asterisk/asterisk.git
6 years agoClean up Makefile "warning" clutter when makeopts doesn't exist.
Walter Doekes [Mon, 8 Apr 2013 18:24:50 +0000 (18:24 +0000)]
Clean up Makefile "warning" clutter when makeopts doesn't exist.

Review: https://reviewboard.asterisk.org/r/2304

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDon't attempt a websocket protocol removal if res_http_websocket isn't there
Matthew Jordan [Mon, 8 Apr 2013 15:38:34 +0000 (15:38 +0000)]
Don't attempt a websocket protocol removal if res_http_websocket isn't there

This patch sets the protocols container provided by res_http_websocket to NULL
when the module gets unloaded and adds the necessary checks when adding/
removing a websocket protocol. This prevents some FRACKing on an invalid
pointer to the disposed container if a module that uses res_http_websocket is
unloaded after it.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd multi-channel Stasis messages; refactor Dial AMI events to Stasis
Matthew Jordan [Mon, 8 Apr 2013 14:26:37 +0000 (14:26 +0000)]
Add multi-channel Stasis messages; refactor Dial AMI events to Stasis

This patch does the following:
 * A new Stasis payload has been defined for multi-channel messages. This
   payload can store multiple ast_channel_snapshot objects along with a single
   JSON blob. The payload object itself is opaque; the snapshots are stored
   in a container keyed by roles. APIs have been provided to query for and
   retrieve the snapshots from the payload object.
 * The Dial AMI events have been refactored onto Stasis. This includes dial
   messages in app_dial, as well as the core dialing framework. The AMI events
   have been modified to send out a DialBegin/DialEnd events, as opposed to
   the subevent type that was previously used.
 * Stasis messages, types, and other objects related to channels have been
   placed in their own file, stasis_channels. Unit tests for some of these
   objects/messages have also been written.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoStasis application WebSocket support
David M. Lee [Mon, 8 Apr 2013 13:27:45 +0000 (13:27 +0000)]
Stasis application WebSocket support

This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.

This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.

Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.

Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.

Other changes along for the ride are:
 * An ast_frame_dtor function that's RAII_VAR safe
 * Some common JSON encoders for name/number, timeval, and
   context/extension/priority

Review: https://reviewboard.asterisk.org/r/2361/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd a res_sorcery_astdb module which uses the astdb to persist objects.
Joshua Colp [Sat, 6 Apr 2013 16:00:20 +0000 (16:00 +0000)]
Add a res_sorcery_astdb module which uses the astdb to persist objects.

Review: https://reviewboard.asterisk.org/r/2420/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix For Not Overriding The Default Settings In chan_sip
Michael L. Young [Fri, 5 Apr 2013 20:41:27 +0000 (20:41 +0000)]
Fix For Not Overriding The Default Settings In chan_sip

The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting.  Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.

This patch works similar to what occurs in build_peer().  We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.

In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.

This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.

(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
  asterisk-21225-handle-options-default-prob_v4.diff
Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2385/
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Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoSeparate some event struct definitions from instantiation.
Richard Mudgett [Thu, 4 Apr 2013 18:15:34 +0000 (18:15 +0000)]
Separate some event struct definitions from instantiation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_dahdi: Change inband_on_proceeding option default to no/disabled.
Richard Mudgett [Wed, 3 Apr 2013 20:27:11 +0000 (20:27 +0000)]
chan_dahdi: Change inband_on_proceeding option default to no/disabled.

(issue ASTERISK-21151)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_dahdi: Add inband_on_proceeding compatibility option.
Richard Mudgett [Wed, 3 Apr 2013 20:20:09 +0000 (20:20 +0000)]
chan_dahdi: Add inband_on_proceeding compatibility option.

The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.

Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio.  However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.

ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.

(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUpdate documentation for CHANNEL function
Matthew Jordan [Wed, 3 Apr 2013 17:17:33 +0000 (17:17 +0000)]
Update documentation for CHANNEL function

Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
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Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoastobj2: Fix rbtree duplicate handling.
Richard Mudgett [Wed, 3 Apr 2013 16:01:51 +0000 (16:01 +0000)]
astobj2: Fix rbtree duplicate handling.

OBJ_PARTIAL_KEY searching a rbtree did not find all possible matches if
the container did not accept duplicates.

Added matching node bias to indicate which matching node is being searched
for: first, last, any.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixed spurious rebuilds of func_version.
David M. Lee [Tue, 2 Apr 2013 17:35:45 +0000 (17:35 +0000)]
Fixed spurious rebuilds of func_version.

func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that was
added in r384410 to fix parallel make bugs.

Now build.h will only be created if it does not exist, which was the
original behavior of the Makefile.
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Merged revisions 384544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384545 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoPass the object type name to the configuration framework.
Joshua Colp [Tue, 2 Apr 2013 12:18:50 +0000 (12:18 +0000)]
Pass the object type name to the configuration framework.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384518 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMake things work again
Matthew Jordan [Tue, 2 Apr 2013 11:40:05 +0000 (11:40 +0000)]
Make things work again

Sorry folks. ',' are still greater than '|'.

