Commit graph

23885 commits

Author SHA1 Message Date
Richard Mudgett
3d63833bd6 Merge in the bridge_construction branch to make the system use the Bridging API.
Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:00:22 +00:00
David M. Lee
e1e1cc2dee Fixed some extra field assertion when the event WebSocket is connected
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 14:17:24 +00:00
Matthew Jordan
d8aec72494 Set the AST_CDR_FLAG_ORIGINATED flag on originated channel's CDRs
This may alleviate some of the CDR woes with originated channels, as CDRs
do like to know when a channel was originated. Eventually this will get
converted to be a channel flag, so its location is still good to know
post the great CDR shakeup of 2013.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 19:24:16 +00:00
Richard Mudgett
d8f40ed39a Fixup svn:keywords in all *.c and *.h files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 18:03:22 +00:00
Richard Mudgett
81171a4f4b Fixup svn:keywords in all *.c and *.h files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 17:53:24 +00:00
Jason Parker
a042955a24 Add doxygen.log to svn:ignore property.
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Merged revisions 389244 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 389245 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 17:44:41 +00:00
Kinsey Moore
e9a0ab42da Add missing exports file
This exposes stasis_app_control_answer and allows
res_stasis_http_channels to load properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 14:21:39 +00:00
Joshua Colp
734a154eef In Sorcery pass the name of the object being allocated to the allocator.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 14:02:37 +00:00
Kinsey Moore
eb06c505f4 Add documentation for record_file_append
When this option was added, it was noted in CHANGES, but was missing
the XML documentation that this patch adds.

(closes issue ASTERISK-21780)
Patch-by: Brad Latus (snuffy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-20 13:45:50 +00:00
Alexandr Anikin
62e64134fa add ast_publish_channel_state according new event framework
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19 20:52:34 +00:00
Damien Wedhorn
8d6b0c9f3b Add transfer softkey to ringout state to enable blond transfers.
(closes issue ASTERISK-21327)
Reported by: wedhorn
Tested by: myself
Patches: 
    skinny-blindxfer01.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19 19:45:14 +00:00
Kinsey Moore
1b5a3069f9 Add base XML documentation for res_sip
Thanks to Brad Latus, this patch adds a significant amount much-needed
documentation to res_sip. It should cover all existing configuration
options currently in Asterisk trunk.

Patch-by: Brad Latus (snuffy)
Review: https://reviewboard.asterisk.org/r/2471/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19 17:45:42 +00:00
Joshua Colp
b46840ae3e Don't hold the outgoing lock for a prolonged period of time as it may block the originator.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19 02:21:44 +00:00
Joshua Colp
4d8c35abf2 If the caller of the originate API calls wants the channel ensure it has been requested and dialed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19 00:49:15 +00:00
Damien Wedhorn
01d6e8dbc9 Add call forward no answer to skinny and cleanup general callfwd handling.
CallforwardNoAnswer uses a sched to determine when to forward the call. 
Defaults to 20secs but configurable in skinny.conf.

Adds dialType to each subchannel structure to be used to differentiate
between normal dials that result in a call being placed (default) and
other uses for the skinny_dialer (such as cfwd digit collection).
Restructured all cfwd handling to use this new arrangement.

(closes issue ASTERISK-21292)
Reported by: wedhorn
Tested by: myself
Patches: 
    skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18 23:20:53 +00:00
Joshua Colp
7316abeb8f Fix a bug where synchronous origination (oddly enough triggered by doing an async manager Originate) would not work properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18 22:49:14 +00:00
Joshua Colp
4e38a4eb64 Move origination to use the dialing API and send Stasis messages on dial begin and end.
(closes issue ASTERISK-21549)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2512/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18 19:47:24 +00:00
David M. Lee
b97c71bb11 Fix shutdown assertions in stasis-core
In r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either before
it was initialized or after it had been cleaned up. It turns out that
this assertion fires during shutdown.

This actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message types
used by the subscription could be freed before the final message of
the subscription was processed.

This patch adds stasis_subscription_join(), which blocks until the
last message has been processed by the subscription. Since joining was
most commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also added.

Similar functions were also added to the stasis_caching_topic and
stasis_message_router, since they wrap subscriptions and have similar
problems.

