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

ASTERISK-29099

Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6

Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and
AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice,
resulting in to 181 being generated.

Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab

Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e

When constructing a stream name based on the media type
and position the allocated name was not being freed
causing a leak.

Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de

Only track our sample offset if we are playing a non-announcement file,
otherwise we will skip that number of samples when we start playing the
first MoH file.

ASTERISK-24329 #close

Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc

The ast_sip_dialog_get_session function returns the session
with reference count increased. This was not taken into
account and was causing sessions to remain around when they
should not be.

ASTERISK-29089

Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8

Sometimes not play MOH on bridge.

ASTERISK-29081
Reported-by: Michal Hajek <michal.hajek@daktela.com>

Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232

The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.

Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21

Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276

Implemention of History-Info capable of interworking with Diversion
Header following RFC7544

ASTERISK-29027 #close

Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6

The recent 491 changes introduced a check to determine if the active
and pending topologies were equal and to suppress the re-invite if they
were. When a re-invite is sent for a COLP-only change, the pending
topology is NULL so that check doesn't happen and the re-invite is
correctly sent. Of course, sending the re-invite sets the pending
topology.  If a 491 is received, when we resend the re-invite, the
pending topology is set and since we didn't request a change to the
topology in the first place, pending and active topologies are equal so
the topologies-equal check causes the re-invite to be erroneously
suppressed.

This change checks if the topologies are equal before we run the media
state resolver (which recreates the pending topology) so that when we
do the final topologies-equal check we know if this was a topology
change request.  If it wasn't a change request, we don't suppress
the re-invite even though the topologies are equal.

ASTERISK-29014

Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314

Added to:
 * bridges/bridge_softmix.c
 * channels/chan_pjsip.c
 * include/asterisk/res_pjsip_session.h
 * main/channel.c
 * res/res_pjsip_session.c

There NO functional changes in this commit.

Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3

When both Asterisk and a UA send re-invites at the same time, both
send 491 "Transaction in progress" responses to each other and back
off a specified amount of time before retrying. When Asterisk
prepares to send its re-invite, it sets up the session's pending
media state with the new topology it wants, then sends the
re-invite.  Unfortunately, when it received the re-invite from the
UA, it partially processed the media in the re-invite and reset
the pending media state before sending the 491 losing the state it
set in its own re-invite.

Asterisk also was not tracking re-invites received while an existing
re-invite was queued resulting in sending stale SDP with missing
or duplicated streams, or no re-invite at all because we erroneously
determined that a re-invite wasn't needed.

There was also an issue in bridge_softmix where we were using a stream
from the wrong topology to determine if a stream was added.  This also
caused us to erroneously determine that a re-invite wasn't needed.

Regardless of how the delayed re-invite was triggered, we need to
reconcile the topology that was active at the time the delayed
request was queued, the pending topology of the queued request,
and the topology currently active on the session.  To do this we
need a topology resolver AND we need to make stream named unique
so we can accurately tell what a stream has been added or removed
and if we can re-use a slot in the topology.

Summary of changes:

 * bridge_softmix:
   * We no longer reset the stream name to "removed" in
     remove_all_original_streams().  That was causing  multiple streams
     to have the same name and wrecked the checks for duplicate streams.

   * softmix_bridge_stream_sources_update() was checking the old_stream
     to see if it had the softmix prefix and not considering the stream
     as "new" if it did.  If the stream in that slot has something in it
     because another re-invite happened, then that slot in old might
     have a softmix stream but the same stream in new might actually
     be a new one.  Now we check the new_stream's name instead of
     the old_stream's.

 * stream:
   * Instead of using plain media type name ("audio", "video", etc) as
     the default stream name, we now append the stream position to it
     to make it unique.  We need to do this so we can distinguish multiple
     streams of the same type from each other.

   * When we set a stream's state to REMOVED, we no longer reset its
     name to "removed" or destroy its metadata.  Again, we need to
     do this so we can distinguish multiple streams of the same
     type from each other.

