[asterisk-commits] file: branch file/logqueue r43962 - in /team/file/logqueue: ./ apps/ channels...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu Sep 28 18:38:34 MST 2006


Author: file
Date: Thu Sep 28 20:38:33 2006
New Revision: 43962

URL: http://svn.digium.com/view/asterisk?rev=43962&view=rev
Log:
Merged revisions 43522,43525-43526,43539-43540,43546,43554,43565,43571-43575,43583,43590,43597,43604,43617,43627,43641,43643,43651,43658-43659,43666-43667,43675,43677,43687,43696,43698,43701-43702,43704,43709,43715,43721,43727,43733,43740,43747,43754,43757,43763,43776-43777,43780,43785,43792,43799,43802,43804,43814,43817,43827,43829-43830,43836,43845,43853-43855,43863,43865,43875,43888,43894,43900,43902,43912,43914,43917,43920-43921,43934,43945,43953-43954 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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r43522 | file | 2006-09-22 19:34:27 -0400 (Fri, 22 Sep 2006) | 2 lines

You see nothing...

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r43525 | kpfleming | 2006-09-22 19:45:14 -0400 (Fri, 22 Sep 2006) | 10 lines

Merged revisions 43524 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43524 | kpfleming | 2006-09-22 18:44:47 -0500 (Fri, 22 Sep 2006) | 2 lines

don't output the 'build complete' message when the target being run is already going to do an installation

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r43526 | kpfleming | 2006-09-22 19:46:02 -0400 (Fri, 22 Sep 2006) | 2 lines

file forgot one :-)

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r43539 | pcadach | 2006-09-23 14:25:13 -0400 (Sat, 23 Sep 2006) | 1 line

Correct behavior on Cisco's DTMF
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r43540 | pcadach | 2006-09-23 14:28:23 -0400 (Sat, 23 Sep 2006) | 1 line

Avoid possible deadlock on channel destruction
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r43546 | pcadach | 2006-09-24 08:15:49 -0400 (Sun, 24 Sep 2006) | 1 line

Small Cisco's RTP DTMF update
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r43554 | russell | 2006-09-24 09:57:19 -0400 (Sun, 24 Sep 2006) | 20 lines

Merged revisions 43553 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43553 | russell | 2006-09-24 09:53:35 -0400 (Sun, 24 Sep 2006) | 12 lines

Merged revisions 43552 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 Sep 2006) | 4 lines

Check to see if the channel that is activating the IAXPEER function is actually
an IAX2 channel before proceeding to process it to avoid crashing.
(issue #8017, reported by admott, fixed by myself)

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r43565 | russell | 2006-09-24 11:01:09 -0400 (Sun, 24 Sep 2006) | 13 lines

Merged revisions 43564 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 Sep 2006) | 5 lines

Fix a CLI command registration issue where an erroneous message claiming that
"iax2 show provisioning" was already registered.  This was because this command
was registering itself as both the command, as well as the command it is
deprecating.  (issue #8022, reported by bjweeks, fixed by myself)

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r43571 | pcadach | 2006-09-24 14:12:35 -0400 (Sun, 24 Sep 2006) | 1 line

Define DTMF payload types
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r43572 | pcadach | 2006-09-24 14:16:44 -0400 (Sun, 24 Sep 2006) | 1 line

Specify non-standard data independedly on OpenH323's codec name (it can be easily changed)
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r43573 | pcadach | 2006-09-24 14:19:56 -0400 (Sun, 24 Sep 2006) | 1 line

Define Cisco RTP capability
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r43574 | pcadach | 2006-09-24 14:53:44 -0400 (Sun, 24 Sep 2006) | 1 line

DTMF mode is bitmask, not valued field
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r43575 | pcadach | 2006-09-24 14:59:38 -0400 (Sun, 24 Sep 2006) | 1 line

Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated)
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r43583 | pcadach | 2006-09-25 02:19:46 -0400 (Mon, 25 Sep 2006) | 1 line

Block revision 43582
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r43590 | pcadach | 2006-09-25 03:50:25 -0400 (Mon, 25 Sep 2006) | 1 line

Disable fastStart if requested by remote side
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r43597 | pcadach | 2006-09-25 05:03:14 -0400 (Mon, 25 Sep 2006) | 1 line

