asterisk/asterisk.git
7 years agoPickup: Pickup() and PickupChan() parameter parsing improvements.
Richard Mudgett [Thu, 14 Nov 2013 21:36:25 +0000 (21:36 +0000)]
Pickup: Pickup() and PickupChan() parameter parsing improvements.

* Made Pickup() and PickupChan() tollerate empty pickup values.  i.e., You
can now have Pickup(&&exten@context).

* Made PickupChan() use the standard option flag parsing code.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPickup: Ensure using PICKUPMARK never considers the picking channel.
Richard Mudgett [Thu, 14 Nov 2013 20:53:52 +0000 (20:53 +0000)]
Pickup: Ensure using PICKUPMARK never considers the picking channel.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402820 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoSay: If SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF
Jonathan Rose [Thu, 14 Nov 2013 20:32:45 +0000 (20:32 +0000)]
Say: If SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF

Similar to how background works, if a say application is called with
this variable set to 'true', 'yes', 'on', etc. then using DTMF while
the say action is in progress will result in the channel jumping to
that extension in the dialplan.

Review: https://reviewboard.asterisk.org/r/3011/

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7 years agores_ari_channels: Add the ability to stop locally generated ringing on a channel.
Joshua Colp [Wed, 13 Nov 2013 23:11:32 +0000 (23:11 +0000)]
res_ari_channels: Add the ability to stop locally generated ringing on a channel.

Using the 'ring' operation it is possible to start locally generated ringback if
the channel is answered. This change adds the ability to stop it by using DELETE.
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7 years agoari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Kevin Harwell [Tue, 12 Nov 2013 23:17:45 +0000 (23:17 +0000)]
ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404

Was returning a 404 on a valid technology with an empty list of endpoints.
Now checking against the channel tech to make sure the tech itself is valid
and not just an empty list of endpoints.

(issue ASTERISK-22803)
Reported by: David M. Lee
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7 years agoari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Kevin Harwell [Tue, 12 Nov 2013 22:17:28 +0000 (22:17 +0000)]
ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404

Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found.  Fixed so a "404 Not Found" will be returned
instead.

(closes issue ASTERISK-22803)
Reported by: David M. Lee
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7 years agoSwitch to a scoped lock to avoid missing unlocks in failure returns.
Mark Michelson [Tue, 12 Nov 2013 19:38:03 +0000 (19:38 +0000)]
Switch to a scoped lock to avoid missing unlocks in failure returns.
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7 years agoMove a NULL check to a place that makes more sense.
Mark Michelson [Tue, 12 Nov 2013 19:08:14 +0000 (19:08 +0000)]
Move a NULL check to a place that makes more sense.

Two variables were being checked for NULLity immediately
after being declared NULL. I moved the NULL check until
after the variables are allocated.

This allows for the "channelvars" option in manager.conf
to work as intended again.
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7 years agopjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences
Kevin Harwell [Tue, 12 Nov 2013 16:49:17 +0000 (16:49 +0000)]
pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences

Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to
crash because they were trying to dereference a NULL pointer.

In the case of res_pjsip_messaging it was attempting to "print" a contact
header that did not exist.  In fact contact headers should not be part of
a SIP MESSAGE, so the offending code was simply removed.

In the case of res_pjsip_header_funcs a null private channel tech was being
passed to the function and then later dereferenced.  Added null checks (and
error logging) to the read/write function handlers to guard against crashing.

(closes issue ASTERISK-22821)
Reported by: Anthony Messina
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7 years agoCELGenUserEvent: Fix error message from ast_json_pack
Kinsey Moore [Tue, 12 Nov 2013 16:34:31 +0000 (16:34 +0000)]
CELGenUserEvent: Fix error message from ast_json_pack

This prevents NULL from being passed into an ast_json_pack call when no
extra information is passed to the application which prevents an error
message about NULL arguments from being generated.
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7 years agoFixed a typ.
David M. Lee [Tue, 12 Nov 2013 15:27:00 +0000 (15:27 +0000)]
Fixed a typ.
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7 years agochan_dahdi: Fix crash during caller ID read
Kinsey Moore [Tue, 12 Nov 2013 15:03:18 +0000 (15:03 +0000)]
chan_dahdi: Fix crash during caller ID read

Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.

