This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.

ASTERISK-26470 #close

Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d

The "Q" option will set the cause on the unanswered channels when
another channel answers.  It overrides the default of
ANSWERED_ELSEWHERE.

NOTE:  chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.

ASTERISK-26446 #close

Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47

This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.

ASTERISK-26398 #close

Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d

sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).

To use this, use a systemd unit with 'Type=notify' for Asterisk.

This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.

Also adds support for libsystemd detection in the configure script.

Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811

This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn

Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762

Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.

This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.

Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
  conference (if the channel and conference use the same language)

ASTERISK-26289 #close
Reported by Mark Michelson

Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a

Following the Encrypt-all-the-things paradigm:

The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone
determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts
the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile
(sRTP is preferred aka optional; not mandatory). If the VoIP server does not
support sRTP and TLS, the phone shows an open padlock icon.

This paradigm is supported by several VoIP/SIP clients on default. Some
implementations even cannot be changed to RTP/sAVP. Therefore here, this
change allows Preferred sRTP for ingress. For egress, please, create a dial
plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP.

ASTERISK-20234 #close
Reported by: tootai
Tested by: tootai, Alexander Traud
patches:
 srtp_patches.diff submitted by Matt Jordan

Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd

The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect.  Any that are selected will automatically be
downloaded and installed when "make install" is run.  Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.

Example use with codecs:

The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included.  Their support levels are 'external', which
triggers the download and install, and defaultenabled is no.  Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name.  You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory.  In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.

A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.

To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball.  The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.

bash and xmlstarlet are required for downloader operation.  If they're
not installed, the external items in menuselect will be unavailable.

Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a

We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals.  Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object.  With a
memory cache for realtime, there is about the same amount of overhead as
for config files.  Either way, it is still fairly expensive to access the
sorcery object that much.

* Cache the global config options so we can access them faster.  You must
now always perform a res_pjsip reload to change the global options.

Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7

Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth
is not supported in IAX2 protocol. Please refer to section 8.6.13 of
RFC 5456.

But plaintext auth is still supported by Asterisk implementation of IAX2.
This support should be dropped.

Patch, based on asterisk-dev discussion, adds deprecation warning on
startup if 'auth' is set to 'plaintext', changes default values of
'auth' from 'md5, plaintext' to 'md5'.

Patch is safe in terms of backwards compatibility, will work even if
remote peers have auth=plaintext and we have defaults.

auth=plaintext setting will remain deprecated in Asterisk 14 and 15,
and IAX2 plaintext support will be removed in Asterisk 16.

ASTERISK-22820 #close

Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf

contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.

Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4

Make it clear that we're talking about device state hints and add
an entry to the sample config.

Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433

This patch adds a new PJSIP specific dialplan function,
PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
session will be refreshed via either an UPDATE or re-INVITE request.
When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
the formats in use on a PJSIP channel can be re-negotiated and changed
dynamically after call setup.

ASTERISK-26277 #close

Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
(cherry picked from commit eec60dd7)

The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.

This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.

This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.

ASTERISK-26230 #close

Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a

On heavy loaded system with IMAP or DB storage,
'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
It could happen when the IMAP or DB server dies or is unreachable.
It could happen on startup when there are many (thousands)
realtime endpoints configured with unsolicited mwi.
If the taskprocessor queue reaches the high water level
then the alert is triggered and pjsip stops processing new requests
until the queue reaches the low water level to clear the alert.

This patch adds 2 new 'general' configuration options
to tune taskprocessor alert levels:
'tps_queue_high' - Taskprocessor high water alert trigger level.
'tps_queue_low' - Taskprocessor low water clear alert level

ASTERISK-26229 #close

Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8

Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
Session-Timers for SIP over TCP (and for SIP over TLS).

However with longer international calls via TCP, the SIP channel might break,
because all hops on the Internet route must stay online (have not a single power
outage, for example). Therefore with Session-Timers enabled (which are enabled
at default), you might see dropped calls. Consequently even with this change,
you might be better-off going for session-timers=refuse in your sip.conf.

ASTERISK-19968 #close

Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957

Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
this.

ASTERISK-26218 #close
ASTERISK-26221 #close
Reported by: Aaron Meriwether

Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa

Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc

The new option allows the channel driver's faxdetect option to timeout on
a call after the specified number of seconds into a call.  The new feature
is disabled if the timeout is set to zero.  The option is disabled by
default.

* Don't clear dsp_features after passing them to the dsp code in
my_pri_ss7_open_media().  We should still remember them especially for the
new faxdetect_timeout option.

ASTERISK-26214
Reported by: Richard Mudgett

Change-Id: Ieffd3fe788788d56282844774365546dce8ac810

The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call.  The new feature is disabled if the timeout is set
to zero.  The option is disabled by default.

ASTERISK-26214
Reported by: Richard Mudgett

Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d

Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
for DTLS. The source code from main/tcptls.c should have been re-used to ease
security audits. Therefore, this change rolls back the change from July 2015 and
re-uses the code from July 2014. This has the additional benefits to work under
CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.

ASTERISK-25659 #close
Reported by: StefanEng86, urbaniak, pay123
Tested by: sarumjanuch, traud
patches:
res_rtp_asterisk.patch submitted by sarumjanuch
dtls_centos_step_1.patch submitted by traud
dtls_centos_step_2.patch submitted by traud

Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c

The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.