Support for negotiation and receiption of Cisco's RTP DTMF
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r43604 | murf | 2006-09-25 10:49:56 -0400 (Mon, 25 Sep 2006) | 1 line

This tiny fix prevents asterisk from crashing if trying to play an OGG moh file.
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r43617 | qwell | 2006-09-25 11:24:48 -0400 (Mon, 25 Sep 2006) | 12 lines

Merged revisions 43616 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43616 | qwell | 2006-09-25 08:23:31 -0700 (Mon, 25 Sep 2006) | 4 lines

One more fix for sounds installation - this time for portability.

Reported to asterisk-dev mailing list.

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r43627 | pcadach | 2006-09-25 16:30:17 -0400 (Mon, 25 Sep 2006) | 1 line

Block revision 43626 from 1.4 tree - already here
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r43641 | tilghman | 2006-09-25 18:06:26 -0400 (Mon, 25 Sep 2006) | 20 lines

Merged revisions 43640 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43640 | tilghman | 2006-09-25 17:04:47 -0500 (Mon, 25 Sep 2006) | 12 lines

Merged revisions 43634 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43634 | tilghman | 2006-09-25 16:14:41 -0500 (Mon, 25 Sep 2006) | 4 lines

Two bugs when forwarding voicemail (Issue 7824):
1) delete=yes was ignored
2) maxmessages was ignored

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r43643 | tilghman | 2006-09-25 18:09:13 -0400 (Mon, 25 Sep 2006) | 10 lines

Merged revisions 43642 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43642 | tilghman | 2006-09-25 17:07:44 -0500 (Mon, 25 Sep 2006) | 2 lines

Should have moved these lines up in the merge, instead of removing them

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r43651 | qwell | 2006-09-26 11:35:37 -0400 (Tue, 26 Sep 2006) | 19 lines

Merged revisions 43650 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines

Add proper codec support to chan_skinny.  Works with at least ulaw, alaw, and g729a.

This is technically a "new feature", but there are justifications for it.

I found a bug with the recent rtp packetization changes, which caused the media setup to
fail under certain circumstances, particularly when using allow=all, or having no allow=
statements (globally or on the device).

I could have either removed the rtp packetization features, or I could add proper codec
support (which, without, I think most people would consider to be a bug anyways).

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r43658 | tilghman | 2006-09-26 12:28:01 -0400 (Tue, 26 Sep 2006) | 2 lines

These three are not defined on all platforms that we support

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r43659 | tilghman | 2006-09-26 12:56:07 -0400 (Tue, 26 Sep 2006) | 2 lines

MailboxExists should be a dialplan function, not an application (Issue 7503)

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r43666 | qwell | 2006-09-26 13:09:01 -0400 (Tue, 26 Sep 2006) | 5 lines

Add optional queue_log_name config option for logger.conf, to change the
name of the queue_log file.

Issue #7363, patch by Steve Davies, slightly modified by me.

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r43667 | tilghman | 2006-09-26 13:25:27 -0400 (Tue, 26 Sep 2006) | 2 lines

Gosub arguments (Issue 7780)

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r43675 | qwell | 2006-09-26 14:10:07 -0400 (Tue, 26 Sep 2006) | 12 lines

Merged revisions 43674 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43674 | qwell | 2006-09-26 11:08:51 -0700 (Tue, 26 Sep 2006) | 4 lines

Issue #8015, patch by Dan Austin.

Maximum values were incorrect, which is why this is being put in 1.4

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r43677 | kpfleming | 2006-09-26 14:39:01 -0400 (Tue, 26 Sep 2006) | 10 lines

Merged revisions 43676 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43676 | kpfleming | 2006-09-26 13:34:27 -0500 (Tue, 26 Sep 2006) | 2 lines

update to use 1.4.3 core sounds, with corrected beep/beeperr/tt-monkeys files

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r43687 | kpfleming | 2006-09-26 15:37:48 -0400 (Tue, 26 Sep 2006) | 2 lines

start a CHANGES file for trunk... no need to force people to have to review commit logs after branching

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r43696 | file | 2006-09-26 16:11:44 -0400 (Tue, 26 Sep 2006) | 10 lines