(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
    chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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7 years agoConfbridge: add test events for dynamic menus test
Jonathan Rose [Mon, 11 Nov 2013 20:28:38 +0000 (20:28 +0000)]
Confbridge: add test events for dynamic menus test

Adds a couple of test events for conference menu actions so that it's
easy to discern when those menu actions have been triggered.

(issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2999/

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7 years agoGet rid of some inaccurate comments.
Mark Michelson [Mon, 11 Nov 2013 19:31:40 +0000 (19:31 +0000)]
Get rid of some inaccurate comments.

I'm doing some unrelated work in app_confbridge and finding
these "invalid pin" comments to be annoying. Get out!
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7 years agoapp_queue: Honor penalty limits of 0
Kinsey Moore [Mon, 11 Nov 2013 15:37:03 +0000 (15:37 +0000)]
app_queue: Honor penalty limits of 0

In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.

(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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7 years agochan_sip: keep same local (from) tag for outgoing register requests
Scott Griepentrog [Fri, 8 Nov 2013 23:07:50 +0000 (23:07 +0000)]
chan_sip: keep same local (from) tag for outgoing register requests

For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal.  That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...".  This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.

(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
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7 years agores_stasis.c: Fix locking issues with the app_bridge_moh container.
Richard Mudgett [Fri, 8 Nov 2013 20:37:08 +0000 (20:37 +0000)]
res_stasis.c: Fix locking issues with the app_bridge_moh container.

* Fix unlinking from the app_bridges_moh container in remove_bridge_moh()
without a lock under normal circumstances.

* Made check ast_bridge_set_after_callback() return value in
bridge_moh_create() to handle failure.

* Fixed SCOPED_AO2LOCK() locking over too much scope in
stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop().

* Fixed unusual usage of ao2_unlink_flag() in control_unlink().

* Fixed orphaned bridge from off nominal path in
stasis_app_bridge_create().

* Fixed strange construct in stasis_app_unsubscribe().  From a bad merge?

* Made load_module() cleanup on failure.

Review: https://reviewboard.asterisk.org/r/2962/
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7 years agosecurity_events: Push out security events over AMI events
Jonathan Rose [Fri, 8 Nov 2013 19:33:48 +0000 (19:33 +0000)]
security_events: Push out security events over AMI events

Security Events will now be written to any listener of the new 'security' class

Review: https://reviewboard.asterisk.org/r/2998/
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7 years agoClarify an ambiguous error message.
Mark Michelson [Fri, 8 Nov 2013 19:22:53 +0000 (19:22 +0000)]
Clarify an ambiguous error message.
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7 years agores_pjsip: Print a helpful error message if sorcery registration fails
David M. Lee [Fri, 8 Nov 2013 18:53:14 +0000 (18:53 +0000)]
res_pjsip: Print a helpful error message if sorcery registration fails
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7 years agoChanges from make ari-stubs after r402560
David M. Lee [Fri, 8 Nov 2013 18:52:19 +0000 (18:52 +0000)]
Changes from make ari-stubs after r402560
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7 years agoARI playback: Rename ARI Playback to Playbacks
Kevin Harwell [Fri, 8 Nov 2013 17:59:16 +0000 (17:59 +0000)]
ARI playback: Rename ARI Playback to Playbacks

Before playback was the only non plural resource.  It has been renamed to
playbacks for consistency.

(closes issue ASTERISK-22737)
Reported by: Paul Belanger
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7 years agoari: Add application/x-www-form-urlencoded parameter support
David M. Lee [Fri, 8 Nov 2013 17:29:53 +0000 (17:29 +0000)]
ari: Add application/x-www-form-urlencoded parameter support

ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.

