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

Review: https://reviewboard.asterisk.org/r/3440/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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/
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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

Not sure why that change to ast_channel_alloc was made but ... okay.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for the Japanese language to both the say family of
applications, as well as for VoiceMail and VoiceMailMain. A new pack of
language sounds will be released at the same time as the next major version
of Asterisk to support the new language features.

The language features can be enabled using a language code of 'ja'.

Review: https://reviewboard.asterisk.org/r/3477

ASTERISK-23324 #close
Reported by: Kevin McCoy
patches:
  app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
  say.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417591 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is essentially a backport of a small portion of r397526 from
ASTERISK-21981. In that patch, pass through support and format attribute
negotiation was added for Opus. Part of that included being more tolerant to
whitespace in the fmtp line of an SDP; that part of the patch is being
applied here.

As the author of the backport pointed out, in SDP, the fmtp line is allowed to
include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001)
includes an example for this. This was not removed in the updated RFC 4867 in
2007.

Review: https://reviewboard.asterisk.org/r/3658

#ASTERISK-23916 #close
Reported by: Alexander Traud
patches:
  sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417590 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In v12+ the type values from the table are only used by the CEL unit
tests.  Since the unit tests were only comparing a generated expected
event with a real event to see if the ie contents matched and using the
same table IE_PLTYPE values to read the event contents, the type
mismatches were not detected.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change ensures that __ao2_ref_debug writes to ref_log when given a
non-NULL pointer to an invalid ao2 object.  This is to ensure that we
record any attempt manipulate references of already freed objects.

ASTERISK-23948 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3677/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When processing a 212MB refs file, refcounter.py used over 3GB of RAM.
This change greatly reduces memory usage in two ways:

* Saving object history in whole lines instead of separated values.
* Not saving normal/skewed/leaked object lists unless they are requested.

ASTERISK-23921 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3668/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417485 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A number of various PJSIP AMI actions were failing to parse out and place the
ActionID into their responses. This patch updates the various PJSIP actions
such that the passed in ActionID is emitted on any event list complete events,
as well as any intermediate events created as a result of the action.

#ASTERISK-23947 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3675/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417461 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update the CEL unit tests that handle BRIDGE_ENTER and BRIDGE_EXIT
events to expect the "bridge_technology" extra field key.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add missing CHANGES changes from r417361 and r417383.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417423 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks to Sean Bright for pointing out that this was missed in #asterisk-dev.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This will drive the test on review r3419. Note that the patch for this was
done by Ben Ford, although it was slightly modified for this commit.

ASTERISK-23562
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add the "bridge_technology" extra field key to BRIDGE_ENTER and
BRIDGE_EXIT CEL events to convey the bridge technology in use at the
time the record was generated.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417383 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows the current owner of a channel to define various
feature hooks to be made available once the channel has entered a
bridge. This includes any hooks that are setup on the
ast_bridge_features struct such as DTMF hooks, bridge event hooks
(join, leave, etc.), and interval hooks.

Review: https://reviewboard.asterisk.org/r/3649/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch enables the jack-audiohook to cope with dynamic sampling rates from
and to Asterisk. Information from the channel is taken to derive the channel's
sampling rate, suiting SLINxx format and frame->datalen.

There are stil a few limitations after this patch:
* Required information is taken from the channel during initialization as
  the audiohook does not provide this information.
  Audiohook.internal_sampl_rate(...) is set later, but no callback is available
  to inform app_jack.

* Frame.datalen is computed using "rate / 50" assuming a ptime of 20ms.
  There is no internal API available to determine datalen for a SLINxx.

* Ringbuffer size is now dynamic depending on the value of frame.datalen
  (see above) and the number of frames, which are in RINGBUFFER_FRAME_CAPACITY,
  that need to fit.

Review: https://reviewboard.asterisk.org/r/3618

Note that the patch being committed here is based on the patch posted on
ASTERISK-23836. However, Matthis Schmieder also provided a patch to enable
this functionality, and that patch is noted below.

ASTERISK-20696 #close
Reported by: Matthis Schmieder
patches:
  app_jack.patch uploaded by Matthis Schmieder (License 6445)

ASTERISK-23836 #close
Reported by: Dennis Guse
patches:
  patch-app_jack.c uploaded by Dennis Guse (License 6513)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using FEC, with span=3 and entries=4 Asterisk will attempt to repair
the packet with sequence number 5, as it will see that packet -4 is
missing. The result is Asterisk sending garbage packets that can kill a
fax.

This patch adds a check to see if the sequence number is valid before
checking if the packet is missing.

Review: https://reviewboard.asterisk.org/r/3657/

#ASTERISK-23908 #close
Reported by: Torrey Searle
patches:
  udptl_fec.patch uploaded by Torrey Searle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417327 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
   websocket to respond to pings. As such, Asterisk maintains a reference to
   the session during the loop. When ast_http_websocket_write fails, it may
   cause the session to decrement its ref count, but this in and of itself
   does not break the read loop. The read loop's write, on the other hand,
   does not break the loop if it fails. This causes the socket to get in a
   'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
   fails with a large volume of data when the client takes awhile to process
   the information. When it does fail, it fails writing only a portion of
   the bytes. With some debugging, it was shown that this was failing in a
   similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
   with a long enough timeout solved the problem.

Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.

#ASTERISK-23917 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3624/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

From headers were processed using a 256 character buffer on the stack.
This change replaces that with a heap allocation by ast_strdup.