Merged revisions 43695 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines

Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980)

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r43698 | file | 2006-09-26 16:20:34 -0400 (Tue, 26 Sep 2006) | 10 lines

Merged revisions 43697 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep 2006) | 2 lines

Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo)

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r43701 | russell | 2006-09-26 16:26:00 -0400 (Tue, 26 Sep 2006) | 22 lines

Merged revisions 43700 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43700 | russell | 2006-09-26 16:24:39 -0400 (Tue, 26 Sep 2006) | 14 lines

Merged revisions 43699 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26 Sep 2006) | 6 lines

When parsing the sections of voicemail.conf that contain mailbox definitions,
don't introduce a length limit on the definition by using a 256 byte temporary
storage buffer.  Instead, make the temporary buffer just as big as it needs
to be to hold the entire mailbox definition.
(fixes BE-68)

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r43702 | qwell | 2006-09-26 16:30:18 -0400 (Tue, 26 Sep 2006) | 2 lines

update CHANGES file to reflect codec support in chan_skinny

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r43704 | russell | 2006-09-26 16:33:37 -0400 (Tue, 26 Sep 2006) | 10 lines

Blocked revisions 43703 via svnmerge

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r43703 | russell | 2006-09-26 16:30:36 -0400 (Tue, 26 Sep 2006) | 3 lines

Add missing newline character in the warning message about deprecated TOS values
in configuration.

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r43709 | file | 2006-09-26 16:51:36 -0400 (Tue, 26 Sep 2006) | 18 lines

Merged revisions 43707 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43707 | file | 2006-09-26 16:47:26 -0400 (Tue, 26 Sep 2006) | 10 lines

Merged revisions 43705 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2 lines

Use proper type to represent the group variable (issue #8025 reported by makoto)

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r43715 | russell | 2006-09-26 17:01:02 -0400 (Tue, 26 Sep 2006) | 25 lines

Merged revisions 43710 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43710 | russell | 2006-09-26 16:56:42 -0400 (Tue, 26 Sep 2006) | 17 lines

(This was actually BE-65)

Merged revisions 43708 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | 7 lines

Back in revision 4798, this message was changed from using ast_cli() to directly
calling write().  During this change, checking if this was a remote console was
removed.  This caused this message about using "exit" or "quit" to exit an
Asterisk console to come up in times where it did not make sense.  This change
restores the check to see if this is a remote console before printing the
message.  (fixes BE-4)

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r43721 | qwell | 2006-09-26 17:30:41 -0400 (Tue, 26 Sep 2006) | 5 lines

Add QueueSummary manager action.
Gives "at a glance" information about a single queue, or all queues.

Issue #8035, patch by rgollent, slightly modified (formatting) by me.

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r43727 | qwell | 2006-09-26 18:18:04 -0400 (Tue, 26 Sep 2006) | 4 lines

Fire a manager event when a meetme is started/stopped.

Issue #7891, patch by suhler.

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r43733 | file | 2006-09-26 21:39:39 -0400 (Tue, 26 Sep 2006) | 2 lines

Clean up code and convert last two things (firmware/dialplan cache) to linked list macros.

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r43740 | murf | 2006-09-26 22:46:50 -0400 (Tue, 26 Sep 2006) | 9 lines

Merged revisions 43739 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1 line

This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse.
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r43747 | murf | 2006-09-26 23:45:22 -0400 (Tue, 26 Sep 2006) | 1 line

This commits the changes to AEL to use the gosub-with-args from Tilghman to perform macro calls. This results in substantially smaller stack footprint, which allows macro call depths in excess of 100,000 levels, rather than the limit of 7 calls deep, which the Macro app is subject to.
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r43754 | russell | 2006-09-27 00:28:37 -0400 (Wed, 27 Sep 2006) | 6 lines

remote an unused buffer in mm_login() (issue #8038, selsky)

In passing, I have cleaned up some formatting to better comply with our
guidelines.  I have also changed one place to use S_OR(), and a couple of
places to use ast_strlen_zero() as appropriate.

