asterisk/asterisk.git
6 years agochan_sip: Revert r398835 due to failing tests involving originate
Jonathan Rose [Thu, 12 Sep 2013 20:27:56 +0000 (20:27 +0000)]
chan_sip: Revert r398835 due to failing tests involving originate

(issue ASTERISK-22424)
Reported by: Jonathan Rose
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6 years agocore_local: Fix memory corruption race condition.
Richard Mudgett [Thu, 12 Sep 2013 16:44:34 +0000 (16:44 +0000)]
core_local: Fix memory corruption race condition.

The masquerade super test is failing on v12 with high fence violations and
crashing.  The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function.  The invalid string
values happen to be the freed memory fill pattern.

After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value.  The copying thread is using the old string pointer
being freed by the updating thread.  A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.

A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes.  :)

(issue ASTERISK-22221)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2839/
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6 years agoFix symbol collision with pjsua.
David M. Lee [Thu, 12 Sep 2013 15:23:54 +0000 (15:23 +0000)]
Fix symbol collision with pjsua.

We shouldn't be exporting any symbols that start with pjsip_.
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6 years ago'queue add member' help text correction
Rusty Newton [Thu, 12 Sep 2013 00:04:57 +0000 (00:04 +0000)]
'queue add member' help text correction

You are adding dial strings to the queue, not channels. An aribitrary string
could be used, but you are typically referencing a channel. Correcting the
command help text.

(issue ASTERISK-22263)
(closes issue ASTERISK-22263)
Reported By: Rusty Newton
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6 years agoDocumentation fix - waitfordialtone is not boolean, it's time in milliseconds
Rusty Newton [Wed, 11 Sep 2013 23:52:49 +0000 (23:52 +0000)]
Documentation fix - waitfordialtone is not boolean, it's time in milliseconds

Changing text in chan_dahdi.conf sample to be accurate.

(issue ASTERISK-22308)
(closes issue ASTERISK-22308)
Reported By: Malcolm Davenport
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6 years agochan_sip: Reject calls without prior SDP on 200 OK
Jonathan Rose [Wed, 11 Sep 2013 20:03:19 +0000 (20:03 +0000)]
chan_sip: Reject calls without prior SDP on 200 OK

If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.

(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
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6 years agoFix typo in confbridge.conf.sample
Russell Bryant [Wed, 11 Sep 2013 18:03:30 +0000 (18:03 +0000)]
Fix typo in confbridge.conf.sample

The denoise filter requires func_speex, not codec_speex.  Fix this in the
description of the denoise=yes option in confbridge.conf.
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6 years agopjsip: reinvite for connected line updates occurs when it should not
Kevin Harwell [Wed, 11 Sep 2013 14:23:28 +0000 (14:23 +0000)]
pjsip: reinvite for connected line updates occurs when it should not

Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:

1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.

Also added an SDP when an update is sent out.

(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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6 years agoFix incorrect usages of ast_realloc().
Richard Mudgett [Tue, 10 Sep 2013 18:05:47 +0000 (18:05 +0000)]
Fix incorrect usages of ast_realloc().

There are several locations in the code base where this is done:
buf = ast_realloc(buf, new_size);

This is going to leak the original buf contents if the realloc fails.

Review: https://reviewboard.asterisk.org/r/2832/
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6 years agoFixed utils directory breakage from r398748, this time with extra hate.
David M. Lee [Tue, 10 Sep 2013 17:50:13 +0000 (17:50 +0000)]
Fixed utils directory breakage from r398748, this time with extra hate.
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6 years agoFixed utils directory breakage from r398648
David M. Lee [Tue, 10 Sep 2013 17:26:19 +0000 (17:26 +0000)]
Fixed utils directory breakage from r398648
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6 years agoMALLOC_DEBUG: Change fence magic number to be completely different from the freed...
Richard Mudgett [Mon, 9 Sep 2013 23:29:44 +0000 (23:29 +0000)]
MALLOC_DEBUG: Change fence magic number to be completely different from the freed magic number.

Race conditions between freeing a nul terminated string and
ast_strdup()'ing it are more likely to be detected if the fence and freed
magic numbers are completely different.
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6 years agoAdd extra debugging to res_pjsip_endpoint_identifier_ip
Mark Michelson [Mon, 9 Sep 2013 22:00:44 +0000 (22:00 +0000)]
Add extra debugging to res_pjsip_endpoint_identifier_ip
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6 years agoFix DEBUG_THREADS when lock is acquired in __constructor__
David M. Lee [Mon, 9 Sep 2013 20:13:40 +0000 (20:13 +0000)]
Fix DEBUG_THREADS when lock is acquired in __constructor__

This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.

With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).

This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).

