When an object is being unlinked from its container *and* being returned to
the caller, we do not want to decrement the reference count after unlinking
it from the container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining reference to
the object, that could result in the object being destroyed.



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  r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
  
  Fix documentation for ast_softhangup() and correct the misuse thereof.
  (closes issue #16103)
   Reported by: majorbloodnok
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.

(closes issue #15604)
Reported by: alecdavis
Patches:
      asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
      Some minor modificatons were made.
Tested by: alecdavis, rmudgett

Review: https://reviewboard.asterisk.org/r/405/


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  r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
  
  IAX2: VNAK loop caused by signaling frames with no destination call number
  
  It is possible for the PBX thread to queue up signaling frames before
  a destination call number is received.  This can result in signaling
  frames being sent out with no destination call number. Since recent
  versions of Asterisk require accurate destination callnumbers for all
  Full Frames, this can cause a VNAK loop to occur.  To resolve this
  no signaling frames are sent until a destination callnumber is received,
  and destination call numbers are now only required for iax_pvt matching
  when the frame is an ACK.
  
  Review: https://reviewboard.asterisk.org/r/413/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIPShowPeer AMI action.

(closes issue #15990)
Reported by: _brent_
Patches:
      sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388)

Review: https://reviewboard.asterisk.org/r/381/



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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/



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  r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines
  
  Revert 225169, as this doesn't account for the possibility of a list of frames.
........


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  r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines
  
  Isolate the frame returned from ast_translate().
........


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  r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines
  
  Suffix is not needed for a match
........


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This is the second commit for this and documents the text stream using the configured
IP address and fixes a bug in the original patch where the UDPTL stream would also
use the different IP address.

(closes issue #14729)
Reported by: _brent_
Patches:
      media_address.patch uploaded by brent (license 388)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Fixes SWP-238)


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........
  r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
  
  IAX/SIP shrinkcallerid option
  
  The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
  and '-' from the string.  This means values such as 555.5555 and
  test-test result in 555555 and testtest.  There are instances,
  such as Skype integration, where a specific value is passed via
  caller id that must be preserved unmodified.  This patch makes
  the shrinking of caller id optional in chan_sip and chan_iax in
  order to support such cases.  By default this option is on to
  preserve previous expected behavior.
  
  (closes issue #15940)
  Reported by: dimas
  Patches:
        v2-15940.patch uploaded by dimas (license 88)
        15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
  Tested by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/408/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14729)
Reported by: _brent_
Patches:
      media_address.patch uploaded by brent (license 388)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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........
  r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
  
  Isolate frames returned from a DSP instance or codec translator.
  
  The reasoning for these changes are the same as what I wrote in the commit
  message for rev 222878.
........


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........
  r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
  
  Pay attention to the return value of the manipulate function.
  While this looks like an optimization, it prevents a crash from occurring
  when used with certain audiohook callbacks (diagnosed with SVN trunk,
  backported to 1.4 to keep the source consistent across versions).
........


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  r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines
  
  Add support for relaying early media in the features attended transfer option.
  
  (closes issue #14828)
  Reported by: licedey
........


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........
  r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines
  
  Correct timestamp calculations when RTP sample rates over 8kHz are used.
  
  While testing some endpoints that support 16kHz and 32kHz sample rates, some
  log messages were generated due to calc_rxstamp() computing timestamps in a way
  that produced odd results, so this patch sanitizes the result of the
  computations.
........


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(closes issue AST-29)


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........
  r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines
  
  Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
  
  (closes issue #14763)
  Reported by: cupotka
........


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Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This corrects an issue reported on the -users list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions.


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........
  r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines
  
  Fix stale caller id data from being reported in AMI NewChannel event
  
  The problem here is that chan_dahdi is designed in such a way to set
  certain values in the dahdi_pvt only once. One of those such values
  is the configured caller id data in chan_dahdi.conf. For PRI, the
  configured caller id data could be overwritten during a call. Instead
  of saving the data and restoring, it was decided that for all non-analog
  channels it was simply best to not set the configured caller id in the
  first place and also clear it at the end of the call.
  
  (closes issue #15883)
  Reported by: jsmith
........


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........
  r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines
  
  Never released PRI channels when using Busy() or Congestion() dialplan apps.
  
  When the Busy() or Congestion() application is used towards ISDN (an ISDN
  progress is sent), the responding ISDN Disconnect or Release may contain
  the ISDN cause user busy or one of the congestion causes.  In chan_dahdi.c
  these causes will only set the needbusy or needcongestion flags and not
  activate the softhangup procedure.  Unfortunately only the latter can
  interrupt the endless wait loop of Busy()/Congestion().
  
  Result: PRI channels staying in state busy for the rest of asterisk life
  or until the other end times out and forces the call to clear.
  
  (issue #14292)
  Reported by: tomaso
  Patches:
        disc_rel_userbusy.patch uploaded by tomaso (license 564)
        (This patch is unrelated to the issue.)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Option o)

(closes issue #15675)
Reported by: john8675309
Patches:
      issue15675patchtrunk.txt uploaded by dbrooks (license 790)
Tested by: jgutierrez on users list:
 http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Explains new options for detecting DTMF CID on fxo lines

(issue #9096)
Reported by: fleed
Patches:
      chan_dahid_sample_config.patch uploaded by sum (license 766)



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Ability has been added to both manager command SIPnotify as well as console
command sip notify. Message body is stored in the "Content" variable. An 
example is present in sip_notify.conf.

(closes issue #13926)
Reported by: jthurman
Patches:
      sip-notify-svn189463.diff uploaded by gareth (license 208)
Tested by: gareth


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