This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.

(closes issue ASTERISK-22743)
Review: https://reviewboard.asterisk.org/r/2986/
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7 years agoapp_dahdiras: Use waitpid instead of wait4.
Kevin Harwell [Fri, 8 Nov 2013 14:58:13 +0000 (14:58 +0000)]
app_dahdiras: Use waitpid instead of wait4.

Several places in the code were using wait4 while other places were using
waitpid.  This change makes all places use waitpid in order to make things
more consistent and since the 'rusage' object passed in/out of wait4 was
never used.

(closes issue ASTERISK-22557)
Reported by: YvesGael
Patches:
     asterisk-11.5.1-wait4.patch uploaded by hurdman (license 6537)

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7 years agoPJSIP: Improve error handling in digest authenticator
Jonathan Rose [Thu, 7 Nov 2013 23:42:31 +0000 (23:42 +0000)]
PJSIP: Improve error handling in digest authenticator

Previously, regardless of whether failure to authenticate was due to
lacking any authentication or actually failing authentication, the
Digest Authenticator would simply return that a challenge was still
needed. It will continue to do that when no authentication information
is in the received SIP digest, but when authentication information
is present and does not pass authentication, that will be treated as
an authentication error. This is to ensure that PJSIP will issue
security events indicated failed auths.
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7 years agoari: User better nicknames for ARI operations
David M. Lee [Thu, 7 Nov 2013 21:10:31 +0000 (21:10 +0000)]
ari: User better nicknames for ARI operations

While working on building client libraries from the Swagger API, I
noticed a problem with the nicknames.

    channel.deleteChannel()
    channel.answerChannel()
    channel.muteChannel()

Etc. We put the object name in the nickname (since we were generating C
code), but it makes OO generators redundant.

This patch makes the nicknames more OO friendly. This resulted in a lot
of name changing within the res_ari_*.so modules, but not much else.

There were a couple of other fixed I made in the process.

 * When reversible operations (POST /hold, POST /unhold) were made more
   RESTful (POST /hold, DELETE /unhold), the path for the second operation
   was left in the API declaration. This worked, but really the two
   operations should have been on the same API.
 * The POST /unmute operation had still not been REST-ified.

Review: https://reviewboard.asterisk.org/r/2940/
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7 years agoapp_queue: crash if first agent is "busy"
Kevin Harwell [Wed, 6 Nov 2013 21:58:17 +0000 (21:58 +0000)]
app_queue: crash if first agent is "busy"

If the first agent/member (via CLI "queue show") in a queue is "busy" (dnd,
circuit busy, etc...) and no agents answered then app_queue would crash.
This occurred because while the calling of agent(s) remained valid the channel
on "busy" agent would be set to NULL and then later dereferenced upon a second
"rna" function call.  The original intention of the code is to have only valid
"call attempt" objects (channels != NULL) checked while attempting to call
agent(s).  It does this by building a "call_next" list of valid "call attempt"
objects.  In the case of the "busy" agent subsequent builds of the valid "call
attempt" list would sometimes include (the case mentioned above) an invalid
"call attempt" object.

The fix was to make sure the "call attempt" list was appropriately built on
every iteration.  A NULL sanity check was also added at the original offending
spot of the crash just in case another one slipped by somehow.

(closes issue ASTERISK-22644)
Reported by: Marco Signorini
Review: https://reviewboard.asterisk.org/r/2983/
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7 years agochan_sip: Use AST_AF* defined constant when calling ast_get_ip
Matthew Jordan [Tue, 5 Nov 2013 21:17:30 +0000 (21:17 +0000)]
chan_sip: Use AST_AF* defined constant when calling ast_get_ip

While the structure passed to ast_get_ip should be set memset to 0, thus
initializing the ss_family member to 0, explicitly setting it to AST_AF_UNSPEC
is more portable.
........

