Commit Graph

300 Commits

Author SHA1 Message Date
Kinsey Moore 59753b1ea1 Strip down the old event system
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.

Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 14:39:27 +00:00
Mark Michelson f8622e7c5c Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels.
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
2013-08-02 14:05:07 +00:00
Matthew Jordan 0d86021842 Add pickup.h include lines for chan_dahdi and chan_mgcp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 02:51:33 +00:00
Matthew Jordan 5c4b482471 Support externally initiated parking requests; remove some dead code
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
2013-08-01 20:55:17 +00:00
David M. Lee e1b959ccbb Split caching out from the stasis_caching_topic.
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
2013-08-01 13:49:34 +00:00
Matthew Jordan cafc115896 A great big renaming patch
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
2013-07-25 04:06:32 +00:00
Richard Mudgett b4dd4ffbc4 Some chan_dahdi protected function renaming.
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
2013-07-23 16:26:36 +00:00
Matthew Jordan d0a55fa52d Refactor RTCP events over to Stasis; associate with channels
This patch does the following:

* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
  information in the RTCP events. Because Stasis provides a cache, Jaco's
  patch was modified to pass the channel uniqueid to the RTP layer as
  opposed to a pointer to the channel. This has the following benefits:
  (1) It keeps the RTP engine 'clean' of references back to channels
  (2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
  Potentially, other implementations (such as res_rtp_multicast) could also
  raise RTCP information. The engine provides structs to represent RTCP headers
  and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
  RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
  but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
  assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
  raise an event when we sent a RR report.

Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.

Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.

Review: https://reviewboard.asterisk.org/r/2603/

(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
  asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)

(closes issue ASTERISK-21471)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
Matthew Jordan 6258bbe7bd Update Asterisk's CDRs for the new bridging framework
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
    This means CDRs track well with what an actual channel is doing - which
    is useful in transfer scenarios (which were previously difficult to pin
    down). It does, however, mean that CDRs cannot be 'fooled'. Previous
    behavior in Asterisk allowed for CDR applications, channels, and other
    properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
    be what everyone wants, but it is a defined behavior and as such, it is
    predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
    changes have been made to ResetCDR and ForkCDR in particular. Many of the
    options for these two applications no longer made any sense with the new
    framework and the (slightly) more immutable nature of CDRs.

There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.

(closes issue ASTERISK-21196)

Review: https://reviewboard.asterisk.org/r/2486/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
Mark Michelson 2dc8a06006 Refactor the features configuration scheme.
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.

In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.

Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.

Review: https://reviewboard.asterisk.org/r/2578/

(issue ASTERISK-21542)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06 21:40:35 +00:00
Mark Michelson fac3839e68 Adds support for a core attended transfer function plus adds some hiding of masquerades.
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.

The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.

Review: https://reviewboard.asterisk.org/r/2511

(closes issue ASTERISK-21334)
Reported by Matt Jordan

(closes issue Asterisk-21336)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-28 14:45:31 +00:00
Jason Parker 154fbf8cae Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:21:25 +00:00
Matthew Jordan 06be8463b6 Migrate a large number of AMI events over to Stasis-Core
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
 * ChanSpyStart/Stop
 * MonitorStart/Stop
 * MusicOnHoldStart/Stop
 * FullyBooted/Reload
 * All Voicemail/MWI related events

In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.

Review: https://reviewboard.asterisk.org/r/2532

(closes issue ASTERISK-21462)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 20:44:07 +00:00
Richard Mudgett 3d63833bd6 Merge in the bridge_construction branch to make the system use the Bridging API.
Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:00:22 +00:00
Richard Mudgett 213834ffd4 Remove some unnecessary calls to ast_bridged_channel() in chan_mgcp.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01 20:01:27 +00:00
Kinsey Moore 1a2a4578d2 Convert MWI state message type to the new stasis naming convention
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27 22:42:06 +00:00
Kinsey Moore 6300aa6ae4 Make sure things compile...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16 16:00:40 +00:00
Kinsey Moore 99aa02d17f Transition MWI to Stasis-core
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.

Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16 15:45:58 +00:00
Mark Michelson da85f8489f Make evaluation of channel variables consistently case-sensitive.
Due to inconsistencies in how variable names were evaluated, the
decision was made to make all evaluations case-sensitive. See the
UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity
for more details.

(closes issue ASTERISK-20163)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2160


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-29 21:27:09 +00:00
Andrew Latham 3820f1586e Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:47:40 +00:00
Andrew Latham 99e1174bfa Doxygen Cleanup
Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-01 23:24:35 +00:00
Andrew Latham fd98835f1f Doxygen Updates Janitor Work
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-22 20:43:30 +00:00
Kinsey Moore b5a6de76fc Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-14 19:44:27 +00:00
Kinsey Moore a485f44022 Add missing newlines to CLI logging
........

Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 18:19:03 +00:00
Terry Wilson 0e5c761c28 Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01 22:09:18 +00:00
Terry Wilson a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 16:52:47 +00:00
Sean Bright 51c24c88a1 Prefer ast_set_qos() over ast_netsock_set_qos()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27 14:13:58 +00:00
Terry Wilson ebaf59a656 Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24 00:32:20 +00:00
Terry Wilson 57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Terry Wilson 34c55e8e7c Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-13 17:27:06 +00:00
Richard Mudgett 23bc964e1c Constify some more channel driver technology callback parameters.
Review: https://reviewboard.asterisk.org/r/1707/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-01 19:53:38 +00:00
Terry Wilson 99cae5b750 Opaquify channel stringfields
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 20:12:09 +00:00
Terry Wilson 04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Richard Mudgett 10de040b6e More parking issues.
* Fix potential deadlocks in SIP and IAX blind transfer to parking.

* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter).  Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.

* Made masq_park_call() handle a failed ast_channel_masquerade() setup.

* Reduced excessive struct parkeduser.peername[] size.
........

Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-18 21:15:45 +00:00
Matthew Jordan 8b5ba33fe0 Merged revisions 335078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
  
  Merged revisions 335064 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
    
    Updated SIP 484 handling; added Incomplete control frame
    
    When a SIP phone uses the dial application and receives a 484 Address 
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.
    
    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to 
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.
    
    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan
    
    Review: https://reviewboard.asterisk.org/r/1416/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09 16:28:23 +00:00
Leif Madsen a525edea59 Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 20:28:54 +00:00
Tilghman Lesher 7d179abfd4 Merged revisions 326411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
  
  Add the attribute "type" to each "<use>" for menuselect.
  
  This matters only when autoconf fails to detect that weak linking is supported.
  External optional dependencies will become optional in both cases, as they are
  removed at compile time when not detected.  However, runtime-optional modules
  are made mandatory when weak linking is not found.  This change affects only
  the external optional dependencies; previously, they were incorrectly required
  when weak linking support was not detected.
  
  Patches:
  	20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
  
  Tested by: iasgoscouk
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-05 22:11:40 +00:00
Russell Bryant 0938974902 Merged revisions 317478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
  
  Fix some consistency issues with jitterbuffer config.
  
  Store the defaults noted in the sample config files in the jitterbuffer config
  data structure.  This makes the CLI commands that output these settings show
  the right thing.  Also only show the settings that are relevant in the settings
  CLI commands, based on which jitterbuffer is selected and whether it's enabled.
  
  (closes issue #19083)
  Reported by: rgagnon
  Patches:
        issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-05 22:55:09 +00:00
Russell Bryant 95561bd37a Merged revisions 316336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) | 8 lines
  
  Use htons() instead of ntohs() in some places.
  
