asterisk/asterisk.git
8 years agoDestroy the generic_monitors container after the core_instances in ccss
Matthew Jordan [Wed, 3 Oct 2012 17:27:53 +0000 (17:27 +0000)]
Destroy the generic_monitors container after the core_instances in ccss

For each item in core_instances disposed of in the shutdown of ccss, any
generic monitor instances referenced by the objects will be removed from
generic_monitors during their destruction.  Hilarity ensues if
generic_monitors no longer exists.

Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly
when it ran into this.
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Merged revisions 374300 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374301 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374302 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMissed an astobj2.c debug tag.
Richard Mudgett [Tue, 2 Oct 2012 23:23:30 +0000 (23:23 +0000)]
Missed an astobj2.c debug tag.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago* Add ref debug tags to astobj2.c ref usage.
Richard Mudgett [Tue, 2 Oct 2012 22:39:47 +0000 (22:39 +0000)]
* Add ref debug tags to astobj2.c ref usage.

* Make container nodes not show up in the ref debug log.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure Shutdown AMI event is still fired during Asterisk shutdown
Matthew Jordan [Tue, 2 Oct 2012 21:26:27 +0000 (21:26 +0000)]
Ensure Shutdown AMI event is still fired during Asterisk shutdown

Richard pointed out that having the manager dispose of itself gracefully
during shutdown meant that the Shutdown event will no longer get fired.
This patch moves the AMI event just prior to running the atexit callbacks.
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Merged revisions 374230 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374231 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374248 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoModify hashtest2 to compile after r374213. Someone, somewhere, may care.
Matthew Jordan [Tue, 2 Oct 2012 20:45:22 +0000 (20:45 +0000)]
Modify hashtest2 to compile after r374213.  Someone, somewhere, may care.

Because hashtest2 has to provide symbols for things in asterisk that items
it includes may use, when astobj2 decided to use ast_register_atexit it needed
to provide a declaration for that as well.  Otherwise - no linky.

On a related note, ASTERISK-20505 was filed to convert hashtest/hashtest2 into
actual unit tests, so we don't run into this problem again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix findings from check-in on r374177
Matthew Jordan [Tue, 2 Oct 2012 17:16:20 +0000 (17:16 +0000)]
Fix findings from check-in on r374177

Richard pointed out two problems with the check-in from r374177:
* The ast_msg_shutdown function declaration doesn't match the prototype
  in main/message.c.
* The ref/alloc function usage in astobj2 (in trunk) can use the ao2_t_*
  variants of the functions to allow the REF_DEBUG flag to enable/disable
  their debug counterparts.
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Merged revisions 374210 from http://svn.asterisk.org/svn/asterisk/branches/10
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8 years agoFix a variety of ref counting issues
Matthew Jordan [Tue, 2 Oct 2012 01:47:16 +0000 (01:47 +0000)]
Fix a variety of ref counting issues

This patch resolves a number of ref leaks that occur primarily on Asterisk
shutdown.  It adds a variety of shutdown routines to core portions of
Asterisk such that they can reclaim resources allocate duringd initialization.

Review: https://reviewboard.asterisk.org/r/2137
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Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:39:45 +0000 (23:39 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:24:35 +0000 (23:24 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:24:10 +0000 (23:24 +0000)]
Doxygen Cleanup

Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374165 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Cleanup
Andrew Latham [Mon, 1 Oct 2012 23:22:50 +0000 (23:22 +0000)]
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

8 years agoapp_queue: Support persisting and loading of long member lists.
Sean Bright [Mon, 1 Oct 2012 20:36:25 +0000 (20:36 +0000)]
app_queue: Support persisting and loading of long member lists.

Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10.  dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case.  This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.

The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.

As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.

Review: https://reviewboard.asterisk.org/r/2136/
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Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUse ast_copy_string instead of strncpy to guarantee a NUL terminated string.
Sean Bright [Mon, 1 Oct 2012 17:28:41 +0000 (17:28 +0000)]
Use ast_copy_string instead of strncpy to guarantee a NUL terminated string.
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Merged revisions 374132 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374133 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange core show help output format.
Richard Mudgett [Mon, 1 Oct 2012 17:05:37 +0000 (17:05 +0000)]
Change core show help output format.

The CLI "core show help" output leaves something to be desired.
1) The command is truncated to a maximum of 30 characters.
2) The output columns are mirrored from the 31st column.

Current output format:
                   logger mute Toggle logging output to a console
                 logger reload Reopens the log files
                 logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging level for this console
          logger show channels List configured log channels

New format:
logger mute                    -- Toggle logging output to a console
logger reload                  -- Reopens the log files
logger rotate                  -- Rotates and reopens the log files
logger set level {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} -- Enables/Disables a specific logging level for this console
logger show channels           -- List configured log channels

Review: https://reviewboard.asterisk.org/r/2133/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDon't destroy confbridge config when error is encountered during a reload.
Mark Michelson [Mon, 1 Oct 2012 16:26:23 +0000 (16:26 +0000)]
Don't destroy confbridge config when error is encountered during a reload.

Not panicking means that the old config is kept.

(closes issue ASTERISK-20458)
Reported by: Leif Madsen
Patches:
ASTERISK-20458.patch uploaded by Mark Michelson(license #5049)
Tested by Leif Madsen
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Merged revisions 374106 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374107 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd support for retrieving engine specific settings using the speech API and from...
Joshua Colp [Mon, 1 Oct 2012 12:29:04 +0000 (12:29 +0000)]
Add support for retrieving engine specific settings using the speech API and from dialplan.