Other code in trunk was refactored to join() where appropriate, or at
least verify that the subscription was complete before being
destroyed.

Review: https://reviewboard.asterisk.org/r/2540


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 21:10:32 +00:00
Michael L. Young
91bab76422 Remove Character Limit On "inkeys" For IAX2
Currently, the buffer for processing "inkeys" is limited to 256 characters.  If
the user has many keys and the names of those key files are long, the 256
character limit is not enough.

* Change inkeys buffer to be dynamic

(closes issue ASTERISK-21398)
Reported by: Pavel Kopchyk
Tested by: Pavel Kopchyk, Michael L. Young
Patches:
    asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
					by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2501/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 20:24:56 +00:00
Matthew Jordan
d04f1fd60a Publish the outbound channel's application/data when dialing
This patch does two things:
* It fixes a bug where the outbound channel's application/data set by the
  dialing API/app_dial is not communicated until the channel is hung up.
  If that happens, AMI would incorrectly send a NewExten event immediately
  after a Hangup. This isn't really AMI's fault, as the dialing APIs never
  communicated the 'helpful' app/data on the outbound channel until it was
  hungup.
* It makes public sending a stasis message about a change in channel state.
  This is useful enough that - for now at least - it should be public. If
  operations on a channel go to being more coarse-grained, this function
  could be made private again.

Review: https://reviewboard.asterisk.org/r/2548

Note that this problem was found and reported by Matt DiMeo.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 17:43:58 +00:00
Jonathan Rose
b90bba7a30 Stasis: Update security events to use Stasis
Also moves ACL messages to the security topic and gets rid of the
ACL topic

(closes issue ASTERISK-21103)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2496/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 17:36:10 +00:00
David M. Lee
15945a7185 Fixed inverted logic in app_add_channel().
Also added some missing doc comments for stasis/app.h.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-15 21:13:29 +00:00
Kevin Harwell
2eebab3992 Fix for segfault in __ast_rwlock_destroy with DEBUG_THREADS
If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying
to access a possible NULL t->track object.  A NULL check has been added before
trying to access the memory.

(closes issue ASTERISK-21724)
Reported by: Corey Farrell
Fixed by: Corey Farrell
Patches:
	ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909)
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Merged revisions 388838 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388839 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-15 15:58:56 +00:00
Jason Parker
d8d1def22c Fix VM snapshot handling for combined INBOX.
The snapshot API contains an option that allow for combining of new 
and old messages within a single snapshot. New messages, however, 
include options beyond just 'INBOX' - it also includes the Urgent 
folder. A previous patch that combined INBOX and Urgent accidentally 
impacted snapshots that attempted to gain messages from just the Old 
folder. This patch fixes the snapshot gathering such that the API 
returns the appropriate messages for the folder selected, with and 
without the combine option.

This should make it more clear about what's happening.

Review: https://reviewboard.asterisk.org/r/2539/
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Merged revisions 388816 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-15 15:03:40 +00:00
Kinsey Moore
1ead1853f2 Use srtp_shutdown when available
This allows the SRTP library to be shut down properly when the
functionality is offered by libsrtp.

Review: https://reviewboard.asterisk.org/r/2538/
(closes issue ASTERISK-21719)
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Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388769 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-15 12:42:04 +00:00
David M. Lee
9648e258c7 Refactored the rest of the message types to use the STASIS_MESSAGE_TYPE_*
macros.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-15 02:37:22 +00:00
David M. Lee
e8f4ac6c61 Break res_stasis into smaller files.
When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.

This patch breaks the major components of res_stasis.c into individual
files.

 * res/stasis/app.c - Stasis application tracking
 * res/stasis/control.c - Channel control objects
 * res/stasis/command.c - Channel command object

This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.

The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.

Review: https://reviewboard.asterisk.org/r/2530/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-14 21:45:08 +00:00
Richard Mudgett
d1d1425327 Make ao2 global objects not always use the debug version of the ao2_ref() calls.
The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.

(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
      jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck
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Merged revisions 388700 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-14 19:03:26 +00:00
Kinsey Moore
2a9cbd693e Move JSON event generators into separate modules
This moves the JSON event generators out of the Stasis-HTTP modules and
into standalone JSON-related counterparts so that Stasis-HTTP and
res_stasis can depend on them without creating dependency cycles. This
also provides a future location for Swagger Model validator functions
once the generators for that code are written.