Thanks for playing along :-)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384514 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoinstall_prereq: Build jansson from source, when necessary
David M. Lee [Mon, 1 Apr 2013 20:10:47 +0000 (20:10 +0000)]
install_prereq: Build jansson from source, when necessary

When r383579 was committed, it made Jansson a required dependency.

While libjansson-dev and jansson-devel are available on recent
distros, some older (but still supported) distros don't have
it. There's a pull request[1] to get it into repoforge, but that still
doesn't help everyone. (And helps no one until the pull request is
merged and packages are built).

This patch adds Jansson install from source to the install_unpackaged()
function. There are a few gotcha's, which makes this change not
completely trivial.

 * Since Jansson may be installed by a package, don't install from
   source if a package installation can be found
   * libresample may also be installed via package, so I added a
     similar check to that.
 * Since Jansson installs into /usr/local, this patch also adds
   /usr/local/lib to /etc/ld.so.conf.d so that the library can be
   found.
   * The alternative was to install into /usr, but then it gets
     complicated having to deal with EL's /usr/lib{32,64} shenanigans.

 [1]: https://github.com/repoforge/rpms/pull/250

Review: https://reviewboard.asterisk.org/r/2414/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384488 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMake appropriate items parse using '|' instead of ','
Matthew Jordan [Mon, 1 Apr 2013 14:44:30 +0000 (14:44 +0000)]
Make appropriate items parse using '|' instead of ','

This patch fixes a bug introduced in r76703, wherein Asterisk could only parse
arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
syntax of NoOp,foo|bar is now parsed correctly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoRemove silly use of strncmp.
Joshua Colp [Mon, 1 Apr 2013 14:10:46 +0000 (14:10 +0000)]
Remove silly use of strncmp.
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Merged revisions 384414 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agostasis: Fixed message ordering issues when forwarding
David M. Lee [Mon, 1 Apr 2013 13:37:51 +0000 (13:37 +0000)]
stasis: Fixed message ordering issues when forwarding

This patch fixes an issue of message ordering that occurs when
multiple topics are forwarded to an aggregator topic (such as
ast_channel_topic_all()).

It is (very reasonably) expected that the rules governing message
dispatch order still apply, so long as the messages start from the
same thread, and are received by the same subscription. Because the
existing code had an additional layer of dispatching via the Stasis
thread pool for forwards, those promises couldn't be kept.

Forwarding subscriptions no longer have their own mailbox, and now
dispatch directly from the forwarding topic's stasis_publish()
call. This means that the topic's lock is held for the duration of not
only a message's dispatch, but the dispatch of all the forwards. This
shouldn't be a problem right now, but if an aggregator topic had many
subscribers, it could become a problem. But I figure we can write more
clever code when the time comes, if necessary.

Review: https://reviewboard.asterisk.org/r/2419/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384413 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix parallel make problems.
David M. Lee [Mon, 1 Apr 2013 13:34:51 +0000 (13:34 +0000)]
Fix parallel make problems.

Occasionally, make -j would fail due to missing includes, or other
unusual errors.

This was due to the 'cleantest' target, which was designed to force a
make clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile did
not depend upon cleantest, hence would run in parallel to it. By
adding the dependency, make -j runs happily now.

Review: https://reviewboard.asterisk.org/r/2418/
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Merged revisions 384411 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384412 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoProperly format an intmax_t value
Matthew Jordan [Sat, 30 Mar 2013 05:15:42 +0000 (05:15 +0000)]
Properly format an intmax_t value

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384390 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoConvert TestEvent AMI events over to Stasis Core
Matthew Jordan [Sat, 30 Mar 2013 05:06:54 +0000 (05:06 +0000)]
Convert TestEvent AMI events over to Stasis Core

This patch migrates the TestEvent AMI events to first be dispatched over the
Stasis-Core message bus. This helps to preserve the ordering of the events
with other events in the AMI system, such as the various channel related
events.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384389 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoapp_voicemail: Add blank argument to externnotify if no context argument
Jonathan Rose [Fri, 29 Mar 2013 16:37:23 +0000 (16:37 +0000)]
app_voicemail: Add blank argument to externnotify if no context argument

At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.

(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
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Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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6 years agoAdd uuid wrapper API call ast_uuid_generate_str().
Richard Mudgett [Thu, 28 Mar 2013 23:59:20 +0000 (23:59 +0000)]
Add uuid wrapper API call ast_uuid_generate_str().

* Updated test_uuid.c to test the new API call.

* Made system use the new API call to eliminate "10's of lines" where
used.

* Fixed untested ast_strdup() return in stasis_subscribe() by eliminating
the need for it.  struct stasis_subscription now contains the uniqueid[]
string.