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r43757 | russell | 2006-09-27 00:37:33 -0400 (Wed, 27 Sep 2006) | 17 lines

Blocked revisions 43756 via svnmerge

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r43756 | russell | 2006-09-27 00:35:18 -0400 (Wed, 27 Sep 2006) | 10 lines

Backport revision 43754 from the trunk, which removes an unused buffer from
mm_login to close bug 8038, as well as addresses some formatting and coding
guidelines issues in passing.

Originally, I did not commit this to 1.4 since it is not necessarily fixing a
bug.  However, since the IMAP storage code is brand new, I decided it would
be better to make the change here as well, in case someone has to work on this
code to address issues in the very near future.  I don't want to make
unnecessary merge problems going to the trunk.

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r43763 | pcadach | 2006-09-27 08:32:06 -0400 (Wed, 27 Sep 2006) | 1 line

Use ast_strdupa() instead of strdup(), thanks to sergee
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r43776 | file | 2006-09-27 12:25:45 -0400 (Wed, 27 Sep 2006) | 10 lines

Merged revisions 43774 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2 lines

Make rfc2833compensate a global option. 

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r43777 | mattf | 2006-09-27 12:49:01 -0400 (Wed, 27 Sep 2006) | 2 lines

Add CLI block and unblock circuit commands for SS7.

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r43780 | russell | 2006-09-27 12:57:44 -0400 (Wed, 27 Sep 2006) | 58 lines

Merged revisions 43779 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines

Merged revisions 43778 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines

Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period.  For example, if blind
transfer is configured as '##', and a user presses just '#'.  In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
 valuable input provided by mneuhauser and kuj.  Fixed by me, with testing help
 and peer review from Joshua Colp).

There are a couple of issues involved in this fix:

1) When ast_generic_bridge determines that there has been a timeout, it returned
   AST_BRIDGE_RETRY.  Then, when ast_channel_bridge gets this result, it calls
   ast_generic_bridge over again with the same timestamp for the next event.
   This results in an endless loop of nothing until the call is terminated.
   This is resolved by simply changing ast_generic_bridge to return 
   AST_BRIDGE_COMPLETE when it sees a timeout.

2) I also changed ast_channel_bridge such that if in the process of calculating
   the time until the next event, it knows a timeout has already occured, to
   immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
   channels anyway.

3) In the process of testing the previous two changes, I ran into a problem in
   res_features where ast_channel_bridge would return because it determined
   that there was a timeout.  However, ast_bridge_call in res_features would
   then determine by its own calculation that there was still 1 ms before the
   timeout really occurs.  It would then proceed, and since the bridge broke
   out and did *not* return a frame, it interpreted this as the call was over
   and hung up the channels.

   The reason for this was because ast_bridge_call in res_features and
   ast_channel_bridge in channel.c were using different times for their
   calculations.  channel.c uses the start_time on the bridge config, which
   is the time that the feature digit was recieved.  However, res_features
   had another time, 'start', which was set right before calling 
   ast_channel_bridge.  'start' will always be slightly after start_time in the
   bridge config, and sometimes enough to round up to one ms.

   This is fixed by making ast_bridge_call use the same time as 
   ast_channel_bridge for the timeout calculation.

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r43785 | mattf | 2006-09-27 13:00:48 -0400 (Wed, 27 Sep 2006) | 2 lines

Fix some little things

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r43792 | file | 2006-09-27 13:02:28 -0400 (Wed, 27 Sep 2006) | 10 lines

Merged revisions 43783 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 lines

Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right.

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r43799 | file | 2006-09-27 15:12:40 -0400 (Wed, 27 Sep 2006) | 10 lines

Merged revisions 43798 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2 lines

Compensate for out of order packets better if RFC2833 compensation is turned on.

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r43802 | file | 2006-09-27 15:39:39 -0400 (Wed, 27 Sep 2006) | 2 lines

I *think* this is the last list in chan_iax2

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r43804 | qwell | 2006-09-27 15:45:24 -0400 (Wed, 27 Sep 2006) | 12 lines

Merged revisions 43803 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines

Fix an issue with PLAYBACKSTATUS not being set under certain circumstances.
Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string.
Fix Background() to return -1 like Playback(), if no args are specified.

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r43814 | qwell | 2006-09-27 16:03:23 -0400 (Wed, 27 Sep 2006) | 4 lines

Add BACKGROUNDSTATUS to Background()

Issue #7835, original patch by bcnit - redone by me.