ASTERISK-23790 #close
Reported by: uniken1
Tested by: uniken1
Review: https://reviewboard.asterisk.org/r/3669/
Patches:
    chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417251 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This helps to pave the way for RLS work that is to come.
Since this is a self-contained change and subscription
tests still pass, this work is being committed directly
to trunk instead of a working branch.

ASTERISK-23865 #close
Review: https://reviewboard.asterisk.org/r/3628



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417233 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Almost every reference operation against container node's uses
__ao2_alloc or __ao2_ref, thereby preventing ref logging for
the nodes.  One node reference is released with ao2_t_ref, causing
refcounter.py to falsely report skews and leaks for many nodes.

ASTERISK-23922 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3670/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Move eid functions from netsock.c to utils.c.  These functions were
already published by utils.h.  Flag netsock.h as deprecated and switch
res_pjsip_session.h to use netsock2.h.  The only code that still uses
netsock.h is chan_iax2.

ASTERISK-23920 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23834 #close
Reported by: Richard Kenner
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Appending the ;2 to the user supplied ;1 uniqueid to create the ;2 version
if the user did not also supply an extra uniqueid for the ;2 channel
resulted in allocating a buffer that was one byte too small.

* Fix off by one error in ast_unreal_new_channels() when generating the ;2
uniqueid from the user suppled ;1 version.

* Pulled some long assignment lines from if tests to improve line break
readability in ast_unreal_new_channels().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a DAHDI span disappears, we wish for its representation in Asterisk
to be destroyed as well.

The information about the span's removal may come from several paths:

1. DAHDI sends DAHDI_EVENT_REMOVE on every channel.
2. An extra DAHDI_EVENT_REMOVED is sent on every subsequent call to
   DAHDI_GET_EVENT.
3. Every read (including the internal one by libpri on the D-channel)
   returns -ENODEV.

Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by destroying it.

Destroying a channel requires holding the channel list lock (iflock).
Destroying a channel that is part of a span requires holding the span's
lock. Destroying a channel from a context that holds the span lock,
while at the same time another channel is destroyed directly, leads to a
deadlock. Solution: don't destroy span while holding the channels list
lock.

Thus changes in this patch:
* Deferring removal of PRI spans in response to events: doomed spans
  are collected on a list.
* Doomed spans are removed periodically by the monitor thread.
* ENODEV reads from the D-channel will warant the same deferred removal.

Review: https://reviewboard.asterisk.org/r/3548/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417059 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This macro replaces one object reference with another cleaning up the original.

param dst Pointer to the object that will be cleaned up.
param src Pointer to the object replacing it.

src's ref count is bumped if it's non-NULL.
dst's ref count is decremented if it's non-NULL.
src is assigned to dst,

This patch was reviewed on IRC by coreyfarrell and mjordan.
 
Tested by: George Joseph
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416996 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_ext_tool_check.m4 isn't handling cases where a path to a package is
provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config
tool (E.G. mysql_config) and the package has its own subdirectories in include
or lib.  For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql
but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib.
libxml2 has the same problem with its includes.  They're in 
${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include.
Both cause configure to fail and there are others in the same boat.

The problem is caused by logic in ast_ext_tool_check that overrides the result
of the config tool's --cflags and --libs options if package_DIR is set.

This patch prepends package_DIR (if specified) to the -L and -I results from
the package's config tool instead of overriding them.

A regenerated ./configure and include/asterisk/autoconfig.h.in are included
but can be regenerated by running ./bootstrap.sh at any time.

Tested by: George Joseph
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3550/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416935 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_ext_tool_check.m4 isn't handling cases where a path to a package is
provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config
tool (E.G. mysql_config) and the package has its own subdirectories in include
or lib.  For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql
but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib.
libxml2 has the same problem with its includes.  They're in 
${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include.
Both cause configure to fail and there are others in the same boat.

The problem is caused by logic in ast_ext_tool_check that overrides the result
of the config tool's --cflags and --libs options if package_DIR is set.

This patch prepends package_DIR (if specified) to the -L and -I results from
the package's config tool instead of overriding them.

Tested by: George Joseph
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3550/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416872 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously module information was not included due to an oversight.

Review: https://reviewboard.asterisk.org/r/3626/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Part of a series of AMI command equivalents to existing CLI
commands

Review: https://reviewboard.asterisk.org/r/3651/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Extract the sayname API call to its own registerd callback.  This allows
the app_directory and app_chanspy applications to say a mailbox owner's
name using an alternate provider when app_voicemail is not available
because you are using res_mwi_external.  app_directory still uses the
voicemail.conf file.

AFS-64 #close
Reported by: Mark Michelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Move some implementation specific code from astobj2_container.c into
astobj2_hash.c and astobj2_rbtree.c.  This completely removes the need for
astobj2_container to switch on RTTI and it no longer has any knowledge of
the implementation details.

Also adds AO2_DEBUG as a new compile option in menuselect which controls
astobj2 debugging independently of AST_DEVMODE and REF_DEBUG.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3593/
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Merged revisions 416806 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr.
* Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask
  instead of ast_sockaddr_stringify_addr.
* Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead
  of ast_ha_join() for the CLI output.

This is a CLI change only.  AMI was not affected.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3652/
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Merged revisions 416737 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416738 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 416733 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 416734 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416736 65c4cc65-6c06-0410-ace0-fbb531ad65f3