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7 years agochan_iax2: Fix incorrect usage of ast_get_ip involving uninitialized struct
Matthew Jordan [Tue, 5 Nov 2013 21:14:42 +0000 (21:14 +0000)]
chan_iax2: Fix incorrect usage of ast_get_ip involving uninitialized struct

This started off as a fix for the failing IAX2 acl_call test in the Asterisk
Test Suite. When inspecting why that test was failing, it became clear that all
attempts to bind to any local loopback address was failing:

[Nov  2 15:56:28] VERBOSE[15787] chan_iax2.c:   == Binding IAX2 to address
                                 127.0.0.1:4569
[Nov  2 15:56:28] DEBUG[15787] netsock2.c: Splitting '127.0.0.1' into...
[Nov  2 15:56:28] DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''.
[Nov  2 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", "(null)",
                               ...): ai_family not supported
[Nov  2 15:56:28] WARNING[15787] acl.c: Unable to lookup '127.0.0.1'

While there's conceivably other ways for getaddrino to return EAI_FAMILY, the
most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not provided as the
desired family. The culprit was the call to ast_get_ip, defined in acl.h. This
function uses the family from the passed in addr object (which it will also
populate when it returns!) when it eventually calls getaddrinfo.

This patch fixes the use of ast_get_ip that were not specifying the family in
chan_iax2. This prevents uninitialized use of the structure, so that the
addresses resolve correctly.

Review: https://reviewboard.asterisk.org/r/2991
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7 years agonetsock2: Define AST_AF_* enum constants to their AF_* equivalents
Matthew Jordan [Tue, 5 Nov 2013 21:06:25 +0000 (21:06 +0000)]
netsock2: Define AST_AF_* enum constants to their AF_* equivalents

This patch explicitly defines AST_AF_* enum constants to their sys/socket.h
defined equivalents. It is certainly unclear why these constants actually have
to exist, given that netsock2.h includes sys/socket.h; however, since the code
base is already liberally sprinkled with the usage of AST_AF_* (as well as with
direct calls to AF_*), this will at least keep the semantics consistent between
their usage across systems.
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7 years agostasis_channels: Don't give preference to ANI info in channel snapshots
Matthew Jordan [Tue, 5 Nov 2013 20:59:39 +0000 (20:59 +0000)]
stasis_channels: Don't give preference to ANI info in channel snapshots

When publishing channel snapshots, we currently compute the caller ID name and
number by giving preference first to ani.{name|number}, then to
id.{name|number}. However, when a channel driver (such as chan_sip) updates the
caller ID, it typically only updates the caller ID stored in id.{name|number}.
This means that we are currently giving preference to stale information.

When looking at the rest of the code base, the only other place where we appear
to use this same logic is in app_amd. Everywhere else, we treat the party
information in ani as being separate to the party information in id.

This patch publishes only the caller ID name and number in the snapshot field
for caller_name and caller_num. Note that the information in ANI is still
available in caller_ani.

Review: https://reviewboard.asterisk.org/r/2992/
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7 years agochan_sip: notify dialog info ignores presentation indicator in callerid
Kevin Harwell [Mon, 4 Nov 2013 21:02:18 +0000 (21:02 +0000)]
chan_sip: notify dialog info ignores presentation indicator in callerid

The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring.  Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow.  If they are restricted then "anonymous" is used instead.

(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
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Merged revisions 402450 from http://svn.asterisk.org/svn/asterisk/branches/11
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7 years agovector: Uppercase API to follow C convention.
Richard Mudgett [Sat, 2 Nov 2013 04:30:49 +0000 (04:30 +0000)]
vector: Uppercase API to follow C convention.

C does not support templates like C++.
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7 years agovector: Update API to be more flexible.
Richard Mudgett [Sat, 2 Nov 2013 04:12:36 +0000 (04:12 +0000)]
vector: Update API to be more flexible.