  (closes issue #19200)
  Reported by: wdoekes
  Patches:
        issue19200-trunk.patch uploaded by wdoekes (license 717)
        issue19200-1.8.x.patch uploaded by wdoekes (license 717)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03 22:16:23 +00:00
Richard Mudgett e2b21c4942 Merged revisions 315349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 Apr 2011) | 9 lines
  
  When using MGCP realtime gateway definitions, random crashes occur.
  
  Fixed incorrect linked list node removal for realtime gateways.
  
  (closes issue #18291)
  Reported by: nahuelgreco
  Patches:
        dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-25 21:55:00 +00:00
David Vossel c26c190711 Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:22:10 +00:00
Russell Bryant cc0b7e7df5 Some scheduler API cleanup and improvements.
Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation.  However, if you used it, it required using different
functions for modifying scheduler contents.  This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there.  This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.

In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.

Review: https://reviewboard.asterisk.org/r/1007/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20 17:15:54 +00:00
Jeff Peeler 41b95ee887 Merged revisions 286931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
  
  Add parking extension for non-default parking lots.
  
  This is a new feature that allows for parking to custom parking lots to be
  accessed directly, rather than with channel variables or by changing the
  default parking lot. The extension is set with the parkext option just as the
  default parking lot is done. Also, the manager action has been updated to
  optionally allow a specified parking lot.
  
  (closes issue #14882)
  Reported by: vmikhnevych
  Patches: 
        patch_14882.txt uploaded by mnick (license 874)
        modified by me
  
  Review: https://reviewboard.asterisk.org/r/884/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 19:23:56 +00:00
Tilghman Lesher 8190e96fad Merged revisions 284610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
  
  When optional_api is non-optional, force dependent modules to be loaded.
  
  (closes issue #17707)
   Reported by: ira
   Patches: 
         20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman
   
  Review: https://reviewboard.asterisk.org/r/876/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02 05:27:53 +00:00
Tilghman Lesher ca661f4702 Merged revisions 280879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 Aug 2010) | 14 lines
  
  Check cur value before attempting a deref.
  
  (closes issue #17775)
   Reported by: svinson
   Patches: 
         20100804__issue17775.diff.txt uploaded by tilghman (license 14)
   Tested by: svinson
  
  (closes issue #17743)
   Reported by: tgruenberg
   Patches: 
         20100804__issue17775.diff.txt uploaded by tilghman (license 14)
   Tested by: tgruenberg
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-04 14:05:11 +00:00
Tilghman Lesher b4e18d5660 Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20 19:35:02 +00:00
Richard Mudgett cf7bbcc4c6 Expand the caller ANI field to an ast_party_id
Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.

This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/

Review: https://reviewboard.asterisk.org/r/744/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 16:58:03 +00:00
Richard Mudgett ec37ffbdaf ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
	char *str;
	int char_set;
	int presentation;
	unsigned char valid;
};
struct ast_party_number {
	char *str;
	int plan;
	int presentation;
	unsigned char valid;
};
struct ast_party_subaddress {
	char *str;
	int type;
	unsigned char odd_even_indicator;
	unsigned char valid;
};
struct ast_party_id {
	struct ast_party_name name;
	struct ast_party_number number;
	struct ast_party_subaddress subaddress;
	char *tag;
};
struct ast_party_dialed {
	struct {
		char *str;
		int plan;
	} number;
	struct ast_party_subaddress subaddress;
	int transit_network_select;
};
struct ast_party_caller {
	struct ast_party_id id;
	char *ani;
	int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.


The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review:	https://reviewboard.asterisk.org/r/702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
Mark Michelson cd4ebd336f Add IPv6 to Asterisk.
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.

Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.

(closes issue #17565)
Reported by: russell
Patches: 
      asteriskv6-test-report.pdf uploaded by russell (license 2)

Review: https://reviewboard.asterisk.org/r/743



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08 22:08:07 +00:00
Tzafrir Cohen c613897d1c Fix various typos reported by Lintian
(Also fix the typos in the comments)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-02 15:57:02 +00:00