git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Remove some dead code dealing with: AST_BRIDGE_REC_CHANNEL_0, AST_BRIDGE_REC_CHANNEL_1, and AST_BRIDGE_IGNORE_SIGS.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This resolves several deadlocks in chan_sip relating to usage of
ast_channel_bridge_peer and improves accessibility of lock debugging
function calls.

Review: https://reviewboard.asterisk.org/r/2756/
(closes issue ASTERISK-22215)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396723 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396713 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_vpb: Effectively remove native support.  Left enough bread crumbs to be able to convert later if needed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-21944)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_misdn: Effectively remove native support.  Left enough bread crumbs to be able to convert later if needed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patches:
    reviewboard-2377.patch uploaded by Paul Belanger
Review: https://reviewboard.asterisk.org/r/2377/
........

Merged revisions 396582 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 396583 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patches:
    reviewboard-2155.patch uploaded by Paul Belanger
Review: https://reviewboard.asterisk.org/r/2155/
........

Merged revisions 396579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 396580 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This makes it so that we can detect failures to originate as with
earlier versions of Asterisk, which restores the Asterisk 11 behavior
for the originate manager action. This was causing the ACL tests for
SIP and IAX2 to fail since those tests expected originate failures
when ACLs would cause rejections. Also, this patch fixes crashes in
chan_sip when ACLs rejected peers during registration verification.

(closes issue ASTERISK-22212)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This code adds chan_dahdi the command 'dahdi create channels <range>'
(where <range> is a single <n>-<m> or 'new') and updates 'dahdi destroy
channel' with a similar 'dahdi destroy channels'. It allows DAHDI
channels and spans to be added after the initial channel load
(without destroying all other channels as in 'dahdi restart').

It also includes some fixes to the D-Channel / span destruction code
(r394552).

This change is intended to provide a hook for a script running from
udev once a span has been assigned ("registered") / unassigned
("unregistered") for its channels. The udev hook configures the span's
channels with dahdi_cfg -S, and can then ask Asterisk to create ethe
channels. See the scripts added to DAHDI-tools in 2.7.0.

Review: https://reviewboard.asterisk.org/r/1598/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396474 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
   "implementation detail flag" on the channel technology. This tells
   consumers of Stasis that the creation of this channel is an implementation
   detail in Asterisk and can be ignored (if they so choose). This
   consolidates the conference recorder/announcer flags as well - these flags
   had no additional meaning beyond "ignore this channel please".

2. It modifies allocation of a channel in two ways:
   (a) If a channel technology can be determined from the name, we set it
       directly in the allocation routine. This prevents the initial
       publication of the message from going out with a NULL channel technology
       where possible. This lets Stasis consumers get the right channel
       technology on the first publication.
   (b) It reorganizes allocation to make use of the 'finalized' property on the
       channel. This was already used to know that a channel had completely
       finished its construction in the masquerade routine; now we also use it
       to know whether or not the setting of certain channel properties is
       occurring during or post construction. The various set routines were
       modified accordingly as well.

3. The masquerade event is now dead, Jim. It no longer served any purpose
   whatsoever - if you perform a call pickup you'll get a Pickup event;
   if you perform an attended transfer you will still get those events; if you
   steal a channel to put it elsewhere you'll get the corresponding NewExten or
   BridgeEnter events.

Review: https://reviewboard.asterisk.org/r/2740


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 - Fix different issues with call transfer cancel. In case 3rd party busy or congestion call was not returned.
 - Fix displaying soft button 'Redial' in case of no redial number exists
........

Merged revisions 396377 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.

This patch sets the dialog's transport based on the transport that was defined
in the register line.  If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.

(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
    asterisk-21964-set-reg-dialog-transport.diff
					by Michael L. Young (license 5026)
........

Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 396248 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396253 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396145 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.

