Commit Graph

23905 Commits

Author SHA1 Message Date
Kinsey Moore 1223199b3d Remove a junk define
BLOB_HANDLER_BUCKETS is a remnant of using "type" fields in
JSON/snapshot blobs and is no longer used.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:01:30 +00:00
Matthew Jordan 06be8463b6 Migrate a large number of AMI events over to Stasis-Core
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
 * ChanSpyStart/Stop
 * MonitorStart/Stop
 * MusicOnHoldStart/Stop
 * FullyBooted/Reload
 * All Voicemail/MWI related events

In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.

Review: https://reviewboard.asterisk.org/r/2532

(closes issue ASTERISK-21462)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 20:44:07 +00:00
Matthew Jordan c1b51fd265 Print all logger messages on shutdown
When Asterisk shuts down and shuts down the loggin gsubsystem, any
messages currently in flight will not get logged. This patch prevents the
loop writing messages from breaking out prematurely, such that all of the
messages are logged.

(closes issue ASTERISK-21716)
Reported by: Corey Farrell
patches:
  logger-process-all-messages.patch uploaded by Corey Farrell (license 5909)
........

Merged revisions 389676 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 389677 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 11:57:48 +00:00
Igor Goncharovskiy 1fb6f365ec Fix several problems caused by multiple line usage with i2004 phones.
Reported by: Daniel Bohling, MihaiMircea

(closes issue ASTERISK-21061)
(closes issue ASTERISK-21120)
........

Merged revisions 389661 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 10:23:48 +00:00
David M. Lee 32a86f902a stasis-http: Provide a response body for 201 created responses
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 21:46:38 +00:00
Jonathan Rose 9c7c1897e5 res_parking: Add a verbose message when a channel is parked
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 21:11:18 +00:00
Jonathan Rose dd6923d0a9 res_parking: Fix some simple bugs
Both of them are covered in the dynamic parking review on
https://reviewboard.asterisk.org/r/2550 - Remove unref against
parking lot that the bridge did on dissolve since the reference
wasn't taken in the first place. On a swap, reapply bridge roles
in order to get music on hold and such playing on the channel that
swaps into the bridge.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 20:48:41 +00:00
Joshua Colp 814fa7fe11 Fix a crash due to the INVITE session being destroyed before the session.
This change ensures that the INVITE session remains valid for the lifetime
of the session object itself by increasing the session count on the dialog that
the INVITE session is allocated from. Once this reaches zero (normally as a result
of decrementing it within the session destructor) the dialog, and INVITE session,
are destroyed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 20:25:48 +00:00
David M. Lee 557125664d This patch adds support for controlling a playback operation from the
Asterisk REST interface.

This adds the /playback/{playbackId}/control resource, which may be
POSTed to to pause, unpause, reverse, forward or restart the media
playback.

Attempts to control a playback that is not currently playing will
either return a 404 Not Found (because the playback object no longer
exists) or a 409 Conflict (because the playback object is still in the
queue to be played).

This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource.

(closes issue ASTERISK-21587)
Review: https://reviewboard.asterisk.org/r/2559


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 20:21:16 +00:00
David M. Lee 10ba6bf8a8 This patch implements the REST API's for POST /channels/{channelId}/play
and GET /playback/{playbackId}.

This allows an external application to initiate playback of a sound on a
channel while the channel is in the Stasis application.

/play commands are issued asynchronously, and return immediately with
the URL of the associated /playback resource. Playback commands queue up,
playing in succession. The /playback resource shows the state of a
playback operation as enqueued, playing or complete. (Although the
operation will only be in the 'complete' state for a very short time,
since it is almost immediately freed up).

(closes issue ASTERISK-21283)
(closes issue ASTERISK-21586)
Review: https://reviewboard.asterisk.org/r/2531/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 20:11:35 +00:00
Richard Mudgett 3464e0919a Fix inverted test preventing DTMF disconnect from working.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 18:40:50 +00:00
Joshua Colp 61927f7c82 Fix a bug where the DTMF mode was not set on newly created RTP instances in the res_sip_sdp_rtp module.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 18:39:05 +00:00
Joshua Colp 7f43acc49c Fix a bug with applying the end result of the codec negotiation to the Asterisk channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 18:19:27 +00:00
Joshua Colp 9d742946db Fix a bug where the codec order as configured was not being obeyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 15:51:05 +00:00
David M. Lee 8a5a09e62c Fixed startup race condition which caused occasional stasis_mwi_state_type assertions.
The caching topic (which refers to the message type) was created before the
message type. If the initial subscription message gets processed before
the type can be initialized, the assertion about using an uninitialized type
fires.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-22 19:15:16 +00:00
Jason Parker 1355e5d34f Remove bad props, before anybody notices.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-22 18:20:53 +00:00
Jason Parker b6aac885be Add dial events to app_queue and app_followme.
Also fixes an issue in app_dial, where the channels were swapped on dial events.

(closes issue ASTERISK-21551)
(closes issue ASTERISK-21550)

Review: https://reviewboard.asterisk.org/r/2549/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-22 18:11:57 +00:00
David M. Lee 054efbc45a Fix destruction order assert for stasis_bridging
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 22:49:23 +00:00
Richard Mudgett 908ac3507a Conditional out more app_queue logging that needs to be reworked.
Fixes crash because app_queue was unconditionally freeing a datastore that
was still on a channel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 21:08:19 +00:00
Matthew Jordan afb1d96068 Raise the ConfBridgeMute/Unmute events when a CLI or AMI action triggers the change
New in 12 are the ConfBridgeMute/Unmute events, which are triggered when a user
changes their mute/unmute state. This was typically triggered when a user hit a
DTMF key that triggered the mute/unmute menu handler. Forgotten in this is when an
AMI action or CLI command triggers the mute/unmute. This patch now raises the
events in those situations as well.

(closes issue ASTERISK-21802)
Reported by: Birger "WIMPy" Harzenetter

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:45:57 +00:00
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.
........

Merged revisions 389244 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

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)
........

Merged revisions 388838 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

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/
........

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)
........

Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

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
........

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/
........

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