https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-29)


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.


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This corrects an issue reported on the -users list.


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


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.


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(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


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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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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines
  
  Ensure ringing continues for branched calls after progress is received
  
  While waiting for an answer, don't send progress for branched calls
  for which ringing was sent.
  
  (closes issue #15028)
  Reported by: fnordian
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SWP-151



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct 2009) | 13 lines
  
  Remove automatic switching from T.38 to voice mode in chan_sip.
    
  chan_sip has some code to automatically switch from T.38 mode to voice mode when
  a voice frame is written to the channel while it is in T.38 mode; this was
  intended to handle the situation when a FAX transmission has ended and the channel
  is not yet hung up, but is causing problems at the beginning of FAX sessions as
  well when there are still voice frames 'in flight' at the time the T.38 negotiation
  completes. This patch removes the automatic switchover.
    
  (issue #16025)
  Reported by: jamicque
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.

(closes issue #16025)
Reported by: jamicque


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a pending reinvite were sent, we might not properly
send connected party info since we were checking the wrong
flag. This was a rare occurrence, but could still happen
nevertheless.



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009) | 2 lines
  
  Remove a duplicate ao2_iterator_destroy().
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  r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines
  
  Don't use data outside of its scope.
  
  The purpose of this code was to have a hangup frame put on the list of deferred
  frames.  However, the code that read the hangup frame was outside of the scope
  of where the hangup frame was declared.
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  r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines
  
  Remove some unnecessary code.
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CONFIRMED status doesn't imply busy or free, that is handled with the TRANSP
field. Luckily, libical already sets the is_busy status on the span for us.


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SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
to initiate a switchover to T.38, not the direction that the FAX transfer will
take place.

(closes issue #16039)
Reported by: jamicque


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines
  
  Signal timeouts by returning AST_CONTROL_RINGING when originating calls.
  (closes issue #15104)
  Reported by: nblasgen
  Patches:
        manager-timeout1.diff uploaded by mnicholson (license 96)
  Tested by: nblasgen, mnicholson
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