(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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6 years agoMultiple revisions 398638-398639
David M. Lee [Mon, 9 Sep 2013 19:09:21 +0000 (19:09 +0000)]
Multiple revisions 398638-398639

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  r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line

  Added note about expected behavior of originate
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  r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line

  Added note about expected behavior of originate (the rest of the commit)
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6 years agoBlocked revisions 398559,398578
David M. Lee [Mon, 9 Sep 2013 19:08:15 +0000 (19:08 +0000)]
Blocked revisions 398559,398578

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Multiple revisions 398559,398578

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  r398559 | kmoore | 2013-09-06 14:32:03 -0500 (Fri, 06 Sep 2013) | 20 lines

  Blocked revisions 398558

  ........
  Fix Jabber/XMPP distributed MWI

  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.

  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........

  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398578 | kmoore | 2013-09-06 16:03:45 -0500 (Fri, 06 Sep 2013) | 1 line

  Unblock r398558

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6 years agoUpdate CDR Unit tests to reflect container changes in r398579
Matthew Jordan [Sun, 8 Sep 2013 23:30:39 +0000 (23:30 +0000)]
Update CDR Unit tests to reflect container changes in r398579

When a channel joins a multi-party bridge, the ordering of the CDRs that is
created is determined by the ordering of the channels who happen to be in that
bridge. When r398579 changed the number of buckets in the container to
something sensible, it changed the ordering that the CDRs was created in,
causing one of the multiparty tests to fail. This fixes the test with the
now expected ordering.
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6 years agoPrevent XMPP timeout on blank responses
Kinsey Moore [Sat, 7 Sep 2013 01:03:07 +0000 (01:03 +0000)]
Prevent XMPP timeout on blank responses

Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.

This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.

Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.

(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
    xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
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6 years agoMultiple revisions 398558,398577
Kinsey Moore [Fri, 6 Sep 2013 21:23:02 +0000 (21:23 +0000)]
Multiple revisions 398558,398577

........
  r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines

  Fix Jabber/XMPP distributed MWI

  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.

  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........

  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines

  Commit the remainder of r398523

  This is a missing part of the commit in revision 398523 that corrects
  the name of a variable.

  (issue ASTERISK-22435)
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6 years agocdr: Change the number of container buckets to be similar to the channels container.
Richard Mudgett [Fri, 6 Sep 2013 21:17:45 +0000 (21:17 +0000)]
cdr: Change the number of container buckets to be similar to the channels container.

* Fix the temporary cdr candidate containers to use a prime number of
buckets.
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6 years agocore_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot.
Richard Mudgett [Fri, 6 Sep 2013 20:21:21 +0000 (20:21 +0000)]
core_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot.

* Fix the LocalOptimizationBegin AMI event by eliminating an artificial
buffer size limitation that is too small anyway.
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6 years agocdr: Fix some ref leaks.
Richard Mudgett [Fri, 6 Sep 2013 20:03:01 +0000 (20:03 +0000)]
cdr: Fix some ref leaks.

* Added missing unregister of the cdr container in cdr_engine_shutdown().

* Fixed ref leak in off nominal path of cdr_object_alloc().

* Removed some unnecessary NULL checks in cdr_object_dtor().
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6 years agoastobj2: Add warn unused attribute to some functions.
Richard Mudgett [Fri, 6 Sep 2013 19:26:48 +0000 (19:26 +0000)]
astobj2: Add warn unused attribute to some functions.

* Fixed resulting warnings with improper use of ao2_global_obj_replace().

* Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the
equivalent and more appropriate ao2_global_obj_release() call.
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6 years agoFix build warnings
Kinsey Moore [Fri, 6 Sep 2013 18:53:32 +0000 (18:53 +0000)]
Fix build warnings

When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.

(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
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Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix chan_h323 compilation
Kinsey Moore [Fri, 6 Sep 2013 16:01:05 +0000 (16:01 +0000)]
Fix chan_h323 compilation

This fixes the things in chan_h323 that were missed or ignored in the
great channel opaquification and gets chan_h323 back into a compiling
state.

(closes issue ASTERISK-22365)
Reported by: Dmitry Melekhov
Patches:
    chan_h323.patch uploaded by Dmitry Melekhov
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Merged revisions 398510 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398511 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoastobj2: Only define ao2_bt() once.
Richard Mudgett [Thu, 5 Sep 2013 21:48:02 +0000 (21:48 +0000)]
astobj2: Only define ao2_bt() once.

* Make ao2_bt() not use single char variable names.

* Fix ao2_bt() formatting.
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Merged revisions 398498 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_iax2: Reduce indentation in __attempt_transmit().
Richard Mudgett [Thu, 5 Sep 2013 19:18:10 +0000 (19:18 +0000)]
chan_iax2: Reduce indentation in __attempt_transmit().

* Reduce indentation in __attempt_transmit().

* Don't update the static last error time variable every time in
__schedule_action() and socket_read().
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Merged revisions 398456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398457 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398458 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_iax2: Fix stray reference to worker thread idle_list.
Richard Mudgett [Thu, 5 Sep 2013 17:31:29 +0000 (17:31 +0000)]
chan_iax2: Fix stray reference to worker thread idle_list.

* Fix stray reference to idle_list in cleanup_thread_list().  This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.

* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
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Merged revisions 398416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398417 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398418 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_iax2: Fix bridgecallno deadlock avoidance.
Richard Mudgett [Thu, 5 Sep 2013 17:17:53 +0000 (17:17 +0000)]
chan_iax2: Fix bridgecallno deadlock avoidance.

* Fix bridgecallno deadlock avoidance.  When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.

* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.

* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list.  defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
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Merged revisions 398379 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398380 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398381 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoClarify server_uri and client_uri registration settings.
Mark Michelson [Thu, 5 Sep 2013 14:10:45 +0000 (14:10 +0000)]
Clarify server_uri and client_uri registration settings.

Used some of Rusty's suggested language plus also included
more SIPesque descriptions of where the URIs are actually
used in an outgoing REGISTER.

(closes issue ASTERISK-22390)
reported by Rusty Newton
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Merged revisions 398368 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398369 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_iax2: Add missing control frame names to debug frame decode output.
Richard Mudgett [Wed, 4 Sep 2013 23:07:41 +0000 (23:07 +0000)]
chan_iax2: Add missing control frame names to debug frame decode output.
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Merged revisions 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398302 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398303 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoGive more detail regarding failures to create request with auth credentials.
Mark Michelson [Wed, 4 Sep 2013 22:49:25 +0000 (22:49 +0000)]
Give more detail regarding failures to create request with auth credentials.

(issue ASTERISK-22386)
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Merged revisions 398299 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agounit tests: test_voicemail_api leaks stringfields from snapshots
Jonathan Rose [Wed, 4 Sep 2013 21:37:01 +0000 (21:37 +0000)]
unit tests: test_voicemail_api leaks stringfields from snapshots

(closes issue ASTERISK-22414)
Reported by: Corey Farrell
Patches:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398285 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398286 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoapp_voicemail: Fix leaking config objects when msg_id doesn't match
Jonathan Rose [Wed, 4 Sep 2013 21:20:02 +0000 (21:20 +0000)]
app_voicemail: Fix leaking config objects when msg_id doesn't match

(issues ASTERISK-22414)
Reported by: Corey Farrell
Patch:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398281 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398283 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_misdn: Fix misdn debug output printed with arbitrary verbose levels.
Richard Mudgett [Wed, 4 Sep 2013 16:03:14 +0000 (16:03 +0000)]
chan_misdn: Fix misdn debug output printed with arbitrary verbose levels.

Fix the misdn debug output to remote consoles.  chan_misdn uses
ast_console_puts() which doesn't know about verbose levels.  Better to use
ast_verbose() instead.  Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e.  any undefined level.

(closes issue AST-1218)
Reported by: Guenther Kelleter
Patches:
      misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter
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Merged revisions 398235 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398236 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398237 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDebug messages for pjsip outbound registration
Kevin Harwell [Wed, 4 Sep 2013 14:32:25 +0000 (14:32 +0000)]
Debug messages for pjsip outbound registration

Added debug messages indicating that an outbound registration attempt was made
and it was successful in pjsip.

(closes issue ASTERISK-22388)
Reported by: Rusty Newton
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Merged revisions 398226 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix remote tcs sequence handling on empty tcs received
Alexandr Anikin [Tue, 3 Sep 2013 20:28:01 +0000 (20:28 +0000)]
Fix remote tcs sequence handling on empty tcs received
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Merged revisions 398214 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398215 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoPrevent a crash in res_pjsip_dtmf_info.c
Kinsey Moore [Tue, 3 Sep 2013 18:09:02 +0000 (18:09 +0000)]
Prevent a crash in res_pjsip_dtmf_info.c

This change makes sure that a content type header exists before
checking the contents of the header against known SIP INFO DTMF content
types.
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Merged revisions 398206 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398207 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixed 'make clean' for wiki docs
David M. Lee [Tue, 3 Sep 2013 17:19:30 +0000 (17:19 +0000)]
Fixed 'make clean' for wiki docs
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Merged revisions 398198 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoBe a little more verbose when loading cel_custom.conf.
Walter Doekes [Tue, 3 Sep 2013 14:29:52 +0000 (14:29 +0000)]
Be a little more verbose when loading cel_custom.conf.

Review: https://reviewboard.asterisk.org/r/2805/
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Merged revisions 398167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398168 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398196 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix graceful shutdown crash.
David M. Lee [Fri, 30 Aug 2013 20:58:59 +0000 (20:58 +0000)]
Fix graceful shutdown crash.

The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
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Merged revisions 398149 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398150 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoNew pjsip.conf.sample
Rusty Newton [Fri, 30 Aug 2013 20:37:54 +0000 (20:37 +0000)]
New pjsip.conf.sample

(issue ASTERISK-22145)
(closes issue ASTERISK-22145)
Reported By: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2811/
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Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd a reloadable option for sorcery type objects
Kevin Harwell [Fri, 30 Aug 2013 19:55:56 +0000 (19:55 +0000)]
Add a reloadable option for sorcery type objects

Some configuration objects currently won't place nice if reloaded.
Specifically, in this case the pjsip transport objects.  Now when
registering an object in sorcery one may specify that the object is
allowed to be reloaded or not.  If the object is set to not reload
then upon reloading of the configuration the objects of that type
will not be reloaded.  The initially loaded objects of that type
however will remain.

