Fix compile error in main/utils.c because strdup was used in dummy_start

Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793

The spandspflow2pcap.py creates pcap files from fax.log files, generated
through 'fax set debug on' when receiving a fax. An example fax.log is
included as spandspflow2pcap.log.

The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a
recent version of SIPp.

ASTERISK-25660 #close

Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526

* changes:
  main/pbx: Move hangup handler routines to pbx_hangup_handler.c.
  main/pbx: Move dialplan application management routines to pbx_app.c.
  main/pbx: Move switch routines to pbx_switch.c.

The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter
y not the address of y.

I capture the radius UDP packet via tcpdump, and the AV pairs are not correct,
then i review the source code and compare it with cdr/cdr_radius.c. Fix it and
 it works.

ASTERISK-25647 #close
Reported by: Aaron An
Tested by: Aaron An

Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0

This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves hangup handler management functions to their own source.

Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104

This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves dialplan application management functions to their own source.

Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c

This is the fifth patch in a series meant to reduce the bulk of pbx.c.
This moves ast_switch functions to their own source.

Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e

This is the fourth patch in a series meant to reduce the bulk of pbx.c.
This moves pbx timing functions to their own source.

Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6

The menuselect conflict between app_voicemail and res_mwi_external
makes it hard to package 1 version of Asterisk.  There no actual
build dependencies between the 2 so moving this check to runtime
seems like a better solution.

The ast_vm_register and ast_vm_greeter_register functions in app.c
were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
is already a voicemail module registered. The modules' load_module
functions were then modified to return DECLINE instead of -1 to the
loader.  Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
the modules were incorrectly causing Asterisk to stop so this needed
to be cleaned up anyway.

Now you can build both and use modules.conf to decide which voicemail
implementation to load.

The default menuselect options still build app_voicemail and not
res_mwi_external but if both ARE built, res_mwi_external will load
first and become the voicemail provider unless modules.conf rules
prevent it.  This is noted in CHANGES.

Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247

This is the third patch in a series meant to reduce the bulk of pbx.c.
This moves channel and global variable routines to their own source.

Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6

If a caller hangs up before dial is executed within an AGI then the AGI
has likely eaten all queued frames before executing the dial in DeadAGI
mode.  With the caller hung up and no pending frames from the caller's
read queue, dial would not know that the call has hung up until a called
channel answers.  It is rather annoying to whoever just answered the
non-existent call.

Dial should not continue execution in DeadAGI mode, hangup handlers, or
the h exten.

* Added a check early in dial to abort dialing if the caller has hungup.

ASTERISK-25307 #close
Reported by: David Cunningham

Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418

Prior to this patch, we explicitly disallowed setting any properties on a
finalized CDR. This seemed like a good idea at the time; in practice, it was
more restrictive.

There are weird and strange scenarios where setting a property on a finalized
CDR is definitely wrong. For example, we may Fork a CDR, finalizing the
previous one, then change a property. In said case, the old CDR is supposed
to now be 'immutable' (so to speak), and should not be updated. From the
perspective of the code, a forked CDR that is finalized is just finalized.
Hence why we decided these should not be updated.

In practice, it is much more common to want to set a property on a CDR in
the h extension or in a hangup handler. Disallowing a common scenario to make
an esoteric behaviour work isn't good. This patch fixes this by allowing
callers to set a property IF we are the last CDR in the chain. This preserves
the finalized CDR if it was forked, while allowing the more common case to
function.

ASTERISK-25458 #close

Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9

Prior to this patch, the CDR engine attempted to set the end time on a CDR
that was executing hangup logic and with endbeforehexten set to Yes by
calling a function that inspects the properties on the Party A snapshot to
determine if we are ready to set the end time. That always failed. This is
because a Party A snapshot is not updated for CDRs that are executing hangup
logic with endbeforehexten=Yes.

Instead of calling a function that looks at the Party A snapshot, we just
simply set the end time on the CDR. This is safe to call multiple times, and is
safe to call at this point as we know that (a) we are executing hangup logic,
and (b) we are supposed to set the end time at this point.

ASTERISK-25458

Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3

This is the second patch in a series meant to reduce the bulk of pbx.c.
This moves custom function management routines to their own source.

Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177

Change-Id: I22d3c90f6f27df82e915bbf81c1d91221f7a945e

This patch adds a new module, res_pjsip_history, that provides a slightly
better way of debugging SIP message traffic on a busy Asterisk system. The
existing mechanisms all rely on passively dumping a SIP message to the CLI.
While this is perfectly fine for logging purposes and well controlled
environments, on many installations, the amount of SIP messages Asterisk
receives will quickly swamp the CLI. This makes it difficult to view/capture
those messages that you want to diagnose in real time.

This patch provides another way of handling this. When enabled, the module
will store SIP message traffic in memory. This traffic can then be queried
at leisure.

In order to make the querying useful, a CLI command has been implemented,
'pjsip show history', that supports a basic expression syntax similar to
SQL or other query languages. A small number of useful fields have been
added in this initial patch; additional fields can easily be added in
later improvements. Those fields are:
 - number: The entry index in the history
 - timestamp: The time the message was recieved
 - addr: The source/destination address of the message
 - sip.msg.request.method: The request method
 - sip.msg.call-id: The Call-ID header

Note - this is a resurrection of the module initially proposed on Review Board
here: https://reviewboard.asterisk.org/r/4053/

Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36

We joked about splitting pbx.c into multiple files but this first step was
fairly easy.  All of the pbx_builtin dialplan applications have been moved
into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
is called by asterisk.c just after load_pbx().

A few functions were renamed and are cross-exposed between the 2 source files.

Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a

The test_timezone_watch unit test is written to expect a
condition to be signaled when the inotify daemon thread runs.
There exists a small window where the test_timezone_watch
thread can signal the inotify daemon thread while it is not
reading on the underlying file descriptor. If this occurs
the test_timezone_watch thread will wait indefinitely for a
signal that will never arrive.

This change adds a timeout to the condition so it will return
regardless after a period of time.

Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390

The cache_clear test was written to expect duplicate Stasis messages
sent from the technology endpoint to the all caching topic. This patch
fixes the test to no longer expect these duplicate messages.

ASTERISK-25137

Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981

Updated ast_websocket_write to encode the entire frame in to one
write operation, to ensure that we don't end up with a situation
where the websocket header has been sent, while the body can not
be written.

Previous to August's patch in commit b9bd3c14, certain network
conditions could cause the header to be written, and then the
sub-sequent body to fail - which would cause the next successful
write to contain a new header, and a new body (resulting in
the peer receiving two headers - the second of which would be
read as part of the body for the first header).

This was patched to have both write operations individually fail
by closing the websocket.

In a case available to the submitter of this patch, the same
body which would consistently fail to write, would succeed
if written at the same time as the header.

This update merges the two operations in to one, adds debug messages
indicating the reason for a websocket connection being closed during
a write operation, and clarifies some variable names for code legibility.

Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598

When an endpoint is created, its messages are forwarded to both the tech
endpoint topic and the all endpoints topic. This is done so that various
parties interested in endpoint messages can subscribe to just the tech
endpoint and receive all messages associated with that particular technology,
as opposed to subscribing to the all endpoints topic. Unfortunately, when the
tech endpoint is created, it also forwards all of its messages to the all
topic. This results in duplicate messages whenever an endpoint publishes its
messages.

This patch resolves the duplicate message issue by creating a new function
for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts
as a normal caching topic, save that it no longer forwards messages it receives
to the all endpoints topic. This allows it to act as an aggregation "sink",
while preserving the necessary caching behaviour.

ASTERISK-25137 #close
Reported-by: Vitezslav Novy

ASTERISK-25116 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>

Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b

Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b

websocket_write_timeout was not being set to its default value
during sip config reload, which meant that prior to this commit,
1) the default value of 100 was not used, unless an invalid value
(or 1) was specified in sip.conf for websocket_write_timeout, and
2) if the websocket_write_timeout directive was removed from sip.conf
without a full restart of asterisk, then the previous value would
continue to be used indefinitely.

This essentially lead to a 0ms write timeout (the first write attempt
in ast_careful_fwrite must have succeeded) in websocket write requests
from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf.

Changes to websocket_write_timeout still only apply to new websocket
sessions, after the sip reload -- timeouts on existing sessions are
not adjusted during sip reload.

Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953

Use of GOTO_ON_BLINDXFR would not work at all.  The target location would
never be executed by the transferring channel.

* Made feature_blind_transfer() call ast_bridge_set_after_go_on() with
valid context, exten, and priority parameters from the transferring
channel.

* Renamed some feature_blind_transfer() local variables for clarity.

ASTERISK-25641 #close
Reported by Dmitry Melekhov

Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac