(closes issue #10277, patches by mvanbaak)

Basically, this changes ...

if (option_verbose > 2)
   ast_verbose(VERBOSE_PREFIX_3, "Something\n");

to ...

ast_verb(3, "Something\n");


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

where verbatim text went off the end of the page on the PDF, and various
other improvements
(closes issue #10307, IgorG)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 * random() is defined to return a positive result, and now ast_random()
   will always do so as well


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #10308, reported by yehavi, patched by me)


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and why this does not overwrite useful information.



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The upgrade of application argument separators to comma has an effect on AEL; I commented out the code that substitutes commas with vertbars, so we can get apps to parse their args correctly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77218 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line

This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines

(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

At first sight (but the function is very large so i am not 100% sure)
the code seems correct, so maybe my compiler is just not smart
enough to figure that out at the optimization level it has.

Not worthwhile merging to 1.4 i believe.



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r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines

chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string


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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

+ 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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 - 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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76706 65c4cc65-6c06-0410-ace0-fbb531ad65f3

change to the Local channel name.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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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