While the transport objects will not longer be reloaded it is still
possible for a user to configure an endpoint to an invalid transport.
A couple of log messages were added to help diagnose this problem if
it occurs.

(closes issue ASTERISK-22382)
Reported by: Rusty Newton
(closes issue ASTERISK-22384)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2807/
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Merged revisions 398139 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398140 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix various memory leaks
Kevin Harwell [Fri, 30 Aug 2013 19:22:59 +0000 (19:22 +0000)]
Fix various memory leaks

main/config.c - cleanup cache fie includes
res/res_security_log.c - unregister logger level
channesl/chan_sip.c - cleanup io context and notify_types
main/translator.c - cleanup at shutdown
main/named_acl.c - cleanup cli commands
main/indications.c - ast_get_indication_tone() unref default_tone_zone if used

(closes issues ASTERISK-22378)
Reported by: Corey Farrell
Patches:
     config_shutdown.patch uploaded by coreyfarrell (license 5909)
     res_security_log.patch uploaded by coreyfarrell (license 5909)
     chan_sip-11.patch uploaded by coreyfarrell (license 5909)
     indications_refleak.patch uploaded by coreyfarrell (license 5909)
     named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license 5909)
     translate_shutdown.patch uploaded by coreyfarrell (license 5909)

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Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398103 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398116 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398124 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUpdate UPGRADE.txt file for Asterisk 12
Matthew Jordan [Fri, 30 Aug 2013 18:38:00 +0000 (18:38 +0000)]
Update UPGRADE.txt file for Asterisk 12

This simply pulls in the changes that were breaking from the CHANGES file
and updates a few other areas accordingly. It also removes the 10 => 11
notes, which are traditionally removed from each major version and stored
in the appropriate UPGRADE-X.txt file.
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Merged revisions 398100 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398101 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agofeatures_config: Ignore parkinglots in features.conf instead of failing to load
Jonathan Rose [Fri, 30 Aug 2013 18:30:01 +0000 (18:30 +0000)]
features_config: Ignore parkinglots in features.conf instead of failing to load

Parkinglots are defined in res_features.conf now, but this patch fixes
features_config so that features don't fail to load when parkinglots
are present in features.conf

Review: https://reviewboard.asterisk.org/r/2801/
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Merged revisions 398068 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398099 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agofeatures_config: Don't require features.conf to be present for Asterisk to load
Jonathan Rose [Fri, 30 Aug 2013 18:04:41 +0000 (18:04 +0000)]
features_config: Don't require features.conf to be present for Asterisk to load

(closes issue ASTERISK-22426)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2806/
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Merged revisions 398020 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398064 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMemory leak fix
Kevin Harwell [Fri, 30 Aug 2013 17:59:06 +0000 (17:59 +0000)]
Memory leak fix

ast_xmldoc_printable returns an allocated block that must be freed by the
caller.  Fixed manager.c and res_agi.c to stop leaking these results.

(closes issue ASTERISK-22395)
Reported by: Corey Farrell
Patches:
     manager-leaks-12.patch uploaded by coreyfarrell (license 5909)
     res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398061 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398062 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agotest_substitution: Fix failing test.
Richard Mudgett [Fri, 30 Aug 2013 17:11:06 +0000 (17:11 +0000)]
test_substitution: Fix failing test.

Revert the -r392190 change.  The original test was correct.  The CDR code
was actually returning an unititialized buffer.
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Merged revisions 398025 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398026 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agotest_substituition: Fix failed test reporting to actually report failure.
Richard Mudgett [Fri, 30 Aug 2013 17:03:09 +0000 (17:03 +0000)]
test_substituition: Fix failed test reporting to actually report failure.

You cannot put the "Testing <blah> pass/fail" on a single line before
actually performing the test.  Now any additional failure information is
logged before the test pass/fail announcement.

* Added an additional CDR(answer,u) test.
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Merged revisions 398018 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398019 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398023 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix memory leaks
Kevin Harwell [Fri, 30 Aug 2013 16:27:57 +0000 (16:27 +0000)]
Fix memory leaks

(closes issue ASTERISK-22368)
Reported by: Corey Farrell
Patches:
     issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes (license 5674)
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Merged revisions 398004 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 398011 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398016 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCheck return value on fwrite
Kevin Harwell [Fri, 30 Aug 2013 15:39:09 +0000 (15:39 +0000)]
Check return value on fwrite
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Merged revisions 398000 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398002 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agooptional_api: Fix linking problems between modules that export global symbols
David M. Lee [Fri, 30 Aug 2013 13:40:27 +0000 (13:40 +0000)]
optional_api: Fix linking problems between modules that export global symbols

With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].

This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.

For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.

Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)

The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.

