asterisk/asterisk.git
8 years agoARI: MOH start and stop for a channel
Jonathan Rose [Fri, 19 Jul 2013 19:40:27 +0000 (19:40 +0000)]
ARI: MOH start and stop for a channel

(issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2680/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394810 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI: Bridge Playback, Bridge Record
Jonathan Rose [Fri, 19 Jul 2013 19:35:21 +0000 (19:35 +0000)]
ARI: Bridge Playback, Bridge Record

Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.

(closes issue ASTERISK-21592)
Reported by: Matt Jordan

(closes issue ASTERISK-21593)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2670/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFilter channels used as internal mechanisms
Kinsey Moore [Fri, 19 Jul 2013 19:23:39 +0000 (19:23 +0000)]
Filter channels used as internal mechanisms

This adds new flags to the channel tech properties that flag it as
different types of implementation detail used exclusively to provide a
feature. Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are the
only two marked as such by this patch.

Review: https://reviewboard.asterisk.org/r/2633/
(closes issue ASTERISK-21873)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix crash when using temporary peers
Kinsey Moore [Fri, 19 Jul 2013 18:09:43 +0000 (18:09 +0000)]
Fix crash when using temporary peers

Temporary peers do not have an associated Stasis endpoint and quite a
bit of code in chan_sip assumes that all peers have a Stasis endpoint.
All endpoint accesses in chan_sip are now wrapped in an endpoint
NULL-check.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394795 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoConvert CCSS manager events to stasis.
Jason Parker [Fri, 19 Jul 2013 18:00:35 +0000 (18:00 +0000)]
Convert CCSS manager events to stasis.

(closes issue ASTERISK-21473)

Review: https://reviewboard.asterisk.org/r/2682/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMade audiohooks, framehooks, and monitor prevent local channel optimization.
Richard Mudgett [Fri, 19 Jul 2013 17:55:49 +0000 (17:55 +0000)]
Made audiohooks, framehooks, and monitor prevent local channel optimization.

Audiohooks, framehooks, and monitor represent state on a local channel
that will go away if it is optimized out.

(closes issue ASTERISK-21954)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2685/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixup doxygen on ast_hangup().
Richard Mudgett [Thu, 18 Jul 2013 22:01:42 +0000 (22:01 +0000)]
Fixup doxygen on ast_hangup().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd a bunch of options from sip.conf to res_sip.conf
Mark Michelson [Thu, 18 Jul 2013 19:25:51 +0000 (19:25 +0000)]
Add a bunch of options from sip.conf to res_sip.conf

For a complete list of the options added, see the review linked
at the bottom of this commit message.

(closes issue ASTERISK-21506)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2671

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed null dereference when WebSocket subprotocol isn't specified
David M. Lee [Thu, 18 Jul 2013 18:05:07 +0000 (18:05 +0000)]
Fixed null dereference when WebSocket subprotocol isn't specified

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394744 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agobridge_holding/app_bridgewait: Add new entertainment options
Jonathan Rose [Thu, 18 Jul 2013 16:49:44 +0000 (16:49 +0000)]
bridge_holding/app_bridgewait: Add new entertainment options

This patch adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)

(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI: Add support for suppressing media streams.
Jason Parker [Thu, 18 Jul 2013 16:03:12 +0000 (16:03 +0000)]
ARI: Add support for suppressing media streams.

Also convert res_mutestream to use the core feature behind this.

(closes issue ASTERISK-21618)

Review: https://reviewboard.asterisk.org/r/2652/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTweak debug statements
Matthew Jordan [Thu, 18 Jul 2013 14:50:56 +0000 (14:50 +0000)]
Tweak debug statements

This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols being
   compared after the string length calculation such that it shows the correct
   string length in the output
2. It adds some additional debug that displays when it matches on a
   sub-protocol and when it fails

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix caching topic shutdown assertions
David M. Lee [Thu, 18 Jul 2013 14:08:21 +0000 (14:08 +0000)]
Fix caching topic shutdown assertions

The recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing asserts
in the /stasis/core tests.

If the caching topic's subscription is the last reference to the
caching topic, it will destroy the caching topic after the final
message has been processed. When dispatching to a different thread,
this usually gave the unsubscribe enough time to finish before
destruction happened. Now, however, it consistently destroys before
unsubscription is complete.

This patch adds an extra reference to the caching topic, to hold it
for the duration of the unsubscription.

This patch also removes an extra unref that was happening when the
final message was received by the caching topic. It was put there
because of an extra ref that was put into the caching topic's
constructor. Both have been removed, which makes the destructor a bit
less confusing.

