Commit Graph

74 Commits

Author SHA1 Message Date
Kinsey Moore abd3e4040b Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Corey Farrell fbe0dfaf44 Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
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Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27 19:21:44 +00:00
Matthew Jordan ce423d2ea4 func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
 * RTP information, including source/destination media addresses, whether or
   not the media is secure, held, and other properties.
 * RTCP information. This includes sets of parseable information, as well as
   individual statistic attriutes.
 * PJSIP information. This includes URIs, local/remote signalling addresses,
   whether or not the signalling is secure, and other properties.
 * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
   function to obtain more detailed endpoint information.

Review: https://reviewboard.asterisk.org/r/3038/
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Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11 13:06:30 +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 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
Matthew Jordan 9d8a5ceb02 Move after bridge callbacks into their own file
One more major refactoring to go.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-25 02:20:23 +00:00
Jonathan Rose f3fcf0aa2e func_channel: dtmf_features setting
Allows reading andsetting dtmf features via a channel function
CHANNEL(dtmf_features)

(closes issue ASTERISK-21876)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2648/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23 21:32:33 +00:00
Matthew Jordan 88e344d2df Clean up documentation
This patch cleans up documentation in func_channel for the following items:
* rtpsource
* secure_signaling
* secure_media
* various OOH323 parameters

(closes issue ASTERISK-20969)
Reported by: snuffy
patches:
  func_chan-update.diff uploaded by snuffy (License 5024)
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Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394981 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-21 22:09:23 +00:00
Jonathan Rose d014ad2261 func_channel: Read/Write after_bridge_goto option
Allows reading and setting of a channel's after_bridge_goto datastore

(closes issue ASTERISK-21875)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2628/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-26 20:59:14 +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
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
Matthew Jordan 1cafccdf87 Update documentation for CHANNEL function
Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
........

Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-03 17:17:33 +00:00
Richard Mudgett da5944fc56 Named call pickup groups. Fixes, missing functionality, and improvements.
* ASTERISK-20383
Missing named call pickup group features:

CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.

* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up.  In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.

Regression because of the named call pickup group feature.

* See ASTERISK-20386 for the implementation improvements.  These are the
changes in channel.c and channel.h.

* Fixed some locking issues in CHANNEL().

(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2112/
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Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20 17:22:41 +00:00
Kinsey Moore 9b16c8b0f6 Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 20:21:43 +00:00
Richard Mudgett ac35b92b62 Hangup handlers - Dialplan subroutines that run when the channel hangs up.
Hangup handlers are an alternative to the h extension.  They can be used
in addition to the h extension.  The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up.  Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel.  You
can attach multiple handlers that will execute in the order of most
recently added first.

(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2002/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29 17:02:32 +00:00
Michael L. Young a7a3050de9 Add documentation to function CHANNEL for options echocan_mode and buffers
The ability to set "echocan_mode" and "buffers" through the dialplan was added
to chan_dahdi some time ago.  This patch adds some documentation to
func_channel.

(Closes issue ASTERISK-19911)
Reported by: Dale Noll
Tested by: Michael L. Young
Patches: 
  asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1949/
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Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368093 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-01 03:30:01 +00:00
Kinsey Moore a485f44022 Add missing newlines to CLI logging
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Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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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 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
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
Gregory Nietsky fca9962766 Merged revisions 338995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines
  
  Remove the channel function OOH323() and place its options into
  CHANNEL()
  
  channel drivers should not have there own dialplan functions.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03 14:24:45 +00:00
Leif Madsen a525edea59 Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  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
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 20:28:54 +00:00
Gregory Nietsky e789eb8b2d CHANNEL(pickupgroup)
Allow Setting / Reading the pickupgroup of a channel with func_channel.c
  
  (closes issue #19045)
  Reported by: irroot
  
  Review: https://reviewboard.asterisk.org/r/1148/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25 15:43:28 +00:00
Leif Madsen fd2f95b3c4 Merged revisions 314206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines
  
  Merged revisions 314205 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines
    
    Remove duplicate documentation from func_channel.c
    
    (closes issue #18970)
    Reported by: IgorG
    Patches: 
          func_channel.c.doc.diff uploaded by IgorG (license 20)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-19 14:28:46 +00:00
Richard Mudgett 928ec2b990 Merged revisions 309445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
  
  Get real channel of a DAHDI call.
  