Other changes made as a part of this patch:
 * The stubs for http_websocket that wrap system calls set errno to
   ENOSYS.

 * res_http_websocket now properly increments module use count.

 * In loader.c, the while() wrappers around dlclose() were removed. The
   while(!dlclose()) is actually an anti-pattern, which can lead to
   infinite loops if the module you're attempting to unload exports a
   symbol that was directly linked to.

 * The special handling of nonoptreq on systems without weak symbol
   support was removed, since we no longer rely on weak symbols for
   optional_api.

 [1]: https://wiki.asterisk.org/wiki/x/wACUAQ

(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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6 years agoARI: Implement /recordings/stored API's
David M. Lee [Fri, 30 Aug 2013 13:28:50 +0000 (13:28 +0000)]
ARI: Implement /recordings/stored API's

his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.

The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.

There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.

The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)

(closes issue ASTERISK-21582)
Review: https://reviewboard.asterisk.org/r/2693/
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6 years agoMultiple revisions 397975-397976
David M. Lee [Fri, 30 Aug 2013 13:27:52 +0000 (13:27 +0000)]
Multiple revisions 397975-397976

........
  r397975 | rmudgett | 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line

  pbx.c: Make ast_str_substitute_variables_full() not mask variables.
........
  r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013) | 1 line

  Revert last commit.
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6 years agopbx.c: Make pbx_substitute_variables_helper_full() not mask variables.
Richard Mudgett [Fri, 30 Aug 2013 01:20:05 +0000 (01:20 +0000)]
pbx.c: Make pbx_substitute_variables_helper_full() not mask variables.
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6 years agoSanitize XML output for PIDF bodies.
Mark Michelson [Fri, 30 Aug 2013 00:11:22 +0000 (00:11 +0000)]
Sanitize XML output for PIDF bodies.

PJSIP's PIDF API does not replace angle brackets with
their appropriate counterparts for XML. So we have to
do it ourself. In this particular case, the problem had
to do with attempting to place an unsanitized SIP URI
into an XML node. Now we don't get a 488 from recipients
of our PIDF NOTIFYs.
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6 years agoFix method for creating activities string in PIDF bodies.
Mark Michelson [Thu, 29 Aug 2013 22:54:05 +0000 (22:54 +0000)]
Fix method for creating activities string in PIDF bodies.

The previous method did not allocate enough space to create
the entire string, but adjusted the string's slen value to
be larger than the actual allocation. This resulted in garbled
text in NOTIFY requests from Asterisk.

This method allocates the proper amount of space first and then
writes the content into the buffer.
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6 years agoVerbose logging discrepancies
Kevin Harwell [Thu, 29 Aug 2013 22:49:24 +0000 (22:49 +0000)]
Verbose logging discrepancies

Refactored cases where a combination of ast_verbose/options_verbose were
present.  Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used.  Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.

(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
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6 years agoFix when the subscription_terminated callback is called for subscription handlers.
Mark Michelson [Thu, 29 Aug 2013 22:26:03 +0000 (22:26 +0000)]
Fix when the subscription_terminated callback is called for subscription handlers.

The previous placement would result in the resubscribe() callback called instead of
the subscription_terminated() callback being called when a subscription was ended
via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw
an assertion.
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6 years agoFix a race condition where a canceled call was answered.
Mark Michelson [Thu, 29 Aug 2013 22:25:16 +0000 (22:25 +0000)]
Fix a race condition where a canceled call was answered.

RFC 5407 section 3.1.2 details a scenario where a UAC sends
a CANCEL at the same time that a UAS sends a 200 OK for the
INVITE that the UAC is canceling. When this occurs, it is the
role of the UAC to immediately send a BYE to terminate
the call.

This scenario was reproducible by have a Digium phone with two lines
place a call to a second phone that forwarded the call to the second
line on the original phone. The Digium phone, upon realizing that it
was connecting to itself, would attempt to cancel the call. The timing
of this happened to trigger the aforementioned race condition about
80% of the time. Asterisk was not doing its job of sending a BYE
when receiving a 200 OK on a cancelled INVITE. The result was that
the ast_channel structure was destroyed but the underlying SIP
session, as well as the PJSIP inv_session and dialog, were still
alive. Attempting to perform an action such as a transfer, once in
this state, would result in Asterisk crashing.

The circumstances are now detected properly and the session is ended
as recommended in RFC 5407.

