Commit Graph

1007 Commits

Author SHA1 Message Date
Jonathan Rose 0fd34b8c0a app_voicemail: Improve msg_id handling
app_voicemail will no longer issue error messages when it retrieves an msg_id
with a NULL value from realtime and will instead simply populate the msg_id
field with a newly generated msg_id. In addition, this patch changes the way
msg_ids are generated to eliminate certain causes of duplicate IDs appearing
within a single system. In addition, when messages are copied, they will now
receive a new msg_id.

(closes issue ASTERISK-20717)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/2220/
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Merged revisions 379460 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-18 18:25:56 +00:00
Rusty Newton 21a0dadac0 Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists      
and is the correct sound file. Previous behavior was silence and a warning on the CLI.

(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
    asterisk20280.patch uploaded by Rusty Newton (license 5829)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 02:29:40 +00:00
Andrew Latham cfc6f60ca3 Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking to the application.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:45:16 +00:00
Kinsey Moore 0eab8b669d Avoid a segfault on invalid format names
If a format name was not found by ast_getformatbyname, a NULL pointer
would be passed into ast_format_rate and immediately dereferenced.
This ensures that a valid pointer is used since the structure is
already allocated on the stack.

(closes issue DPH-523)
Reported-by: Steve Pitts
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Merged revisions 374932 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-12 21:58:29 +00:00
Andrew Latham 14be2a5514 Doxygen Cleanup
Start adding configuration file linking and pages.  Add module loading doxygen block.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-01 23:22:50 +00:00
Kinsey Moore 5bde2dbc34 Add VoicemailRefresh AMI Action
Currently, if there are modifications to mailboxes that Asterisk is
not aware of, the user needs to add "pollmailboxes" to their mailbox
configuration, which repeatedly polls the subscribed mailboxes for
changes. This results in a lot of extra work for the CPU. This patch
introduces the AMI command VoicemailRefresh which permits external
applications to trigger the refresh themselves. The refresh can apply
to a specified mailbox only, an entire context, or all configured
mailboxes. Even a refresh performed on every mailbox would not consume
as much CPU as the pollmailboxes option, given that pollmailboxes runs
continuously and this only runs on demand.

(closes issue ASTERISK-17206)
(closes issue ASTERISK-19908)
Reported-by: Jeff Hutchins
Reported-by: Tilghman Lesher
Patch-by: Tilghman Lesher


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-27 17:02:13 +00:00
Mark Michelson 7bfa978495 Fix error where improper IMAP greetings would be deleted.
(closes issue ASTERISK-20435)
Reported by: fhackenberger
Patches:
	asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026)
	(with suggested modification made by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-25 21:14:21 +00:00
Andrew Latham fd98835f1f Doxygen Updates Janitor Work
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-22 20:43:30 +00:00
Richard Mudgett 7c1003de84 Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported values.
The AMI action VoicemailUsersList VoicemailUserEntry event headers
ServerEmail and MailCommand did not report the global values if they were
not overridden.  The VoicemailUserEntry event header ServerEmail was not
populated with the global value if the voicemail user did not override it.
The VoicemailUserEntry event header MailCommand was never populated with a
value.

* Removed unused struct ast_vm_user member mailcmd[].

(closes issue AST-973)
Reported by: John Bigelow
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07 21:30:17 +00:00
Matthew Jordan e965020d0c Fix memory leaks in app_voicemail when using IMAP storage or realtime config
This patch fixes two memory leaks:

1. When find_user is called with NULL as its first parameter, the voicemail
   user returned is allocated on the heap.  The inboxcount2 function uses
   find_user in such a fashion when counting new messages, and fails to free
   the resulting voicemail user object.

2. When populate_defaults is called on a voicemail user, it wipes whatever
   flags have been set on the object by copying over the global flags object.
   If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
   that flag is removed.  This leaks the voicemail user when free_user is later
   called.

(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
  asterisk.patch2 uploaded by Filip Jenicek (license 6277)

Patch slightly modified for this commit.