(closes issue ASTERISK-17136)
Reported by: kenner

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix ref leak when adding ICE candidates to an SDP
Matthew Jordan [Sat, 29 Sep 2012 03:56:49 +0000 (03:56 +0000)]
Fix ref leak when adding ICE candidates to an SDP

There was a missing decrement to the reference count for the current ICE
candidate when local candidates are being added to an outbound SDP.  This
patch corrects that.
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Merged revisions 374085 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoInclude channel uniqueid in "AsyncAGI" and "AGIExec" events.
Richard Mudgett [Fri, 28 Sep 2012 22:11:19 +0000 (22:11 +0000)]
Include channel uniqueid in "AsyncAGI" and "AGIExec" events.

* Added AMI event documentation for AsyncAGI and AGIExec events.

(closes issue ASTERISK-20318)
Reported by: Dan Cropp
Patches:
      res_agi_patch.txt (license #6422) patch uploaded by Dan Cropp
      modified for trunk.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_jabber: Remove CLI command 'jabber test'
Jonathan Rose [Fri, 28 Sep 2012 19:37:22 +0000 (19:37 +0000)]
res_jabber: Remove CLI command 'jabber test'

The opinion of development was that it is both improper to have Matt's
personal email address used in the source and that the command wouldn't
be useful without it.

(closes issue AST-467)
Reported by: Malcolm Davenport
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Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374045 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374059 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd pause one second W dial modifier.
Richard Mudgett [Fri, 28 Sep 2012 18:27:02 +0000 (18:27 +0000)]
Add pause one second W dial modifier.

* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second.  Dial, ExternalIVR, and SendDTMF.

* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'.  The 'w' pauses dialing for half a
second.  The 'W' pauses dialing for one second.

* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.

(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
      jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
      Expanded patch to add support in chan_dahdi.
Tested by: rmudgett

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoReset hangup flags on channels created through messages and cleanup globals
Brent Eagles [Fri, 28 Sep 2012 13:04:11 +0000 (13:04 +0000)]
Reset hangup flags on channels created through messages and cleanup globals
in res_xmpp on unload.

This patch fixes an issue where hangup flags were not being reset on a
channel, affecting subsequent use of that channel. The patch also adds some
additional cleanup to res_xmpp to fix an issue with reloading the module.

(closes ASTERISK-20360)
Reported by: Noah Engelberth
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2134/
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Merged revisions 374019 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate documentation to make it explicit that "stream file" will not restart musiconhold.
Joshua Colp [Fri, 28 Sep 2012 12:17:41 +0000 (12:17 +0000)]
Update documentation to make it explicit that "stream file" will not restart musiconhold.

(issue ASTERISK-17367)
Reported by: oej
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Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373990 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373991 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd Duration header for PlayDTMF AMI Action
Matthew Jordan [Fri, 28 Sep 2012 03:06:53 +0000 (03:06 +0000)]
Add Duration header for PlayDTMF AMI Action

This patch adds an optional header to the PlayDTMF AMI action, Duration.
It allows the duration of the DTMF digit to be played on the channel to be
specified in milliseconds.

(closes issue ASTERISK-18172)
Reported by: Renato dos Santos

patches:
  send-dtmf.patch uploaded by Renato dos Santos (license #6267)

Modified slightly for this commit for Asterisk 12.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTweak app_dial documentation.
Richard Mudgett [Thu, 27 Sep 2012 22:43:27 +0000 (22:43 +0000)]
Tweak app_dial documentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373967 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCleanup ast_dtmf_stream()
Richard Mudgett [Thu, 27 Sep 2012 22:33:15 +0000 (22:33 +0000)]
Cleanup ast_dtmf_stream()

* Made ast_dtmf_stream() wait after starting the silence generator rather
than before.

* Made ast_dtmf_stream() put the peer in autoservice for the whole time
things are being done to the chan.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373966 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix SendDTMF crash and channel reference leak using channel name parameter.
Richard Mudgett [Thu, 27 Sep 2012 22:25:34 +0000 (22:25 +0000)]
Fix SendDTMF crash and channel reference leak using channel name parameter.

The SendDTMF channel name parameter has two issues.
1) Crashes if the channel name does not exist.
2) Leaks a channel reference if the channel is the current channel.
Problem introduced by ASTERISK-15956.

* Updated SendDTMF documentation.

* Renamed app to senddtmf_name and tweaked the type.
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Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373946 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373954 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373965 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake res_http_websocket an optional dependency on supported platforms for chan_sip.
Joshua Colp [Thu, 27 Sep 2012 17:12:08 +0000 (17:12 +0000)]
Make res_http_websocket an optional dependency on supported platforms for chan_sip.

(closes issue ASTERISK-20439)
Reported by: sruffell
Patches:
     0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417)
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Merged revisions 373914 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd VoicemailRefresh AMI Action
Kinsey Moore [Thu, 27 Sep 2012 17:02:13 +0000 (17:02 +0000)]
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

8 years agoloader: Ensure dependent modules are properly initialized.
Joshua Colp [Thu, 27 Sep 2012 16:53:19 +0000 (16:53 +0000)]
loader: Ensure dependent modules are properly initialized.

If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it
is possible for Asterisk to skip calling the modules's .load function.
Asterisk was loading and linking the module via load_dynamic_module() but was
not adding the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules in the heap.

Now use load_resource() instead of load_dynamic_module() for non-optional
requirement. This will add the module to the resource_heap so the module can
be properly initialized in the correct order.

This is required if there are any module global data structures initialized in
the .load() callback for the module on platforms which do not support weak
references.

