This patch adds a new operation for stored recordings, copy. It takes an
existing stored recording and makes a copy of it in the same directory
or a relative directory under the stored recording directory.

/ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}

This is particularly useful for voicemail-esque applications, which may need to
copy or move recordings around a directory structure.

Review: https://reviewboard.asterisk.org/r/3768/

ASTERISK-24036 #close
Reported by: Sam Galarneau
Tested by: Sam Galarneau
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a reference count goes negative, instead of
just logging that fact, be more helpful with a
backtrace and an assert that will DO_CRASH.

This patch also removes the duplicate ao2_bt()
function and cleans up extraneous usage of the
ast_log_backtrace() call.

Review: https://reviewboard.asterisk.org/r/3765/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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

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@418587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 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

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418397 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

* 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

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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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

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

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/
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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

........
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

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

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

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

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

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

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

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

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

There was a problem when reading a string from the websocket. It assumed the
received data had a null terminator and tried to write the data to an ast_str.
This of course could/would read past the end of the given buffer while
writing the data to the internal buffer of ast_str. Modified the the code to
correctly place a null terminator on the result string.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416394 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
 * AGI execution
 * Returning objects for ARI commands
 * During some Local channel operations
 * During some dialling operations
 * During variable setting
 * During some bridging operations
And more.

This patch does the following:
 - It removes a number of fields from channel snapshots. These fields were
   rarely used, were expensive to have on the snapshot, and hurt performance.
   This included formats, translation paths, Log Call ID, callgroup, pickup
   group, and all channel variables. As a result, AMI Status,
   "core show channel", "core show channelvar", and "pjsip show channel" were
   modified to either hit the live channel or not show certain pieces of data.
   While this is unfortunate, the performance gain from this patch is worth
   the loss in behaviour.
 - It adds a mechanism to publish a cached snapshot + blob. A large number of
   publications were changed to use this, including:
   - During Dial begin
   - During Variable assignment (if no AMI variables are emitted - if AMI
     variables are set, we have to make snapshots when a variable is changed)
   - During channel pickup
   - When a channel is put on hold/unhold
   - When a DTMF digit is begun/ended
   - When creating a bridge snapshot
   - When an AOC event is raised
   - During Local channel optimization/Local bridging
   - When endpoint snapshots are generated
   - All AGI events
   - All ARI responses that return a channel
   - Events in the AgentPool, MeetMe, and some in Queue
 - Additionally, some extraneous channel snapshots were being made that were
   unnecessary. These were removed.
 - The result of ast_hashtab_hash_string is now cached in stasis_cache. This
   reduces a large number of calls to ast_hashtab_hash_string, which reduced
   the amount of time spent in this function in gprof by around 50%.

#ASTERISK-23811 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3568/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23673 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/3617/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection.  Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.

A similar problem exists if a HTTP request is started but never finished.

* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything.  Defaults to 30000 ms.

* Removed the undocumented manager.conf block-sockets option.  It
interferes with TCP/TLS inactivity timeouts.

* AMI and SIP TLS connections now have better authentication timeout
protection.  Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.

* chan_sip can now handle SSL certificate renegotiations in the middle of
a session.  It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.

* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.

The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability.  This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.

This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.

ASTERISK-23673 #close
Reported by: Richard Mudgett
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This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default
this uses the local astdb but it can also be configured to store within an outside
database. When Asterisk is started these subscriptions are recreated if they have not
expired. Notifications are sent to the devices which have subscribed and they are none
the wiser that the system has restarted.

Review: https://reviewboard.asterisk.org/r/3598/
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This patch is a re-do of r414122.

When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft
hangup flags have a catastrophic effect on the pbx core if they leak out from
the bridge layer: the channel gets hung up. With the number of threads
involved in a blind transfer, and with the initial patch, it was likely that
this would occur. This caused a large number of test failures

This patch is nearly identical with the one proposed in r414122, save for the
following changes:
 - We explicitly clear the UNBRIDGE flag when setting an after goto on a
   channel in a bridge
 - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it

https://reviewboard.asterisk.org/r/3585/
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Prior to this patch, sequential variables would be ordered in reverse
from the order specified in the manager action.

Review: https://reviewboard.asterisk.org/r/3588/
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Split astobj2.c into the following files to improve maintainability.

astobj2.c - object primitives, object primitive misc and initialization code.
astobj2_private.h - internal object declarations needed by the containers.
astobj2_container.c - generic conainer and container misc code.
astobj2_container_hash.c - hash container specific code.
astobj2_container_rbtree.c - rbtree container specific code.
astobj2_container_private.h - generic container definitions and rtti prototypes.

https://reviewboard.asterisk.org/r/3576/
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