* Fixed some issues in exchangecal_write_event():
  Create uid with enough space for a UUID string to avoid a realloc.
  Fix off by one error if the calendar event provided a UUID string.
  There is no need to check for NULL before calling ast_free().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoBreak the world. Stasis message type accessors should now all be named correctly.
Kinsey Moore [Thu, 28 Mar 2013 15:45:18 +0000 (15:45 +0000)]
Break the world. Stasis message type accessors should now all be named correctly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoConvert MWI state message type to the new stasis naming convention
Kinsey Moore [Wed, 27 Mar 2013 22:42:06 +0000 (22:42 +0000)]
Convert MWI state message type to the new stasis naming convention

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdded a doxygen group for Stasis messages and topics
David M. Lee [Wed, 27 Mar 2013 21:52:43 +0000 (21:52 +0000)]
Added a doxygen group for Stasis messages and topics

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAddress uninitialized conditional that valgrind found
Kinsey Moore [Wed, 27 Mar 2013 19:52:19 +0000 (19:52 +0000)]
Address uninitialized conditional that valgrind found
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6 years agoFix a file descriptor leak in off nominal path
Matthew Jordan [Wed, 27 Mar 2013 18:52:16 +0000 (18:52 +0000)]
Fix a file descriptor leak in off nominal path

While looking at the security vulnerability in ASTERISK-20967, Walter noticed
a file descriptor leak and some other issues in off nominal code paths. This
patch corrects them.

Note that this patch is not related to the vulnerability in ASTERISK-20967,
but the patch was placed on that issue.

(closes issue ASTERISK-20967)
Reported by: wdoekes
patches:
  issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix white noise on SRTP decryption
Kinsey Moore [Wed, 27 Mar 2013 17:07:44 +0000 (17:07 +0000)]
Fix white noise on SRTP decryption

When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.

(closes issue ASTERISK-21323)
Reported by: andrea
Tested by: Kinsey Moore, andrea, John Bigelow
Patches:
    whitenoise_fix.diff uploaded by Kinsey Moore
........

Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 384049 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384050 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAST-2013-003: Prevent username disclosure in SIP channel driver
Matthew Jordan [Wed, 27 Mar 2013 15:27:31 +0000 (15:27 +0000)]
AST-2013-003: Prevent username disclosure in SIP channel driver

When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
 * A "407 Proxy Authentication Required" response is sent instead of a
   "401 Unauthorized" response
 * The presence or absence of additional tags occurs at the end of "403
   Forbidden" (such as "(Bad Auth)")
 * A "401 Unauthorized" response is sent instead of "403 Forbidden" response
   after a retransmission
 * Retransmission are sent when a matching peer did not exist, but not when a
   matching peer did exist.

This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.

This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.

(closes issue ASTERISK-21013)
Reported by: wdoekes
Tested by: wdoekes, kmoore
patches:
  AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
  AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
  AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
........

Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAST-2013-002: Prevent denial of service in HTTP server
Matthew Jordan [Wed, 27 Mar 2013 14:39:11 +0000 (14:39 +0000)]
AST-2013-002: Prevent denial of service in HTTP server

AST-2012-014, fixed in January of this year, contained a fix for Asterisk's
HTTP server for a remotely-triggered crash. While the fix put in place fixed
the possibility for the crash to be triggered, a denial of service vector still
exists with that solution if an attacker sends one or more HTTP POST requests
with very large Content-Length values. This patch resolves this by capping
the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with
Content-Length greater than this cap will not result in any memory allocation.
The POST will be responded to with an HTTP 413 "Request Entity Too Large"
response.

This issue was reported by Christoph Hebeisen of TELUS Security Labs

(closes issue ASTERISK-20967)
Reported by: Christoph Hebeisen
patches:
  AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
  AST-2013-002-10.diff uploaded by mmichelson (License 5049)
  AST-2013-002-11.diff uploaded by mmichelson (License 5049)
........

Merged revisions 383978 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383980 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAST-2013-001: Prevent buffer overflow through H.264 format negotiation
Matthew Jordan [Wed, 27 Mar 2013 14:28:36 +0000 (14:28 +0000)]
AST-2013-001: Prevent buffer overflow through H.264 format negotiation

The format attribute resource for H.264 video performs an unsafe read against a
media attribute when parsing the SDP. The value passed in with the format
attribute is not checked for its length when parsed into a fixed length buffer.
This patch resolves the vulnerability by only reading as many characters from
the SDP value as will fit into the buffer.

(closes issue ASTERISK-20901)
Reported by: Ulf Harnhammar
patches:
  h264_overflow_security_patch.diff uploaded by jrose (License 6182)
........

Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383975 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix skinny encall button to not blind xfer.
Damien Wedhorn [Wed, 27 Mar 2013 07:24:37 +0000 (07:24 +0000)]
Fix skinny encall button to not blind xfer.

The softbutton endcall should not turn a transfer into a blind transfer but
hangup the exten being called and leave the original call on hold. This does
that.