(issue ASTERISK-20439)
Reported by: sruffell
Patches:
     0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417)
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8 years agoFix an issue where Local channels dialed by app_queue are considered in use immediately.
Joshua Colp [Thu, 27 Sep 2012 11:33:54 +0000 (11:33 +0000)]
Fix an issue where Local channels dialed by app_queue are considered in use immediately.

The chan_local channel driver returns a device state of in use even if a created Local
channel has not yet been dialed. This fix changes the logic to return a state of not
in use until the channel itself has been dialed.

(closes issue ASTERISK-20390)
Reported by: tim_ringenbach

Review: https://reviewboard.asterisk.org/r/2116/
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8 years agoMove handling of 408 response so there is no misleading warning message.
Mark Michelson [Wed, 26 Sep 2012 21:17:16 +0000 (21:17 +0000)]
Move handling of 408 response so there is no misleading warning message.

(closes issue ASTERISK-20060)
Reported by: Walter Doekes
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8 years agoFixed meetme tab completion and command documentation.
Richard Mudgett [Wed, 26 Sep 2012 18:23:37 +0000 (18:23 +0000)]
Fixed meetme tab completion and command documentation.

* Removed unnecessary case sensitivity in meetme list, lock, unlock, mute,
unmute, and kick commands.

* Separated meetme lock/unlock, mute/unmute, and kick commands into their
own registered commands to simplify tab completion and parameter checking.
meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd()

* Simplified meetme_show_cmd()

(closes issue AST-1006)
Reported by: John Bigelow
Tested by: rmudgett
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8 years agoapp_queue: 'agent available' hint, cleanup restart, and initial state
Alec L Davis [Wed, 26 Sep 2012 08:31:46 +0000 (08:31 +0000)]
app_queue: 'agent available' hint, cleanup restart, and initial state

Fix previously untested senarios;

1). On queue initialisation set queue_avail devstate to INUSE.
    Previously was unavailable, which indicated an agent was available.

2). When removing members, if there are no other members available, set queue_avail to INUSE.
    Previously, if a member interface had become 'unavailable', they were never going to be removed, particularly when persistant queues is enabled.

3). When adding a member, check that they are available, if they are set queue_avail to NOT_INUSE.
 Previously on reloaded, members may have been 'unavailable'.

4). When pausing or unpausing a member, set appropriate queue availability.

alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/2129/
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8 years agoFix saying of date in Dutch.
Mark Michelson [Tue, 25 Sep 2012 23:10:22 +0000 (23:10 +0000)]
Fix saying of date in Dutch.

The Dutch say the date before the month.

(closes issue ASTERISK-20353)
Reported by: Teun Ouwehand
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8 years agoRemove dead code and documentation for nonexistent feature.
Mark Michelson [Tue, 25 Sep 2012 22:57:56 +0000 (22:57 +0000)]
Remove dead code and documentation for nonexistent feature.

multiplelogin was removed from chan_agent back in 1.6.0 when
AgentCallbackLogin() was removed.

(closes issue AST-948)
reported by Steve Pitts
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8 years agoFix error where improper IMAP greetings would be deleted.
Mark Michelson [Tue, 25 Sep 2012 21:14:21 +0000 (21:14 +0000)]
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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8 years agoFix T.38 support when used with chan_local in between.
Joshua Colp [Tue, 25 Sep 2012 20:14:13 +0000 (20:14 +0000)]
Fix T.38 support when used with chan_local in between.

Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the
channel indicate a T.38 negotiation with the parameters present on the channel. The return
value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with
chan_local involved this could never occur.

This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If
the underlying channel technology on the other side does not support T.38 this would have
been determined ahead of time using ast_channel_get_t38_state and an indication would
not occur.

(closes issue ASTERISK-20229)
Reported by: wdoekes
Patches:
     ASTERISK-20229.patch uploaded by wdoekes (license 5674)

Review: https://reviewboard.asterisk.org/r/2070/
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8 years agoAllow for redirecting reasons to be set to arbitrary strings.
Mark Michelson [Tue, 25 Sep 2012 19:29:14 +0000 (19:29 +0000)]
Allow for redirecting reasons to be set to arbitrary strings.

This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.

The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.

(closes issue AST-942)
reported by Malcolm Davenport

(closes issue AST-943)
reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/2101

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoProperly handle UAC/UAS roles for SIP session timers
Terry Wilson [Tue, 25 Sep 2012 19:08:02 +0000 (19:08 +0000)]
Properly handle UAC/UAS roles for SIP session timers

The SIP session timer mechanism contains a mandatory 'refresher' parameter
(included in the Session-Expires header) which is used in the session timer
offer/answer signaling within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of client and
server (caller is uac, callee is uas). The standard rfc 4028 however assigns
the client role to the ((RE)-Invite) requester, the server role to the
((RE)-Invite) responder.

This patch has Asterisk track the actual refresher as "us" or "them" as opposed
to relying on just the configured "uas" or "uac" properties.

(closes issue AST-922)
Reported by: Thomas Airmont

Review: https://reviewboard.asterisk.org/r/2118/
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8 years ago"show" completion option for "queue" shouldn't appear twice
Kinsey Moore [Tue, 25 Sep 2012 18:33:59 +0000 (18:33 +0000)]
"show" completion option for "queue" shouldn't appear twice

When tab-completing CLI commands starting with "queue", "show" appeared
twice in the list due to the way that Asterisk's tab completion
functions and the order in which the commands were registered. The
registration order has been altered to resolve this issue.

(closes issue AST-940)
Reported-by: Steve Pitts
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8 years agoFix valgrind found memcpy issues in codec_ilbc.
Richard Mudgett [Tue, 25 Sep 2012 17:22:25 +0000 (17:22 +0000)]
Fix valgrind found memcpy issues in codec_ilbc.