  Starting with Asterisk v1.8, the DAHDI channel name format was changed for
  ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
  
  There were several reasons that the channel name had to change.
  
  1) Call completion requires a device state for ISDN phones.  The generic
  device state uses the channel name.
  
  2) Calls do not necessarily have B channels.  Calls placed on hold by an
  ISDN phone do not have B channels.
  
  3) The B channel a call initially requests may not be the B channel the
  call ultimately uses.  Changes to the internal implementation of the
  Asterisk master channel list caused deadlock problems for chan_dahdi if it
  needed to change the channel name.  Chan_dahdi no longer changes the
  channel name.
  
  4) DTMF attended transfers now work with ISDN phones because the channel
  name is "dialable" like the chan_sip channel names.
  
  For various reasons, some people need to know which B channel a DAHDI call
  is using.
  
  * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
  CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
  in use by the channel.  Use CHANNEL(no_media_path) to determine if the
  channel even has a B channel.
  
  * Added AMI event DAHDIChannel to associate a DAHDI channel with an
  Asterisk channel so AMI applications can passively determine the B channel
  currently in use.  Calls with "no-media" as the DAHDIChannel do not have
  an associated B channel.  No-media calls are either on hold or
  call-waiting.
  
  (closes issue #17683)
  Reported by: mrwho
  Tested by: rmudgett
  
  (closes issue #18603)
  Reported by: arjankroon
  Patches:
        issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: stever28, rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04 15:28:20 +00:00
Richard Mudgett ffe9e4acfc Merged revisions 309170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines
  
  Document CHANNEL(keypad_digits) and CHANNEL(no_media_path).
  
  * Added XML documentation for CHANNEL(keypad_digits) and
  CHANNEL(no_media_path).
  
  * Tweaked XML documentation for CHANNEL(reversecharge).
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-01 21:57:58 +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
Terry Wilson d04046fbe7 Merged revisions 286189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines
  
  Merged revisions 286115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
    
    Merged revisions 286059 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
      
      Inherit CHANNEL() writes to both sides of a Local channel
      
      Having Local (/n) channels as queue members and setting the language in the
      extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
      channel. Hold time report playbacks happen on the Local/...,1 channel and
      therefor do not play in the specified language.
      
      This patch modifies func_channel_write to call the setoption callback and pass
      the CHANNEL() write info to the callback. chan_local uses this information to
      look up the other side of the channel and apply the same changes to it.
      
      (closes issue #17673)
      Reported by: Guggemand
      
      Review: https://reviewboard.asterisk.org/r/903/
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-10 22:15:47 +00:00
Tilghman Lesher 1c12ca0407 Merged revisions 285484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 Sep 2010) | 2 lines
  
  Documentation only
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-08 07:15:19 +00:00
Tilghman Lesher 2302618bb7 Merged revisions 285373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07 Sep 2010) | 7 lines
  
  Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up.
  
  (closes issue #17652)
   Reported by: kobaz
   Patches: 
         func_channel.patch uploaded by kobaz (license 834)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-07 21:14:54 +00:00
Terry Wilson 857814f435 Add SRTP support for Asterisk
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08 05:29:08 +00:00
Mark Michelson 7acfebf2b8 Adjust XML for func_channel to indicate that rtpdest can take a "text" argument.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-04 20:12:26 +00:00
Russell Bryant 34317fb0d3 Fix the ability to specify an OSP token for an outbound IAX2 call.
When this patch was originally submitted, the code allowed for the token to be
set via a channel variable.  I decided that a cleaner approach would be to
integrate it into the CHANNEL() function.  Unfortunately, that is not a suitable
approach.  It's not possible to get the value set on the channel soon enough
using that method.  So, go back to the simple channel variable method.

