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.



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

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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.
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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

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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

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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


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

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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)

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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


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

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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.

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

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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)

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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)


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

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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.

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

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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.


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

Initially, this was to be a new feature, with a patch from Switchvox,
 but after discussions, it was noted that this feature already existed in trunk.

The resulting discussions ended in a comment that was along the lines of
 "the patch provided here is a lot smaller than what is already in trunk,
 because it doesn't create a new application and duplicate existing code"

It was decided that these two applications could be easily merged to reduce
 code duplication.  SO, that's what this does.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114660 65c4cc65-6c06-0410-ace0-fbb531ad65f3

if using IMAP storage for voicemail. The comment will be recorded and attached
as a second attachment in addition to the original message. This will be invoked
if you choose to prepend a message the way you would with file or ODBC storage


(closes issue #12028)
Reported by: jaroth
Patches:
      forward_with_comment_v2.patch uploaded by jaroth (license 50)
Tested by: jaroth



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add some additional features to the core park_call_full() function, and expose
them as options to the Park() application.  The functionality being added is the
ability to specify a custom return extension/context/priority, a custom timeout,
and a couple of options.  The options are to play ringing instead of MOH to the
parked caller, and to randomize parking spot selection.

(code inspired by the patch in AST-17, code from switchvox)


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pointer.



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r114649 | tilghman | 2008-04-25 10:53:52 -0500 (Fri, 25 Apr 2008) | 6 lines

Reference documentation files that actually exist.
(closes issue #12516)
 Reported by: linuxmaniac
 Patches: 
       diff_rev114611.patch uploaded by linuxmaniac (license 472)

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(closes issue #11328)
Reported by: rain
Patches:
      20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12519)
Reported by: falves11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines

Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.

(closes issue #12513)
Reported by: mneuhauser
Patches:
      asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)


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

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r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr 2008) | 8 lines

Output of channel variables when eventwhencalled=vars was set
was being truncated two characters. This patch corrects the
problem.

(closes issue #12493)
Reported by: davidw


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r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines

Resolve a deadlock in chan_local by releasing the channel lock
temporarily.

(closes issue #11712)
Reported by: callguy
Patches:
      11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham


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r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines

Ensure that when we set the accountcode, it actually shows up in the CDR.
(Fix for AMI Originate)
(Closes issue #12007)

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(Closes issue #12292)
 Reported by: DEA
 Patches: 
       app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3)


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

(closes issue #12496)
Reported by: daniele
Patches:
      misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele

Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.

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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line

added fix from #8899
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r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines

Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
up very quickly.
(issue #12515)

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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines

Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12514)


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