Valgrind found codec_ilbc using memcpy instead of memmove for overlapping
memory blocks.

(issue ASTERISK-19890)
(closes issue ASTERISK-20231)
Reported by: Walter Doekes
Patches:
      ASTERISK-20231.patch (license #5674) patch uploaded by Walter Doekes
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8 years agoMake rebuild GSM, ilbc, or lpc10 codecs if the respective sources change.
Richard Mudgett [Tue, 25 Sep 2012 17:02:21 +0000 (17:02 +0000)]
Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change.
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8 years agochan_sip: Set Quality of Service for video rtp instance
Jonathan Rose [Tue, 25 Sep 2012 16:45:02 +0000 (16:45 +0000)]
chan_sip: Set Quality of Service for video rtp instance

(closes issue ASTERISK-20201)
Reported by: ddkprog
Patches:
    chan_sip.c.diff uploaded by ddkprog (license 6008)
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8 years agores_agi: async_agi responsiveness improvement on datastore problems
Jonathan Rose [Tue, 25 Sep 2012 14:53:42 +0000 (14:53 +0000)]
res_agi: async_agi responsiveness improvement on datastore problems

This patch changes get_agi_cmd so that the return can be checked
to differentiate between an empty list success and something that
triggered an error. This in turn allows launch_asyncagi to detect
these errors and break free from the command processing loop so
that the async agi can be ended more cleanly

(closes issue ASTERISK-20109)
Reported by: Jeremiah Gowdy
Patches: jgowdy-7-9-2012.diff uploaded by Jeremiah Gowdy (license 6358)
           (Modified by me to fix some logical issues and apply to trunk)
Review: https://reviewboard.asterisk.org/r/2117/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago"He who go through turnstile sideways is going to Bangkok"
Mark Michelson [Tue, 25 Sep 2012 14:13:08 +0000 (14:13 +0000)]
"He who go through turnstile sideways is going to Bangkok"
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8 years agoFix documentation for default username in res_odbc
Kinsey Moore [Tue, 25 Sep 2012 13:29:37 +0000 (13:29 +0000)]
Fix documentation for default username in res_odbc

This was previously stated to be "root", but is actually the name of
the context if unspecified.

(closes issue ASTERISK-20258)
Reported by: Stefan x
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8 years agoFix an issue where a caller to ast_write on a MulticastRTP channel would determine...
Joshua Colp [Tue, 25 Sep 2012 12:12:20 +0000 (12:12 +0000)]
Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not.

When sending RTP packets via multicast the amount of data sent is stored in a variable and returned
from the write function. This is incorrect as any non-zero value returned is considered a failure while
a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing went wrong and it was actually a success.

The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should.

(closes issue ASTERISK-17254)
Reported by: wybecom
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8 years agoBe consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>
Richard Mudgett [Mon, 24 Sep 2012 22:14:28 +0000 (22:14 +0000)]
Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>

When setting CALLERID(pres)=unavailable in the dialplan, the From header
in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>.
For consistency, Asterisk should use a lowercase a in the userpart of the
URI.

* Make the From header use a lowercase A in the userpart of the anonymous
URI.

(closes issue ASTERISK-19838)
Reported by: Antti Yrjola
Patches:
      chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola
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8 years agofunc_audiohookinherit: Document some missed sources.
Jonathan Rose [Mon, 24 Sep 2012 21:19:49 +0000 (21:19 +0000)]
func_audiohookinherit: Document some missed sources.

This patch also mentions that AUDIOHOOK_INHERIT can be used to
transfer MixMonitor audiohooks. There is also wiki that addresses
audiohooks and the use of AUDIOHOOK_INHERIT at the following link:
https://wiki.asterisk.org/wiki/display/AST/Audiohooks

(closes issue ASTERISK-18220)
Reported by: Ishfaq Malik
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8 years agoFix potential reentrancy problems in chan_sip.
Richard Mudgett [Mon, 24 Sep 2012 21:15:26 +0000 (21:15 +0000)]
Fix potential reentrancy problems in chan_sip.

Asterisk v1.8 and later was not as vulnerable to this issue.

* Made find_call() lock each private as it processes the found dialogs.
(Primary cause of ABE-2876)

* Made the other functions that traverse the dialogs container lock each
private as it examines them.

* Fix race condition in sip_call() if the thread that sent the INVITE is
held up long enough for a response to be processed.  The p->initid for the
INVITE retransmission could be added after it was canceled by the response
processing.

* Made __sip_destroy() clean up resource pointers after freeing.  This is
primarily defensive in case someone has a stale private pointer.

* Removed redundant memset() in reqprep().  The call to init_req() already
does the memset() and is the first reference to req in reqprep().

* Removed useless set of req.method in transmit_invite().  The calls to
initreqprep() and reqprep() have to do this because they memset() the req.

JIRA ABE-2876

..........

Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373466 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373469 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix a deadlock caused by a race condition between removing a hint and reloading the...
Joshua Colp [Mon, 24 Sep 2012 19:23:32 +0000 (19:23 +0000)]
Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint.

If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other
wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it
had the SIP dialog lock and wanted the contexts lock.

This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock
it. Once the extension state is retrieved the SIP dialog is locked again and life carries on.

As the SIP dialog is reference counted it is not possible for it to go away after unlocking.

(closes issue ASTERISK-20437)
Reported by: jhutchins
........

Merged revisions 373438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373440 from http://svn.asterisk.org/svn/asterisk/branches/10
........

Merged revisions 373454 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373456 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix an issue with H.264 format attribute comparison and fix an issue with improper...
Joshua Colp [Mon, 24 Sep 2012 14:27:17 +0000 (14:27 +0000)]
Fix an issue with H.264 format attribute comparison and fix an issue with improper SDP being produced.