(closes issue #16711)
Reported by: homesick
Patches:
      iax-svn.diff uploaded by homesick (license 91)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-27 17:32:07 +00:00
Kevin P. Fleming cdd1f9e296 Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.

During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.

Review: https://reviewboard.asterisk.org/r/379/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 21:08:47 +00:00
Tilghman Lesher 2cfddf8cb6 Add MASTER_CHANNEL() dialplan function, as well as a useful usage.
(closes issue #13140)
 Reported by: cpina
 Patches: 
       20090807__issue13140.diff.txt uploaded by tilghman (license 14)
 Tested by: lmadsen
 Change-type: feature


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-01 23:41:06 +00:00
Tilghman Lesher 642bec4d6f AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:20:57 +00:00
Sean Bright 719917fe59 Support setting and receiving Reverse Charging Indication over ISDN PRI.
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI.  This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:

...,Dial(DAHDI/g1/C4445556666)

And to read it off an inbound channel:

exten => s,1,Set(RCI=${CHANNEL(reversecharge)})

Thanks again to rmudgett for the thorough review.

(closes issue #13760)
Reported by: mrgabu

Review: https://reviewboard.asterisk.org/r/303/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02 17:46:14 +00:00
Russell Bryant 0264eef115 Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 15:28:53 +00:00
Kevin P. Fleming 82fb56886e More 'static' qualifiers on module global variables.
The 'pglobal' tool is quite handy indeed :-)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15 17:34:30 +00:00
Leif Madsen 9408242796 Recorded merge of revisions 193544 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines
  
  Document CHANNEL(transfercapability) in CLI documentation.
  
  (issue #15073)
  Reported by: pkempgen
  Patches:
        20090511__issue15073.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-11 18:01:44 +00:00
Russell Bryant cba19c8a67 Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
Russell Bryant 16fc1993ef Add support for the "name" option in the CHANNEL() function.
Review: http://reviewboard.digium.com/r/199/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 21:28:04 +00:00
Russell Bryant 4ec301360c Merge a large set of updates to the Asterisk indications API.
This patch includes a number of changes to the indications API.  The primary
motivation for this work was to improve stability.  The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.

The changes included are:

1) Remove the module res_indications.  This included the critical functionality
   that actually loaded the indications configuration.  I have seen many people
   have Asterisk problems because they accidentally did not have an
   indications.conf present and loaded.  Now, this code is in the core,
   and Asterisk will fail to start without indications configuration.

   There was one part of res_indications, the dialplan applications, which did
   belong in a module, and have been moved to a new module, app_playtones.

2) Object management has been significantly changed.  Tone zones are now
   managed using astobj2, and it is no longer possible to crash Asterisk by
   issuing a reload that destroys tone zones while they are in use.

3) The API documentation has been filled out.

4) The API has been updated to follow our naming conventions.

5) Various bits of code throughout the tree have been updated to account
   for the API update.

6) Configuration parsing has been mostly re-written.

7) "Code cleanup"

The code is from svn/asterisk/team/russell/indications/.

Review: http://reviewboard.digium.com/r/149/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
Russell Bryant ef6ad2b53c Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines

Revert unnecessary indications API change from rev 122314

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13 19:22:13 +00:00
Russell Bryant 5b168ee34b Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00
Tilghman Lesher b2ef18dab4 Add some more IAX2-specific information about the channel to the CHANNEL()
function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
 Reported by: mostyn
 Patches: 
       iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
       (with some additional cleanup by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-15 15:21:16 +00:00
Brett Bryant c1451b5537 This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function 
for any channel that uses RTP.

(closes issue #10590)
Reported by: gasparz
Patches:
      chan_sip_c.diff uploaded by gasparz (license 219)
      rtp_c.diff uploaded by gasparz (license 219)
      rtp_h.diff uploaded by gasparz (license 219)
      audioqos-trunk.diff uploaded by snuffy (license 35)
      rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05 16:24:19 +00:00