(closes issue ASTERISK-21321)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
    skinny-xferendcall01.diff uploaded by wedhorn (license 5019)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoRemove the noop handler from sorcery so it does not produce an empty value.
Joshua Colp [Tue, 26 Mar 2013 23:34:43 +0000 (23:34 +0000)]
Remove the noop handler from sorcery so it does not produce an empty value.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383925 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoResolve deadlock between SIP registration and channel based functions
Matthew Jordan [Tue, 26 Mar 2013 02:30:10 +0000 (02:30 +0000)]
Resolve deadlock between SIP registration and channel based functions

In r373424, several reentrancy problems in chan_sip were addressed. As a
result, the SIP channel driver is now properly locking the channel driver
private information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by functions called
by register_verify. This includes:
 * Holding the private lock while calling sip_send_mwi_to_peer. This can create
   a new sip_pvt via sip_alloc, which will obtain the channel container lock.
   This is a locking inversion, as any channel related lock must be obtained
   prior to obtaining the SIP channel technology private lock.

   Note that this issue was already fixed in Asterisk 11.

 * Holding the private lock while calling sip_poke_peer. In the same vein as
   sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
   the same locking inversion.

Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.

(issue ASTERISK-21068)
Reported by: Nicolas Bouliane

(issue ASTERISK-20550)
Reported by: David Brillert

(issue ASTERISK-21314)
Reported by: Badalian Vyacheslav

(issue ASTERISK-21296)
Reported by: Gabriel Birke
........

Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoResolve deadlock between pending CDR and batch CDR locks
Matthew Jordan [Tue, 26 Mar 2013 01:58:45 +0000 (01:58 +0000)]
Resolve deadlock between pending CDR and batch CDR locks

r375757 attempted to resolve a race condition between multiple submissions of
CDRs while in batch mode from attempting to destroy the scheduled batch
submission by extending the batch CDR lock. Unfortunately, this causes a
deadlock between the pending CDR lock and the batch CDR lock. This patch
resolves the intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is kept to protect
manipulation of the batch CDR settings, but has been placed such that it
is not held when the pending lock is held.

Thanks to Chase Venters for providing lock analysis on the issue.

(issue ASTERISK-21162)
Reported by: Chase Venters
........

Merged revisions 383839 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383840 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383841 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoSuppress compiler warning.
Russell Bryant [Tue, 26 Mar 2013 01:46:39 +0000 (01:46 +0000)]
Suppress compiler warning.

This code caused a compiler warning when --enable-dev-mode was not used.
The warning was that this variable was set but not used.  That was indeed
the case as the only place this is used is as an argument to SKINNY_DEBUG
which is compiled out when not in dev mode.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix multi-station answer race condition.
Russell Bryant [Tue, 26 Mar 2013 01:38:56 +0000 (01:38 +0000)]
Fix multi-station answer race condition.

When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk.  If more than
one station answers the call at the same time, all channels other than
the first one to answer are left in a bad state.  The channel gets
leaked, is not connected to anything, and there's no way to get rid of
it.

We now properly clean up these losing channels by hanging up on them.
Since they lost the race, as we process their answer, there is no
ringing trunk for them to answer.
........

Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383837 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoSet the CALLERID(dnid-num-plan) for incoming ISDN calls.
Richard Mudgett [Mon, 25 Mar 2013 23:25:32 +0000 (23:25 +0000)]
Set the CALLERID(dnid-num-plan) for incoming ISDN calls.

The CALLEDTON channel variable is set for incoming ISDN calls to the lower
7 bits of the Q.931 type-of-number/numbering-plan octet.  The
CALLERID(dnid-num-plan) should have the same value.

(closes issue ASTERISK-21248)
Reported by: rmudgett
........

Merged revisions 383796 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383798 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix typo
Kinsey Moore [Mon, 25 Mar 2013 20:15:09 +0000 (20:15 +0000)]
Fix typo

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383754 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix missing ' ' around '='
Kinsey Moore [Mon, 25 Mar 2013 20:07:00 +0000 (20:07 +0000)]
Fix missing ' ' around '='

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383753 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoinstall_prereq: removed some out-of-date comments
David M. Lee [Mon, 25 Mar 2013 19:28:04 +0000 (19:28 +0000)]
install_prereq: removed some out-of-date comments

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383747 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoinstall_prereq: Adding jansson-devel to RH packages
David M. Lee [Mon, 25 Mar 2013 17:12:03 +0000 (17:12 +0000)]
install_prereq: Adding jansson-devel to RH packages

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383728 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMove NewCallerid, HangupRequest and SoftHangupRequest to Stasis
David M. Lee [Mon, 25 Mar 2013 16:19:55 +0000 (16:19 +0000)]
Move NewCallerid, HangupRequest and SoftHangupRequest to Stasis

HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.

NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.

Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.

There a a few other changes wrapped up in here as well.

 * When ast_channel_topic() is given NULL for a channel, it returns
   the ast_channel_topic_all() topic instead of NULL. This can clean
   up a lot of NULL checking we're doing currently.
 * The fields Cause and Cause-txt were removed from the base channel
   information and put only on the Hangup events, since those fields
   are meaningless outside of a Hangup event.
 * Removed the pipe-delimiter processing of the channelvars field,
   since that's been deprecated forever.

