* Remove duplicate res_parking.conf courtesytone config option
documentation.

ASTERISK-24596 #close
Reported by:  Philip Correia

ASTERISK-24605
Reported by:  Philip Correia
Patches:
      call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia

Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5

The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again.  Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.

In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'.  Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip.  This should preserve the current behavior.

Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf

Previously you could add [!dnid] to the SIP dial string to alter the To:
header. This change allows you to alter the From header as well.

SIP dial string extra options now look like this:

    [![touser[@todomain]][![fromuser][@fromdomain]]]

INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To:
header, that is no longer possible.

ASTERISK-25803 #close

Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7

Warnings and errors in the pjproject libraries are generally handled by
Asterisk.  In many cases, Asterisk wouldn't even consider them to be warnings
or errors so the messages emitted by pjproject directly are either superfluous
or misleading.  A good exampe of this are the level-0 errors pjproject emits
when it can't open a TCP/TLS socket to a client to send an OPTIONS.  We don't
consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS
client be treated any differently?

A config file for res_pjproject has bene added (pjproject.conf) and a new
log_mappings object allows mapping pjproject levels to Asterisk levels
(or nothing).  The defaults if no pjproject.conf file is found are the same
as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR,
2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level>

Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898

ps_systems needed disable_tcp_switch
ps_registrations needed line and endpoint

ASTERISK-25737 #close

Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19

During ICE negotiation the IPs of the local interfaces are sent to the remote
peer as host candidates. In many cases Asterisk is behind a static one-to-one
NAT, so these host addresses will be internal IP addresses.

To help in hiding the topology of the internal network, this patch adds the
ability to override the host candidates by matching them against a
user-defined list of replacements.

Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f

This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.

The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.

ASTERISK-24972 #close

Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8

ASTERISK-20987 #close
Reported by: hristo

Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38

Since res_pjsip now depends on res_pjproject, this has been added to
basic-pbx modules.conf.

Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d

Update the version number in the comments from Asterisk 12 to Asterisk 12+

Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b

Added new global option (regcontext) to pjsip. When set, Asterisk will
dynamically create and destroy a NoOp priority 1 extension
for a given endpoint who registers or unregisters with us.

ASTERISK-25670 #close
Reported-by: Daniel Journo

Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62

On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
 is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address.  This happens because
 res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).

The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address.  This causes the packets to originate from
the specified address.

ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo

Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88

In the sample sip.conf this is written with regard to notifyringing:
;notifyringing = no ; Control whether subscriptions already INUSE get sent
RINGING when another call is sent (default: yes)

However, this setting changes whether or not any RINGING indications are sent
to subscriptions. There is no separate configurable setting that allows
to control whether INUSE subscriptions also get sent RINGING. This is however
a useful option, to see (using BLF) if somebody else is able to handle an
incoming call or if everybody is busy.

This patch corrects the documentation for notifyringing (so the documentation
matches the functionality) and make notifyringing a tri-state option, by adding
the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing =
notinuse, only subscriptions that are not INUSE are sent the RINGING signal.

The default setting for notifyringing remains set to yes, so the default
behaviour is not affected.

ASTERISK-25558

Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa

- The maximum_number_of_words was previously documented as being
the number of words that when exceeded, would result in the AMD
application returning that the audio represents a machine.

This was inconsistent with its actual functionality - it was
a number of words that when REACHED, would result in determination
as a machine.

This update corrects the functionality to match the previously
documented functionality.  This is a backwards incompatible change
in configuration file, and has been added to UPGRADE.txt as a result.

The sample configuration file and application defaults have been updated
so that the default value is now 2, which reflects the same default
functionality as previous versions.

- Update documentation for silence_threshold, which previously implied
that it was measuring time, rather than noise averages in the sample.

- Update the comments in amd.conf.sample.

ASTERISK-25639 #close
Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093

This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".

The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.

ASTERISK-25549 #close
Reported by Mark Michelson

Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98

This patch adds a new setting "websockets_enabled" to sip.conf.
Setting this to false allows chan_sip to be used without causing
conflicts with res_pjsip_transport_websocket.

ASTERISK-24106 #close
Reported by: Andrew Nagy

Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7

During outbound registration it is possible to receive a fatal (any permanent/
non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due
to a problem with the registrar itself. Upon receiving the failure response
Asterisk terminates outbound registration for the given endpoint.

This patch adds an option, 'fatal_retry_interval', that when set continues
outbound registration at the given interval up to 'max_retries' upon receiving
a fatal response.

