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move automatic user interface selection for menuselect to this Makefile



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r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines

When we get a 408 Timeout, don't stop trying to re-register.
(closes issue #12863)
 Reported by: ricvil

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wasn't working.  :)


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r125893 | tilghman | 2008-06-27 11:46:05 -0500 (Fri, 27 Jun 2008) | 7 lines

Since HAVE_DAHDI is defined to HAVE_ZAPTEL in dahdi_compat.h, we must first
check for HAVE_ZAPTEL.
(closes issue #12938)
 Reported by: opticron
 Patches: 
       tonezone_compat.diff uploaded by opticron (license 267)

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Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used 
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on 
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.

(issue #12799)


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that it isn't evaluated every time through the loop.



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Thanks to Russell for pointing out the problem



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Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html


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r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines

In this debugging function, copy to a buffer instead of using potentially unsafe pointers.

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r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines

Add proper deadlock avoidance.
(closes issue #12914)
 Reported by: ozan
 Patches: 
       20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
 Tested by: ozan

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Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list

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MixMonitor to mix audio. However, it was pointed out to me that because
of this, the command set for the MONITOR_EXEC variable is ignored as well.
This means that people can't do their own custom mixing commands at the end
of recordings in order to make, for instance, stereo recordings of calls.

With this patch, app_queue will set the "joinfiles" variable for the channel's
monitor if MONITOR_EXEC is not zero-length. This means that for normal audio
mixing, MixMonitor is still the preferred choice, but we allow custom
mixing to be done with the two Monitor streams if desired.

(closes issue #12923)
Reported by: snyfer



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to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.

After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the 
previous behavior of app_dial if desired.

(closes issue #12489)
Reported by: bcnit



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This is now handled in skinny_register and load_config.

part two of chan_skinny cleanup


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This has never been used, and noone could give us info about what
it is used for.


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r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | 1 line

Make sure to unlock the lock_info lock (huh?).  Possible deadlock?
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r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines

Add the interface of a queue member to the output of the "queue show" command
so that it can easily be associated with a queue member's name. This helps
so that the appropriate queue member can be removed or paused since the 
interface is required, not the member's name.

(closes issue #12783)
Reported by: davevg
Patches:
      app_queue.diff uploaded by davevg (license 209) with small mod from me


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r125530 | mmichelson | 2008-06-26 17:02:55 -0500 (Thu, 26 Jun 2008) | 10 lines

Backport of attended transfer queue_log patch from trunk.

This patch allows for attended transfers to be logged in the
queue_log the same way that blind transfers have always been.

It was decided by popular opinion on the asterisk-dev mailing
list that this should be backported to 1.4. Thanks to everyone
who gave an opinion.


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r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines

Prior to this patch, the "queue show" command used cached
information for realtime queues instead of giving up-to-date
info. Now realtime is queried for the latest and greatest in
queue info.

(closes issue #12858)
Reported by: bcnit
Patches:
      queue_show.patch uploaded by putnopvut (license 60)


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(closes issue #12893)
 Reported by: jaroth
 Patches: 
       duplicate_saved.patch uploaded by jaroth (license 50)


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(closes issue #12932)
 Reported by: snuffy
 Patches: 
       bug_12932_20080627.diff uploaded by snuffy (license 35)


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Merged revisions 125384 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines

Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked)


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r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines

ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number

(issue AST-80)


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- change ast_settimeout() to honor max rate in edge cases of file playback
  (this will make some warning messages go away at the end of playing back
   a file)


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