Review: https://reviewboard.asterisk.org/r/2675/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoProperly indicate failure to open an audio stream in res_agi
Michael L. Young [Thu, 18 Jul 2013 12:54:50 +0000 (12:54 +0000)]
Properly indicate failure to open an audio stream in res_agi

If there is an error streaming an audio file, the current return status makes it
difficult for an AGI script to determine that there was an error with the audio
file.

This patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS.  From looking at other parts of
res_agi, this would appear to be the proper way to handle an error.

(closes issue ASTERISK-21903)
Reported by: Ariel Wainer
Tested by: Ariel Wainer
Patches:
asterisk-21903-return-stream-res_1.8.diff
by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2625/
........

Merged revisions 394640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 394641 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394642 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange ast_hangup() to return void and be NULL safe.
Richard Mudgett [Wed, 17 Jul 2013 22:30:28 +0000 (22:30 +0000)]
Change ast_hangup() to return void and be NULL safe.

Since ast_hangup() is effectively a channel destructor, it should be a
void function.

* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.

* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove some completed and no longer relevant BUGBUG notes.
Richard Mudgett [Wed, 17 Jul 2013 19:27:26 +0000 (19:27 +0000)]
Remove some completed and no longer relevant BUGBUG notes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoapp_confbridge: Eliminate a reference leak for confbridge announcer channels
Jonathan Rose [Wed, 17 Jul 2013 18:26:19 +0000 (18:26 +0000)]
app_confbridge: Eliminate a reference leak for confbridge announcer channels

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoLeft over spacing issues of review 726.
Tzafrir Cohen [Wed, 17 Jul 2013 17:49:54 +0000 (17:49 +0000)]
Left over spacing issues of review 726.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agohandle DAHDI_EVENT_REMOVED on a pri D-Channel
Tzafrir Cohen [Wed, 17 Jul 2013 16:56:14 +0000 (16:56 +0000)]
handle DAHDI_EVENT_REMOVED on a pri D-Channel

When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.

This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.

This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).

It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.

Review: https://reviewboard.asterisk.org/r/726/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd 'kick all' capability to ConfBridge CLI command
Matthew Jordan [Tue, 16 Jul 2013 22:33:27 +0000 (22:33 +0000)]
Add 'kick all' capability to ConfBridge CLI command

This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".

Note that this patch was modified slightly to conform to trunk.

(closes issue ASTERISK-21827)
Reported by: dorianlogan
patches:
  kickall-patch_v2.diff uploaded by dorianlogan (License 6504)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRe-order handlers in CEL to ensure that HANGUP events happen after APP_END
Matthew Jordan [Tue, 16 Jul 2013 22:25:33 +0000 (22:25 +0000)]
Re-order handlers in CEL to ensure that HANGUP events happen after APP_END

When a channel is hungup, both an APP_END event and a HANGUP event can be
fired. To ensure that HANGUP events occur after APP_END events, the method
callbacks for the APP_END event should be processed prior to the callbacks
for the HANGUP event.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDebug logging to help with WebSocket connection problems
David M. Lee [Tue, 16 Jul 2013 21:44:12 +0000 (21:44 +0000)]
Debug logging to help with WebSocket connection problems

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394513 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agochan_gulp: Fix gulp_indicate() handling of AST_CONTROL_PVT_CAUSE_CODE.
Richard Mudgett [Tue, 16 Jul 2013 20:00:25 +0000 (20:00 +0000)]
chan_gulp: Fix gulp_indicate() handling of AST_CONTROL_PVT_CAUSE_CODE.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent crash from trying to end a session in an invalid way.
Mark Michelson [Tue, 16 Jul 2013 19:13:04 +0000 (19:13 +0000)]
Prevent crash from trying to end a session in an invalid way.

This ensures that code that was only meant to be run on a reinvite failure
only runs on a reinvite failure.

(closes issue ASTERISK-22061)
reported by Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394473 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove some dead code dealing with old bridging method.
Richard Mudgett [Tue, 16 Jul 2013 18:49:26 +0000 (18:49 +0000)]
Remove some dead code dealing with old bridging method.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSimplify bridge_simple chan join code.
Richard Mudgett [Tue, 16 Jul 2013 18:48:49 +0000 (18:48 +0000)]
Simplify bridge_simple chan join code.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRe-order cleanup
Matthew Jordan [Tue, 16 Jul 2013 18:22:07 +0000 (18:22 +0000)]
Re-order cleanup

This patch attempts to fix some possible race conditions in shutdown of the
CDR engine. It:
* Adds a cleanup handler to only unsubscribe and join on stasis messages during
  graceful shutdown. The cleanup handler should execute before the regular atexit
  handler, as we want to unsubscribe for any further messages before dispatching
  the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed null dereference when WebSocket protocol is omitted
David M. Lee [Tue, 16 Jul 2013 15:30:09 +0000 (15:30 +0000)]
Fixed null dereference when WebSocket protocol is omitted

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394442 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoReplace chan_agent with app_agent_pool.
Richard Mudgett [Mon, 15 Jul 2013 23:20:55 +0000 (23:20 +0000)]
Replace chan_agent with app_agent_pool.