(closes issue AST-1209)
reported by John Bigelow
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6 years agoMemory leaks fix
Kevin Harwell [Thu, 29 Aug 2013 21:37:29 +0000 (21:37 +0000)]
Memory leaks fix

(closes ASTERISK-22376)
Reported by: John Hardin
Patches:
     memleak.patch uploaded by jhardin (license 6512)
     memleak2.patch uploaded by jhardin (license 6512)
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6 years agoRevert r394939 due to (numerous) objections
Matthew Jordan [Thu, 29 Aug 2013 20:22:08 +0000 (20:22 +0000)]
Revert r394939 due to (numerous) objections

The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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6 years agoAccount for {} in Swagger notes
David M. Lee [Thu, 29 Aug 2013 16:21:31 +0000 (16:21 +0000)]
Account for {} in Swagger notes
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6 years agoRecursively search for '.c' files when making documentation with 'make full'
Matthew Jordan [Thu, 29 Aug 2013 16:05:23 +0000 (16:05 +0000)]
Recursively search for '.c' files when making documentation with 'make full'

Without this, documentation defined in sub-folders is ignored. Since having
properly generated documentation is especially important in Asterisk 12 -
not having it can cause a module to not load - 'make full' needs to look in
all .c files.
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6 years agoMultiple revisions 397921-397922
Mark Michelson [Thu, 29 Aug 2013 15:43:23 +0000 (15:43 +0000)]
Multiple revisions 397921-397922

........
  r397921 | mmichelson | 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines

  Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events.

  Attempting to transfer an unbridged call would result in crashes in either CEL code or
  in the conversion to AMI messages.
........
  r397922 | mmichelson | 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) | 3 lines

  Remove extra debug message.
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6 years agoActually *add* the database schema management utilities
Matthew Jordan [Thu, 29 Aug 2013 12:30:07 +0000 (12:30 +0000)]
Actually *add* the database schema management utilities

In r397874, the scripts were removed... but not replaced. Thanks to
Michael Young for noticing this!
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6 years agoFix some uninitialized buffers for CDR handling valgrind found.
Richard Mudgett [Wed, 28 Aug 2013 23:15:43 +0000 (23:15 +0000)]
Fix some uninitialized buffers for CDR handling valgrind found.

* Made ast_strftime_locale() ensure that the output buffer is initialized.
The std library strftime() returns 0 and does not touch the buffer if it
has an error.  However, the function can also return 0 without an error.

(closes issue ASTERISK-22412)
Reported by: rmudgett
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6 years agoFixed problems with ast_cdr_serialize_variables().
Richard Mudgett [Wed, 28 Aug 2013 22:56:03 +0000 (22:56 +0000)]
Fixed problems with ast_cdr_serialize_variables().

* Fixed return value of ast_cdr_serialize_variables() on error.  It needs
to return 0 indicating no CDR variables found.

* Made ast_cdr_serialize_variables() check the return value of
cdr_object_format_property() and assert if nonzero.  A member of the
cdr_readonly_vars[] was not handled.

* Removed unused elements from cdr_readonly_vars[]: total_duration,
total_billsec, first_start, and first_answer.
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6 years agoMade the on/off in CLI "cdr set debug [on|off]" case insensitive.
Richard Mudgett [Wed, 28 Aug 2013 22:43:14 +0000 (22:43 +0000)]
Made the on/off in CLI "cdr set debug [on|off]" case insensitive.
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6 years agoMake CDR variable name chandling consistently case insensitive.
Richard Mudgett [Wed, 28 Aug 2013 22:38:30 +0000 (22:38 +0000)]
Make CDR variable name chandling consistently case insensitive.
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6 years agoMake CDR code deal with channel names case insensitively.
Richard Mudgett [Wed, 28 Aug 2013 22:35:25 +0000 (22:35 +0000)]
Make CDR code deal with channel names case insensitively.
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6 years agoSome CDR code optimization.
Richard Mudgett [Wed, 28 Aug 2013 22:24:01 +0000 (22:24 +0000)]
Some CDR code optimization.
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6 years agoWhitespace and curly braces.
Richard Mudgett [Wed, 28 Aug 2013 21:38:39 +0000 (21:38 +0000)]
Whitespace and curly braces.
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6 years agoImprove detection of answer on SIP blind transfer.
Mark Michelson [Wed, 28 Aug 2013 21:09:43 +0000 (21:09 +0000)]
Improve detection of answer on SIP blind transfer.

A problem encountered during testing was that res_pjsip_refer would
not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook
that was supposed to send the NOTIFY would never be told that an answer
had occurred. This happened for two reasons:

1) The transferee channel on which the framehook was on was already up.
2) Answers are rarely if ever written to channels. Rather, the ast_answer()
or ast_raw_answer() function is used to answer channels.

Thanks to a suggestion by Matt Jordan, the best way to detect that the call
had been answered was to find out when the transferee channel joined a bridge.
With stasis this is an easy task. So now, in addition to the framehook logic,
there is a stasis subscription used to determine when the transferee has entered
a bridge. Once it has entered, an appropriate NOTIFY is sent.
........

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6 years agoAdd database schema management using Alembic
Matthew Jordan [Wed, 28 Aug 2013 20:55:53 +0000 (20:55 +0000)]
Add database schema management using Alembic

This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk.  In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.

With this the schema only needs to be maintained in one place instead of
once per database.  The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes).  Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.

See contrib/ast-db-manage/README.md for more details.

Review: https://reviewboard.asterisk.org/r/2731

patch by Russell Bryant (license 6300)
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6 years agoUpdate CHANGES file for Asterisk 12
Matthew Jordan [Wed, 28 Aug 2013 20:49:02 +0000 (20:49 +0000)]
Update CHANGES file for Asterisk 12

This updates the Asterisk 12 CHANGES file with the things that were missed
during the development cycle.