 * res_pjsip_session:
   * Added resolve_refresh_media_states() that takes in 3 media states
     and creates an up-to-date pending media state that includes the changes
     that might have happened while a delayed session refresh was in the
     delayed queue.

   * Added is_media_state_valid() that checks the consistency of
     a media state and returns a true/false value. A valid state has:
     * The same number of stream entries as media session entries.
         Some media session entries can be NULL however.
     * No duplicate streams.
     * A valid stream for each non-NULL media session.
     * A stream that matches each media session's stream_num
       and media type.

   * Updated handle_incoming_sdp() to set the stream name to include the
     stream position number in the name to make it unique.

   * Updated the ast_sip_session_delayed_request structure to include both
     the pending and active media states and updated the associated delay
     functions to process them.

   * Updated sip_session_refresh() to accept both the pending and active
     media states that were in effect when the request was originally queued
     and to pass them on should the request need to be delayed again.

   * Updated sip_session_refresh() to call resolve_refresh_media_states()
     and substitute its results for the pending state passed in.

   * Updated sip_session_refresh() with additional debugging.

   * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE
     to pjproject if a transaction is in progress.  This stops us from
     creating a partial pending media state that would be invalid later on.

   * Updated reschedule_reinvite() to clone both the current pending and
     active media states and pass them to delay_request() so the resolver
     can tell what the original intention of the re-invite was.

   * Added a large unit test for the resolver.

ASTERISK-29014

Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb

Currently, it was not possible to create bridge with video_mode single.
This made hard to put the bridge in a vidoe_single mode.
So, added video_single option for Bridge creation using the ARI.
This allows create a bridge with video_mode single.

ASTERISK-29055

Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae

Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again.

Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6

When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.

ASTERISK-29042

Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a

Properly bump reference on format object to avoid memory corruption on double free

ASTERISK-29040 #close

Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3

Adapt the response handler so it also called when 181 is received.
In the case 181 is received, also generate the 181 response.

ASTERISK-29001 #close

Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df

Per the RFC when an outgoing re-INVITE is done we should
only terminate the dialog if a 481 or 408 is received.

ASTERISK-29033

Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503

Two changes of note in this patch:

* Use ast_file_read_dir instead of opendir/readdir/closedir

* If the files list should be sorted, do that at the end rather than as
  we go which improves performance for large lists

Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f

The MoH class internal file vector is potentially being manipulated by
multiple threads at the same time without sufficient locking. Switch to
a reference counted list and operate on copies where necessary.

ASTERISK-28927 #close

Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217

When reading in a codec preference configuration option
the value would be set on the respective option before
applying any default adjustments, resulting in the
configuration not being as expected.

This was exposed by the REST API push configuration as
it used the configuration returned by Asterisk to then do
a modification. In the case of codec preferences one of
the options had a transcode value of "unspecified" when the
defaults should have ensured it would be "allow" instead.

This also renames the options in other places that were
missed.

Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964

This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.

Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d

Fixed a memory allocation that was not passing in the correct size for
the struct in curl.c.

Change-Id: I5fb92fbbe84b075fa6aefa2423786df80e114c3a

When a bundled stream is removed, its bundle_group is reset to -1.
If that stream is later reused, the bundle parameters on session
media need to be reset correctly it could mistakenly be rebundled
with a stream that was removed and never reused.  Since the removed
stream has no rtp instance, a crash will result.

Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7

Statically configured contacts on an AOR don't have an expiration
time so when adding them to the resulting 200 OK if an endpoint
registers ensure they are marked as such.

ASTERISK-28995

Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596

Currently, if the bridge has created by the ARI, the video_mode
parameter was
not shown in the BridgeCreated event correctly.

Fixed it and added video_mode shown in the 'bridge show <bridge id>'
cli.

ASTERISK-28987

Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295

When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.

The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.

Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01

We read beyond the end of the buffer when copying the string out of the
buffer when we used ast_copy_string() because the original string was
not null terminated. Instead switch to ast_strndup() which does not
exhibit the same behavior.

ASTERISK-28975 #close

Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a

Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.

Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.

Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.

Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1