The functionality this patch attempts to achieve should already
be possible using [general](+) in the config file.

issue #17957



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The original implementation of this did the merging of all contexts with the
same name in the realtime layer, but that implementation severely breaks
drivers which use the same context name (e.g. iax.conf, type={peer,user}).
Therefore, the implementation needs to do the merging for particular entries
only, based upon what contexts would allow that in the channel driver itself.
This implementation is for chan_sip only, but others could be added in the
future.

(closes issue #17957)
 Reported by: marcelloceschia
 Patches: 
       chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079)
 Tested by: tilghman


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Includes a new highly optimized and customizable
ConfBridge application capable of mixing audio at
sample rates ranging from 8khz-192khz.

Review: https://reviewboard.asterisk.org/r/1147/



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Review: https://reviewboard.asterisk.org/r/1157/


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(closes issue #19076)
Reported by: lmadsen
Patches: 
      __20110408-channel-description.txt uploaded by lmadsen (license 10)
Tested by: lmadsen

Review: https://reviewboard.asterisk.org/r/1163/

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If the user invokes 'dialplan add extension' into a non-existing context, the context will be created
and a message informing the user of the context being created will be issued in cli.

(closes issue #17431)
Reported by: leearcher
Patches:
      context_auto_create.diff uploaded by kobaz (license 834)
Tested by: leearcher, kobaz, jrose


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In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


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Adds an option to FollowMe that isn't useful for the bug it was made to solve.  Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost.

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Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.

Review: https://reviewboard.asterisk.org/r/1134/


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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


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From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.

(closes issue #17905)
Reported by: rcasas
Patches: 
      app_meetme.c.patch uploaded by rcasas (license 641)

Review: https://reviewboard.asterisk.org/r/874/



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(closes issue #17946)
Reported by: bluecrow76
Patches:
      asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff


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Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.

The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.

JIRA SWP-2845
JIRA ABE-2736


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The display ie handling can be controlled independently in the send and
receive directions with the following options:

* Block display text data.

* Use display text in SETUP/CONNECT messages for name.

* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).

* Pass arbitrary display text during a call.  Sent in INFORMATION
messages.  Received from any message that the display text was not used as
a name.

If the display options are not set then the options default to legacy
behavior.

The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.

To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.

JIRA SWP-2688
JIRA ABE-2693


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Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of:  asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands.  thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1075/


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Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.

Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.

JIRA SWP-2687
JIRA ABE-2691

Review:	https://reviewboard.asterisk.org/r/1063/


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(closes issue #17979)
 Reported by: tilghman
 Patches: 
       20100911__for_blitzrage.diff.txt uploaded by tilghman (license 14)
 Tested by: lmadsen


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(closes issue #17668)
 Reported by: tilghman
 Patches: 
       20100718__logger_reload_altconf__2.diff.txt uploaded by tilghman (license 14)
 
Review: (by lmadsen, russell within comments on issue tracker)


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Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file.  If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.

Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
	meetme_final_patch_v.diff uploaded by parisioa (license 1153)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/984/


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........
  r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) | 2 lines
  
  Update CHANGES to reflect new gtalk.conf options.
........


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........
  r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) | 2 lines
  
  Add note about the checkhangup option of ${CHANNEL()}
........


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........
  r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
  
  Addition of the FrameHook API (AKA AwesomeHooks)
  
  So far all our tools for viewing and manipulating media streams
  within Asterisk have been entirely focused on audio.  That made
  sense then, but is not scalable now.  The FrameHook API lets us
  tap into and manipulate _ANY_ type of media or signaling passed
  on a channel present today or in the future.  This tool is a step
  in the direction of expanding Asterisk's boundaries and will help
  generate some rather interesting applications in the future.
  
  In addition to the FrameHook API, a simple dialplan function
  exercising the api has been included as well.  This function
  is called FRAME_TRACE().  FRAME_TRACE() allows for the internal
  ast_frames read and written to a channel to be output.  Filters
  can be placed on this function to debug only certain types of frames.
  This function could be thought of as an internal way of doing
  ast_frame packet captures.
  
  Review: https://reviewboard.asterisk.org/r/925/
........



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........
  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/
........


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........
  r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line
  
  Added missing documentation for ExternalIVR feature added in January 2010
........


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........
  r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
  
  authenticate OPTIONS requests just like we would an INVITE
  
  OPTIONS requests should be treated the same as an INVITE
  This includes authentication.  This patch adds the ability for
  incoming out of dialog OPTION requests to be authenticated
  before providing a response indicating whether an extension
  is available or not.  The authentication routine works the
  exact same way as it does for incoming INVITEs.  This means
  that if a peer has 'insecure=invite' in their peer definition,
  the same will be true for the processing of the OPTIONS request.
  
  Review: https://reviewboard.asterisk.org/r/881/
........


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........
  r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
  
  remove current STUN support from chan_sip.c
  
  This patch removes the current broken/useless stun
  support from chan_sip.
  
  (closes issue #17622)
  Reported by: philipp2
  
  Review: https://reviewboard.asterisk.org/r/855/
........


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  r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
  
  res_stun_monitor and corresponding options CHANGES documentation
........


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........
  r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
  
  Add a "core reload" CLI command.
  
  Review: https://reviewboard.asterisk.org/r/859/
........


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........
  r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
  
  improved translation paths for wideband codecs
  
  The problem I'm addressing is that Asterisk's current
  method of building the least cost translation paths
  between codecs does not take into account sample rate.
  For instance, it was possible for siren14 (a 32khz codec),
  to contain the a translation path to siren7 (a 16khz
  audio codec) that goes through slin at 8khz.  In this
  case Asterisk takes a 32khz codec, down samples it to
  8khz and then up samples it to 16khz which is terrible
  regardless if it is computationally less expensive.  This
  patch now builds translation paths that give priority to
  maintaining the best possible sample rate before taking
  into consideration computational cost.  This patch also
  adds cli commands to expose what translation paths are
  actually being used.
  
  Changes:
  1. Translation paths will never contain a step that changes
  the sample rate unless absolutely necessary.
  2. When choosing the best codec to make two channels compatible.
  Shared codecs with the highest sample rate are given priority.
  3. A new cli command to show all translation paths available
  for a specific codec 'core show translation paths [codec name]'
  has been added.
  4. 'core show translation' which displays the translation
  matrix now includes the new higher bit audio codecs in the table.
  5. 'core show channel [channel name]'  now displays the
  translation paths if translation is used.
  
  (closes issue #16841)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/842/
........


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........
  r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines
  
  Sneak FIELDNUM() into 1.8.  Returns a 1-based index into a list of a specified item.
  
  Matches up with FIELDQTY() and CUT().
  
  (closes issue #17713)
   Reported by: gareth
   Patches: 
         svn-279754.diff uploaded by gareth (license 208)
   Tested by: gareth, tilghman
  
   Review: https://reviewboard.asterisk.org/r/810/
........


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(closes issue #17730)
Reported by: jkroon
Patches:
      iax2-peerstate-address.patch uploaded by jkroon (license 714)
Tested by: lmadsen


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........
  r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines
  
  Updated documentation for FAX logger level.
........


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........
  r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines
  
  Add documentation for FAX logger level.
  
  (closes issue #17715)
  Reported by: vrban
  Patches:
        17715.patch uploaded by pabelanger (license 224)
  Tested by: vrban
........


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