(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoProperly delimit post data in res_config_curl.
Sean Bright [Mon, 25 Mar 2013 12:38:15 +0000 (12:38 +0000)]
Properly delimit post data in res_config_curl.
........

Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383668 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixed another issue from r383579.
David M. Lee [Fri, 22 Mar 2013 20:51:33 +0000 (20:51 +0000)]
Fixed another issue from r383579.

Core modules don't honor <depend> flags in MODULEINFO, which broke jansson
if specified --with-jansson to configure.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel
Michael L. Young [Fri, 22 Mar 2013 20:43:24 +0000 (20:43 +0000)]
Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel

A regression was accidentally introduced when allowing an optional ID to be used
when calling StopMixMonitor.  When we are unable to stop MixMonitor on a
channel, -1 is being returned which triggers the hangup of the channel.

This patch restores the prior behavior by returning 0 whether we were successful
or not.  It also allows the call from the manager to use the return code when
the action fails.

(closes issue ASTERISK-21294)
Reported by: daroz
Tested by: daroz
Patches:
  asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2404/
........

Merged revisions 383631 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCorrected some module issues introduced by r383579.
David M. Lee [Fri, 22 Mar 2013 19:26:37 +0000 (19:26 +0000)]
Corrected some module issues introduced by r383579.

When I moved res_json.c to json.c, I left the MODULE_INFO stuff in there,
which was interesting if you ran module show. I also forgot to call what
was in module_load() from asterisk main().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383611 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMove more channel events to Stasis; move res_json.c to main/json.c.
David M. Lee [Fri, 22 Mar 2013 14:06:46 +0000 (14:06 +0000)]
Move more channel events to Stasis; move res_json.c to main/json.c.

This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.

To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.

I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.

 * Move JSON support from res_json.c to main/json.c
   * Made libjansson-dev a required dependency
 * Added an ast_channel_blob message type, which has a channel
   snapshot and JSON blob of data.
 * Changed UserEvent and Newexten events so that they are dispatched
   via ast_channel_blob messages on the channel's topic.
 * Got rid of the ast_channel_varset message; used ast_channel_blob
   instead.
 * Extracted the manager functions converting Stasis channel events to
   AMI events into manager_channel.c.

(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix skinny voicemail indication issues.
Damien Wedhorn [Fri, 22 Mar 2013 06:32:03 +0000 (06:32 +0000)]
Fix skinny voicemail indication issues.

Unsubscribe from MWI stasis event on channel reload.

(closes issue ASTERISK-21216)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
    skinny-mwiind02.diff uploaded by snuffy (license 5024)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoCorrected doc error for Stasis. I guess the mutex isn't necessary.
David M. Lee [Thu, 21 Mar 2013 20:09:11 +0000 (20:09 +0000)]
Corrected doc error for Stasis. I guess the mutex isn't necessary.

Thanks, rmudgett!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383541 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix astobj2 doxygen comment.
Richard Mudgett [Thu, 21 Mar 2013 17:41:52 +0000 (17:41 +0000)]
Fix astobj2 doxygen comment.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoHave func_curl log a warning when a curl request fails.
Walter Doekes [Wed, 20 Mar 2013 20:27:37 +0000 (20:27 +0000)]
Have func_curl log a warning when a curl request fails.

Review: https://reviewboard.asterisk.org/r/2403/
........

Merged revisions 383460 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383461 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMinor cleanup in func_curl near hashcompat code.
Walter Doekes [Wed, 20 Mar 2013 20:18:40 +0000 (20:18 +0000)]
Minor cleanup in func_curl near hashcompat code.

Review: https://reviewboard.asterisk.org/r/2402/
........

Merged revisions 383457 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoResolve a race condition in Stasis
Kinsey Moore [Wed, 20 Mar 2013 16:01:30 +0000 (16:01 +0000)]
Resolve a race condition in Stasis

Because of the way that topics were handled when publishing, it was
possible to dispatch a message to a subscription after that
subscription had been unsubscribed such that the dispatched message
arrived at the callback after the callback had received its final
message. In callbacks that cleaned up user data, this would often cause
a segfault. This has been resolved by locking the topic during the
entirety of dispatch. To prevent long publishing and topic locking
times, forwarding subscriptions have been made to be standard
subscriptions instead of mailboxless subscriptions which were
dispatched at publishing time.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383422 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoPass the sorcery instance to wizards for CUD operations as well as retrieve.
Joshua Colp [Wed, 20 Mar 2013 14:52:23 +0000 (14:52 +0000)]
Pass the sorcery instance to wizards for CUD operations as well as retrieve.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383405 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix lock destruction/unlock inversion
Kinsey Moore [Tue, 19 Mar 2013 19:07:46 +0000 (19:07 +0000)]
Fix lock destruction/unlock inversion

When using scoped locks, the unref of an AO2 object should happen after
the unlock occurs which requires usage of scoped refs.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMultiple revisions 383341-383342
David M. Lee [Tue, 19 Mar 2013 16:00:22 +0000 (16:00 +0000)]
Multiple revisions 383341-383342

........
  r383341 | dlee | 2013-03-19 10:57:29 -0500 (Tue, 19 Mar 2013) | 5 lines

  Removed codecs/g722/*.i on make clean
  ........