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r43817 | tilghman | 2006-09-27 16:27:03 -0400 (Wed, 27 Sep 2006) | 18 lines

Merged revisions 43816 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43816 | tilghman | 2006-09-27 15:21:54 -0500 (Wed, 27 Sep 2006) | 10 lines

Merged revisions 43815 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27 Sep 2006) | 2 lines

Avoid inability to lock directory log message by creating the directory ahead of time. (Issue 7631)

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r43827 | file | 2006-09-27 17:39:12 -0400 (Wed, 27 Sep 2006) | 2 lines

Do the directory walk dance instead of repeated stat calls as it seems to be faster (issue #7507 reported by Corydon76)

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r43829 | file | 2006-09-27 17:41:25 -0400 (Wed, 27 Sep 2006) | 2 lines

Finish up last commit

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r43830 | file | 2006-09-27 17:48:01 -0400 (Wed, 27 Sep 2006) | 2 lines

Add ability to set high ToS bits as non-root on Linux using libcap (issue #7047 reported by maddison)

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r43836 | file | 2006-09-27 18:01:32 -0400 (Wed, 27 Sep 2006) | 2 lines

Update CHANGES to reflect libcap capability that was added.

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r43845 | pcadach | 2006-09-28 06:41:38 -0400 (Thu, 28 Sep 2006) | 1 line

Handle HOLD/RETRIEVE notifications
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r43853 | pcadach | 2006-09-28 07:12:58 -0400 (Thu, 28 Sep 2006) | 21 lines

Merged revisions 43635,43843-43844,43846 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (?\208?\146?\209?\130?\209?\128, 26 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Do not open transmit channel until TCS is received
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r43854 | crichter | 2006-09-28 07:27:09 -0400 (Thu, 28 Sep 2006) | 9 lines

Merged revisions 43775 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) | 1 line

removed the chan_misdn versioning, since asterisk has it's own
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r43855 | crichter | 2006-09-28 07:32:32 -0400 (Thu, 28 Sep 2006) | 17 lines

Merged revisions 43852 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43852 | crichter | 2006-09-28 13:03:05 +0200 (Do, 28 Sep 2006) | 9 lines

Merged revisions 43764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 Sep 2006) | 1 line

fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed.
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r43863 | pcadach | 2006-09-28 09:02:30 -0400 (Thu, 28 Sep 2006) | 13 lines

Merged revisions 43861-43862 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Put attribute tag at correct place
........
r43862 | pcadach | 2006-09-28 18:58:22 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Force remote side to start media on outgoing PROGRESS message
........

................
r43865 | bweschke | 2006-09-28 09:35:04 -0400 (Thu, 28 Sep 2006) | 11 lines

Merged revisions 43864 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r43864 | bweschke | 2006-09-28 09:24:10 -0400 (Thu, 28 Sep 2006) | 3 lines

 Autopause not working for queue members. (#8042 - jmls reported and patch)


........

................
r43875 | bweschke | 2006-09-28 11:32:48 -0400 (Thu, 28 Sep 2006) | 19 lines

Merged revisions 43873 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r43873 | bweschke | 2006-09-28 11:29:21 -0400 (Thu, 28 Sep 2006) | 11 lines

Merged revisions 43871 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28 Sep 2006) | 3 lines

 Fix race condion crash with get_member_status (#7864 - tim_ringenbach reported and patched)


........

................

................
r43888 | pcadach | 2006-09-28 12:01:48 -0400 (Thu, 28 Sep 2006) | 9 lines

Merged revisions 43877 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r43877 | pcadach | 2006-09-28 21:56:21 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Fix compiler warning
........

................
r43894 | file | 2006-09-28 12:19:22 -0400 (Thu, 28 Sep 2006) | 18 lines

Merged revisions 43893 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r43893 | file | 2006-09-28 12:17:36 -0400 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43891 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2 lines

Stop the stream after waitstream returns so that our formats get restored. (issue #7370 reported by kryptolus)

........

................

