Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Page.  Simplify dialplan by making Page
handle these conditions gracefully.

* Made tolerate empty positions in the paged device list.

* Reduced some warnings associated with the 's' option to verbose
messages.  The warning level for those messages really serves no purpose
as that is why the 's' option exists.

ASTERISK-28638

Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3

The db_init() function ultimately calls db_sync() which signals the
condition before it is initialized.

Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab

ASTERISK-28667 #close

Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2

When testing for the existance of a file, the media cache is searched even if
the file has no chance of being in it.  This can cause performance issues
as the media cache size increases.

As a result, calls to applications like Read and Playback using local files
must scan through the media cache before playing.  Under load and with a
large cache, this can delay the playback of those files.

This patch updates the function that checks for the existance of a file to
only consult the media cache database if the requested file is a remote path.
It introduces a new is_remote_path() function in main/file.c.

ASTERISK-28625  #close
Reported-by:  <kevin@phoneburner.com>

Change-Id: If91137493732d9034dafa381c081c69274a7dcc9

ASTERISK-28664 #close

Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6

Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.

ASTERISK-28636

Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074

Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.

ASTERISK-28604

Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7

ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee

A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.

This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.

ASTERISK-28659

Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355

A previous patch:

Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.

This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).

This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.

The default is disabled keeping with the old behavior.

ASTERISK-28660

Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a

Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies.  It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.

Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

add missing argument "rtt" and "status" to the documentation

The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.

This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)

ASTERISK-28626

Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d

This reverts commit 7e3015d7.

Reason for revert: Regression in XML validation.

validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)

As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.

Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f

In Asterisk 16+, there are a few places in ast_rtp_read where we've
allocated a frame list but return a null frame instead of the list.
In these cases, any frames left in the list won't be freed.  In the
vast majority of the cases, the list is empty when we return so
there's nothing to free but there have been leaks reported in the
wild that can be traced back to frames left in the list before
returning.

The escape paths now all have logic to free frames left in the
list.

ASTERISK-28609
Reported by: Ted G

Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a

chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to
213.150.203.60:1492 returned -2: Interrupted system call

returned -2 implies this wasn't actually an OS error, so errno makes no
sense either.  Internal error was already logged higher up, and -2
generally means that either there isn't a valid connection available, or
the pipe notification failed, and that is already correctly logged.

ASTERISK-28651 #close

Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

RFC3261 Section 10 "Registrations", specifically paragraph
"10.2.4: Refreshing Bindings", states that a user agent compares
each contact address (in a 200 REGISTER response) to see if it
created the contact.  If the Asterisk endpoint has the
rewrite_contact option set however, the contact host and port sent
back in the 200 response will be the rewritten one and not the
one sent by the user agent.  This prevents the user agent from
matching its own contact.  Some user agents get very upset when
this happens and will not consider the registration successful.
While this is rare, it is acceptable behavior especially if more
than 1 user agent is allowed to register to a single endpoint/aor.

This commit updates res_pjsip_nat (where rewrite_contact is
implemented) to store the original incoming Contact header in
a new "x-ast-orig-host" URI parameter before rewriting it, and to
restore the original host and port to the Contact headers in the
outgoing response.

This is only done if the request is a REGISTER and rewrite_contact
is enabled.

pjsip_message_filter was also updated to ensure that if a request
comes in with any existing x-ast-* URI parameters, we remove them
so they don't conflict.  Asterisk will never send a request
with those headers in it but someone might just decide to add them
to a request they craft and send to Asterisk.

NOTE: If a device changes its contact address and registers again,
it's a NEW registration.  If the device didn't unregister the
original registration then all existing behavior based
on aor/remove_existing and aor/max_contacts apply.

ASTERISK-28502
Reported-by: Ross Beer

Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e