  Merged revisions 383340 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r383342 | dlee | 2013-03-19 10:58:33 -0500 (Tue, 19 Mar 2013) | 1 line

  Remove codecs/speex/*.i on make clean
........

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6 years agoMake sure things compile...
Kinsey Moore [Sat, 16 Mar 2013 16:00:40 +0000 (16:00 +0000)]
Make sure things compile...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoTransition MWI to Stasis-core
Kinsey Moore [Sat, 16 Mar 2013 15:45:58 +0000 (15:45 +0000)]
Transition MWI to Stasis-core

Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.

Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd support for using XMPP buddy state via device state.
Joshua Colp [Sat, 16 Mar 2013 15:40:31 +0000 (15:40 +0000)]
Add support for using XMPP buddy state via device state.

This change allows you to use XMPP buddy state in places where device state
can be used be used, such as dialplan hints. If at least one resource is
available the buddy is considered available. Now your phone can reflect
their IM status too!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix a bug where resources were not found due to hashing on the priority itself.
Joshua Colp [Sat, 16 Mar 2013 15:15:44 +0000 (15:15 +0000)]
Fix a bug where resources were not found due to hashing on the priority itself.
........

Merged revisions 383266 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoA simplistic router for stasis_message's.
David M. Lee [Fri, 15 Mar 2013 17:35:16 +0000 (17:35 +0000)]
A simplistic router for stasis_message's.

Often times, when subscribing to a topic, one wants to handle
different message types differently. While one could cascade if/else
statements through the subscription handler, it is much cleaner to
specify a different callback for each message type. The
stasis_message_router is here to help!

A stasis_message_router is constructed for a particular stasis_topic,
which is subscribes to. Call stasis_message_router_unsubscribe() to
cancel that subscription.

Once constructed, routes can be added using
stasis_message_router_add() (or stasis_message_router_set_default()
for any messages not handled by other routes). There may be only one
route per stasis_message_type. The route's callback is invoked just as
if it were a callback for a subscription; but it only gets called for
messages of the specified type.

(issue ASTERISK-20887)
Review: https://reviewboard.asterisk.org/r/2390/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383242 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoSample config file for stasis-core.
David M. Lee [Fri, 15 Mar 2013 16:42:05 +0000 (16:42 +0000)]
Sample config file for stasis-core.

(issue ASTERISK-20887)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383225 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoTake advantage of the fact that stasis_unsubscribe now returns NULL
Kinsey Moore [Fri, 15 Mar 2013 13:04:52 +0000 (13:04 +0000)]
Take advantage of the fact that stasis_unsubscribe now returns NULL

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383169 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMake stasis unsubscription functions return NULL
Kinsey Moore [Fri, 15 Mar 2013 12:58:23 +0000 (12:58 +0000)]
Make stasis unsubscription functions return NULL

Unsubscribing things in Asterisk seems to very commonly follow with
NULLing out the variable that was unsubscribed. This change makes that
a bit simpler.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383168 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agotcptls: Prevent unsupported options from being set
Kinsey Moore [Fri, 15 Mar 2013 12:53:03 +0000 (12:53 +0000)]
tcptls: Prevent unsupported options from being set

AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.

This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.

Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
........

Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383166 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383167 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoWhen a session timer expires during a T.38 call, re-invite with correct SDP
Matthew Jordan [Fri, 15 Mar 2013 01:38:53 +0000 (01:38 +0000)]
When a session timer expires during a T.38 call, re-invite with correct SDP

When a session timer expires during a dialog that has re-negotiated to T.38
and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the poor fax
session under weigh.

This patch corrects that by sending T.38 parameters if we are in the middle of
a T.38 session.

(closes issue ASTERISK-21232)
Reported by: Nitesh Bansal
patches:
  dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418)
........

Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383125 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix processing of call files when using KQueue on OS X
Matthew Jordan [Fri, 15 Mar 2013 01:24:23 +0000 (01:24 +0000)]
Fix processing of call files when using KQueue on OS X

In certain situations, call files are not processed when using KQueue with
pbx_spool. Asterisk was sending an invalid timeout value when the spool
directory is empty, causing the call to kevent to error immediately. This
can create a tight loop, increasing the CPU load on the system.

(closes issue ASTERISK-21176)
Reported by: Carlton O'Riley
patches:
  kqueue_osx.patch uploaded by coriley (License 6473)
........

Merged revisions 383120 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383121 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix whitespace in AST_EXT_LIB_CHECK macro.
Jason Parker [Thu, 14 Mar 2013 16:57:36 +0000 (16:57 +0000)]
Fix whitespace in AST_EXT_LIB_CHECK macro.
........