................
r43900 | kpfleming | 2006-09-28 12:43:40 -0400 (Thu, 28 Sep 2006) | 18 lines

Merged revisions 43898 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r43898 | kpfleming | 2006-09-28 11:38:25 -0500 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43895 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28 Sep 2006) | 2 lines

eliminate compiler warning introduced by recent changes

........

................

................
r43902 | bweschke | 2006-09-28 12:54:01 -0400 (Thu, 28 Sep 2006) | 19 lines

Merged revisions 43899 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r43899 | bweschke | 2006-09-28 12:41:05 -0400 (Thu, 28 Sep 2006) | 11 lines

Merged revisions 43897 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 Sep 2006) | 3 lines

 app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach)


........

................

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r43912 | file | 2006-09-28 13:07:37 -0400 (Thu, 28 Sep 2006) | 2 lines

Add jitterbuffer information to sip list settings (issue #7945 reported by sergee)

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r43914 | file | 2006-09-28 13:15:38 -0400 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43913 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43913 | file | 2006-09-28 13:14:07 -0400 (Thu, 28 Sep 2006) | 2 lines

Warning be gone!

........

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r43917 | file | 2006-09-28 13:32:46 -0400 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43915 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2 lines

Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg)

........

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r43920 | pcadach | 2006-09-28 13:36:33 -0400 (Thu, 28 Sep 2006) | 9 lines

Merged revisions 43918 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43918 | pcadach | 2006-09-28 23:34:19 +0600 (?\208?\167?\209?\130?\208?\178, 28 ?\208?\161?\208?\181?\208?\189 2006) | 1 line

Extend call establishment timeout
........

................
r43921 | kpfleming | 2006-09-28 13:38:07 -0400 (Thu, 28 Sep 2006) | 18 lines

Merged revisions 43919 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r43919 | kpfleming | 2006-09-28 12:35:42 -0500 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43916 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28 Sep 2006) | 2 lines

fix buggy (and overly complex) loop used during reload of app_queue for static member list updating

........

................

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r43934 | file | 2006-09-28 14:09:01 -0400 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43933 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines

Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)

........

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r43945 | qwell | 2006-09-28 16:13:09 -0400 (Thu, 28 Sep 2006) | 12 lines

Merged revisions 43944 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r43944 | qwell | 2006-09-28 13:11:22 -0700 (Thu, 28 Sep 2006) | 4 lines

Fix incorrect argument order for member names, on persisted members.

Issue 8047, patch by jmls.

........

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r43953 | kpfleming | 2006-09-28 18:43:24 -0400 (Thu, 28 Sep 2006) | 10 lines

Merged revisions 43952 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r43952 | kpfleming | 2006-09-28 17:42:18 -0500 (Thu, 28 Sep 2006) | 2 lines

eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h

........

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r43954 | file | 2006-09-28 18:49:55 -0400 (Thu, 28 Sep 2006) | 2 lines

Cache the application pointer so we don't have to needlessly search for it over and over. This should yield a suitable performance increase.

................