The ill conceived chan_agent is no more.  It is now replaced by
app_agent_pool.

Agents login using the AgentLogin() application as before.  The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan.  (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)

Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()

Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001

Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
   basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
   the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.

To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support.  The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback.  The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.

(closes issue ASTERISK-21554)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2657/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove misleading documentation for channel snapshot creation.
Mark Michelson [Mon, 15 Jul 2013 22:05:57 +0000 (22:05 +0000)]
Remove misleading documentation for channel snapshot creation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394402 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDocument the ari.conf allowed_origins setting
David M. Lee [Mon, 15 Jul 2013 21:22:12 +0000 (21:22 +0000)]
Document the ari.conf allowed_origins setting

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove some callbacks and functions which are not needed.
Joshua Colp [Mon, 15 Jul 2013 13:43:37 +0000 (13:43 +0000)]
Remove some callbacks and functions which are not needed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoProvide error message for QUEUE_MEMBER when member is not in queue
Matthew Jordan [Sun, 14 Jul 2013 02:41:43 +0000 (02:41 +0000)]
Provide error message for QUEUE_MEMBER when member is not in queue

When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.

(closes issue ASTERISK-21980)
Reported by: Avraam David
........

Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394346 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove redundant code in dns.c
Matthew Jordan [Sun, 14 Jul 2013 02:05:04 +0000 (02:05 +0000)]
Remove redundant code in dns.c

Peter J Philipp pointed out that there are two checks that ensure that len is
not less than 0. If len is less than 0, the function returns. Having both of
them is clearly redundant.

This removes the second and attempts to clarify (slightly) the error condition.

(closes issue ASTERISK-21772)
Reported by: Peter J Philipp

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394305 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoClarify documentation for function PASSTHRU
Matthew Jordan [Sun, 14 Jul 2013 02:00:02 +0000 (02:00 +0000)]
Clarify documentation for function PASSTHRU

It is not apparent to the average user that the PASSTHRU function should not
be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions
which take a variable name and not its contents.

This patch clarifies the behavior in the documentation and provides an example.

(closes issue ASTERISK-21717)
Reported by: Richard Miller
patches:
  func_strings.diff uploaded by Richard Miller (license 5685)
........

Merged revisions 394302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 394303 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix FRACK message from external redirects; handle outbound channels better
Matthew Jordan [Sat, 13 Jul 2013 23:28:23 +0000 (23:28 +0000)]
Fix FRACK message from external redirects; handle outbound channels better

This patch does the following:
 * It simplifies the Dial handling in CDRs. As a rule, the caller in a dial
   relationship is always the Party A. There was some logic present in the
   handling of the dial message that could, conceivably, pick the caller
   as Party A for the beginning of the dial and the peer as Party A for the
   end of the dial. This shouldn't have happened if the code in the bridging
   framework was doing its job; however, that was broken and it led to the
   FRACK. As it is, this code was overly ocmplex and not needed: the caller,
   if present, should always be Party A. Period.
 * It properly checks to see if a channel will continue on in the dialplan.
   ast_check_hangup - much like cake at the end - is a lie. It will tell
   you that you are hungup when you are not. Do not believe it.

   I would make this function tell the truth, but I'm nervous that we've been
   depending on it sitting on its throne of lies for far too long, and it would
   probably break lots of things. So I'm just checking the "internal" soft
   hangup flags, like everyone else.

(closes issue ASTERISK-22060)
Reported by: Mark Michelson

(issue ASTERISK-21831)
Reported by: Matt Jordan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394290 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPretty up a debug message if the referred-by-uri isn't available
Matthew Jordan [Sat, 13 Jul 2013 22:14:04 +0000 (22:14 +0000)]
Pretty up a debug message if the referred-by-uri isn't available

Instead of formatting a NULL pointer into a "%s" format string (which is
usually not a good thing to do), we instead print "Unknown".

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix a longstanding issue with MFC-R2 configuration that prevented users
Moises Silva [Fri, 12 Jul 2013 22:35:50 +0000 (22:35 +0000)]
Fix a longstanding issue with MFC-R2 configuration that prevented users
from mixing different variants or general MFC-R2 settings within the same E1 line.

Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or
the whole E1 has the same country variant and R2 settings.

In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1.

This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every
time a new channel => section is found and the configuration was changed.

(closes issue ASTERISK-21117)
Reported by: Rafael Angulo
Related Elastix issue: http://bugs.elastix.org/view.php?id=1612
........

Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 394173 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd support to the bridging core for performing COLP updates when channels join a...
Joshua Colp [Fri, 12 Jul 2013 21:42:53 +0000 (21:42 +0000)]
Add support to the bridging core for performing COLP updates when channels join a 2 party bridge.

(closes issue ASTERISK-21829)

Review: https://reviewboard.asterisk.org/r/2636/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent potential race condition in multiparty basic bridges.
Mark Michelson [Fri, 12 Jul 2013 21:01:51 +0000 (21:01 +0000)]
Prevent potential race condition in multiparty basic bridges.

For more details about the race condition see the linked review
at the bottom of this commit

(closes issue ASTERISK-21882)
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2663

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix a compiler warning.
Jason Parker [Fri, 12 Jul 2013 19:35:08 +0000 (19:35 +0000)]
Fix a compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394216 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed intermittent crash when loading test_json.so
David M. Lee [Fri, 12 Jul 2013 18:23:39 +0000 (18:23 +0000)]
Fixed intermittent crash when loading test_json.so

The JSON test attempted an overly clever use of RAII_VAR to run code
at the beginning and end of each test, in order to validate that no
JSON objects were leaked during the test.

The problem is that the validation code would run during the initial
load, when the tests were initialized. This happens during startup,
when other parts of the system might actively be allocating and
freeing JSON objects.

This patch changes the RAII_VAR to use the new
ast_test_register_{init,cleanup} functions to run the validations
properly.

(closes issue ASTERISK-21978)
Review: https://reviewboard.asterisk.org/r/2669/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI: Add support for Cross-Origin Resource Sharing (CORS), origin headers
Jason Parker [Fri, 12 Jul 2013 17:52:52 +0000 (17:52 +0000)]
ARI: Add support for Cross-Origin Resource Sharing (CORS), origin headers

This rejects requests from any unknown origins.

(closes issue ASTERISK-21278)

Review: https://reviewboard.asterisk.org/r/2667/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394189 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix bridge tech write callback parameter name.
Richard Mudgett [Thu, 11 Jul 2013 21:01:09 +0000 (21:01 +0000)]
Fix bridge tech write callback parameter name.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed chan_skinny for systems were pthread_t isn't an int.
David M. Lee [Thu, 11 Jul 2013 20:59:10 +0000 (20:59 +0000)]
Fixed chan_skinny for systems were pthread_t isn't an int.

I'm looking at you, OS X.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394156 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRefactor and cleanup of skinny session handling.
Damien Wedhorn [Thu, 11 Jul 2013 20:17:53 +0000 (20:17 +0000)]
Refactor and cleanup of skinny session handling.

Major changes are to pull all packet reading functions into skinny_session
and move timeout handling to scheduling arrangements. Thread cancelling is
now undertaken directly rather than waiting for the read to timeout
(cleanup is popped on thread cancel). Also added some keepalive timings in
debugging messages.

Keepalive timeout has been increased from 1.1 by keepalive to 3 times
keepalive. This seems to align (after keepalives stabilise) with when
devices reset after not receiving keepalives. Probably needs more work,
especially around the first and/or second keepalives that vary
significantly by device and firmware version.

Review: https://reviewboard.asterisk.org/r/2611/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394147 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTweak the subscription failure warning message to include endpoint name and context.
Joshua Colp [Thu, 11 Jul 2013 16:23:41 +0000 (16:23 +0000)]
Tweak the subscription failure warning message to include endpoint name and context.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394103 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrect test_cel cleanup.
David M. Lee [Thu, 11 Jul 2013 15:37:51 +0000 (15:37 +0000)]
Correct test_cel cleanup.

When I corrected the CEL test crash in r394037, I didn't quite pay attention
to how the globals and locals were being shuffled around in the cleanup
callback. I removed the nulling of the global variables, which caused them
to be double cleaned.

This patch puts the global nulling code back (since the vars are cleaned up
by RAII_VARs), and removes the explicit ao2_cleanup() (since they were no-ops,
because the variables had just been nulled).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange ARI user config to use a type field
David M. Lee [Thu, 11 Jul 2013 14:39:55 +0000 (14:39 +0000)]
Change ARI user config to use a type field

When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e., [user-admin]).

