The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa77
failed to add ENUM support for Postgres databases. This requires a
specific import from the sqlalchemy.dialects.postgresql package. This
patch corrects this error, which allows for Postgres update scripts to
be generated.

ASTERISK-24706

Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015

This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
columns added in Asterisk 1.8. The columns are:
 * peeraccount
 * linkedid
 * sequence
When enabled, the columns in the database entry will be populated with the data
from the CDR.

ASTERISK-24976 #close

Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b

Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options.
This extra space prevented any of the dependencies from being seen by
menuselect, so building with default options would fail if PJSIP was
not installed.

This also makes the tool that extracts information for menuselect
tolerant of multiple spaces in the future.

ASTERISK-25033 #close
Reported by: Peter Whisker

Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698

When using the force black background command-line option or configuration
option an invalid reset sequence is sent following a coloured output item 
in the CLI, the result is that the colour is not 'turned off' and continues
until the next non-default coloured text output.

A reset sequence is already defined in term.c, but the ast_term_reset
function doesn't use it, instead building it's own invalid sequence and 
returning that.

This patch changes that behaviour, removing the building of a reset sequence
and instead using the pre-built constant 'enddata' which is a suitable reset
sequence for this purpose.

ASTERISK-24896 #close
Reported by: Dan Tucny

Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43

This ensures that refdebug is initialized before AO2_DEBUG if
both are enabled, since AO2_DEBUG allocates a container.

This change also makes AO2_DEBUG initialization critical, a
failure will abort Asterisk startup.  This is needed since
the failure would be caused by reg_containers allocation
failure, and that would result in a segmentation fault by
ao2_container_register later in startup.

ASTERISK-25048 #close
Reported by: Corey Farrell

Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244

The PBX core maintains two hash tables for hints: a container of the
actual hints (hints), along with a container of devices that are watching that
hint (hintdevices). When a dialplan reload occurs, each hint in the hints
container is destroyed; this requires a lookup in the container of devices to
find the device => hint mapping object. In the current code, this performs an
ao2_callback, iterating over each of the device to hint objects in the
hintdevices container. For a large number of hints, this is extremely
expensive: dialplan reloads with 20000 hints could take several minutes
in just this phase.

This patch improves the performance of this step in the dialplan reloads
by caching which devices are watching a hint on the hint object itself.
Since we don't want to create a circular reference, we just cache the
name of the device. This allows us to perform a smarter ao2_callback on
the hintdevices container during hint removal, hashing on the name of the
device and returning an iterator to the matching names. The overall
performance improvement is rather large, taking this step down to a number of
seconds as opposed to minutes.

In addition, this patch also registers the hint containers in the PBX
core with the astobj2 library. This allows for reasonable debugging to
hash collisions in those containers.

ASTERISK-25040 #close
Reported by: Matt Jordan

Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360

The sample pjsip.conf has a few comment lines that are missing the
semicolons at the start of the comment, causing the config to fail
load.

Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352

Scenario:
Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects
the incoming call (or some other immediate circumstance causes Carol not
to answer the call)

What occurs in this case is that when the bridge between Alice and Bob
breaks, Alice is told to masquerade into Bob's channel that had placed
the call to Carol. The actual masquerade goes down without a hitch.
However, a channel fixup callback that attempts to publish dial events
over Stasis has a crash. The reason for this crash is that the datastore
on Bob's channel that placed the outbound call to Carol only had a bare
pointer to Carol's channel. Since Carol rejected the incoming call,
Carol's channel has been hung up and freed, meaning accessing her
channel results in a crash.

The fix here is simple. The dial fixup code has been altered to hold
references to the involved channels and to drop those references when
freeing data.

ASTERISK-25025 #close
Reported by Chet Stevens

Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197

The build system now scans additional sources when generating
moduleinfo for menuselect.  Unfortunately the extra sources
for format_mp3 only exist if downloaded.

Use the Makefile macro 'wildcard' to allow moduleinfo generator
to ignore sources that do not exist.

Change-Id: I596604713b7345ce994f32197f8f6bfd9bcf4170

The res_ari_device_states module depends on res_stasis_device_state,
not res_stasis_device_states.

Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de

Some telco switches occasionally ignore ISDN RESTART requests.  The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request.  If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco.  The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.

chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code.  Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards.  I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.

* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.