Modified:
    team/file/logqueue/   (props changed)
    team/file/logqueue/.cleancount
    team/file/logqueue/CHANGES
    team/file/logqueue/Makefile
    team/file/logqueue/apps/app_alarmreceiver.c
    team/file/logqueue/apps/app_chanspy.c
    team/file/logqueue/apps/app_festival.c
    team/file/logqueue/apps/app_followme.c
    team/file/logqueue/apps/app_meetme.c
    team/file/logqueue/apps/app_playback.c
    team/file/logqueue/apps/app_queue.c
    team/file/logqueue/apps/app_rpt.c
    team/file/logqueue/apps/app_stack.c
    team/file/logqueue/apps/app_voicemail.c
    team/file/logqueue/channels/Makefile
    team/file/logqueue/channels/chan_h323.c
    team/file/logqueue/channels/chan_iax2.c
    team/file/logqueue/channels/chan_local.c
    team/file/logqueue/channels/chan_mgcp.c
    team/file/logqueue/channels/chan_misdn.c
    team/file/logqueue/channels/chan_sip.c
    team/file/logqueue/channels/chan_skinny.c
    team/file/logqueue/channels/chan_zap.c
    team/file/logqueue/channels/h323/ast_h323.cxx
    team/file/logqueue/channels/h323/ast_h323.h
    team/file/logqueue/channels/h323/caps_h323.cxx
    team/file/logqueue/channels/h323/caps_h323.h
    team/file/logqueue/channels/h323/chan_h323.h
    team/file/logqueue/channels/h323/cisco-h225.asn
    team/file/logqueue/channels/h323/cisco-h225.cxx
    team/file/logqueue/channels/h323/cisco-h225.h
    team/file/logqueue/channels/iax2-provision.c
    team/file/logqueue/channels/misdn/isdn_lib.c
    team/file/logqueue/channels/misdn/isdn_lib.h
    team/file/logqueue/configs/extensions.ael.sample
    team/file/logqueue/configs/h323.conf.sample
    team/file/logqueue/configs/logger.conf.sample
    team/file/logqueue/configure
    team/file/logqueue/configure.ac
    team/file/logqueue/doc/rtp-packetization.txt
    team/file/logqueue/doc/security.txt
    team/file/logqueue/formats/format_ogg_vorbis.c
    team/file/logqueue/funcs/func_timeout.c
    team/file/logqueue/include/asterisk/channel.h
    team/file/logqueue/include/asterisk/compiler.h
    team/file/logqueue/include/asterisk/lock.h
    team/file/logqueue/include/asterisk/logger.h
    team/file/logqueue/main/Makefile
    team/file/logqueue/main/asterisk.c
    team/file/logqueue/main/channel.c
    team/file/logqueue/main/cli.c
    team/file/logqueue/main/file.c
    team/file/logqueue/main/frame.c
    team/file/logqueue/main/logger.c
    team/file/logqueue/main/pbx.c
    team/file/logqueue/main/rtp.c
    team/file/logqueue/main/slinfactory.c
    team/file/logqueue/main/udptl.c
    team/file/logqueue/makeopts.in
    team/file/logqueue/pbx/ael/ael-test/ael-test11/extensions.ael
    team/file/logqueue/pbx/ael/ael-test/ael-test3/extensions.ael
    team/file/logqueue/pbx/ael/ael-test/ael-test5/extensions.ael
    team/file/logqueue/pbx/ael/ael-test/ref.ael-ntest10
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test1
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test2
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test3
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test4
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test5
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test6
    team/file/logqueue/pbx/ael/ael-test/ref.ael-test7
    team/file/logqueue/pbx/ael/ael-test/ref.ael-vtest13
    team/file/logqueue/pbx/pbx_ael.c
    team/file/logqueue/res/res_features.c
    team/file/logqueue/res/res_limit.c
    team/file/logqueue/res/res_monitor.c
    team/file/logqueue/res/res_musiconhold.c
    team/file/logqueue/res/res_smdi.c
    team/file/logqueue/sounds/Makefile

Propchange: team/file/logqueue/
            ('automerge-email' removed)

Propchange: team/file/logqueue/
------------------------------------------------------------------------------
--- branch-1.4-blocked (original)
+++ branch-1.4-blocked Thu Sep 28 20:38:33 2006
@@ -1,1 +1,1 @@
-/branches/1.4:43484,43510
+/branches/1.4:43484,43510,43582,43626,43703,43756

Propchange: team/file/logqueue/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Propchange: team/file/logqueue/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Sep 28 20:38:33 2006
@@ -1,1 +1,1 @@
-/trunk:1-43519
+/trunk:1-43960

Modified: team/file/logqueue/.cleancount
URL: http://svn.digium.com/view/asterisk/team/file/logqueue/.cleancount?rev=43962&r1=43961&r2=43962&view=diff
==============================================================================
--- team/file/logqueue/.cleancount (original)
+++ team/file/logqueue/.cleancount Thu Sep 28 20:38:33 2006
@@ -1,1 +1,1 @@
-24
+25

Modified: team/file/logqueue/CHANGES
URL: http://svn.digium.com/view/asterisk/team/file/logqueue/CHANGES?rev=43962&r1=43961&r2=43962&view=diff
==============================================================================
--- team/file/logqueue/CHANGES (original)
+++ team/file/logqueue/CHANGES Thu Sep 28 20:38:33 2006
@@ -1,127 +1,19 @@
-Changes since Asterisk 1.2.0-beta2:
+Changes since Asterisk 1.4-beta was branched:
 