Review: https://reviewboard.asterisk.org/r/2096
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 14:44:36 +00:00
Kinsey Moore 9b16c8b0f6 Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 20:21:43 +00:00
Kinsey Moore 163d3b05d4 AST-2012-011: Resolve heap corruption issue with voicemail
The heard and deleted arrays in the voicemail state structure were not
handled properly following the memory leak fix in r354890 and a fix for
an invalid free in r356797.  This could result in accessing and writing
into freed memory.  The allocation for these arrays has been reworked
to avoid the possibility of invalid frees, access of freed memory, and
crashes that were occurring as a result of this.

Locking around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added to
prevent simultaneous modification and access when IMAP storage is in
use.  If IMAP storage is not in use, this locking is not compiled in.

Review: https://reviewboard.asterisk.org/r/1994/
(closes issue ASTERISK-19923)
Reported by: Dan Delaney
Tested by: Dan Delaney, Julian Yap
Patches:
  vm_alloc_fix.diff uploaded by kmoore (license 6273)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05 19:36:21 +00:00
Matthew Jordan 82a7409c15 Add AMI event documentation
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules.  Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.

The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation.  Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event.  The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files.  It generates
the final core-[lang].xml file.

As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.

Review: https://reviewboard.asterisk.org/r/1967/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25 17:59:34 +00:00
Jason Parker 88c9c6bef8 Fix voicemail API tests by using the correct argument order for create/destroy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:30:58 +00:00
Jason Parker 6334142050 Multiple revisions 368963,368965
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  r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | 14 lines
  
  Remove global symbol requirement from app_voicemail.
  
  This uses the existing "function installation" stuff that already existed for
  other functions, like getting message counts.
  
  (closes issue AST-807)
  (issue AST-901)
  (issue AST-908)
  
  Review: https://reviewboard.asterisk.org/r/1965/
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  r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun 2012) | 11 lines
  
  These functions that were moved need to be static.
  
  Also wrap test functions in a #ifdef.
  
  (issue AST-807)
  (issue AST-901)
  (issue AST-908)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-14 19:40:11 +00:00
Kinsey Moore c6142cf2cc Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them and
in some cases the variable being saved to could be removed as well.

(issue ASTERISK-19672)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-11 15:23:30 +00:00
Kinsey Moore bd958c037f Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.

(closes issue ASTERISK-19876)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-05 15:23:43 +00:00
Mark Michelson 14a985560e Merge changes dealing with support for Digium phones.
Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.

Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.

Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.

chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.

Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.

Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
Matthew Jordan 7b51320642 Fix a variety of memory leaks
This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool.  A brief summary of the changes:

* app_minivm:       free ast_str objects on off nominal paths
* app_page:         free the ast_dial object if the requested channel technology
                    cannot be appended to the dialing structure
* app_queue:        if a penalty rule failed to match any existing rule list
                    names, the created rule would not be inserted and its memory
                    would be leaked
* app_read:         dispose of the created silence detector in the presence of
                    off nominal circumstances
* app_voicemail:    dispose of an allocated unique ID field for MWI event
                    un-subscribe requests in off nominal paths; dispose of
                    configuration objects when using the secret.conf option
* chan_dahdi:       dispose of the allocated frame produced by ast_dsp_process
* chan_iax2:        properly unref peer in CLI command "iax2 unregister"
* chan_sip:         dispose of the allocated frame produced by sip_rtp_read's
                    call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup:   properly deref ao2 object grhead in nominal path of
                    dialgroup_read
* func_odbc:        free resultset in off nominal paths of odbc_read
* cli:              free match_list in off nominal paths of CLI match completion
* config:           free comment_buffer/list_buffer when configuration file load
                    is unchanged; free the same buffers any time they were
                    created and config files were processed
* data:             free XML nodes in various places
* enum:             free context buffer in off nominal paths
* features:         free ast_call_feature in off nominal paths of applicationmap
                    config processing
* netsock2:         users of ast_sockaddr_resolve pass in an ast_sockaddr struct
                    that is allocated by the method.  Failures in
                    ast_sockaddr_resolve could result in the users of the method
                    not knowing whether or not the buffer was allocated.  The
                    method will now not allocate the ast_sockaddr struct if it
                    will return failure.
* pbx:              cleanup hash table traversals in off nominal paths; free
                    ignore pattern buffer if it already exists for the specified
                    context
* xmldoc:           cleanup various nodes when we no longer need them
* main/editline:    various cleanup of pointers not being freed before being
                    assigned to other memory, cleanup along off nominal paths
* menuselect/mxml:  cleanup of value buffer for an attribute when that attribute
                    did not specify a value
* res_calendar*:    responses are allocated via the various *_request method
                    returns and should not be allocated in the various
                    write_event methods; ensure attendee buffer is freed if no
                    data exists in the parsed node; ensure that calendar objects
                    are de-ref'd appropriately
* res_jabber:       free buffer in off nominal path
* res_musiconhold:  close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
                    the rtp object
* res_srtp:         if we fail to create the session in libsrtp, destroy the
                    temporary ast_srtp object