This is neither idiomatic Asterisk configuration, nor is it really
that user friendly. This patch replaces the category prefix with a
type field in the section, which is much cleaner.

Review: https://reviewboard.asterisk.org/r/2664/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoApply defaults to ari.conf's general section
David M. Lee [Thu, 11 Jul 2013 13:56:26 +0000 (13:56 +0000)]
Apply defaults to ari.conf's general section

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agotest_voicemail_api: fix warning found by gcc-4.8
David M. Lee [Thu, 11 Jul 2013 04:34:49 +0000 (04:34 +0000)]
test_voicemail_api: fix warning found by gcc-4.8

The voicemail_api test had code like strncmp(a, b, sizeof(a)), but a was a
char pointer, instead of a literal or char array. This meant that sizeof was
the size of the pointer, not the length of the string.

Since the string is in a stringfield and should be null terminated, I just
changed it to a plain strcmp.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394050 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed some CEL test crashes
David M. Lee [Thu, 11 Jul 2013 02:02:48 +0000 (02:02 +0000)]
Fixed some CEL test crashes

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPSJIP - sip.conf to res_sip.conf script
Kevin Harwell [Wed, 10 Jul 2013 22:26:13 +0000 (22:26 +0000)]
PSJIP - sip.conf to res_sip.conf script

** This script is in no way finished.

Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit".  This script
and supporting files should be executable against python version 2.5.

An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.

Currently the scripts supports the majority of endpoint options found in
res_sip.conf.  Support has also been added for Aor(s) and the ACL/security
sections.  Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.

Also items that still need to be done: templates, includes, parsing '=>'
delimiter.  Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them).  However, the
parsing of and adding of the section needs to be done.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoHandle outbound registration failures that do not occur as a result of a real response.
Joshua Colp [Wed, 10 Jul 2013 20:02:59 +0000 (20:02 +0000)]
Handle outbound registration failures that do not occur as a result of a real response.

(closes issue ASTERISK-22064)
Reported by: Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDocument the 400 error response for originate
David M. Lee [Wed, 10 Jul 2013 17:13:21 +0000 (17:13 +0000)]
Document the 400 error response for originate

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrected api-docs for channel variables
David M. Lee [Wed, 10 Jul 2013 13:50:48 +0000 (13:50 +0000)]
Corrected api-docs for channel variables

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoastobj2-ify the SLA code
Russell Bryant [Wed, 10 Jul 2013 01:56:15 +0000 (01:56 +0000)]
astobj2-ify the SLA code

The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk.  Worse, support for reloads did not exist at first
and was added later as a bolt-on feature.  I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle.  Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.

This patch converts various SLA objects to be reference counted objects
using astobj2.  This allows reloads to be processed while the system is
in use.  The code ensures that the objects will not disappear while one
of the other threads is using them.  However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.

Review: https://reviewboard.asterisk.org/r/2581/
........

Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake SCOPED_LOCK use RAII_VAR.
Jason Parker [Tue, 9 Jul 2013 21:40:38 +0000 (21:40 +0000)]
Make SCOPED_LOCK use RAII_VAR.

This fixes an issue with requiring SCOPED_LOCK to be the last variable
declaration and removes duplicate code in the process.

Review: https://reviewboard.asterisk.org/r/2665/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix printf NULL string (null) substituion for NULL config framework default.
Richard Mudgett [Tue, 9 Jul 2013 21:06:21 +0000 (21:06 +0000)]
Fix printf NULL string (null) substituion for NULL config framework default.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUse correct function for getting bridged peer when doing direct media checks.
Mark Michelson [Tue, 9 Jul 2013 20:07:21 +0000 (20:07 +0000)]
Use correct function for getting bridged peer when doing direct media checks.

(closes issue ASTERISK-21947)
reported by Matt Jordan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393897 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix some stasis doxygen comments.
Richard Mudgett [Tue, 9 Jul 2013 19:38:00 +0000 (19:38 +0000)]
Fix some stasis doxygen comments.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure all pjsip_regc_* access occurs within a pjlib thread.
Joshua Colp [Tue, 9 Jul 2013 11:05:48 +0000 (11:05 +0000)]
Ensure all pjsip_regc_* access occurs within a pjlib thread.

(closes issue ASTERISK-22054)
Reported by: Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTweak log message slightly.
Joshua Colp [Mon, 8 Jul 2013 21:27:18 +0000 (21:27 +0000)]
Tweak log message slightly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTreat the authentication object as invalid if digest configuration is chosen and...
Joshua Colp [Mon, 8 Jul 2013 21:26:37 +0000 (21:26 +0000)]
Treat the authentication object as invalid if digest configuration is chosen and the digest is not of the correct length.