ASTERISK-25485 #close

Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2

When Asterisk is part of a larger distributed system, log files are often
gathered using tools (such as logstash) that prefer to consume information
and have it rendered using other tools (such as Kibana) that prefer a
structured format, e.g., JSON. This patch adds support for JSON formatted
logs by adding support for an optional log format specifier in Asterisk's
logging subsystem. By adding a format specifier of '[json]':

full => [json]debug,verbose,notice,warning,error

Log messages will be output to the 'full' channel in the following
format:

{
  "hostname": Hostname or name specified in asterisk.conf
  "timestamp": Date/Time
  "identifiers": {
    "lwp": Thread ID,
    "callid": Call Identifier
  }
  "logmsg": {
    "location": {
      "filename": Name of the file that generated the log statement
      "function": Function that generated the log statement
      "line": Line number that called the logging function
    }
    "level": Log level, e.g., DEBUG, VERBOSE, etc.
    "message": Actual text of the log message
  }
}

ASTERISK-25425 #close

Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238

This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.

ASTERISK-25259 #close

Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9

Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19

 * In sip.conf.sample fix sentence where we said that WS or WSS are supported
   transports for use in an outbound register definition. They are not
   supported in that case.
 * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used
   to enable CDR on a channel.

ASTERISK-24867 #close
Reported by: Rusty Newton

ASTERISK-24853 #close
Reported by: PSDK

Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca

Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.

ASTERISK-25158 #close
Reported by: Steve Pitts

Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615

When a client connects to a server via SSL/TLS, the server commonly utilizes an
RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if
the server socket is configured with a certificate for either one of those, it
would lose its compatibility with RSA-only clients.

Now, the server socket can be configured with up to one RSA, ECDSA and DSA key
each. For example, if a client is not compatible with SHA-2 hashed certificates
like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy
clients and ECDSA/SHA-2 for everyone else.

ASTERISK-24815 #close
Reported by: Alexander Traud
patches:
  tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520)

Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a

The main IVR was playing demo-congrats. I've switched it over to the
basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
has Allison prompting the user with the actual IVR menu.

ASTERISK-24892 #close

Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d

ASTERISK-25034
Reported by: Richard Mudgett

Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30

Add feature to set optional schema parameter on configuration file via
'schema' setting.

Fix query to get columns from table while considering schema. If in
the database there exists two tables with same name in distinct schemas
it will return an error when inserting record.

ASTERISK-24967 #close

Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c

Added the ability to set the character to quote identifiers. This
allows adding the character at the start and end of table and column
names. This setting is configurable for cdr_adaptive_odbc via the
quoted_identifiers in configuration file cdr_adaptive_odbc.conf.

ASTERISK-25006

Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8

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

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

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

This switches files used to generate other sources to use the new
ASTERISK_REGISTER_FILE macro.

ASTERISK-25026 #close
Reported by: Corey Farrell

Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740

* The REF_DEBUG compiler flag no longer has any effect on code that uses
  Astobj2.  It is used to determine if reference debugging is enabled by
  default.  Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
  This was possible now that we no longer require a dual ABI.

ASTERISK-24974 #close
Reported by: Corey Farrell

Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1

Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
 * peeraccount
 * linkedid
 * sequence
This feature is configurable in cdr_odbc.conf using a new configuration
option, 'newcdrcolumns'.

ASTERISK-24976 #close

Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127

Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup.  So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.

This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies.  This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.

If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random().  If not set,
qualify_timeout is used.

The default is "0" (disabled).

ASTERISK-24863 #close

Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>

This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html



The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint.  Only dynamic contact add/delete actions
update the endpoint.  Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.

This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...

1.  A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message.  The default is 3000ms.  When the timer expires, the contact is
marked unavailable.

2.  Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'.  When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'.  The
existing endpoint events are generated appropriately.

ASTERISK-24863 #close

Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>

After the "progressinband" value setting of "never" was updated to never send a
183 this separated its use from the "no" value. Since "never" was the default,
but most users probably expect "no" this patch updates the default for the
"progressinband" setting to "no."

ASTERISK-24835 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4606/
........

Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new configuration parameter, 'calltokenexpiration', that
controls how long before an authentication call token is expired. The default
maintains the RFC specified 10 seconds. Setting it to a higher value may be
useful in lossy networks.

Review: https://reviewboard.asterisk.org/r/4588

ASTERISK-24939 #close
Reported by: Y Ateya
patches:
  ctoken_configuration.diff submitted by Y Ateya (License 6693)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps
to be logged in GMT.

Review: https://reviewboard.asterisk.org/r/4571/

ASTERISK-23186 #close
Reported by: Rodrigo Ramirez Norambuena
patches:
  cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/4504/
........

Merged revisions 433656 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company

Patch 4488 includes all functionality needed for SAC's outside connectivity
and some externally accessed features, as well as outbound dialing.

Reported by: Malcolm Davenport
Tested by: Rusty Newton

Review: https://reviewboard.asterisk.org/r/4488/
........

Merged revisions 433624 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433637 65c4cc65-6c06-0410-ace0-fbb531ad65f3