(issue ASTERISK-19665)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1922
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
Kinsey Moore dd81b047db Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved.  Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.

(Closes issue ASTERISK-19650)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10 20:56:09 +00:00
Jonathan Rose 8227f70cd7 Coverity Report: Fix issues for error type CHECKED_RETURN for core
(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10 18:35:14 +00:00
Matthew Jordan 11faa15d11 Fix channel opaquification slip-up in r365477
Those channels are opaque now...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07 18:58:40 +00:00
Matthew Jordan 9e7de73fee Support VoiceMail d() option when extension does not exist in channel's context
The VoiceMail d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting.  This option works fine if the
extension being redirected to has an extension with the same initial digit in
the channel's current context.  If that digit did not happen to exist in some
extension, a dialplan match would fail and the user would not be redirected.

This patch fixes it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original context.

(closes issue ASTERISK-18243)
Reported by: mjordan
Tested by: mjordan

Review: https://reviewboard.asterisk.org/r/1892
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07 18:42:48 +00:00
Kinsey Moore 781f4657b9 Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks.  There are a couple
optimizations to remove the need to check for NULL and outboundproxy parsing
in chan_sip.c was rewritten to avoid use of strtok.  Additionally, a bug was
found and fixed with the parsing of outboundproxy when "outboundproxy=," was
set.

(Closes issue ASTERISK-19654)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04 22:17:38 +00:00
Matthew Jordan ebcccf8997 Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the
voicemail message is first stored on disk.  The sound file associated with
the message is read into memory before being transmitted to the database.
When this occurs, a failure in the C library's lseek function would cause a
negative value to be passed to the mmap as the size of the memory map to
create.  This would almost certainly cause the creation of the memory map to
fail, resulting in the message being lost.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:42:12 +00:00
Kinsey Moore a485f44022 Add missing newlines to CLI logging
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 18:19:03 +00:00
Terry Wilson 7876521659 Fix IMAP storage compilation after opaquification changes
(closes issue ASTERISK-19513)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 18:55:14 +00:00
Terry Wilson a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 16:52:47 +00:00
Matthew Jordan 5e40f2cd98 Fix crash in app_voicemail during close_mailbox
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers.  However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL.  In that case, an invalid free would be attempted,
which could crash app_voicemail.  As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers.  This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-25 17:22:55 +00:00
Terry Wilson ebaf59a656 Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24 00:32:20 +00:00
Terry Wilson 57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Terry Wilson 34c55e8e7c Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-13 17:27:06 +00:00
Jason Parker 6749b6e2be Fix a voicemail memory leak with heard/deleted messages.
open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
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2012-02-10 22:44:12 +00:00
Matthew Jordan a8cf4dc2b5 Fix IMAP app_voicemail compilation issue introduced in r354429
This simply fixes the compilation issue introduced in r354429 by
re-adding the 'quote' variable.