(closes issue ASTERISK-22003)
Reported by: Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOh menuconfig, why do you hate margins?
David M. Lee [Mon, 8 Jul 2013 20:31:41 +0000 (20:31 +0000)]
Oh menuconfig, why do you hate margins?

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393843 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBetter structure for the WebSocket validation failure message
David M. Lee [Mon, 8 Jul 2013 20:07:58 +0000 (20:07 +0000)]
Better structure for the WebSocket validation failure message

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoEnsure that a valid bind host is specified for transports.
Joshua Colp [Mon, 8 Jul 2013 19:53:13 +0000 (19:53 +0000)]
Ensure that a valid bind host is specified for transports.

(closes issue ASTERISK-22017)
Reported by: Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRefactor operations to access the stasis cache instead of objects directly when retri...
Joshua Colp [Mon, 8 Jul 2013 19:19:55 +0000 (19:19 +0000)]
Refactor operations to access the stasis cache instead of objects directly when retrieving information.

(closes issue ASTERISK-21883)

Review: https://reviewboard.asterisk.org/r/2645/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_stasis_http doesn't depend on res_stasis any more
David M. Lee [Mon, 8 Jul 2013 16:04:01 +0000 (16:04 +0000)]
res_stasis_http doesn't depend on res_stasis any more

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_parking: Apply ringing role option on swap with a channel that rings
Jonathan Rose [Mon, 8 Jul 2013 15:59:47 +0000 (15:59 +0000)]
res_parking: Apply ringing role option on swap with a channel that rings

(closes issue ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2656/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393815 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix building.
Joshua Colp [Mon, 8 Jul 2013 15:11:07 +0000 (15:11 +0000)]
Fix building.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI: Add support for getting/setting channel and global variables.
Jason Parker [Mon, 8 Jul 2013 14:46:20 +0000 (14:46 +0000)]
ARI: Add support for getting/setting channel and global variables.

This allows for reading and writing of functions on channels.

(closes issue ASTERISK-21868)

Review: https://reviewboard.asterisk.org/r/2641/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393806 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMove channel driver Registry manager events to core.
Jason Parker [Mon, 8 Jul 2013 14:42:57 +0000 (14:42 +0000)]
Move channel driver Registry manager events to core.

This also shuffles the stasis system topic and related handling.

(closes issue ASTERISK-21488)

Review: https://reviewboard.asterisk.org/r/2631/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCreate Local channel messages on the Stasis message bus and produce AMI events
Matthew Jordan [Mon, 8 Jul 2013 14:26:40 +0000 (14:26 +0000)]
Create Local channel messages on the Stasis message bus and produce AMI events

This patch does the following:

* It adds a virtual table of callbacks to core_unreal. These callbacks can be
  supplied by concrete implementations of "unreal" channel drivers, which lets
  the unreal channel driver call specific functionality when it performs some
  action. Currently, this is done to notify implementations when an
  optimization operation has begun, and when an optimization operation has
  succeeded.

* It adds Stasis-Core messages for Local channel bridging and Local channel
  optimization. Local channel optimization is now two events: a Begin and an
  End. Some consumers of Stasis-Core may want to know when an operation is
  beginning so that they can 'prepare' their information; others will be more
  concerned about when the operation has completed, so that they can 'fix up'
  information. Stasis-Core allows for both, as does AMI.

Review: https://reviewboard.asterisk.org/r/2552

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix some broken logic in sending outbound caller ID.
Mark Michelson [Mon, 8 Jul 2013 13:57:28 +0000 (13:57 +0000)]
Fix some broken logic in sending outbound caller ID.

* trust_id_outbound was required even when the caller ID was not marked
private. This is against intentions and documentation.
* We now check both name and number privacy instead of checking name privacy
twice.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoIn a channel destructor dispose of items that raise Stasis message properly
Matthew Jordan [Sun, 7 Jul 2013 21:29:40 +0000 (21:29 +0000)]
In a channel destructor dispose of items that raise Stasis message properly

This patch reorders certain actions that may raise Stasis messages in the
channel destructor such that they occur before the Stasis cache is cleared.
Once the Stasis cache is cleared, its rather a bad idea to be trying to
publish information about a channel.

