progress_notify() now use the
progress serializer, we decided to simplify the locking so we
didn't have to unlock and relock the dialog in
refer_progress_on_evsub_state().

Now, refer_progress_notify() holds the dialog lock for its
duration and since pjproject also holds the dialog lock while
calling refer_progress_on_evsub_state() (which does the cleanup),
there should be no more chances for the subscription to be
cleaned up while still being used to send NOTIFYs.

To be extra safe, we also now increment the session count on
the dialog when we create a progress object and decrement
the count when the progress is destroyed.

ASTERISK-29313

Change-Id: I97a8bb01771a3c85345649b8124507f7622a8480

For some RTCP packet types the report count is actually the packet's subtype.
This was not being reflected in the packet debug output.

This patch makes it so for some RTCP packet types a "Packet Subtype" is
now output in the debug replacing the "Reception reports" (i.e count).

Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8

When PJSIP receives a re-INVITE without an SDP offer the INVITE
session library will first call the on_create_offer callback and
if unavailable then use the active negotiated SDP as the offer.

In some cases this would result in a different SDP then was
previously used without an incremented SDP version number. The two
known cases are:

1. Sending an initial INVITE with a set of codecs and having the
remote side answer with a subset. The active negotiated SDP would
have the pruned list but would not have an incremented SDP version
number.

2. Using re-INVITE for unhold. We would modify the active negotiated
SDP but would not increment the SDP version.

To solve these, and potential other unknown cases, the on_create_offer
callback has now been implemented which produces a fresh offer with
incremented SDP version number. This better fits within the model
provided by the INVITE session library.

ASTERISK-28452

Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1

Also improve the in-process documentation to clarify that the value is
initialised from the DSN and not default false, but that the DSN's value
is default false if unset.

ASTERISK-29311 #close

Change-Id: I46e2379f7b0656034442bce77cb37ccd4e61098d
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Removes an unnecessary check for the conditional that compares the
stream topologies to see if they are equal to suppress re-invites. This
was a problem when a Digium phone received an INVITE that offered codecs
different than what it supported, causing Asterisk to send the
re-invite.

ASTERISK-29303

Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63

Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function.

ASTERISK-29293 #close

Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5

When an endpoint requests to re-negotiate for fax and the incoming
re-invite is received prior to Asterisk sending out the 200 OK for
the initial invite the re-invite gets delayed. When Asterisk does
finally send the re-inivite the SDP includes streams for both audio
and T.38.

This happens because when the pending topology and active topologies
differ (pending stream is not in the active) in the delayed scenario
the pending stream is appended to the active topology. However, in
the fax case the pending stream should replace the active.

This patch makes it so when a delay occurs during fax negotiation,
to or from, the audio stream is replaced by the T.38 stream, or vice
versa instead of being appended.

Further when Asterisk sent the re-invite with both audio and T.38,
and the endpoint responded with a declined T.38 stream then Asterisk
would crash when attempting to change the T.38 state.

This patch also puts in a check that ensures the media state has a
valid fax session (associated udptl object) before changing the
T.38 state internally.

ASTERISK-29203 #close

Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09

Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.

ASTERISK-29260
Reported by: Alexander Traud

Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458

New responses sent within a PJSIP sessions are based on those that were
sent before. Therefore, adding/modifying a header once causes it to be
sent on all responses that follow.

Sending 181 Call Is Being Forwarded many times first adds "histinfo"
duplicated more and more, and eventually overflows past the array
boundary.

This commit adds a check preventing adding "histinfo" more than once,
and skipping it if there is no more space in the header.

Similar overflow situations can also occur in res_pjsip_path and
res_pjsip_outbound_registration so those were also modified to
check the bounds and suppress duplicate Supported values.

ASTERISK-29227
Reported by: Ivan Poddubny

Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322

ASTERISK-29205 #close

Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea

refer_progress_notify wasn't always being called from the progress
serializer.  This could allow clearing notification->progress->sub
in one thread while another was trying to use it.

* Instances where refer_progress_notify was being called in-line,
  have been changed to use ast_sip_push_task().

Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e

if From number contain * or # asterisk will not add user=phone

Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY
this is a problem when you want to send call to ISUP
as they will disregard the From header and either replace From with anonymous or with p-asserted-identity

ASTERISK-29261
Reported by: Mark Petersen
Tested by: Mark Petersen

Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4

Provided a support of variuos URL-schemes for res_musiconhold,
registered by ast_bucket_scheme_register().

ASTERISK-29262 #close

Change-Id: If0ea8697587353dce358a70035d82649fd4632b6

Otherwise, Clang 10 warned because of logical-not-parentheses.

Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79

ASTERISK-29248

Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556

The last argument to ast_copy_string() is the buffer size, not the
number of characters, so we add 1 to avoid stamping out the final \n
in the persisted SUBSCRIBE message.

Change-Id: I019b78942836f57965299af15d173911fcead5b2

function ast_sip_session_media_state_add.

Check ast_media_type matches when a ast_sip_session_media is found
otherwise when transitioning from say image to audio, the wrong
session is returned in the first if statement.

ASTERISK-29220 #close

Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d

When both a tech subscription and an endpoint subscription exist for a given
endpoint, TextMessageReceived events are dispatched to the tech subscription
only.

ASTERISK-29229

Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8

session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.

This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.

ASTERISK-29240

Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755

This reverts commit 2fe76dd8.

Reason for revert: Too many issues reported.  Need to research and correct.

ASTERISK-29230
ASTERISK-29231
Reported by: Michael Maier

Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb

Segfault occurs during outbound UDP registration when all
transport states are being iterated over. The transport object
in the transport is accessed, but flow transports have a NULL
transport object.

Modify to not iterate over any flow transport

ASTERISK-29210 #close

Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a

Add a check to see if the URI is a Tel URI and prevent crashing on
trying to retrieve the reason parameter.

ASTERISK-29191
ASTERISK-29219

Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
(cherry picked from commit a7aea71e)

AST_VECTOR_SIZE() returns a size_t.  This is not always equivalent to an
unsigned long on all machines.

Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338

Added NULL pointer check to prevent Asterisk crash.

ASTERISK-29215

Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95

The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.

    https://wiki.asterisk.org/wiki/x/Xc5uAg

However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.

For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.

ASTERISK-28883 #close

Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f

Fixed wrong null object handle in
/channels/<channel_id>/play request handler.

ASTERISK-29188

Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe

In rewrite_uri asterisk was not making deep copies of strings when
changing the uri. This was in some cases causing garbage in the route
header and in other cases even crashing asterisk when receiving a
message with a record-route header set. Thanks to Ralf Kubis for
pointing out why this happens. A similar problem was found in
res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
to avoid garbage in CANCEL messages.

ASTERISK-29024 #close

Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b

ASTERISK-29211 #close

Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877

This adds support for both Digium and Sangoma user agent strings
for the Sangoma specific body supplement.

Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482