This fixes two reference leaks that would occur when TEST_FRAMEWORK was
enabled and features were successfully executed.
........

Merged revisions 418715 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 418641 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 418649 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 418650 from http://svn.asterisk.org/svn/asterisk/branches/12


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

........

Merged revisions 418586 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 418505 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 418506 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 418465 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 418466 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

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/
........

Merged revisions 418366 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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.
........

Merged revisions 418323 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 418324 from http://svn.asterisk.org/svn/asterisk/branches/12


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.
........

Merged revisions 418261 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 418262 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 418263 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 418225 from http://svn.asterisk.org/svn/asterisk/branches/12


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

........

Merged revisions 418182 from http://svn.asterisk.org/svn/asterisk/branches/12


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
........

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/
........

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
........

Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12


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/
........

Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12


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

........

Merged revisions 418066 from http://svn.asterisk.org/svn/asterisk/branches/12


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

The previous patch (r418034) fixed the 'glitch' that the channels/h323
Makefile no longer existed. Unfortunately, removing the entire line was a bit
of a blunder, as it meant that build_tools/menuselect-deps was never
generated. Hilarity ensued when actually trying to compile.

But hey! At least configure worked.

This patch fixes *that* glitch, and removes some more of the vestiges of h323.
(It had tendrils in the main Makefile? Crazy.)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418035 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This simply removes that check from the configure script, as r418019 removed
chan_h323.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.

Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.

The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.

For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.

And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.

To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.

Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.

We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.

It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.

And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.

Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.

This patch removes:

* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge

It removes the following applications/functions:

* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO

It removes the colon delimiter from the SIPPEER function.

Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.

Review: https://reviewboard.asterisk.org/r/3698/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.

Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP.  It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed.  This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.

NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.

The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.

This commit is a merge of the two patches indicated below.

ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
      pri-4.diff (license #6302) patch uploaded by Pavel Troller
      jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett

Review: https://reviewboard.asterisk.org/r/3633/
........

Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Removed some incorrect newlines on ast_http_error() messages in
manager.c.

* Removed an incorrect newline in res_ari_channels.c.

Addendum to ASTERISK-23552
........

Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417933 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds the following AMI commands:
PRIDebugSet - Set PRI debug levels for a specific span
PRIDebugFileSet - Set the file used for PRI debug message output
PRIDebugFileUnset - Disables file output for PRI debug messages

Review: https://reviewboard.asterisk.org/r/3681/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417916 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds two new manager commands to pbx_config - DialplanExtensionAdd and
DialplanExtensionRemove which allow manager users to create and delete
extensions respectively.

Review: https://reviewboard.asterisk.org/r/3650/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Persistent HTTP connection support is needed due to the increased usage of
the Asterisk core HTTP transport and the frequency at which REST API calls
are going to be issued.

* Add http.conf session_keep_alive option to enable persistent
connections.

* Parse and discard optional chunked body extension information and
trailing request headers.

* Increased the maximum application/json and
application/x-www-form-urlencoded body size allowed to 4k.  The previous
1k was kind of small.

* Removed a couple inlined versions of ast_http_manid_from_vars() by
calling the function.  manager.c:generic_http_callback() and
res_http_post.c:http_post_callback()

* Add missing va_end() in ast_ari_response_error().

* Eliminated unnecessary RAII_VAR() use in http.c:auth_create().

ASTERISK-23552 #close
Reported by: Scott Griepentrog

Review: https://reviewboard.asterisk.org/r/3691/
........

Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417901 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The patch for ASTERISK-23905 that added PFS support in Asterisk depends on the
elliptic curve library support being present in OpenSSL. As it turns out, some
versions of OpenSSL don't have this library - notably the version running on
our build agents.

This patch fixes the build by providing a configure check for the specific
library calls that the PFS patch relies on.

Review: https://reviewboard.asterisk.org/r/3709/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The variables body parameter under the originate and originate with id
operations of the channel resource showed invalid JSON in its description.

The variables body parameter under the userEvent operation of the event
resource made no mention that the custom key/value pairs should be wrapped
in a variables key in order to be added to the custom user event.

ASTERISK-23975 #close

Review: https://reviewboard.asterisk.org/r/3692/
........

Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the api.wiki.mustache template and the swagger_model python
script to understand if an operation has a body parameter. If an operation
does have a body parameter, it will now be displayed in the corresponding
wiki entry.
........

Merged revisions 407389 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Install a hook script for DAHDI to register new spans with Asterisk
automatically by running:

  asterisk -rx 'dahdi create channel FIRST LAST'

Review: https://reviewboard.asterisk.org/r/3157/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch enables Perfect Forward Secrecy (PFS) in Asterisk's core TLS API.
Modules that wish to enable PFS should consider the following:

- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
  specify a ECDHE cipher suite in a module's configuration, for example:
  tlscipher=AES128-SHA:DES-CBC3-SHA

- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
  into the private key file, i.e., tlsprivatekey. For an example, see the
  default dh2048.pem at
http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt

- Because clients expect the server to prefer PFS, and because OpenSSL sorts
  its cipher suites by bit strength, (see "openssl ciphers -v DEFAULT")
  consider re-ordering your cipher suites in the conf file. For example:
tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
  will use PFS when offered by the client. Clients which do not offer PFS
  fall-back to AES-128 (or even 3DES as recommend by RFC 3261).

Review: https://reviewboard.asterisk.org/r/3647/

ASTERISK-23905 #close
Reported by: Alexander Traud
patches:
  tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520)
  tlsPFS.patch uploaded by Alexander Traud (License 6520)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417803 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A loop in ast_careful_fwrite exists that will continually attempt to write to
a file stream, even in the presence of EAGAIN/EINTR errors. However, if a
connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to
fflush will return EOF but not clear errno, resulting in an infinite loop.

This patch clears errno after it is detected and handled the loop, such that
any subsequent call to fflush will not get erroneously stuck.

Review: https://reviewboard.asterisk.org/r/3704

#ASTERISK-23984 #close
Reported by: Steve Davies
patches:
  fflush_loop_fix uploaded by one47 (License 5012)
........

Merged revisions 417797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 417798 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 417799 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417800 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3440/
........

Merged revisions 412653 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 417705 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417706 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.

This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).

ASTERISK-22961 #close
Reported by: Jay Jideliov

Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
........

Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the abstraction effort, this bit of logic got messed up. We
want to recreate the persistence if things go well, not if things
fail.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417663 65c4cc65-6c06-0410-ace0-fbb531ad65f3