asterisk/asterisk.git
7 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

7 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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7 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

7 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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7 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

7 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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7 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

7 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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7 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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7 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

7 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

7 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

7 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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7 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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7 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

7 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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7 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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7 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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7 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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7 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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7 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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7 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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7 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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7 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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7 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

7 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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7 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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7 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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7 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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7 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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7 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

7 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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7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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

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7 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
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7 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
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7 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/
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7 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

7 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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7 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

7 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

7 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

7 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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7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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
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Merged revisions 373132 from http://svn.asterisk.org/svn/asterisk/branches/10
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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

7 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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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.
........

Merged revisions 373090 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373101 from http://svn.asterisk.org/svn/asterisk/branches/10
........

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

7 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
........

Merged revisions 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373062 from http://svn.asterisk.org/svn/asterisk/branches/10
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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

7 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

7 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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Merged revisions 373024 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 373025 from http://svn.asterisk.org/svn/asterisk/branches/10
........

Merged revisions 373029 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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

7 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.
........

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

7 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/
........

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

7 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

7 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.
........

Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372937 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372930 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372931 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372917 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372918 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372885 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372888 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372841 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372886 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372887 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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

7 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

7 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

7 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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Merged revisions 372804 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372805 from http://svn.asterisk.org/svn/asterisk/branches/10
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372795 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372796 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372777 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372767 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372768 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372712 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Recorded merge of revisions 372695 from http://svn.asterisk.org/svn/asterisk/branches/10
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Recorded merge of revisions 372696 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372656 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372657 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372629 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372630 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 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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Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372625 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372626 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372627 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix VoicemailUserEntry event headers ServerEmail and MailCommand reported values.
Richard Mudgett [Fri, 7 Sep 2012 21:30:17 +0000 (21:30 +0000)]
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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Merged revisions 372620 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372621 from http://svn.asterisk.org/svn/asterisk/branches/10
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agosvn:ignore cleanup.
David M. Lee [Fri, 7 Sep 2012 21:04:48 +0000 (21:04 +0000)]
svn:ignore cleanup.

* pjproject bin and lib directories should pretty much ignore everything
* Ignore *.o in codecs/ilbc
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7 years agoFix parallel make for res_asterisk_rtp.
David M. Lee [Fri, 7 Sep 2012 20:53:48 +0000 (20:53 +0000)]
Fix parallel make for res_asterisk_rtp.

Fixes a build regression introduced in r369517 "Add support for ICE/STUN/TURN
in res_rtp_asterisk and chan_sip." [1].

[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=369517

When compiling asterisk in parallel like:
    $ make -j 10

It's possible to get errors like the following:

    .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing separator.  Stop.
    make[4]: *** [depend] Error 2
    make[3]: *** [dep] Error 1
    make[2]: *** [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2
    make[3]: warning: jobserver unavailable: using -j1.  Add `+' to parent make rule.

This is because the build system is trying to build each of the libraries in
pjproject in parallel. Now the build will build pjproject in a single job and
link the results into res_asterisk_rtp.

Parallel builds, on one test system, saves ~1.5 minutes from a default Asterisk
build:

Single job:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make >/dev/null 2>&1 )

    real    2m34.529s
    user    1m41.810s
    sys     0m15.970s

Parallel make:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 )

    real    1m2.353s
    user    2m39.120s
    sys     0m18.850s

(closes issue ASTERISK-20362)
Reported by: Shaun Ruffel
Patches:
    0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch uploaded by Shaun Ruffel (License #5417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFree ast_str objects when temp file fails to be created in MiniVM
Matthew Jordan [Fri, 7 Sep 2012 02:27:42 +0000 (02:27 +0000)]
Free ast_str objects when temp file fails to be created in MiniVM

The previous commit (r372554) was from a patch that was written before
r366880, which ensured that ast_str objects allocated in the sendmail
routine were free'd in off nominal paths.  This commit frees the
string objects in the off nominal path introduced in r372554.

(issue ASTERISK-17133)
Reported by: Tzafrir Cohen
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7 years agoFix file descriptor leak and pointer scope issue in MiniVM when sending mail
Matthew Jordan [Fri, 7 Sep 2012 02:16:54 +0000 (02:16 +0000)]
Fix file descriptor leak and pointer scope issue in MiniVM when sending mail

When MiniVM sends an e-mail and it has the volgain option set, it will spawn
sox in a separate process to handle the manipulation of the sound file.  In
doing so, it creates a temporary file.  There are two problems here:
  1) The file descriptor returned from mkstemp is leaked
  2) The finalfilename character pointer points to a buffer that loses scope
     once volgain processing is finished.

Note that in r316265, Russell fixed some gcc warnings by using the return
value of the mkstemp call.  A warning was placed in minivm that the file
descriptor was going to be leaked.  This patch reverts that change, as it
handles the leak and 'uses' the file descriptor returned from mkstemp.