-   * Cygwin build system portability
-   * Optional generation of outbound silence during channel recording
-
-Changes since Asterisk 1.2.0-beta1:
-
-   * Many, many bug fixes
-   * Documentation and sample configuration updates
-   * Vastly improved presence/subscription support in the SIP channel driver
-   * A new (experimental) mISDN channel driver
-   * A new monitoring application (MixMonitor)
-   * More portability fixes for non-Linux platforms
-   * New dialplan functions replacing old applications
-   * Significant deadlock and performance upgrades for the Manager interface
-   * An upgrade to the 'new' dialplan expression parser for all users
-   * New Zaptel echo cancellers with improved performance
-   * Support for the latest OSP toolkit from TransNexus
-   * Support user-controlled volume adjustment in MeetMe application
-   * More dialplan applications now return status variables instead of priority jumping
-   * Much more powerful ENUM support in the dialplan
-   * SIP domain support for authentication and virtual hosting
-   * Many PRI protocol updates and fixes, including more complete Q.SIG support
-   * New applications: Pickup() and Page()
-
-Changes since Asterisk 1.0:
-
-This list currently only containts changes made from the end of November until
-March 26, 2005.
-
-   * Add new applications:
-     -- AgentMonitorOutgoing
-     -- Curl
-     -- ExecIf
-     -- ExecIfTime
-     -- IAX2Provision
-     -- MacroExit
-     -- MacroIf
-     -- PauseQueueMember
-     -- ReadFile
-     -- SetRDNIS
-     -- SIPAddHeader
-     -- SIPGetHeader
-     -- StartMusicOnHold
-     -- StopMusicOnHold
-     -- UnpauseQueueMember
-     -- WaitForSilence
-     -- While / EndWhile
-   * app Answer
-     -- added delay option
-   * app ChanIsAvail
-     -- added 's' option
-   * app Dial
-     -- add option to specify the class for musiconhold with m option
-   * app EnumLookup
-     -- added "reload enum" for configuration
-   * app Goto
-     -- added relative priorities
-   * app GotoIf
-     -- added relative priorities
-   * app MeetMe
-     -- added 'i' option
-     -- added 'r' option
-     -- added 'T' option
-     -- added 'P' option
-     -- added 'c' option
-     -- added adminpin to meetme.conf
-     -- added reload command
-   * app PrivacyManager
-     -- add config file privacy.conf
-   * app queue
-     -- queues.conf
-        -- added persistentmembers option to queues.conf
-        -- changed music option to musiconhold
-        -- added weight option
-        -- added note about why agent groups probably shouldn't be used
-        -- added timeoutrestart option
-   * app Read
-     -- added attempts parameter
-     -- added timeout parameter
-   * app Record
-     -- added 'q' option
-   * app SendDTMF
-     -- add timeout option
-   * app SMS
-     -- document alternative syntax for queueing messages
-   * app Voicemail
-     -- add info about VM_CATEGORY
-     -- voicemail.conf
-        -- added usedirectory option
-        -- added VM_CIDNUM and VM_CIDNAME in message config
-   * chan IAX2
-     -- new jitterbuffer
-     -- added setvar option
-     -- added regex to iax2 show peers/users
-     -- allow multiple bindaddr lines in iax.conf
-     -- added reload command
-     -- added forcejitterbuffer option
-     -- added note about specifying bindport before bindaddr
-     -- added trunktimestamps option
-   * chan Agent
-     -- added agent logoff CLI command
-   * chan OSS
-     -- added Flash CLI command
-   * chan SIP
-     -- added setvar option
-     -- added compactheaders option
-     -- added usereqphone option
-     -- added registertimeout option
-     -- added externhost option
-     -- added sip notify CLI command
-     -- added sip_notify.conf
-     -- added allowguest option
-   * chan Zap
-     -- added hanguponplarityswitch option
-     -- added sendcalleridafter option
-     -- added priresetinterval option
-     -- added TON/NPI config options (the ones right above the resetinterval option)
-     -- added answeronpolarityswitch option
-     -- added "never" for resetinterval
-   * extensions
-     -- allow '*' when including files (#include "sip-*.conf")
-     -- added eswitch
-   * General

[... 9013 lines stripped ...]


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