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

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r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul 2007) | 2 lines

Fix autoconf logic for finding OpenH323 when it is not in the first place searched (/usr/share/openh323).

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(3-deep nested loop)


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r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 lines

set the sequence number in a frame for all frame types


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remove a couple of entries that got duplicated and snuck into the SIP section.  Also, align the NAT/STUN entry with the others.

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r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue, 24 Jul 2007) | 9 lines

Merged revisions 76978 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line

this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing
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r76937 | tilghman | 2007-07-24 17:12:43 -0500 (Tue, 24 Jul 2007) | 10 lines

Merged revisions 76934 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 Jul 2007) | 2 lines

Oops, res contains the error code, not errno.  I was wondering why a mutex was reporting "No such file or directory"...

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r76891 | tilghman | 2007-07-24 15:42:05 -0500 (Tue, 24 Jul 2007) | 2 lines

Found another place where we should be using the umask (thanks jcmoore)

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r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 lines

Don't create the Asterisk channel until we are starting the PBX on it.
(ASA-2007-018)

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r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul 2007) | 13 lines

Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue.
This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is
roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is
used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone.

As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through
the member list to figure out how many members there are.

Special thanks to blitzrage for helping to test this out.

(closes issue #10127, reported by bcnit, patched by me, tested by blitzrage)


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This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.

Note that this does not remove or deprecate the "old" syntax, as it's still
 quite useful - even moreso if shared lines get implemented.


Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)


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+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload



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You need to put static in front of a static RWLIST declaration to make it really static... and don't call AST_RWLIST_HEAD_DESTROY on a statically declared list.


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Don't bother calling AST_RWLIST_EMPTY on a list before AST_RWLIST_TRAVERSE, it's just a double check.


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r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) | 4 lines

It was our stated intention for 1.4 that files created in app_voicemail should
depend upon the umask.  Unfortunately, mkstemp() creates files with mode 0600,
regardless of the umask.  This corrects that deficiency.

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 - Makes the structures handling external AGI commands a bit more thread-safe
 - Makes AGI transparently work with both live and hungup channels
 - DeadAGI is hence no longer necessary and is deprecated
 - CLI bug fixes
 - Commands will refuse to run if the channel is dead and the command is nonsensical
   for dead channels.


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Clean up res_crypto module. It now uses an rwlist to keep the keys and it should also be thread safe now.


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change to the Local channel name.


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using old methods of parsing arguments to using the standard macros.  However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar).  Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).


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r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul 2007) | 3 lines

Fix some incorrect softkey labels in messages.
Don't try to play dialtone in some unimplemented features.

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r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, 23 Jul 2007) | 12 lines

Merged revisions 76653 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines

(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.

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(closes issue #10276)
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r76620 | qwell | 2007-07-23 12:57:53 -0500 (Mon, 23 Jul 2007) | 4 lines

Don't try to queue up hold/unhold frames on a non-existent channel.

Issue 10276.

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r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2 lines

Allow app_morsecode to build on PPC Linux by putting the value of the digit char in an int.

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does not use  DTMF BEGIN frames.

1.4 seems correct (it does not have the two fields).

However, as this bug shows, the current way of creating the sip_tech
replica is too error-prone, one can easily forget to update one of
the two entries.  Perhaps it would be better to create sip_tech_info
expliclty at module load, by doing

	sip_tech_info = sip_tech;
	sip_tech_info.send_digit_begin = NULL

(in this case, this is something applicable to 1.4 as well).



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r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, 23 Jul 2007) | 14 lines

Merged revisions 76560 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines

(closes issue #10236)
Reported by: homesick
Patches:
      rpid_1.4_75840.patch uploaded by homesick (license 91)
Accept Remote Party ID on guest calls.

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allow building under dev mode


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Reported by: snuffy
Patches:
      doxygen-updates.diff uploaded by snuffy (license 35)

Another big batch of doxygen documentation updates


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Reported by: bbryant
Patches:
      20070720__core_debug_by_file.patch uploaded by bbryant (license 36)
	  (with some modifications by me)
Tested by: russell, bbryant

This set of changes introduces the ability to set the core debug or verbose
levels on a per-file basis.  Interestingly enough, in 1.4, you have the ability
to set core debug for a single file, but that functionality was accidentally
lost in the conversion of the CLI commands to the new format.

This patch improves upon what was in 1.4 by letting you set it for more than 1
file, and by also supporting verbose.

*** Janitor Project ***

This patch also introduces a new macro, ast_verb(), which is similar
to ast_debug().  Setting the per file verbose value only works for messages that
use this macro.  Converting existing uses of ast_verbose() can be done like:

if (option_debug > 2)
   ast_verbose(VERBOSE_PREFIX_3 "Something useful\n");

...

ast_verb(3, "Something useful\n");



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between integers and strings using a single translation table,
and use them in a few places instead of ad-hoc routines
that duplicate the table.

On passing, note that REFER_CONFIRMED is never used, and add a
few comments.

Nothing to backport here.



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Use autoconf logic to determine byte swapping macro presence. This should now also use other macros if present.


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r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul 2007) | 6 lines

(closes issue #10268)
Reported by: mvanbaak
Patches:
      chan_skinny_openbsd.diff uploaded by mvanbaak (license 7)
Add another OS that has to use the Macros for byte ordering.

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