(closes issue ASTERISK-17133)
Reported by: Tzafrir Cohen
patches:
  minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035)
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7 years agoUpdate QueueMemberStatus event documentation to include member status values
Matthew Jordan [Thu, 6 Sep 2012 22:21:12 +0000 (22:21 +0000)]
Update QueueMemberStatus event documentation to include member status values

The Status: header in a QueueMemberStatus event (and other QueueMember* events)
is the numeric value of the device state corresponding to that Queue Member.
As those values are not exactly obvious, listing them in the documentation is
useful.

Matt Riddell reported this indirectly through the wiki page.

(closes issue ASTERISK-20243)
Reported by: Matt Riddell
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7 years agoFix loss of MOH on an ISDN channel when parking a call for the second time.
Richard Mudgett [Thu, 6 Sep 2012 22:14:52 +0000 (22:14 +0000)]
Fix loss of MOH on an ISDN channel when parking a call for the second time.

Using the AMI redirect action to take an ISDN call out of a parking lot
causes the MOH state to get confused.  The redirect action does not take
the call off of hold.  When the call is subsequently parked again, the
call no longer hears MOH.

* Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames
if it is already in a state where it is supposed to be sending MOH.  The
MOH may have been stopped by other means.  (Such as killing the generator.)

This simple fix is done rather than making the AMI redirect action post an
AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus
potentially breaking something with an unexpected AST_CONTROL_UNHOLD.

(closes issue ABE-2873)
Patches:
      jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by rmudgett
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7 years agoEnsure listed queues are not offered for completion
Kinsey Moore [Thu, 6 Sep 2012 21:43:18 +0000 (21:43 +0000)]
Ensure listed queues are not offered for completion

When using tab-completion for the list of queues on "queue reset stats"
or "queue reload {all|members|parameters|rules}", the tab-completion
listing for further queues erroneously listed queues that had already
been added to the list. The tab-completion listing now only displays
queues that are not already in the list.

(closes issue AST-963)
Reported-by: John Bigelow
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7 years agochan_sip: Note change in behavior to how directmediapermit/deny ACL works
Jonathan Rose [Thu, 6 Sep 2012 15:57:51 +0000 (15:57 +0000)]
chan_sip: Note change in behavior to how directmediapermit/deny ACL works

r366547 introduced a change to the directmedia ACL for chan_sip which
modified the behavior significantly. Prior to the patch, this option would
bridge peers with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the bridged peer's
ACL instead. This change has been present since 1.8.14.0. That patched failed
to document the change in Upgrade.txt, so this patch adds mention of that
change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches)

(issue AST-876)
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7 years agoEnsure "rules" is tab-completable for "queue show"
Kinsey Moore [Thu, 6 Sep 2012 14:31:44 +0000 (14:31 +0000)]
Ensure "rules" is tab-completable for "queue show"

Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.

(closes issue AST-958)
Reported-by: John Bigelow
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7 years agoFix DUNDi message routing bug when neighboring peer is unreachable
Matthew Jordan [Thu, 6 Sep 2012 02:52:37 +0000 (02:52 +0000)]
Fix DUNDi message routing bug when neighboring peer is unreachable

Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors,
PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors.  If the connection
is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in
the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send
messages to PBX3.  If it does, PBX2 will assume that PBX3 already received the
message and fail to forward the message on to PBX3 itself.  This patch fixes
this by only including peers in a DPDISCOVER message that are reachable by the
sending node.  This includes all peers with an empty address
(00:00:00:00:00:00) and that are have been reached by a qualify message.

This patch also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers.

The patch uploaded by Peter was modified slightly for this commit.

(closes issue ASTERISK-19309)
Reported by: Peter Racz
patches:
  dundi_routing.patch uploaded by Peter Racz (license 6290)

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7 years agoAllow configured numbers for FollowMe to be greater than 90 characters
Matthew Jordan [Thu, 6 Sep 2012 01:02:17 +0000 (01:02 +0000)]
Allow configured numbers for FollowMe to be greater than 90 characters

When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters.  This can artificially limit some parallel dial scenarios.  This
patch allows for numbers of any length to be defined in the configuration
file.

Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue.  The patch originally expanded the buffer to 256
characters.  Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.

(closes issue ASTERISK-16879)
Reported by: Clod Patry
Tested by: mjordan
patches:
  followme_no_limit.diff uploaded by Clod Patry (license #5138)

Slightly modified for this commit.
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7 years agoRecorded merge of revisions 372373 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Wed, 5 Sep 2012 19:44:32 +0000 (19:44 +0000)]
Recorded merge of revisions 372373 from svn.asterisk.org/svn/asterisk/branches/11

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Fix compile error.
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