Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.

Review: https://reviewboard.asterisk.org/r/3721/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Specifically the following equivalents were created:
fax show session -> FAXSession
fax show sessions -> FAXSessions
fax show stats -> FAXStats

Review: https://reviewboard.asterisk.org/r/3666/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.

This patch changes the behaviour of this setting in the following ways:

1) In sip_write(), do not pass the media unless we have either progressed
   beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
   media is both set-up and wanted. This helps resolve double-ringing on some
   buggy handsets.

2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
   SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
   enabling early media. Avoid sending double ring indications.

NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.

Review: https://reviewboard.asterisk.org/r/3700

ASTERISK-23972 #close
Reported by: Steve Davies
patches:
  inband_never_present_early_media2 uploaded by Steve Davies (License 5012)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The commit that added libxml2 support didn't fully check for the libxml2
development script in the Asterisk configure file. As a result, Asterisk could
be configured, then fail on menuselect. This patch fixes it so that Asterisk
should detect the libxml2 dependency failure first.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the final patch in adding menuselect to Asterisk.
 - The first patch (r418832) added menuselect along with mxml
 - The second patch (r418833) removed mxml from menuselect

This patch adds support for libxml2 to menuselect, and makes libxml2 a
required library for Asterisk.

Note that the libxml2 portion of this patch was written by Sean Bright,
and was made available on a team branch:
  http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/

Review: https://reviewboard.asterisk.org/r/3773/

ASTERISK-20703 #close
patches:
  some_mysterious_team_branch uploaded by seanbright (License 5060)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the second patch that adds menuselect to Asterisk trunk. The previous
commit (r418832) added menuselect along with mxml; this patch removes mxml
completely from Menuselect.

A subsequent patch will switch menuselect over to using libxml2, and make
libxml2 a required dependency for Asterisk.

ASTERISK-20703


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the first patch that adds menuselect to Asterisk trunk, and removes
the svn:externals property. This is being done for two reasons:
(1) The removal of external repositories eases a future migration to git
(2) Asterisk is now the only thing that uses menuselect; as a result, there's
    little need to keep it in an external repository

Subsequent patches will remove the mxml dependency from menuselect and tidy
up the build system.

ASTERISK-20703


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure that three-way transfers can be reported even if featuremap is
non-NULL.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418811 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The include was unneeded, this is split off from r3758 as it applies to 12.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.

This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.

Review: https://reviewboard.asterisk.org/r/3724/

ASTERISK-24000 #close
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for the PostgreSQL application_name connection setting.
When the appropriate PostgreSQL module's configuration is set with an
application name, the name will be passed to PostgreSQL on connection and
displayed in the database's pg_stat_activity view, as well as in CSV logs. This
aids in managing which applications/servers are connected to a PostgreSQL
database, as well as tracing the activity of those connections.

Review: https://reviewboard.asterisk.org/r/3591

ASTERISK-23737 #close
Reported by: Gergely Domodi
patches:
  pgsql_application_name.patch uploaded by Gergely Domodi (License 6610)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Technically, ADPCM is a method that can be applied to several codecs.
Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec.

See http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information
about said codec.

Review: https://reviewboard.asterisk.org/r/3744

patches:
  rb3744.patch uploaded by dennis.guse (License 6513)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418738 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the PresenceState action is executed, the nominal path fails to include
the ActionID in the successful response. This patch adds a call to
astman_start_ack, which guarantees that an ActionID (if provided) will be
sent back to the AMI client.

Unlike the Asterisk 11 and 12 patches, this patch also deprecates the
duplicate Message key in the response to the action, replacing it with the
key 'PresenceMessage'.

Review: https://reviewboard.asterisk.org/r/3776/

ASTERISK-23985 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes two reference leaks that would occur when TEST_FRAMEWORK was
enabled and features were successfully executed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418716 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously these two dialplan functions would issue warnings and
return failure when an empty string is used as the argument. Now
they will not issue a warning and will successfully return an
empty string.

ASTERISK-23911 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3745/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418654 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It's been 2014 for like... 6 months.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Update refcounter.py to use delta's to track the current reference count.
* Use result from internal_ao2_ref to write old_refcount to refs_log.

Review: https://reviewboard.asterisk.org/r/3756/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change maps the ao2_t_cleanup() function to the
correct debug function so that it can be used.

Review: https://reviewboard.asterisk.org/r/3764/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418488 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Cleanup games object in app_skel.
* Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+).

Review: https://reviewboard.asterisk.org/r/3757/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698.

Review: https://reviewboard.asterisk.org/r/3755/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Tags are useful in hunting down ref imbalances; this patch adds tag variants
for these commonly used macros/functions.

Review: https://reviewboard.asterisk.org/r/3750/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change causes ao2_replace to do nothing when src == dst. This
avoids REF_DEBUG logging when we're not actually doing anything.

Review: https://reviewboard.asterisk.org/r/3743/
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Merged revisions 418396 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When updated configuration is written back to the conf
file - for example when a user changes their voicemail
pin, make sure that any config hook that wants to know
of changes is informed.

Review: https://reviewboard.asterisk.org/r/3708/
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Merged revisions 418366 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 418369 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is a whitespace only change.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Square pegs in round holes don't work very well.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418264 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.

* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.

* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.

* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.

AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/3720/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418226 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The dtls_perform_handshake function was mistakenly placed under the guards for
USE_PJPROJECT. If PJPROJECT was not installed, the function would not be
defined, while other functions would attempt to still use it. This prevented
res_rtp_asterisk from being loaded.

ASTERISK-24001 #close
Reported by: Don Fanning
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Merged revisions 418172 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This module implements dialog-info+xml for the purposes of presence. This means
that phones such as Grandstreams can now subscribe to receive presence information
for an extension.

ASTERISK-21443 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3705/
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Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418117 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes two bugs:

1. When originating a channel into a Stasis application, we already create a
   subscription for the channel that is going into our Stasis app.
   Unfortunately, when you create a Local channel and pass it off to a Stasis
   app, you really aren't creating just one channel: you're creating two. This
   patch snags the second half of the Local channel pair (assuming it is a
   Local channel pair, but luckily core_local is kind about such assumptions)
   and subscribes to it as well.

2. Subscriptions are a bit sticky right now. If a subscription is made, the
   'interest' count gets bumped on the Stasis subscription - but unless
   something explicitly unsubscribes the channel, said subscription sticks
   around. This is not much of a problem is a user is creating the subscription
   - if they made it, they must want it. However, when we are creating
   implicit subscriptions, we need to make sure something clears them out.
   This patch takes a pessimistic approach: it watches the cache updates
   coming from Stasis and, if we notice that the cache just cleared out an
   object, we delete our subscription object. This keeps our ao2 container of
   Stasis forwards in an application from growing out of hand; it also is a
   bit more forgiving for end users who may not realize they were supposed to
   unsubscribe from that channel that just hung up.

Review: https://reviewboard.asterisk.org/r/3710/
#ASTERISK-23939 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.

It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.

The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.

This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.

Review: https://reviewboard.asterisk.org/r/3690/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418067 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Egads. Another bad deletion of too much when attempting to remove h323 stuff.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418050 65c4cc65-6c06-0410-ace0-fbb531ad65f3