Merged revisions 383061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 383062 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAlways set the RTP instance data in the RTP engine
Matthew Jordan [Wed, 13 Mar 2013 14:39:54 +0000 (14:39 +0000)]
Always set the RTP instance data in the RTP engine

Not informing the RTP engine of the instance data creates shrapnel.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUpdate Doxygen
Andrew Latham [Tue, 12 Mar 2013 22:43:15 +0000 (22:43 +0000)]
Update Doxygen

Push some cleanups upstream before testing another ticket.

(issue ASTERISK-20259)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix Sorting Order For Parking Lots Stored In Static Realtime
Michael L. Young [Tue, 12 Mar 2013 21:19:39 +0000 (21:19 +0000)]
Fix Sorting Order For Parking Lots Stored In Static Realtime

When retrieving the parking lots from a MySQL database table, the current order
is "filename, cat_metric desc, var_metric asc, category".  If there are multiple
parking lots with the same cat_metric but different categories, everything is
being sorted on cat_metric first resulting in errors when loading the parking
lots.

This patch fixes the problem by sorting on the category field first, then the
cat_metric field.

(closes issue ASTERISK-21035)
Reported by: Alex Epshteyn
Patches:
  asterisk-21035-orderby.diff Michael L. Young (license 5026)
........

Merged revisions 382942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382943 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382954 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoUpdate Contributed Realtime Schema Files - IPv6 Addresses
Michael L. Young [Tue, 12 Mar 2013 20:41:42 +0000 (20:41 +0000)]
Update Contributed Realtime Schema Files - IPv6 Addresses

This commit updates some fields in the contributed realtime schema files to
handle IPv6 addresses.

(closes issue ASTERISK-21173)
Reported by: Torrey Searle
Patches:
  realtime_sql.patch Torrey Searle (license 5334)
  asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026)
........

Merged revisions 382939 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382940 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382941 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix a crash when res_xmpp is configured using a username without a domain.
Joshua Colp [Tue, 12 Mar 2013 20:07:10 +0000 (20:07 +0000)]
Fix a crash when res_xmpp is configured using a username without a domain.

(closes issue ASTERISK-21156)
Reported by: amsoft2001
........

Merged revisions 382923 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoSwitch to using external pjproject libraries.
Jason Parker [Tue, 12 Mar 2013 19:08:59 +0000 (19:08 +0000)]
Switch to using external pjproject libraries.

ICE/STUN/TURN support in res_rtp_asterisk is also now optional.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoInclude the Username field in SIP Registry events when Status is registered
Matthew Jordan [Tue, 12 Mar 2013 16:30:02 +0000 (16:30 +0000)]
Include the Username field in SIP Registry events when Status is registered

In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.

(issue ASTERISK-17888)

(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
  chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
........

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........

Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix core dump on CLI usage
Igor Goncharovskiy [Tue, 12 Mar 2013 08:55:14 +0000 (08:55 +0000)]
Fix core dump on CLI usage

Fix issue with 'unistim show info' CLI command when device connected not configured
........

Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoAdded an option to disallow music on hold
Kevin Harwell [Mon, 11 Mar 2013 15:22:02 +0000 (15:22 +0000)]
Added an option to disallow music on hold

Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event.  This essentially stops telling the peer
to start music on hold.

(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoconfbridge: Rename items for clarity and consistency.
Richard Mudgett [Sat, 9 Mar 2013 00:21:46 +0000 (00:21 +0000)]
confbridge: Rename items for clarity and consistency.

struct conference_bridge_user -> struct confbridge_user
struct conference_bridge -> struct confbridge_conference
struct conference_state -> struct confbridge_state

struct conference_bridge_user *conference_bridge_user -> struct confbridge_user *user
struct conference_bridge_user *cbu -> struct confbridge_user *user
struct conference_bridge *conference_bridge -> struct confbridge_conference *conference

The names are now generally shorter, consistently used, and don't conflict
with the struct names.

This patch handles the renaming part of the issue.

(issue ASTERISK-20776)
Reported by: rmudgett

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_sip: Update the via header when relaying SMS MESSAGE
Jonathan Rose [Fri, 8 Mar 2013 20:26:03 +0000 (20:26 +0000)]
chan_sip: Update the via header when relaying SMS MESSAGE

Prior to this change, certain conditions for sending the message would
result in an address of '(null)' being used in the via header of the
SIP message because a NULl value of pvt->ourip was used when initially
generating the via header. This is fixed by adding a call to build_via
when the address is set before sending the message.

(closes issue ASTERISK-21148)
Reported by: Zhi Cheng
Patches:
700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475)
........

Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11

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6 years agoStasis documentation updates.
David M. Lee [Fri, 8 Mar 2013 16:59:02 +0000 (16:59 +0000)]
Stasis documentation updates.

(issue ASTERISK-20887)
(issue ASTERISK-20959)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoEnsure dummy channels get a stasis topic.
David M. Lee [Fri, 8 Mar 2013 16:25:58 +0000 (16:25 +0000)]
Ensure dummy channels get a stasis topic.

Fixes test failure introduced in r382685.