1) It restarted any OPTIONS RTT ping cycle.

2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.

3) It cleared the RTT time each time the endpoint was refreshed.

4) The cleared RTT time was sent out as a statsd update each time.

5) It created two AMI events for each update.

* Revert the original patch and reimplement it.  Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration.  The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.

ASTERISK-26160 #close
Reported by: Matt Jordan

Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1

If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.

ASTERISK-25471 #close

Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514

ASTERISK-18995 #close

Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a

We should not be processing any incoming messages until we are fully
booted.  We may not have dialplan or other needed configuration loaded
yet.

ASTERISK-26089 #close
Reported by: Scott Griepentrog

ASTERISK-26088
Reported by:  Richard Mudgett

Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264

Testing has shown that our usage of UnixODBC is problematic
due to bugs within UnixODBC itself as well as the heavy weight
cost of connecting and disconnecting database connections, even
when pooling is enabled.

For users of UnixODBC 2.3.1 and earlier crashes would occur due
to insufficient protection of the disconnect operation. This was
fixed in UnixODBC 2.3.2 and above.

For users of UnixODBC 2.3.3 and higher a slow-down would occur
under heavy database use due to repeated connection establishment.
A regression is present where on each connection the database
configuration is cached again, with the cache growing out of
control.

The connection pool implementation present in this change helps
to mitigate these issues by reducing how much we connect and
disconnect database connections. We also solve the issue of
crashes under UnixODBC 2.3.1 by defaulting the maximum number of
connections to 1, returning us to the previous working behavior.
For users who may have a fixed version the maximum concurrent
connection limit can be increased helping with performance.

The connection pool works by keeping a list of active connections.
If the connection limit has not been reached a new connection is
established. If the connection limit has been reached then the
request waits until a connection becomes available before
continuing.

ASTERISK-26074 #close
ASTERISK-26054 #close

Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff

Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.

This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.

The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.

ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson

Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665

Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])

Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.

Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9

Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.

Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.

ASTERISK-26059 #close

Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2

If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192.  While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.

In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.

To facilitate determination of format names, the format name has been
added to "core show codecs".

ASTERISK-26070 #close

Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b

Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended

With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.

AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.

Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.

ASTERISK-25925 #close
Reported by Mark Michelson

Change-Id: I42cbec7730d84640a434d143a0d172a740995543

As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.

Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.

ASTERISK-26011

Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576

Add the option 'enable_callee_prompt' to followme.conf. Enabled by
default. If disabled, a callee is not prompted to accept or reject
the forwarded call.

ASTERISK-26064 #close

Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

func_odbc was changed in Asterisk 13.9.0
to make func_odbc use a single database connection per DSN
because of reported bug ASTERISK-25938
with MySQL/MariaDB LAST_INSERT_ID().

This is drawback in performance when func_odbc is used
very often in dialplan.

Single database connection should be optional.

ASTERISK-26010

Change-Id: I7091783a7150252de8eeb455115bd00514dfe843

This patch adds a new feature to ARI that allows a client to download
the media associated with a stored recording. The new route is
/recordings/stored/{name}/file, and transmits the underlying binary file
using Asterisk's HTTP server's underlying file transfer facilities.

Because this REST route returns non-JSON, a few small enhancements had
to be made to the Python Swagger generation code, as well as the
mustache templates that generate the ARI bindings.

ASTERISK-26042 #close

Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181

Some SBCs require responses to empty SIP INFO packets
after establishing call via INVITE, if not responded to
they may drop your call after unspecified timeout of X minutes.

They are identified by having no Content-Type, check for this
and respond with 200 - OK message.

ASTERISK-24986 #close
Reported-by: Ilya Trikoz, Federico Santulli

Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0

Added a new multi_user option that when specified allows a particular
configuration to be used for multiple users. It does this by replacing
the user portion of the server uri with a dynamically created one.

Two new API calls have been added in order to make use of the new
functionality:

ast_sip_publish_user_send - Sends an outgoing publish message based on the
given user. If state for the user already exists it uses that, otherwise
it dynamically creates new outbound publishing state for the user at that
time.

ast_sip_publish_user_remove - Removes all outbound publish state objects
associated with the user. This essentially stops outbound publishing for
the user.

ASTERISK-25965 #close

Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc

* Provide consistent indenting of lines in bulleted paragraphs
* Respect the 80 character column width
* Group all like items together, e.g., all dialplan applications under
  "Applications", etc.
* Use a single blank line to break up functionality changes within a
  larger section
* Use two blanks lines to delineate larger sections

Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647

Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.

Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.

In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.

It's important to note the following:
 - If an offset is provided to the 'play' operations, it only applies to the
   first media URI, as it would be weird to skip n seconds forward in every
   media resource.
 - Operations that control the position of the media only affect the current
   media being played. For example, once a media resource in the list
   completes, a 'reverse' operation on a subsequent media resource will not
   start a previously completed media resource at the appropiate offset.
 - This patch does not add any new operations to control the list. Hopefully,
   user feedback and/or future patches would add that if people want it.

ASTERISK-26022 #close

Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f