Review: https://reviewboard.asterisk.org/r/2534/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-14 12:47:52 +00:00
Michael L. Young
54424c2ee2 Fix Missing CALL-ID When Logging Through Syslog
The CALL-ID (ie [C-00000074]) is missing when logging to syslog.  This was just
an oversight when this feature was added.

* Add CALL-IDs when using syslog

(closes issue ASTERISK-21430)
Reported by: Nikola Ciprich
Tested by: Nikola Ciprich, Michael L. Young
Patches:
    asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2526/
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Merged revisions 388605 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 21:21:03 +00:00
Michael L. Young
4ff6e61808 Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix
The prior code committed, r385473, failed to take into consideration that not
all outgoing calls will be to a peer.  My fault.

This patch does the following:

* Check if there is a related peer involved.  If there is, check and set NAT 
  settings according to the peer's settings.

* Fix a problem with realtime peers.  If the global setting has auto_force_rport
  set and we issued a "sip reload" while a peer is still registered, the peer's
  flags for NAT are reset to off.  When this happens, we were always setting the
  contact address of the peer to that of the full contact info that we had.

(closes issue ASTERISK-21374)
Reported by: jmls
Tested by: Michael L. Young
Patches:
   asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2524/
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Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 21:07:02 +00:00
Kinsey Moore
5c33714520 Revert r388529 for now
Adding the cleanup function needs some deeper thought since it
apparently doesn't exist for all variants of libsrtp.
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Merged revisions 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388597 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 20:37:11 +00:00
Jonathan Rose
6a257dd534 pbx: Fix lack of cleanup on macrolock and context_table
(closes issue ASTERISK-21723)
Reported by: Corey Farrell
Patches:
    core-pbx-cleanup.patch uploaded by Correy Farrell (license 5909)
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Merged revisions 388532 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388578 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 19:29:56 +00:00
Kinsey Moore
3de63e9de9 Close libsrtp properly
Ensure that libsrtp is shutdown properly when res_srtp is unloaded.

(closes issue ASTERISK-21719)
Reported by: Corey Farrell
Patches:
    res_srtp-library-shutdown.patch uploaded by Corey Farrell
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Merged revisions 388529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388530 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 18:10:22 +00:00
Jonathan Rose
f2ba7c04fb chan_gulp: Minor readability Improvements to chan_gulp
(closes issue ASTERISK-21670)
Reported by: Snuffy
Review: https://reviewboard.asterisk.org/r/2473/
Patches:
    gulp-coding-guide.diff uploaded by snuffy (license 5024)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 17:20:33 +00:00
Richard Mudgett
681b50df0a Fix SendText AMI action to never return non-zero.
AMI actions must never return non-zero unless they intend to close the AMI
connection.  (Which is almost never.)

(closes issue ASTERISK-21779)
Reported by: Paul Goldbaum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13 14:28:50 +00:00
Richard Mudgett
785be439f9 Allow mISDN to send PROGRESS messsage.
* Made isdn_msg_parser.c build a progress message with the mandatory
progress indicator IE.  (The mISDNuser NT state machine rejected sending
the incomplete message.)

Note: The associated mISDN and mISDNuser patches respectively are viewable
here:
http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
http://svnview.digium.com/svn/thirdparty?view=rev&rev=201

(closes issue AST-1153)
Reported by: Guenther Kelleter
Patches:
      progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter
      progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter
      progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter
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Merged revisions 388426 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 22:12:52 +00:00
Mark Michelson
85fda4ca87 Fix memory leak in pbx_dundi
pbx_dundi added an io context without removing
it. This caused a memory leak when the module was
unloaded.

(closes ASTERISK-21718)
Reported by Corey Farrell
Patches:
	pbx_dundi-ast_io_remove.patch uploaded by Corey Farrell (License #5909)
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Merged revisions 388376 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 388378 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 20:50:53 +00:00
Michael L. Young
8f05ed46fe Fix Finding Extensions With Patterns Using ODBC Realtime
After the merge of support for the realtime sorcery module, extensions that
contained a pattern were not being found through odbc realtime.  It was tracked
down to this one line that was advancing to the next variable list before it
should have been.  The removal of this one line fixes this.