The H.264 format attribute module compares two format attribute structures to determine if they are
compatible or not. In some instances it was possible for this check to determine that both structures
were incompatible when they actually should be considered compatible. This check has now been made even
more permissive by assuming that if no attribute information is available the two structures are compatible.
If both structures contain attribute information a base level comparison of the H.264 IDC value is done to
see if they are compatible or not.

The above issue uncovered a secondary issue in chan_sip where the SDP being produced would be incorrect if
the formats were considered incompatible. This has now been fixed by checking that all information required
to produce the SDP is available instead of assuming it is.

(closes issue ASTERISK-20464)
Reported by: Leif Madsen
........

Merged revisions 373413 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_rtp_asterisk: Make TURN and STUN server configurations consistent.
Brent Eagles [Mon, 24 Sep 2012 12:42:19 +0000 (12:42 +0000)]
res_rtp_asterisk: Make TURN and STUN server configurations consistent.

This patch removes the turnport configuration property and changes the
turnaddr property to be a combined host[:port] configuration string. The
patch also modifies the documentation in the example configuration to
reflect the property changes and adds some additional text indicating how
the STUN port is configured.

(closes issue ASTERISK-20344)
Reported by: beagles
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2111/
........

Merged revisions 373403 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Updates Janitor Work
Andrew Latham [Sat, 22 Sep 2012 20:43:30 +0000 (20:43 +0000)]
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

8 years agoiax2-provision: Fix improper return on failed cache retrieval
Jonathan Rose [Fri, 21 Sep 2012 19:35:37 +0000 (19:35 +0000)]
iax2-provision: Fix improper return on failed cache retrieval

(closes issue ASTERISK-20337)
reported by: John Covert
Patches:
    iax2-provision.c.patch uploaded by John Covert (license 5512)
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Merged revisions 373342 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373343 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373368 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373369 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate Doxygen Config Comments
Andrew Latham [Fri, 21 Sep 2012 18:22:05 +0000 (18:22 +0000)]
Update Doxygen Config Comments

This annoying update is almost totally whitespace and updated config comments. I did add Python to the documented file types.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDoxygen Updates - janitor work
Andrew Latham [Fri, 21 Sep 2012 17:14:59 +0000 (17:14 +0000)]
Doxygen Updates - janitor work

Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style.  Some missing txt file links are removed but their content or essense will be included in some later updates.  A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.

Further updates coming.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoStart work on documentation janitor project with a little commit. This adds a link...
Andrew Latham [Fri, 21 Sep 2012 16:06:30 +0000 (16:06 +0000)]
Start work on documentation janitor project with a little commit. This adds a link to the Asterisk wiki at https://wiki.asterisk.org to the README file.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoapp_queue: Make queue reload members and variants of that work
Jonathan Rose [Fri, 21 Sep 2012 15:41:09 +0000 (15:41 +0000)]
app_queue: Make queue reload members and variants of that work

Prior to this patch, 'queue reload members' cli command did not
work at all. This also affects the manager function 'QueueReload'
when supplied with the 'members: yes' field.

(closes issue AST-956)
Reported by: John Bigelow
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Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373300 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373318 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agodsp.c: remove more whitespace mentioned in review2107
Alec L Davis [Fri, 21 Sep 2012 09:11:39 +0000 (09:11 +0000)]
dsp.c: remove more whitespace mentioned in review2107

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agodsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup
Alec L Davis [Fri, 21 Sep 2012 06:51:25 +0000 (06:51 +0000)]
dsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup

janitor cleanup. No functional change.

1). ast_dsp_call_progress: use 'short samp' instead of s[x] inside loop.
    apply same casting as other _init, dsp->energy = (int32_t) samp * (int32_t) samp

2). ast_dtmf_detect_init: move repeated setting of s->energy to outside of loop.
    do goertzel_init loop first before setting s->lasthit and s->current_hit, consistant with ast_dsp_digitreset()

3). ast_mf_detect_init:
    do goertzel_init loop first before setting s->hits[] and s->current_hit, consistant with ast_dsp_digitreset()

4). Don't chain init different variables, as the type may change

Review https://reviewboard.asterisk.org/r/2107/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix incorrect MeetME conference bridge reference count decrementing and sometimes...
Joshua Colp [Thu, 20 Sep 2012 19:16:59 +0000 (19:16 +0000)]
Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction.

When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see
if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one.
This operation left around a pointer to the last created conference bridge still containing participants.
When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of
and the reference count of the conference bridge decremented. If there was only a single participant in the
conference bridge it was ultimately destroyed prematurely.

(closes issue AST-994)
Reported by: John Bigelow
........

Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373245 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373246 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373247 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlocked revisions 373240
Matthew Jordan [Thu, 20 Sep 2012 18:59:39 +0000 (18:59 +0000)]
Blocked revisions 373240

........
app_queue: Support an 'agent available' hint

Sets INUSE when no free agents, NOT_INUSE when an agent is free.

modifes handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate.

Previously exited early if the member was found in the queue.