(closes issue ASTERISK-22001)
Reported by: Jonathan Rose

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoHandle hangup logic in the Stasis message bus and consumers of Stasis messages
Matthew Jordan [Sun, 7 Jul 2013 20:34:38 +0000 (20:34 +0000)]
Handle hangup logic in the Stasis message bus and consumers of Stasis messages

This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
  channel is executing dialplan hangup logic, i.e., the 'h' extension or a
  hangup handler. Stasis messages now also convey the soft hangup flag so
  consumers of the messages can know when a channel is executing said
  hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
  well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
  and other consumers of Stasis have been updated to look for this flag to
  know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
  leaving a bridge. Previously, a new CDR was automatically created when a
  channel left a bridge and put into the 'Pending' state; however, this
  way of handling CDRs made it difficult for the 'endbeforehexten' logic to
  work correctly - there was always a new CDR waiting in the hangup logic
  and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
  hangup routine. This patch completely removes the Pending state and instead
  defers creation of the new CDR until it gets a new message that requires
  a new CDR.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI: return a 503 if Asterisk isn't fully booted
David M. Lee [Fri, 5 Jul 2013 22:08:27 +0000 (22:08 +0000)]
ARI: return a 503 if Asterisk isn't fully booted

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrint error details when set nonblock fails
David M. Lee [Fri, 5 Jul 2013 19:56:50 +0000 (19:56 +0000)]
Print error details when set nonblock fails

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393757 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoDocument MissingParams error message for /ari/events
David M. Lee [Fri, 5 Jul 2013 19:15:27 +0000 (19:15 +0000)]
Document MissingParams error message for /ari/events

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRefactor RTCP events over to Stasis; associate with channels
Matthew Jordan [Fri, 5 Jul 2013 17:33:33 +0000 (17:33 +0000)]
Refactor RTCP events over to Stasis; associate with channels

This patch does the following:

* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
  information in the RTCP events. Because Stasis provides a cache, Jaco's
  patch was modified to pass the channel uniqueid to the RTP layer as
  opposed to a pointer to the channel. This has the following benefits:
  (1) It keeps the RTP engine 'clean' of references back to channels
  (2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
  Potentially, other implementations (such as res_rtp_multicast) could also
  raise RTCP information. The engine provides structs to represent RTCP headers
  and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
  RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
  but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
  assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
  raise an event when we sent a RR report.

Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.

Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.

Review: https://reviewboard.asterisk.org/r/2603/

(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
  asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)

(closes issue ASTERISK-21471)
Reported by: Matt Jordan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOneTouchRecord: Add function defined earlier: ast_bridge_features_do()
Richard Mudgett [Fri, 5 Jul 2013 14:54:27 +0000 (14:54 +0000)]
OneTouchRecord: Add function defined earlier: ast_bridge_features_do()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRemove parkinglot from the channel snapshot
Matthew Jordan [Fri, 5 Jul 2013 03:08:58 +0000 (03:08 +0000)]
Remove parkinglot from the channel snapshot

Legacy channel drivers often include the ability to set a default parking lot
on an endpoint basis; when channels are created for that endpoint, they inherit
the parkinglot option. Parking used to use this option more frequently; while
it is still supported, other options (such as using channel variables or
creation of a custom parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly useful - it
is rarely (if ever) changed on a channel and some consumers of channel
snapshots, such as ARI, will never use the information.

(closes issue ASTERISK-21968)
Reported by: Matt Jordan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393716 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_parking: Replace Parker snapshots with ParkerDialString
Jonathan Rose [Thu, 4 Jul 2013 18:46:56 +0000 (18:46 +0000)]
res_parking: Replace Parker snapshots with ParkerDialString

This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.

(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix int width problem for 32-bit... again
David M. Lee [Thu, 4 Jul 2013 13:37:57 +0000 (13:37 +0000)]
Fix int width problem for 32-bit... again

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393687 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix int width problem for 32-bit
David M. Lee [Thu, 4 Jul 2013 13:18:45 +0000 (13:18 +0000)]
Fix int width problem for 32-bit

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix utils directory breakage.
David M. Lee [Thu, 4 Jul 2013 13:06:15 +0000 (13:06 +0000)]
Fix utils directory breakage.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393675 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd BUGBUG note for ASTERISK-22009
Richard Mudgett [Wed, 3 Jul 2013 23:59:17 +0000 (23:59 +0000)]
Add BUGBUG note for ASTERISK-22009

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRevert accidental overcommit.
Richard Mudgett [Wed, 3 Jul 2013 23:57:37 +0000 (23:57 +0000)]
Revert accidental overcommit.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd BUGBUG note for ASTERISK-22009
Richard Mudgett [Wed, 3 Jul 2013 23:55:53 +0000 (23:55 +0000)]
Add BUGBUG note for ASTERISK-22009

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agochan_dahdi: Fix segfault reloading chan_dahdi when round robin is used.
Richard Mudgett [Wed, 3 Jul 2013 23:34:19 +0000 (23:34 +0000)]
chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used.

* Clear round_robin[] in dahdi_restart().