Tested this fix on my machine.

Received confirmation that this is the right fix from file on IRC.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 20:28:18 +00:00
David M. Lee
4666079b05 Address unload order issues for res_stasis* modules
I've noticed when doing a graceful shutdown that the res_stasis_http.so
module gets unloaded before the modules that use it, which causes some
asserts during their unload.

While r386928 was a quick hack to get it to not assert and die, this
patch increases the use counts on res_stasis.so and res_stasis_http.so
properly. It's a bigger change than I expected, hence the review instead
of just committing it.

Review: https://reviewboard.asterisk.org/r/2489/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 17:12:57 +00:00
David M. Lee
db925c3f06 Avoided __ast names for the private variables created by the
STASIS_MESSAGE_TYPE_*() macros.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 15:55:42 +00:00
Kinsey Moore
7ce05bfb9b Add channel events for res_stasis apps
This change adds a framework in res_stasis for handling events from
channel topics. JSON event generation and validation code is created
from event documentation in rest-api/api-docs/events.json to assist in
JSON event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications.

The userevent application has been refactored along with the code that
handles userevent channel blob events to pass the headers as key/value
pairs in the JSON blob. As a side-effect, app_userevent now handles
duplicate keys by overwriting the previous value.

Review: https://reviewboard.asterisk.org/r/2428/
(closes issue ASTERISK-21180)
Patch-By: Kinsey Moore <kmoore@digium.com>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 13:13:06 +00:00
Sean Bright
2cfedc12ad Fix copy/paste error in one-touch-recording implementation.
........

Merged revisions 388253 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10 11:47:04 +00:00
Matthew Jordan
c81ff6102f Don't expect to pack three tuples when you only have two
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-09 14:41:38 +00:00
Michael L. Young
f758885546 Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit
When we send out a CN packet (for instance, in the case of using rtpkeepalives),
we are not setting the payload code properly.  Also, we are setting the marker
bit when we shouldn't be according to RFC 3389, section 4.

AST_RTP_CN is not defined by AST_FORMAT codes.  Therefore, we should be using
ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup().

11 and trunk already use the appropriate function.

* In 1.8, use ast_rtp_codecs_payload_code()

* Remove the setting of the marker bit

* Fix the debug message by incrementing the seqno after the debug message is set
  in order to display the correct seqno that was sent out

(closes issue ASTERISK-21246)
Reported by: Peter Katzmann
Tested by: Peter Katzmann, Michael L. Young
Patches:
    asterisk-21246-rtp-cng-payload-error_1.8_v2.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2500/
........

Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 388112 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-09 04:11:59 +00:00
Michael L. Young
bb52414990 Fix Segfault In app_queue When "persistentmembers" Is Enabled And Using Realtime
When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse".  We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries.  When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members.  This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.

The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.

(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
    asterisk-21738-rt-ringinuse-field-not-set.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2499/
........

Merged revisions 388108 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-09 03:35:25 +00:00
David M. Lee
180db9c41f Fixed MODFLAG for res_stasis_websocket
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 22:00:53 +00:00
David M. Lee
b007c744a4 Add development flag to disable the inline API.
A GCC bug[1] can, in some cases, pop up an unsuppressible pedwarn when
using a static inline standard library function from a non-static
inline function.

This normally doesn't show up, but can occur if you're running an
upgrade version of GCC (such as GCC 4.8 on OS X, which normally runs
GCC 4.2).

 [1]: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 21:01:33 +00:00
David M. Lee
ec53d7fa87 Removed #if checks for crazy old versions of OS X.
The <arpa/nameser_compat.h> was introduced way back in OS X Panther, which
itself was end-of-lifed back in 2007. We can assume that any OS X machine
we build on will need that header file :-)

Why bother removing it? The flag we're checking (__APPLE_CC__) is actually
Apple's build number. Self-compiled versions of GCC (such as installing the
latest version of GCC from homebrew) sets the value to 0, making it useless
for this sort of compile flaggery.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 20:25:28 +00:00
David M. Lee
07e2eb71e9 Fixed set-but-not-used warning caught by newer GCC
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 19:00:55 +00:00