Now Exits later when both a member was found, and a free agent was found.

alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/2121/

~~~~

Support all ways a member can be available for 'agent available' hints

Alec's patch in r373188 added the ability to subscribe to a hint for when
Queue members are available.  This patch modifies the check that determines
when a Queue member is available by refactoring the availability checks in
num_available_members into a shared function is_member_available.  This
should now handle the ringinuse option, as well as device state values
other than AST_DEVICE_NOT_INUSE.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd queue monitoring hints
Matthew Jordan [Thu, 20 Sep 2012 18:44:26 +0000 (18:44 +0000)]
Add queue monitoring hints

This patch adds support for hints on a queue.  Hints can be added using
the nomenclature 'Queue:name', where name is the name of the queue being
monitored.

This nifty feature was done by Alec Davis.

Review: https://reviewboard.asterisk.org/r/1619

Reported by: Alec Davis
Tested by: alecdavis
patches:
  review1619.diff2 by alecdavis (license 585)
........

Merged revisions 373235 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373239 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
Joshua Colp [Thu, 20 Sep 2012 18:27:28 +0000 (18:27 +0000)]
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.

As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.

Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.

Review: https://reviewboard.asterisk.org/r/2113/
........

Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSupport all ways a member can be available for 'agent available' hints
Matthew Jordan [Thu, 20 Sep 2012 18:02:02 +0000 (18:02 +0000)]
Support all ways a member can be available for 'agent available' hints

Alec's patch in r373188 added the ability to subscribe to a hint for when
Queue members are available.  This patch modifies the check that determines
when a Queue member is available by refactoring the availability checks in
num_available_members into a shared function is_member_available.  This
should now handle the ringinuse option, as well as device state values
other than AST_DEVICE_NOT_INUSE.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373222 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoNamed call pickup groups. Fixes, missing functionality, and improvements.
Richard Mudgett [Thu, 20 Sep 2012 17:22:41 +0000 (17:22 +0000)]
Named call pickup groups. Fixes, missing functionality, and improvements.

* ASTERISK-20383
Missing named call pickup group features:

CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.

* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up.  In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.

Regression because of the named call pickup group feature.

* See ASTERISK-20386 for the implementation improvements.  These are the
changes in channel.c and channel.h.

* Fixed some locking issues in CHANNEL().

(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2112/
........

Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrect handling of unknown SDP stream types
Kinsey Moore [Thu, 20 Sep 2012 13:04:22 +0000 (13:04 +0000)]
Correct handling of unknown SDP stream types

When the patch to handle arbitrary SDP stream arrangements went into
Asterisk, it also included an ability to transparently decline unknown
stream types. The scanf calls used were not checked properly causing
this part of the functionality to be broken.

(closes issue ASTERISK-20203)
........

Merged revisions 373211 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoWhen trying to unload res_curl.so, warn about all dependent modules.
Sean Bright [Thu, 20 Sep 2012 11:05:40 +0000 (11:05 +0000)]
When trying to unload res_curl.so, warn about all dependent modules.

Before this, attempting to unload res_curl.so would warn you about the first
module it found that was dependent.  We now warn about all of the loaded modules
instead.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agodsp.c: remove whitespace mentioned in review2107
Alec L Davis [Thu, 20 Sep 2012 10:41:30 +0000 (10:41 +0000)]
dsp.c: remove whitespace mentioned in review2107

Related https://reviewboard.asterisk.org/r/2107/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373202 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoapp_queue: Support an 'agent available' hint
Alec L Davis [Wed, 19 Sep 2012 22:33:12 +0000 (22:33 +0000)]
app_queue: Support an 'agent available' hint

Sets INUSE when no free agents, NOT_INUSE when an agent is free.

modifes handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate.

Previously exited early if the member was found in the queue.

Now Exits later when both a member was found, and a free agent was found.

alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/2121/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake the casing of CALL_ID in debug messages consistent to satisfy my OCD.
Sean Bright [Tue, 18 Sep 2012 20:19:49 +0000 (20:19 +0000)]
Make the casing of CALL_ID in debug messages consistent to satisfy my OCD.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373142 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDon't crash when passing a NULL message to __astman_get_header.
Sean Bright [Tue, 18 Sep 2012 20:14:33 +0000 (20:14 +0000)]
Don't crash when passing a NULL message to __astman_get_header.

Before this commit, __astman_get_header would blindly dereference the passed in
'struct message *' to traverse the header list.  There are cases, however, such
as '*CLI> sip qualify peer foo' where the message pointer is NULL, so we need
to check for that.
........

Merged revisions 373131 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 373132 from http://svn.asterisk.org/svn/asterisk/branches/10
........

Merged revisions 373133 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd -fnested-functions compile flag, if needed.
David M. Lee [Tue, 18 Sep 2012 15:50:35 +0000 (15:50 +0000)]
Add -fnested-functions compile flag, if needed.

In order to use nested functions on some versions of GCC (e.g. GCC on OS X),
the -fnested-functions flag must be passed to the compiler. This patch adds
detection logic to ./configure to add the flag if necessary. It also adds
a comment to utils.h as to why the nested function needs a prototype.

(closes issue ASTERISK-20399)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2102/
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Merged revisions 373119 from http://svn.asterisk.org/svn/asterisk/branches/11

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8 years agoMade companding law for SS7 calls only determined by SS7 signaling type.
Richard Mudgett [Sat, 15 Sep 2012 00:32:37 +0000 (00:32 +0000)]
Made companding law for SS7 calls only determined by SS7 signaling type.

For SS7, the companding law for a call was chosen inconsistently depending
upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1).
For incoming calls, the companding law was determined by ss7type.  For
outgoing calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts.  An