(closes issue ASTERISK-19337)
Reported by: John Taylor


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-10 14:51:27 +00:00
Walter Doekes db24fc2523 Avoid cppcheck warnings; removing unused vars and a bit of cleanup.
Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-08 20:49:48 +00:00
Paul Belanger 5be89b07e2 Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
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2012-01-25 22:25:30 +00:00
Terry Wilson 99cae5b750 Opaquify channel stringfields
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 20:12:09 +00:00
Terry Wilson 04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Matthew Jordan b0243fb57c Allow overriding of IMAP server settings on a user by user basis
This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user.  It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.

(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1614/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23 21:19:52 +00:00
Jonathan Rose 1b0741c7db Voicemail with the saycid option will now play a caller's name based on cid if available.
In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)

(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
	r uploaded by Russel Brown (license 6182)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16 22:00:37 +00:00
Walter Doekes fd64bb66f9 Add VM_INFO() dialplan function to gather information about a mailbox.
Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname,
language, locale, pager, password, tz.

(closes issue ASTERISK-18634)
Patch by: Kris Shaw
Review: https://reviewboard.asterisk.org/r/1568
Reviewed by: Walter Doekes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06 20:23:13 +00:00
Walter Doekes 7bdaa31d25 Add regression tests for issue ASTERISK-18838.
Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06 19:30:14 +00:00
Walter Doekes 03fd2c0c94 The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.

(closes issue ASTERISK-18838)

Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06 19:28:18 +00:00
Tilghman Lesher 77b670c4ab Allow each logging destination and console to have its own notion of the verbosity level.
Review: https://reviewboard.asterisk.org/r/1599


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29 18:43:16 +00:00
Jonathan Rose 2d67b1b378 Guarantee messages go into the right folders with multiple recipients
Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.

(closes issue ASTERISK-18245)
Reported by: Matt Jordan

(closes issue ASTERISK-18246)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1589/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-16 14:56:03 +00:00
Jonathan Rose ec237d2e4a Moves voicemail setup password entry to the end of the setup process.
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.

(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
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2011-11-14 16:21:06 +00:00
Paul Belanger 2ffea6ddc3 Multiple revisions 341108,341112
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  r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines
  
  Voicemail compiler flags are 'core' support
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  r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines
  
  Fix previous commit
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2011-10-17 16:27:42 +00:00
Richard Mudgett 55b70ae625 Merged revisions 337974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines
  
  Merged revisions 337973 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines
    
    Fix deadlock when using dummy channels.
    
    Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
    ast_channel_unref().  Using ast_channel_release() needlessly grabs the
    channel container lock and can cause a deadlock as a result.
    
    * Analyzed use of ast_dummy_channel_alloc() and made use
    ast_channel_unref() when done with the dummy channel.  (Primary reason for
    the reported deadlock.)
    
    * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
    locks.  Chan_local could not perform deadlock avoidance correctly.
    (Potential deadlock exposed by this issue.  Secondary reason for the
    reported deadlock since the held lock was part of the deadlock chain.)
    
    * Fixed some uses of ast_dummy_channel_alloc() not checking the returned
    channel pointer for failure.
    
    * Fixed some potential chan=NULL pointer usage in func_odbc.c.  Protected
    by testing the bogus_chan value.
    
    * Fixed needlessly clearing a 1024 char auto array when setting the first
    char to zero is enough in manager.c:action_getvar().
    
    (closes issue ASTERISK-18613)
    Reported by: Thomas Arimont
    Patches:
          jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Thomas Arimont
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2011-09-26 19:40:12 +00:00
Tilghman Lesher 90a7ed9901 More silly spacing changes
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2011-09-21 21:26:34 +00:00
Tilghman Lesher 4730309675 ................
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Dumb little spacing fix.
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2011-09-21 21:10:14 +00:00