Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function.  Both
interfaces create an input and output JACK port.  The application makes
these ports the endpoint of the call.  The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
audiohook on the channel.  This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio.  This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.

In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/).  I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...


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audiohooks.  This causes an error when we attempt to destroy the lock later
when freeing the audiohook.


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variable (or function) on an active channel from the CLI.


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functions correctly.  (Closes issue #11749)


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(closes issue #11212)
Reported by: tzafrir
Patches:
      zap_dnd.diff uploaded by tzafrir (modified by me) (license 46)



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r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines

Add a connection timeout attribute, as that was what was intended with the
login timeout, but ODBC divides it up into 2 different timeouts.
(Closes issue #11745)

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run "make asterisk.pdf" when not all of the right packages are installed.

(closes issue #10763)
Reported by: Corydon76
Patches:
      20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76


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(closes issue #11690)
Reported by: tzafrir
Patches:
      signaling_to_signalling.diff uploaded by tzafrir (license 46)
      signalling_cleanup.diff uploaded by tzafrir (license 46)
      zap_auto_default.diff uploaded by tzafrir (license 46)
      zap_no_default_sig.diff uploaded by tzafrir (license 46)
      zap_signal_auto.diff uploaded by tzafrir (license 46)



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r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines

Fix up setting the EID on BSD based systems.

(closes issue #11646)
Reported by: caio1982
Patches:
      dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
      dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22)
Tested by: caio1982, mvanbaak

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r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines

If the incoming RTP stream changes codec force the bridge to break if the other side does not support it.
(closes issue #11729)
Reported by: tsearle
Patches:
      new_codec_patch_udiff.patch uploaded by tsearle (license 373)

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r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines

If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else.
(closes issue #11743)
Reported by: davevg
Patches:
      res_agi.diff uploaded by davevg (license 209)

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r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines

Properly report the hangup cause as no answer when someone does not answer

(closes issue #10574, reported by boch, patched by moy)


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a value from a double, to a float, back to a double.  Sure enough, when I changed
my interim variable back to a double, it still blows up.  Switching all of these
to a float fixes the problem.  This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.

(related to issue #11725)


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with platforms that explode on unaligned access.  I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on.  If it makes
sense to you, feel free to enlighten me.  :)

(closes issue #11725, patched by me)


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r98265 | russell | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines

Backport the ability to set the ToS bits on Linux when not running as root.
Normally, we would not backport features into 1.4, but, I was convinced by the
justification supplied by the supplier of this patch.  He pointed out that this
patch removes a requirement for running as root, thus reducing the potential
impacts of security issues.

(closes issue #11742)
Reported by: paravoid
Patches:
      libcap.diff uploaded by paravoid (license 200)

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r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines

Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely

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framein callbacks different.  However, they are now the same again, so remove
the duplicate code and use the same functions for the lin/lin16 versions.


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r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines

Back out changes from revision 97077, since it wasn't perfect

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r98082 | russell | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines

Fix samples vs. length calculations for g722

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   sample length with g722.  It is _2_ samples per byte, not 1.  This was all
   over the place, and I believed it, and it is what caused me to take so long
   to figure out what was broken.
 - Update copyright information on codec_g722.


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This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.

(closes issue #11722, reported and initially patched by caio1982, final patch by me)



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to set the qualify frequency.

(closes issue #11597)
Reported by: wilder
Patches:
      qualifyfreq5.patch uploaded by wilder (license 362)
	   -- with some mods by me


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r98025 | russell | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines

Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.

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Instead of using is16kHz(), implement a format_rate() function.


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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines

1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running.  Don't just ignore it.
(Closes issue #11698)

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r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines

Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.

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 - The most common fix being made here is to fix all of the places where the
   number of output samples and output bytes gets updated in the translator
   state structure.
 - Fix a number of other places where the number of samples provided as an
   initialization value to a struct was incorrect.


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to fill the buffer is half the buffer size.


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r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines

Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen
to leave a message.

(closes issue #11735, reported and patched by jamessan)


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