A-law/u-law conflict sounds like bad static on the line.

SS7 ITU  signaling with E1 line: ok
SS7 ITU  signaling with T1 line: noise
SS7 ANSI signaling with E1 line: noise
SS7 ANSI signaling with T1 line: ok

* Fix the companding law used to be determined by the SS7 signaling type
only.
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Merged revisions 373107 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373108 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoResolve memory leaks in TLS initialization and TLS client connections
Matthew Jordan [Fri, 14 Sep 2012 19:53:43 +0000 (19:53 +0000)]
Resolve memory leaks in TLS initialization and TLS client connections

This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
   portions of the SSL library.  Asterisk calls SSL_library_init and
   SSL_load_error_strings during SSL initialization; collectively this
   obviates the need for calling any of the following during initialization
   or client connection handling:
   * ERR_load_crypto_strings (handled by SSL_load_error_strings)
   * OpenSSL_add_all_algorithms (synonym for SSL_library_init)
   * SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
   the SSL library for TLS clients.  This included not freeing the SSL_CTX
   object in the SIP channel driver, as well as not clearing the error
   stack when the TLS client exited.

Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.

(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
  (bugAST-889.patch) by Thomas Arimont (license 5525)

Review: https://reviewboard.asterisk.org/r/2105
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Merged revisions 373079 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373080 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed make clean when configured --disable-asteriskssl
David M. Lee [Thu, 13 Sep 2012 20:05:54 +0000 (20:05 +0000)]
Fixed make clean when configured --disable-asteriskssl
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Merged revisions 373047 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix timeouts for ast_waitfordigit[_full].
David M. Lee [Thu, 13 Sep 2012 20:02:56 +0000 (20:02 +0000)]
Fix timeouts for ast_waitfordigit[_full].

ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!

This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.

(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnhance astobj2 to support other types of containers.
Richard Mudgett [Wed, 12 Sep 2012 21:02:29 +0000 (21:02 +0000)]
Enhance astobj2 to support other types of containers.

The new API allows for sorted containers, insertion options, duplicate
handling options, and traversal order options.

* Adds the ability for containers to be sorted when they are created.

* Adds container creation options to handle duplicates when they are
inserted.

* Adds container creation option to insert objects at the beginning or end
of the container traversal order.

* Adds OBJ_PARTIAL_KEY to allow searching with a partial key.  The partial
key works similarly to the OBJ_KEY flag.  (The real search speed
improvement with this flag will come when red-black trees are added.)

* Adds container traversal and iteration order options: Ascending and
Descending.

* Adds an AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats <name>" and
"astobj2 container check <name>".  The channels container is normally
registered since it is one of the most important containers in the system.

* Adds ao2_iterator_restart() to allow iteration to be restarted from the
beginning.

* Changes the generic container object to have a v_method table pointer to
support other types of containers.

* Changes the container nodes holding objects to be ref counted.

The ref counted nodes and v_method table pointer changes pave the way to
allow other types of containers.

* Includes a large astobj2 unit test enhancement that tests the new
features.

(closes issue ASTERISK-19969)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2078/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSkip any non-content information when looking for and handling content.
Joshua Colp [Wed, 12 Sep 2012 20:54:38 +0000 (20:54 +0000)]
Skip any non-content information when looking for and handling content.

This fixes a bug with Jitsi and conference calling. Jitsi implements XEP-0298
which places some conference-info information in the session-initiate request
which chan_motif did not expect to occur.
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Merged revisions 372995 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372996 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_xmpp: Fix a segfault caused by bodyless messages
Jonathan Rose [Wed, 12 Sep 2012 18:33:47 +0000 (18:33 +0000)]
res_xmpp: Fix a segfault caused by bodyless messages

(closes issue ASTERISK-20361)
Reported by: Noah Engelberth
Review: https://reviewboard.asterisk.org/r/2108/
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Merged revisions 372984 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agologger: Add rotatestrategy option of 'none' which does not perform rotations
Jonathan Rose [Wed, 12 Sep 2012 17:13:02 +0000 (17:13 +0000)]
logger: Add rotatestrategy option of 'none' which does not perform rotations

With this option in use, it may be necessary to regulate your log files
externally.

(closes issue ASTERISK-20189)
Reported by: Jaco Kroon
Patches:
    asterisk-logger-norotate-trunk.patch uploaded by Jaco Kroon (license 5671)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd channel name to a warning to make debugging easier.
Mark Michelson [Wed, 12 Sep 2012 15:21:19 +0000 (15:21 +0000)]
Add channel name to a warning to make debugging easier.

The "autodestruct with owner in place" message is typically
indicative of a channel reference leak. Printing out the name
of the channel in the message may be helpful when trying to
debug the issue.
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Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed r372696 when configured --disable-asteriskssl; properly install libasteriskssl...
David M. Lee [Wed, 12 Sep 2012 14:22:54 +0000 (14:22 +0000)]
Fixed r372696 when configured --disable-asteriskssl; properly install libasteriskssl.dylib on OS X.

I didn't realize that libasteriskssl.c was still compiled, even when you
disable asteriskssl; it simple gets statically linked into asterisk.
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8 years agochan_local: Switch from using a random 4 digit hex identifier to unique id
Jonathan Rose [Tue, 11 Sep 2012 22:40:02 +0000 (22:40 +0000)]
chan_local: Switch from using a random 4 digit hex identifier to unique id

Changes chan_local channels to use an 8 digit hex identifier generated
atomically and sequentially in order to eliminate the chance of having
multiple channels with the same name during high call volume situations.

(issue ASTERISK-20318)
Reported by: Dan Cropp
Review: https://reviewboard.asterisk.org/r/2104/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372918 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix inability to shutdown gracefully due to an unending channel reference.