Review: https://reviewboard.asterisk.org/r/2795/
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6 years agopbx.c: Make ast_str_substitute_variables_full() not mask variables.
Richard Mudgett [Wed, 28 Aug 2013 16:13:18 +0000 (16:13 +0000)]
pbx.c: Make ast_str_substitute_variables_full() not mask variables.
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Merged revisions 397859 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoast_free() is null tollerant.
Richard Mudgett [Wed, 28 Aug 2013 16:09:12 +0000 (16:09 +0000)]
ast_free() is null tollerant.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMatch use of ast_free() with ast_calloc() and add some curly braces.
Richard Mudgett [Wed, 28 Aug 2013 16:07:30 +0000 (16:07 +0000)]
Match use of ast_free() with ast_calloc() and add some curly braces.
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Merged revisions 397856 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoFix dialog matching in the SIP distributor.
Mark Michelson [Wed, 28 Aug 2013 15:43:15 +0000 (15:43 +0000)]
Fix dialog matching in the SIP distributor.

Dialog matching is performed in the distributor for the sole
purpose of retrieving an associated serializer so the request
may be serialized.

This patch fixes two problems.

First, incoming CANCEL requests that had no to-tag (which really
should be *all* CANCEL requests) would not match with a dialog.
An earlier bug fix to deal with early CANCEL requests would result
in the CANCEL being replied to with a 481. The fix for this is to
find the matching INVITE transaction and get the dialog from that
transaction.

Second, no SIP responses were matching dialogs. This is because we
were inverting the tags that we were passing into PJSIP's dialog
finding function. This logic has been corrected by setting local
and remote tag variables based on whether the incoming message is
a request or response.
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Merged revisions 397854 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoARI: WebSocket event cleanup
David M. Lee [Tue, 27 Aug 2013 19:19:36 +0000 (19:19 +0000)]
ARI: WebSocket event cleanup

Stasis events (which get distributed over the ARI WebSocket) are created
by subscribing to the channel_all_cached and bridge_all_cached topics,
filtering out events for channels/bridges currently subscribed to.

There are two issues with that. First was a race condition, where
messages in-flight to the master subscribe-to-all-things topic would get
sent out, even though the events happened before the channel was put
into Stasis. Secondly, as the number of channels and bridges grow in the
system, the work spent filtering messages becomes excessive.

Since r395954, individual channels and bridges have caching topics, and
can be subscribed to individually. This patch takes advantage, so that
channels and bridges are subscribed to on demand, instead of filtering
the global topics.

The one case where filtering is still required is handling BridgeMerge
messages, which are published directly to the bridge_all topic.

Other than the change to how subscriptions work, this patch mostly just
moves code around. Most of the work generating JSON objects from
messages was moved to .to_json handlers on the message types. The
callback functions handling app subscriptions were moved from res_stasis
(b/c they were global to the model) to stasis/app.c (b/c they are local
to the app now).

(closes issue ASTERISK-21969)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2754/
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Merged revisions 397816 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397820 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMade MALLOC_DEBUG less CPU intensive by default.
Richard Mudgett [Tue, 27 Aug 2013 18:52:23 +0000 (18:52 +0000)]
Made MALLOC_DEBUG less CPU intensive by default.

Storing a backtrace for each allocation in anticipation of a memory
management problem is very CPU intensive.

* Added the CLI "memory backtrace {on|off}" command to request that the
backtrace be gathered only on request.  The backtrace is off by default.

(issue ASTERISK-22221)
Reported by: Matt Jordan
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Merged revisions 397809 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoAST-2013-005: Fix crash caused by invalid SDP
Matthew Jordan [Tue, 27 Aug 2013 18:10:40 +0000 (18:10 +0000)]
AST-2013-005: Fix crash caused by invalid SDP

If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.

This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.

Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.

(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
  issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
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Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 397758 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 397759 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAST-2013-004: Fix crash when handling ACK on dialog that has no channel
Matthew Jordan [Tue, 27 Aug 2013 17:35:20 +0000 (17:35 +0000)]
AST-2013-004: Fix crash when handling ACK on dialog that has no channel

A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.

This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.

Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.

(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
  issueA21064_fix.patch uploaded by wdoekes (License 5674)
........

Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 397712 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 397713 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397753 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix uninitialized value in struct ast_control_pvt_cause_code usage.
Richard Mudgett [Tue, 27 Aug 2013 16:51:08 +0000 (16:51 +0000)]
Fix uninitialized value in struct ast_control_pvt_cause_code usage.
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Merged revisions 397744 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 397745 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoBetter handle clearing the OUTGOING flag when a channel leaves a bridge
Matthew Jordan [Mon, 26 Aug 2013 23:48:56 +0000 (23:48 +0000)]
Better handle clearing the OUTGOING flag when a channel leaves a bridge

When a channel with the OUTGOING flag leaves a bridge, and it will survive
being pulled from the bridge (either because it will execute dialplan,
go into another bridge, or live in a friendly autoloop), we have to clear
the OUTGOING flag. This is the signal to the CDR engine that this channel
is no longer a second class citizen, i.e., it is not "dialed".

