for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114931 65c4cc65-6c06-0410-ace0-fbb531ad65f3

and denoising to a channel, AGC() and DENOISE(). Also included, is a change 
to the audiohook API to add a new function (ast_audiohook_remove) that can 
remove an audiohook from a channel before it is detached.

This code is based on a contribution from Switchvox.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f3

dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.

(closes issue #12536)
Reported by: bjm
Patches:
      12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114922 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

to announce-position, "limit" and "more," as well as a new option, 
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.

(closes issue #10991)
Reported by: slavon
Patches:
      app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114906 65c4cc65-6c06-0410-ace0-fbb531ad65f3

since another thread could remove them.
(closes issue #12541)
 Reported by: snuffy
 Patches: 
       bug_12156_apps.diff uploaded by snuffy (license 35)
       Several additional changes by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines

Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)

(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114899 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines

Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4

These changes address a critical performance issue introduced in the latest
release.  The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers.  However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls.  On a small embedded platform, it would not be
able to handle a single call.  On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels.  Ouch.

These changes address some performance issues of the find_callno() function
that have bothered me for a very long time.  On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call.  This involved a mutex lock and unlock for each call number
checked.  So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks.  Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.

A second container for IAX2 pvt structs has been added.  It is an astobj2
hash table.  When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number.  Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.

In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114892 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114888 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines

use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114884 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr 2008) | 2 lines

pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114876 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

a colon-delimited list of spygroups to be specified when calling the ChanSpy application
with the 'g' option. Prior to this, you could only specify a single group when using the
'g' option.

I also have upped the maximum number of spygroups to 128 and added a #define so that this
can be easily increased or decreased later.

(closes issue #12497)
Reported by: jsmith
Patches:
      app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license 15)
Tested by: atis, jvandal



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines

Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect 
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.


(closes issue #12549)
Reported by: darren1713
Patches:
      app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
	       (with modifications from me)
Tested by: putnopvut


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114849 65c4cc65-6c06-0410-ace0-fbb531ad65f3

fix this logic to actually be correct... the fd can't be *both* -1 and an array index to be checked in rfds/efds (bug found by gcc-4.3)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

which says the person's name is handled inside app_voicemail now.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114841 65c4cc65-6c06-0410-ace0-fbb531ad65f3

functions in app_directory can be removed since the ODBC-specific lookups are accomplished
within app_voicemail. This change greatly reduces the amount of lines in app_directory that
were solely for the purpose of looking up a name when ODBC_STORAGE is specified for voicemail.

This commit also makes the name-saying interruptable via DTMF.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line

Change warning message to debug, since there are cases where 0 results is perfectly fine.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

................
r114823 | kpfleming | 2008-04-29 07:53:12 -0500 (Tue, 29 Apr 2008) | 10 lines

Merged revisions 114822 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines

stop script from appending source code if run multiple times

........

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114824 65c4cc65-6c06-0410-ace0-fbb531ad65f3

party to be spoken instead of the channel name or number.

This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.

This change comes as a suggestion from Switchvox, which already has this feature. AST-23


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114813 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12351)
 Reported by: Corydon76
 Patches: 
       20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14)
       pbx_incomplete_with_timeout.diff uploaded by fabled (license 448)
 Tested by: Corydon76, fabled


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114773 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114713 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines

When modules are embedded, they take on a different name, without the ".so"
extension.  Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114706 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12214)
Reported by: DEA
Patches:
      chan_skinny-vm-events-v3.txt uploaded by DEA (license 3)
	  Tested by: DEA and me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114700 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines

When we don't explicitly pass a path to the --with-tds configure option, we
may end up finding tds.h in /usr/local/include instead of /usr/include.  If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114696 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114692 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) | 6 lines

Clicking forward without selecting a message leaves an errant .lock file.
(closes issue #12528)
 Reported by: pukepail
 Patches: 
       patch.diff uploaded by pukepail (license 431)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114690 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12524)
 Reported by: ctooley
 Patches: 
       sip_qualify_peer.diff.2 uploaded by ctooley (license 136)
       some modifications for trunk by Corydon76
 Tested by: Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114683 65c4cc65-6c06-0410-ace0-fbb531ad65f3

barge on the call. It is like the existing whisper option, except that
it allows the spy to talk to both sides of the conversation on which
he is spying.

This feature has existed in Switchvox, and this merges the functionality
into Asterisk.

(AST-32)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114678 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12527)
Reported by: mnicholson
Patches:
      pbx_lua4.diff uploaded by mnicholson (license 96)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114676 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines

Use consistent logic for checking to see if a call number has been chosen yet.
Also, remove some redundant logic I recently added in a fix.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114674 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114667 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114665 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr 2008) | 4 lines

Move the unlock of the spyee channel to outside the start_spying() function so that
the channel is not unlocked twice when using whisper mode.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114663 65c4cc65-6c06-0410-ace0-fbb531ad65f3