ASTERISK-25034 #close
Reported by: Richard Mudgett

Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65

When the PJSIP pjsip_regc_send function is invoked and an error
status returned the caller currently decrements the reference count
of the client state that it just incremented, assuming the
registration callback would not have been invoked. In practice
this is not correct. If the failure happens after the transaction
has been set up the callback will still be invoked. This will
cause the reference count to be incorrectly decremented twice, once
by the registration callback and second by the caller of
pjsip_regc_send.

This change makes it so that whether the callback is invoked or
not is known by the caller of pjsip_regc_send. Depending on
this it can know whether it is responsible for decrementing the
reference count of the client state or not.

ASTERISK-25037 #close
Reported by: Joshua Colp

Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de

Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3

The patch in 0b6410c4 did correctly fix a memory leak of the DTLS
structures in the RTP engine. However, when a 'core reload' is issued, a
double free of the memory pointed to by the char *'s in the DTLS
configuration struct can occur, as ast_rtp_dtls_cfg_free does not set
the pointers to NULL when they are freed.

This patch sets those pointers to NULL, preventing a second call to
ast_rtp_dtls_cfg_free from corrupting memory.

ASTERISK-25022

Change-Id: I820471e6070a37e3c26f760118c86770e12f6115

A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so
a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits
per second. This reverts all or some of those patches since according to the
v.27ter standard a rate of 2400 bits per second is also supported.

One of the original patches also added 9600 bits per second support for v.27.
This patch also removes that since v.27ter only supports 2400/4800 bits per
second.

Also, since Asterisk specifically supports v.27ter the enum was renamed to
better reflect this.

ASTERISK-24955 #close
Reported by: Matt Jordan

Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733

This function allows code to run ao2_ref against the real
object associated with a weakproxy.  It is useful when
all of the following conditions are true:
* You have a pointer to weakproxy.
* You do not have or need a pointer to the real object.
* You need to ensure the real object exists and is not
  destroyed during a process.

In this case it's wasteful to store a pointer to the real
object just for the sake of releasing it later.

Change-Id: I38a319b83314de75be74207a8771aab269bcca46

Odd behaviors have been observed during outbound registrations. The most
common problem witnessed has been one where a request with
authentication credentials cannot be created after receiving a 401
response. Other behaviors include apparently processing an incorrect SIP
response.

Inspecting the code led to an apparent issue with regards to how we
handle transactions in outbound registration code. When a response to a
REGISTER arrives, we save a pointer to the transaction and then push a
task onto the registration serializer. Between the time that we save the
pointer and push the task, it's possible for the transaction to be
destroyed due to a timeout. It's also possible for the address to be
reused by the transaction layer for a new transaction.

To allow for authentication of a REGISTER request to be authenticated
after the transaction has timed out, we now hold a reference to the
original REGISTER request instead of the transaction. The function for
creating a request with authentication has been altered to take the
original request instead of the transaction where the original request
was sent.

ASTERISK-25020
Reported by Mark Michelson

Change-Id: I756c19ab05ada5d0503175db9676acf87c686d0a

Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac

Merge "chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR             Sections Exist in pjsip.conf"

* changes:
  res_pjsip: Remove incorrect MODULEINFO from presence_xml.c.
  Git Migration: Create doc/rest-api when needed.

ARI modules that are generated by 'make ari-stubs' are all dependent on
res_ari_model.  Additionally some of the same modules depend on one or more
res_stasis_* modules.

ASTERISK-25027 #close
Reported by: Corey Farrell

Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153

Remove incorrect MODULEINFO block and unneeded header includes
from presence_xml.c.

ASTERISK-25027
Reported by: Corey Farrell

Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb

Create the directory './doc/rest-api' at the start of 'make ari-stubs'
to prevent an error when documentation is generated.  The directory is
also added to git ignores.

ASTERISK-25027
Reported by: Corey Farrell

Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b

* Add AST_DEVMODE to BUILDOPTS
* Use BUILDOPTS to generate AST_BUILDOPT_SUM.
* Remove loop that defined AST_MODULE_*

These changes ensure that only ABI effecting options are considered for
AST_BUILDOPT_SUM.  This also reduces unneeded full system rebuilds caused
by enabling or disabling one module that another is dependent on.

ASTERISK-25028 #close
Reported by: Corey Farrell

Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2