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

In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.

This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.

ASTERISK-29022

Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9

By default libcurl does not follow redirects, so we explicitly enable
it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl
will follow up to CURLOPT_MAXREDIRS redirects, which by default is
configured to be unlimited.

This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If
we determine at some point that this needs to be increased on
configurable it is a trivial change.

ASTERISK-29173 #close

Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30

the 'J' is missing in module description.
"PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk"

ASTERISK-29175 #close

Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a

In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO.

ASTERISK-29148

Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2

RFC 3261 says that the Accept-Encoding header should be present
in an options response. Permitted values according to RFC 2616
are only compression algorithms like gzip or the default identity
encoding. Therefore "text/plain" is not a correct value here.
As long as the header is hard coded, it should be set to "identity".

Without this fix an Alcatel OmniPCX periodically logs warnings like
"[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed"
on a SIP Trunk.

ASTERISK-29165 #close

Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840

* Instead of using the pjproject timer heap, we now use our own
  pjsip_scheduler.  This allows us to more easily debug and allows us to
  see times in "pjsip show/list registrations" as well as being able to
  see the registrations in "pjsip show scheduled_tasks".

* Added the last registration time, registration interval, and the next
  registration time to the CLI output.

* Removed calls to pjsip_regc_info() except where absolutely necessary.
  Most of the calls were just to get the server and client URIs for log
  messages so we now just save them on the client_state object when we
  create it.

* Added log messages where needed and updated most of the existong ones
  to include the registration object name at the start of the message.

Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2

* Added a ONESHOT type that never reschedules.

* Added "like" capability to "pjsip show scheduled_tasks" so you can do
  the following:

  CLI> pjsip show scheduled_tasks like outreg
  PJSIP Scheduled Tasks:

  Task Name                                     Interval  Times Run ...
  ============================================= ========= ========= ...
  pjsip/outreg/testtrunk-reg-0-00000074            50.000   oneshot ...
  pjsip/outreg/voipms-reg-0-00000073              110.000   oneshot ...

* Fixed incorrect display of "Next Start".

* Compacted the displays of times in the CLI.

* Added two new functions (ast_sip_sched_task_get_times2,
  ast_sip_sched_task_get_times_by_name2) that retrieve the interval,
  next start time, and next run time in addition to the times already
  returned by ast_sip_sched_task_get_times().

Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3

ASTERISK-28933
Reported-by: Walter Doekes

Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f

This avoids the inclusion of the OpenSSL headers in the public header,
which avoids one external library dependency in res_pjsip_stir_shaken.

Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4

pjproject returns the dialog locked and with a reference. However,
in Asterisk the method that handles this decrements the reference
and removes the lock prior to returning. This makes it possible,
under some circumstances, for another thread to free said dialog
before the thread that created it attempts to use it again. Of
course when the thread that created it tries to use a freed dialog
a crash can occur.

This patch makes it so Asterisk now returns the newly created
dialog both locked, and with an added reference. This allows the
caller to de-reference, and unlock the dialog when it is safe to
do so.

In the case of a new SIP Invite the lock, and reference are now
held for the entirety of the new invite handling process.
Otherwise it's possible for the dialog, or its dependent objects,
like the transaction, to disappear. For example if there is a TCP
transport error.

ASTERISK-29057 #close

Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
(cherry picked from commit 6baa4b53)

If Asterisk sends out and INVITE and receives a challenge with a
different nonce value each time, it will continually send out INVITEs,
even if the call is hung up. The endpoint must be configured for
outbound authentication in order for this to occur. A limit has been set
on outbound INVITEs so that, once reached, Asterisk will stop sending
INVITEs and the transaction will terminate.

ASTERISK-29013

Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7

ASTERISK-29146

Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41

This patch initializes a couple of local variables to some default values.
Interestingly, in the 'pj_status_t dlg_status' case the value not being
initialized caused memory to grow, and not be recovered, in the off nominal
path (at least on my machine).

Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7

Commit 44bb0858 ("debugging: Add enough
to choke a mule") accidentally removed calls to
ast_sip_message_apply_transport when it was attempting to just add
debugging code.

The kiss of death was saying that there were no functional changes in
the commit comment.

This makes outbound calls that use the 'flow' transport mechanism fail,
since this call is used to relay headers into the outbound INVITE
requests.

ASTERISK-29124 #close

Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a

This changes the outgoing offer call preference
default option to match the behavior of previous
versions of Asterisk.

The additional advanced codec negotiation options
have also been removed from the sample configuration
and marked as reserved for future functionality in
XML documentation.

The codec preference options have also been fixed to
enforce local codec configuration.

ASTERISK-29109

Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2

When handling a send_message request to a non-existing endpoint, the response's
body is overriden and not properly freed.

ASTERISK-29108

Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8

Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:

  dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
  stun, stun_packet

These debug categories can be enable/disable via an Asterisk CLI command.

While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).

ASTERISK-29054 #close

Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849

In the event that the desired extension already exists,
ast_add_extension2_lockopt() will free the 'data' it is passed before
returning an error, so we should not be freeing it ourselves.

Additionally, there were two places where ast_add_extension2_lockopt()
could return an error without also freeing the 'data' pointer, so we
add that.

ASTERISK-29097 #close

Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae

Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
auto but no tel_event was found inside SDP file.

On an incoming call create_rtp will be called and when session->dtmf is
set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
looking at the SDP file.

Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
but continued to advertise RFC2833 support.

This meant the native_rtp bridge would falsely consider the two channels
as compatible. In addition to changing the DTMF mode we now set or
remove the AST_RTP_PROPERTY_DTMF.

The property is checked in ast_rtp_dtmf_compatible and called by
native_rtp_bridge_compatible.

ASTERISK-29051 #close

Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287