(issue ASTERISK-20887)
(issue ASTERISK-20959)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd message dump capability to stasis cache layer
Kinsey Moore [Fri, 8 Mar 2013 16:00:14 +0000 (16:00 +0000)]
Add message dump capability to stasis cache layer

The cache dump mechanism allows the developer to retreive multiple
items of a given type (or of all types) from the cache residing in a
stasis caching topic in addition to the existing single-item cache
retreival mechanism.  This also adds to the caching unit tests to
ensure that the new cache dump mechanism is functioning properly.

Review: https://reviewboard.asterisk.org/r/2367/
(issue ASTERISK-21097)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoThis patch adds a new message bus API to Asterisk.
David M. Lee [Fri, 8 Mar 2013 15:15:13 +0000 (15:15 +0000)]
This patch adds a new message bus API to Asterisk.

For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.

This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:

 - Loosely coupled; new message types can be added in seperate modules.
 - Easy to use; publishing and subscribing are straightforward
   operations.

In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.

(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoRemove unused function
Matthew Jordan [Fri, 8 Mar 2013 04:11:12 +0000 (04:11 +0000)]
Remove unused function

After r382670, get_ip_and_port_from_sdp was no longer used.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoDon't reset the RTP address on a glare re-INVITE
Matthew Jordan [Fri, 8 Mar 2013 03:54:38 +0000 (03:54 +0000)]
Don't reset the RTP address on a glare re-INVITE

Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.

Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.

There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.

Note that the Asterisk Test Suite's connectedline test caught this error.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoChanging log level of "Not changing threadpool size" from notice to debug.
David M. Lee [Thu, 7 Mar 2013 21:55:28 +0000 (21:55 +0000)]
Changing log level of "Not changing threadpool size" from notice to debug.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoLoad sorcery modules earlier, so they can actually be used.
Jason Parker [Thu, 7 Mar 2013 21:14:18 +0000 (21:14 +0000)]
Load sorcery modules earlier, so they can actually be used.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoLet vm_mailbox_snapshot combine "Urgent" when no folder is specified
Matthew Jordan [Thu, 7 Mar 2013 19:14:46 +0000 (19:14 +0000)]
Let vm_mailbox_snapshot combine "Urgent" when no folder is specified

r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.

It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
........

Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix a memory leak in xmldoc
Kinsey Moore [Thu, 7 Mar 2013 16:48:19 +0000 (16:48 +0000)]
Fix a memory leak in xmldoc

Another instance of attribute retrieval not being freed properly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoResolve more memory leaks in xmldoc
Kinsey Moore [Thu, 7 Mar 2013 16:21:52 +0000 (16:21 +0000)]
Resolve more memory leaks in xmldoc

Many places that allocated to pull out an attribute are now freed
properly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd a 'secret' probation strictrtp mode to handle delayed changes in RTP source
Matthew Jordan [Thu, 7 Mar 2013 15:48:06 +0000 (15:48 +0000)]
Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source

Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.

This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.

Review: https://reviewboard.asterisk.org/r/2364

(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow

(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
........

Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix minor memory leak in xmldoc
Kinsey Moore [Thu, 7 Mar 2013 15:36:52 +0000 (15:36 +0000)]
Fix minor memory leak in xmldoc

Strings retrieved via ast_xml_get_text() must be freed with
ast_xml_free_text().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoEnsure that logmsgs are freed properly
Kinsey Moore [Thu, 7 Mar 2013 15:09:01 +0000 (15:09 +0000)]
Ensure that logmsgs are freed properly

Messages sent while the logger thread is shutting down will now have
their associated callid freed properly.
........

Merged revisions 382574 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix ref leak in threadpool.c
Kinsey Moore [Thu, 7 Mar 2013 00:05:16 +0000 (00:05 +0000)]
Fix ref leak in threadpool.c

If ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoResolve a ref leak in threadpool.c
Kinsey Moore [Wed, 6 Mar 2013 15:21:42 +0000 (15:21 +0000)]
Resolve a ref leak in threadpool.c

Ownership of the listener reference is not transferred because the
listener is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd RFC 3327 Path header support to chan_sip
Matthew Jordan [Tue, 5 Mar 2013 13:14:43 +0000 (13:14 +0000)]
Add RFC 3327 Path header support to chan_sip

This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.

A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.

Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/

(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
  path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
  oolong-path-support-trunk in team branch by oej (License 5267)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFix several unreleased mutex locks that cause problem with processing calls
Igor Goncharovskiy [Tue, 5 Mar 2013 03:53:44 +0000 (03:53 +0000)]
Fix several unreleased mutex locks that cause problem with processing calls
Reported by: Daniel Bohling
Tested by: Daniel Bohling

(Closes issue ASTERISK-21119)
........

Merged revisions 382409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 382410 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoFixup some bridge and format capabilities comments and whitespace.
Richard Mudgett [Mon, 4 Mar 2013 21:15:36 +0000 (21:15 +0000)]
Fixup some bridge and format capabilities comments and whitespace.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382392 65c4cc65-6c06-0410-ace0-fbb531ad65f3