(closes issue ASTERISK-22039)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2717



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It adds support for externally initiated parking requests. In particular,
   chan_skinny has a protocol level message that initiates a call park.
   This patch now supports that option, as well as the protocol specific
   mechanisms in chan_dahdi/sig_analog and chan_mgcp.
 * A parking bridge features virtual table has been added that provides
   access to the parking functionality that the Bridging API needs. This
   includes requests to park an entire 'call' (with little or no additional
   information, thank you chan_skinny), perform a blind transfer to a parking
   extension, determine if an extension is a parking extension, as well as the
   actual "do the parking" request from the Bridging API.
 * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
   functions
 * The removal of some - but not all - dead parking code from features.c

This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)

Review: https://reviewboard.asterisk.org/r/2710

(closes issue ASTERISK-22134)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 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

Just a merge error due to a file rename. Grrr...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395971 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.

To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.

In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:

     single_topic  ---------------->  all_topic
           ^
           |
     single_topic_cached  ----+---->  all_topic_cached
                              |
                              +---->  cache

This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.

Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.

(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch makes it so that all registration attempts that fail that
also permanently modify the registration state will raise an appropriate
AMI event.

Note that this patch was forward ported to trunk and the Stasis Core
message bus by mjordan.

(closes issue ASTERISK-21368)
Reported by: Dmitriy Serov
patches:
  chan_sip.c.diff uploaded by Demon (license 6479)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395907 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22135)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2707



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2692/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.

A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.

(closes issue ASTERISK-22130)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22150)

Review: https://reviewboard.asterisk.org/r/2696/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

analog_lib_handles --> dahdi_analog_lib_handles
enable_dtmf_detect --> dahdi_dtmf_detect_enable
disable_dtmf_detect --> dahdi_dtmf_detect_disable
dahdi_enable_ec --> dahdi_ec_enable
dahdi_disable_ec --> dahdi_ec_disable
update_conf --> dahdi_conf_update
dahdi_link --> dahdi_master_slave_link
dahdi_unlink --> dahdi_master_slave_unlink

(closes issue ASTERISK-22129)
Reported by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Created a native_dahdi bridging technology for use with the new bridging
API.

The new bridging technology is part of the chan_dahdi channel driver
because it is very specific to that driver.  Rather than include the new
code directly into chan_dahdi.c the new bridge technology is in its own
file and linked into chan_dahdi.so.  A large part of this change is the
mechanical process of moving declarations around so chan_dahdi.c can be
split up into more files later.

* Changed the bridging core to pass NULL frames into the channel
technologies instead of discarding them.  The channel technologies may
need the proding to determine if their configuration is still valid.

(closes issue ASTERISK-21886)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2681/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch introduces DTLS-SRTP support to chan_pjsip and the options
necessary to configure it including an option to allow choosing between
32 and 80 byte SRTP tag lengths.

During the implementation and testing of this patch, three other bugs
were found and their fixes are included with this patch. The two in
chan_sip were a segfault relating to DTLS setup and mistaken call
rejection. The third bug fix prevents chan_pjsip from attempting to
perform bridge optimization between two endpoints if either of them is
running any form of SRTP.

Review: https://reviewboard.asterisk.org/r/2683/
(closes issue ASTERISK-21419)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This allows modules outside of chan_pjsip itself to get the session given
only an Asterisk channel.

Review: https://reviewboard.asterisk.org/r/2674/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds some of the more recent control frame types to the IAX2
parser. When IAX2 debugging is enabled, it will now show more of the control
frame types.

(closes issue ASTERISK-22120)
Reported by: Birger "WIMPy" Harzenetter
patches:
  iaxcmds.diff uploaded by wimpy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.

This adds tests for blind transfers, several types of attended
transfers, and call pickup.

The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.

Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.

(closes issue ASTERISK-21592)
Reported by: Matt Jordan

(closes issue ASTERISK-21593)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2670/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Temporary peers do not have an associated Stasis endpoint and quite a
bit of code in chan_sip assumes that all peers have a Stasis endpoint.
All endpoint accesses in chan_sip are now wrapped in an endpoint
NULL-check.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394795 65c4cc65-6c06-0410-ace0-fbb531ad65f3