The soft hangup flags are only half the picture. If a channel is being
moved from one bridge to another, the soft hangup flags aren't set; however,
the state of the bridge_channel will not be hung up. Since the channel does
not have one of the two hang up states, that implies that the channel is
still technically alive.

This patch modifies the check so that it checks both the soft hangup flags
as well as the bridge_channel state. If either suggests that the channel
is going to persist, we clear the OUTGOING flag.
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Merged revisions 397690 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixed bucket.c for systems where tv_usec is not an unsigned long.
David M. Lee [Mon, 26 Aug 2013 21:32:13 +0000 (21:32 +0000)]
Fixed bucket.c for systems where tv_usec is not an unsigned long.
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Merged revisions 397673 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agobridging: Fix a livelock with local channel optimization.
Richard Mudgett [Mon, 26 Aug 2013 16:25:39 +0000 (16:25 +0000)]
bridging: Fix a livelock with local channel optimization.

Use a better means of waking up the bridge channel thread.
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Merged revisions 397650 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397651 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_dahdi: Add some missing build cleanup.
Richard Mudgett [Mon, 26 Aug 2013 16:15:02 +0000 (16:15 +0000)]
chan_dahdi: Add some missing build cleanup.
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Merged revisions 397643 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoFix bucket unit tests
Matthew Jordan [Sun, 25 Aug 2013 18:12:42 +0000 (18:12 +0000)]
Fix bucket unit tests

After the review for buckets was completed (r2715), the handling of names in
the bucket core was deferred to the wizards. As such, the bucket unit tests
cannot expect that passing a URI with a scheme specified but no actual resource
name will automatically fail. The tests have been updated to not make this
check.
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Merged revisions 397630 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoFix the config_options_test
Matthew Jordan [Sun, 25 Aug 2013 18:00:46 +0000 (18:00 +0000)]
Fix the config_options_test

The config options test requires the entire configuration item to be transparent from
the documentation system. So we let it do that too.

As an aside, please do not use this power for evil. Documentation is your friend, and
you really should document your configurations. Hiding your module's configuration
information from the system attempting to enforce some sanity in the universe is something
only a Bond villain would contemplate.
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Merged revisions 397628 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoAdd rtpengine configuration parameter
Matthew Jordan [Sun, 25 Aug 2013 01:15:28 +0000 (01:15 +0000)]
Add rtpengine configuration parameter

The rtpengine configuration parameter was documented in the XML documentation,
but it was not actually registered with the sorcery object. This adds the
parameter with a default of "asterisk", such that res_rtp_asterisk is chosen as
the default RTP implementation.

(closes issue ASTERISK-22380)
Reported by: Rusty Newton
Tested by: Rusty Newton
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Merged revisions 397621 from http://svn.asterisk.org/svn/asterisk/branches/12

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6 years agoSet new merge properties on 12
Matthew Jordan [Fri, 23 Aug 2013 22:40:57 +0000 (22:40 +0000)]
Set new merge properties on 12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix building of trunk.
Joshua Colp [Fri, 23 Aug 2013 22:20:39 +0000 (22:20 +0000)]
Fix building of trunk.

Note: This is why I commit on the weekend.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397613 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix channel reference leak in Originated channels
Matthew Jordan [Fri, 23 Aug 2013 22:12:57 +0000 (22:12 +0000)]
Fix channel reference leak in Originated channels

When originating channels, ast_pbx_outgoing_* caused the dialed channel
reference to be bumped twice. Ostensibly, this routine is bumping the channel
lifetime such that the channel doesn't get nuked in between locks/unlocks;
however, since the routine should return the dialed channel with its
reference bumped, it only needs to do this one time.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397606 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoBlocked revisions 397604
Joshua Colp [Fri, 23 Aug 2013 21:59:31 +0000 (21:59 +0000)]
Blocked revisions 397604

........
Make libuuid an optional dependency for res_rtp_asterisk instead of a requirement.

Review: https://reviewboard.asterisk.org/r/2777/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd some clarifying documentation to the rewrite_contact endpoint option.
Mark Michelson [Fri, 23 Aug 2013 21:53:48 +0000 (21:53 +0000)]
Add some clarifying documentation to the rewrite_contact endpoint option.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoBlank line tweaks.
Richard Mudgett [Fri, 23 Aug 2013 21:51:19 +0000 (21:51 +0000)]
Blank line tweaks.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397602 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd the bucket API.
Joshua Colp [Fri, 23 Aug 2013 21:49:47 +0000 (21:49 +0000)]
Add the bucket API.

Bucket is a URI based API for the creation, retrieval, updating, and deletion
of "buckets" and files contained within them.

Review: https://reviewboard.asterisk.org/r/2715/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3