(closes issue ASTERISK-21847)
Reported by: Ivo Andonov
Patches:
      jira_asterisk_21847_v1.8.patch (license #5621) patch uploaded by rmudgett
........

Merged revisions 393627 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 393628 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOneTouchRecord: Make so Monitor/MixMonitor can be toggled/started/stopped.
Richard Mudgett [Wed, 3 Jul 2013 22:36:38 +0000 (22:36 +0000)]
OneTouchRecord: Make so Monitor/MixMonitor can be toggled/started/stopped.

The OneTouchRecord feature has historically been a toggle.  This patch
adds the ability to make the OneTouchRecord hook optionally start/stop
recording only.  If OneTouchRecord is already doing what is requested then
only the invoker hears the courtesy tone and/or start/stop recording
message.

The new feature is written so we could easily add explicit start/stop
recording DTMF hooks for Monitor and MixMonitor.

The majority of the changes in bridge_builtin_features.c is a refactoring
of the OneTouchRecord code (Monitor and MixMonitor versions) so it is easy
to direct the toggle/start/stop functionality.

Review: https://reviewboard.asterisk.org/r/2655/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393612 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMove when bridge channel enter is published so it does not interrupt the thought...
Richard Mudgett [Wed, 3 Jul 2013 22:07:25 +0000 (22:07 +0000)]
Move when bridge channel enter is published so it does not interrupt the thought of some lines of code.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393601 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix some indentation in stasis_config.c.
Richard Mudgett [Wed, 3 Jul 2013 22:05:54 +0000 (22:05 +0000)]
Fix some indentation in stasis_config.c.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix some bugs in CDRs; add some CLI commands to help debugging
Matthew Jordan [Wed, 3 Jul 2013 22:04:08 +0000 (22:04 +0000)]
Fix some bugs in CDRs; add some CLI commands to help debugging

This patch fixes a few minor bugs and one major one: the CDR by bridge
container was less than helpful. The mechanism previously used to try
and find all of the CDRs in a particular bridge ended up missing CDRs,
resulting in incorrect records.

When looking up CDRs in a bridge, we now just bite the bullet and do
a selection across all existing CDRs.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393599 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoLet Stasis load itself with default values
Matthew Jordan [Wed, 3 Jul 2013 21:14:03 +0000 (21:14 +0000)]
Let Stasis load itself with default values

While a Stasis configuration file is nice, it shouldn't be mandatory.
We can carry on with default values.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPublish a bridge enter before pulling on a push-and-swap operation.
Mark Michelson [Wed, 3 Jul 2013 20:41:00 +0000 (20:41 +0000)]
Publish a bridge enter before pulling on a push-and-swap operation.

Prior to this patch, the order of procedures on a bridge push was

* Add new bridge channel to bridge's array.
* Pull the swap channel out of the bridge
* Publish a bridge enter event.

The problem is that when the swap channel was pulled from the bridge,
a bridge leave event would be published. The bridge snapshot
published during the bridge leave showed the new channel that had
been added to the bridge, but there had been no bridge enter event
for that channel.

The fix provided here was to change the order a bit

* Add new bridge channel to bridge's array.
* Publish bridge enter event.
* Pull the swap channel out of the bridge.

This makes it so that the bridge snapshots during the stasis
events are accurate.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix load errors related to the new ari_model_validators.
David M. Lee [Wed, 3 Jul 2013 19:46:50 +0000 (19:46 +0000)]
Fix load errors related to the new ari_model_validators.

The Asterisk strategy of loading modules with RTLD_LAZY to extract metadata
from the module works well enough, until you try to take the address of a
function.

If a module takes the address of a function, that function needs to be
resolved at load time. That kinda defeats RTLD_LAZY.

This patch adds some ari_validator_{id}_fn() wrapper functions for safely
getting the function pointer from a different module.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoViolating the margins to make menuconfig happy
David M. Lee [Wed, 3 Jul 2013 18:24:45 +0000 (18:24 +0000)]
Violating the margins to make menuconfig happy

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393561 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoARI - channel recording support
David M. Lee [Wed, 3 Jul 2013 17:58:45 +0000 (17:58 +0000)]
ARI - channel recording support

This patch is the first step in adding recording support to the
Asterisk REST Interface.

Recordings are stored in /var/spool/recording. Since recordings may be
destructive (overwriting existing files), the API rejects attempts to
escape the recording directory (avoiding issues if someone attempts to
record to ../../lib/sounds/greeting, for example).

(closes issue ASTERISK-21594)
(closes issue ASTERISK-21581)
Review: https://reviewboard.asterisk.org/r/2612/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3