Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.

Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7

This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.

This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.

ASTERISK-28018

Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06

AttendedTransfer queues up attended transfer to the given extension.

This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer

[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
   same => n,Return()

[my_transfer]
include => default
;;;

This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
atxfer => *7

[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer

[custom_atxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,AttendedTransfer(${dest})
   same => n,Return()

[my_transfer]
include => default
;;;

Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b

BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.

This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer

[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
   same => n,Return()
;;;

This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
blindxfer =>

[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer

[custom_blindxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,BlindTransfer(${dest},default)
   same => n,Return()
;;;

Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a

ASTERISK-28234
Reported-by: Kirsty Tyerman

Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1

One of the change files doesn't conform to the format that the release
scripts need in order to parse it.

Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c

This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.

Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1

You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:

[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address

This causes both 192.168.1.1 and 1.2.3.4 to be advertized.

Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db

When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.

This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.

ASTERISK-28401

Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c

Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.

Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.

ASTERISK-28363

Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1

Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.

res_pjsip_refer:  Configures PJSIP norefersub capability accordingly.

Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.

This is useful for Cisco switches that do not follow RFC4488.

ASTERISK-28375 #close
Reported-by: Dan Cropp

Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9

This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a change to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.

There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:

   Subject: res_pjsip
   Subject: Core

   A description that explains the changes made and why. The release
   script will handle the bulleting and section separators!

   You can still separate with new lines within your
   description.

The headers ("Subject" and "Master-Only") are case sensative, but the
value for "Master-Only" ("true" or "True") is not.

For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt

ASTERISK-28111 #close

Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47

This is the first step in changing the release process so that changes
made to the CHANGES and UPGRADE.txt files do not result in merge
conflicts every time multiple people modify these files. The changes
made will go in these new directories: doc/CHANGES-staging and
doc/UPGRADE-staging. The README.md files explain how things will work,
but here's a little overview. When you make a change that would go in
either CHANGES or UPGRADE.txt, this should instead be documented in a
new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory,
respectively. The format will look like this:

   Subject: res_pjsip

   A description that explains the changes made and why. The release
   script will handle the bulleting and section separators! The
   'Subject:' header is case-sensitive.

   You can still separate with new lines within your description.

   Subject: res_ari
   Master-Only: true

   You can have more than one subject, and they don't have to be the
   same! Also, the 'Master-Only' header should always be true and is
   also case-sensitive (but the value is not - you can have 'true' or
   'True'). This header will only ever be present in the master branch.

For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt

This is an initial change for ASTERISK_28111. Functionally, this will
make no difference, but it will prep the directories for when the
changes from CHANGES and UPGRADE.txt are extracted.

Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6

* Consistently use spaces in rest-api-templates/asterisk_processor.py.
* Exclude third-party from docs/full-en_US.xml.
* Add docs/full-en_US.xml to .gitignore.
* Use list() to convert python3 view.
* Use python3 print function.
* Replace cmp() with equivalent equation.
* Replace reference to out of scope subtype variable with name
  parameter.
* Use unescaping triple bracket notation in mustache templates where
  needed.  This causes behavior of Python2 to be maintained when using
  Python3.
* Fix references to has_websocket / is_websocket in
  res_ari_resource.c.mustache.
* Update calculation of has_websocket to use any().
* Use unicode mode for writing output file in transform.py.
* Replace 'from swagger_model import *' with explicit import of required
  symbols.

I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the
print syntax has been fixed.

Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33

This document is out of date and is superseded by content on the
Asterisk wiki.

ASTERISK-24386 #close

Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822

Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a

We should be sending people to secure web URL's where available.
Update README's and docs.

Change-Id: Id5b1e049b0b18b49a784f1254605aefa244ce19a

In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver.  Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel.  Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.

Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea

* Following the example of the PJSIP channel driver, the channel
  technology specific documentation has been moved to the respective
  channel drivers that provide that functionality. This has the benefit
  of locating the documentation of items with those modules that provide
  it.

* Examples of using the CHANNEL function for both standard items as well
  as for PJSIP have been added.

* The 'max_forwards' standard item has been documented.

Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b

We don't need pjproject's documentation embedded in Asterisk's.

Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17

ASTERISK-25189 #close
Reported by: John Hardin

Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac

Previous versions of Asterisk processed command-line options before
processing asterisk.conf.  This meant that if an option was set in
asterisk.conf, it could not be overridden with the equivelent command
line option.  This change causes Asterisk to process the command-line
twice.  First it processes options that are needed to load asterisk.conf,
then it processes the remaining options after the config is read.

This changes the function of -X slightly.  Previously using -X without
disabling execincludes in asterisk.conf caused #exec to be usable in any
config.  Now -X only enables #exec for the load of asterisk.conf, if it
is wanted in the rest of the system it must be enabled with execincludes
in asterisk.conf.  Updated 'asterisk -h' and 'man asterisk' to reflect
the limited function of -X.

ASTERISK-25042 #close
Reported by: Corey Farrell

Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7

Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in

Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly.  Updated clean
targets as well.

Updated /.gitignore and doc/.gitignore.

Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622

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 the .gitignore and .gitreview files to the asterisk repo.

NB:  You can add local ignores to the .git/info/exclude file
without having to do a commit.

Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.

Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph

ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
  escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a manpage for the smsq utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3895/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  smsq.8 uploaded by Jeremy Laine (License 6561)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a manpage for the aelparse utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3896/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  aelparse.8 uploaded by Jeremy Laine (License 6561)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for an <example /> tag in the XML documentation schema.

For CLI help, this doesn't change the formatting too much:
 - Preceeding white space is removed
 - Unlike with para elements, new lines are preserved

However, having an <example /> tag in the XML schema allows for the wiki
documentation generation script to surround the documentation with {code} or
{noformat} tags, generating much better content for the wiki - and allowing us
to put dialplan examples (and other code snippets, if desired) into the
documentation for an application/function/AMI command/etc.

Review: https://reviewboard.asterisk.org/r/3807/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.

Review: https://reviewboard.asterisk.org/r/3812/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027

Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.

(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
 transferred.patch uploaded by Jeremy Laine (license 6561)
 hyphen.patch uploaded by Jeremy Laine (license 6561)
 sip.conf.sample.patch uploaded by Eugene (license 6360)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.

(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.

Review: https://reviewboard.asterisk.org/r/3035
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2898/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2616

(closes issue ASTERISK-21542)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:

1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
   options that use the configuration framework

This patch include configuration documentation for the following modules:
 * chan_motif
 * res_xmpp
 * app_confbridge
 * app_skel
 * udptl

Two new CLI commands have been added:
 * config show help - show configuration help by module, category, and item
 * xmldoc dump - dump the in-memory representation of the XML documentation to
   a new XML file.

Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058

patches:
  on review 2058 uploaded by twilson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3