Mark Michelson [Tue, 11 Sep 2012 21:17:53 +0000 (21:17 +0000)]
Fix inability to shutdown gracefully due to an unending channel reference.

message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.

This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.

(closes issue AST-937)
Reported by Jason Parker
Patches:
AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix bad channel application data reference.
Mark Michelson [Tue, 11 Sep 2012 21:13:26 +0000 (21:13 +0000)]
Fix bad channel application data reference.

When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.

(issue ASTERISK-20335)
Reported by: aragon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372887 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrects the astsbindir setting when installing the sample asterisk.conf.
David M. Lee [Tue, 11 Sep 2012 18:09:22 +0000 (18:09 +0000)]
Corrects the astsbindir setting when installing the sample asterisk.conf.

(closes issue ASTERISK-20406)
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Merged revisions 372863 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372864 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agochan_sip: Fix CHANGES and UPGRADE.txt for r372808
Jonathan Rose [Tue, 11 Sep 2012 14:43:41 +0000 (14:43 +0000)]
chan_sip: Fix CHANGES and UPGRADE.txt for r372808

(issue AST-969)
Reported by John Bigelow

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agochan_sip: Change SIPQualifyPeer to improve initial response time
Jonathan Rose [Mon, 10 Sep 2012 21:15:38 +0000 (21:15 +0000)]
chan_sip: Change SIPQualifyPeer to improve initial response time

Prior to this patch, The acknowledgement wasn't produced until after
executing the sip_poke_peer action actually responsible for
qualifying the peer. Now the response is given immediately once it is
known that a peer will be qualified and a SIPqualifypeerdone event
is issued when the process is finished. Thanks to OEJ for identifying
the problem and helping to come up with a solution.

(issue AST-969)
Reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/2098/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure iax2 debug output is displayed when expected
Kinsey Moore [Mon, 10 Sep 2012 21:00:22 +0000 (21:00 +0000)]
Ensure iax2 debug output is displayed when expected

When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.

(closes issue ASTERISK-20338)
Reported-by: John Covert
Patch-by: John Covert
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDeprecate chan_gtalk, chan_jingle, and res_jabber
Kinsey Moore [Mon, 10 Sep 2012 19:49:30 +0000 (19:49 +0000)]
Deprecate chan_gtalk, chan_jingle, and res_jabber

chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of
using chan_motif and res_xmpp. They are a feature-equivalent
replacement and are written to be more easily maintainable.

(closes issue ASTERISK-20298)
Review: https://reviewboard.asterisk.org/r/2082/
Reported-by: Leif Madsen
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8 years agores_rtp_asterisk: Eliminate "type-punned pointer" build warning.
David M. Lee [Mon, 10 Sep 2012 19:22:54 +0000 (19:22 +0000)]
res_rtp_asterisk: Eliminate "type-punned pointer" build  warning.

Removes "res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer
will break strict-aliasing rules" warning from the build on 32-bit platforms.

The problem is that 'size' was referenced aliased to both (pj_size_t *) and
(pj_ssize_t *). Now just make a copy of size that is the right type so there
isn't any pointer aliasing happening.

It also adds comments and asserts regarding what looks like an inappropriate
use of pj_sock_sendto, but is actually totally fine.

(closes issue ASTERISK-20368)
Reported by: Shaun Ruffell
Tested by: Michael L. Young
Patches:
  0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch uploaded by Shaun Ruffell (license 5417)
    slightly modified by David M. Lee.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoapp_meetme: Document that 'p' option will continue in dialplan.
Jonathan Rose [Mon, 10 Sep 2012 18:58:12 +0000 (18:58 +0000)]
app_meetme: Document that 'p' option will continue in dialplan.

(closes issue AST-991)
Reported by John Bigelow
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8 years agoMasquerade: Retain parkinglot settings made by CHANNEL function.
Jonathan Rose [Mon, 10 Sep 2012 17:41:57 +0000 (17:41 +0000)]
Masquerade: Retain parkinglot settings made by CHANNEL function.

Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.

(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
    masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOnly re-create an SRTP session when needed
Matthew Jordan [Sun, 9 Sep 2012 01:28:31 +0000 (01:28 +0000)]
Only re-create an SRTP session when needed

In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an
SDP offer and the ability to re-create an SRTP session when the crypto keys
changed.  In certain circumstances - most notably when a phone is put on
hold after having been bridged for a significant amount of time - the act
of re-creating the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session regardless
of whether or not the cryptographic keys changed.  Since this is technically
not necessary, this patch modifies the behavior to only re-create the SRTP
session if Asterisk detects that the remote key has changed.  This allows
models of phones that do not handle the SRTP session changing to continue
to work, while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys.

(issue ASTERISK-20194)
Reported by: Nicolo Mazzon
Tested by: Nicolo Mazzon

Review: https://reviewboard.asterisk.org/r/2099
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8 years agoAdd OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.
David M. Lee [Sat, 8 Sep 2012 06:18:48 +0000 (06:18 +0000)]
Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.

Without this flag, those files will compile with the system installed
OpenSSL headers (if they exist). This is a real bummer if a different
path was specified using --with-ssl=

(closes issue ASTERISK-20392)
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8 years agoFix MALLOC_DEBUG version of ast_strndup().
Richard Mudgett [Fri, 7 Sep 2012 23:10:05 +0000 (23:10 +0000)]
Fix MALLOC_DEBUG version of ast_strndup().

(closes issue ASTERISK-20349)
Reported by: Brent Eagles
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8 years agoRemove annoying unconditional debug message from INC/DEC functions.
Richard Mudgett [Fri, 7 Sep 2012 22:10:33 +0000 (22:10 +0000)]
Remove annoying unconditional debug message from INC/DEC functions.

(closes issue AST-1001)
Reported by: Guenther Kelleter
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8 years agoFix exception path typo in app_queue.c try_calling().
Richard Mudgett [Fri, 7 Sep 2012 21:51:31 +0000 (21:51 +0000)]
Fix exception path typo in app_queue.c try_calling().

(closes issue ASTERISK-20380)
Reported by: Jeremy Pepper
Patches:
      fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper
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