Made the vector macro API be more like linked lists.
1) Added a name parameter to ast_vector() to name the vector struct.
2) Made the API take a pointer to the vector struct instead of the struct
itself.
3) Added an element cleanup macro/function parameter when removing an
element from the vector for ast_vector_remove_cmp_unordered() and
ast_vector_remove_elem_unordered().
4) Added ast_vector_get_addr() in case the vector element is not a simple
pointer.

* Converted an inline vector usage in stasis_message_router to use the
vector API.  It needed the API improvements so it could be converted.

* Fixed topic reference leak in router_dtor() when the
stasis_message_router is destroyed.

* Fixed deadlock potential in stasis_forward_all() and
stasis_forward_cancel().  Locking two topics at the same time requires
deadlock avoidance.

* Made internal_stasis_subscribe() tolerant of a NULL topic.

* Made stasis_message_router_add(),
stasis_message_router_add_cache_update(), stasis_message_router_remove(),
and stasis_message_router_remove_cache_update() tolerant of a NULL
message_type.

* Promoted a LOG_DEBUG message to LOG_ERROR as intended in
dispatch_message().

Review: https://reviewboard.asterisk.org/r/2903/
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7 years agoconfbridge: Separate user muting from system muting overrides.
Richard Mudgett [Sat, 2 Nov 2013 03:24:47 +0000 (03:24 +0000)]
confbridge: Separate user muting from system muting overrides.

The system overrides the user muting requests when MOH is playing or a
waitmarked user is waiting for a marked user to join.  System muting
overrides interfere with what the user may wish the muting to be when the
system override ends.

* User muting requests are now independent of the system muting overrides.
The effective muting is now the logical or of the user request and system
override.

* Added a Muted flag to the CLI "confbridge list <conference>" command.

* Added a Muted header to the AMI ConfbridgeList action ConfbridgeList
event.

(closes issue AST-1102)
Reported by: John Bigelow

Review: https://reviewboard.asterisk.org/r/2960/
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7 years agoconfig: Allow ConfBridge DTMF menus to have '#' as the first digit.
Richard Mudgett [Sat, 2 Nov 2013 01:15:11 +0000 (01:15 +0000)]
config: Allow ConfBridge DTMF menus to have '#' as the first digit.

ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:

DTMF-sequence = action,action...

Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.

* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.

(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
      jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)

Review: https://reviewboard.asterisk.org/r/2969/
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7 years agovoicemail: Simplify callback pointer declarations and add doxygen.
Richard Mudgett [Fri, 1 Nov 2013 23:20:54 +0000 (23:20 +0000)]
voicemail: Simplify callback pointer declarations and add doxygen.

* Typedefed and added doxegen for the voicemail callback functions.

* Simplified the prototypes for ast_install_vm_functions() and
ast_install_vm_test_functions() to use the new function typedefs.

* Simplified the voicemail callback function pointer variable declarations
to use the new function typedefs.
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7 years agoapp_confbridge: Make the CONFBRIDGE function be able to create dynamic menus
Jonathan Rose [Fri, 1 Nov 2013 22:48:14 +0000 (22:48 +0000)]
app_confbridge: Make the CONFBRIDGE function be able to create dynamic menus

Also adds the ability to clear all profile items and makes behavior more
consistent with documentation as when choosing whether to use CONFBRIDGE
datastore profiles or the application arguments to the confbridge application.

(closes issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2971/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoManager: Add equivalent AMI actions for the bridge CLI commands.
Scott Griepentrog [Fri, 1 Nov 2013 21:51:20 +0000 (21:51 +0000)]
Manager: Add equivalent AMI actions for the bridge CLI commands.

Adds the following AMI events, closely following their CLI counterparts:

BridgeDestroy
BridgeKick
BridgeTechnologyList
BridgeTechnologySuspend
BridgeTechnologyUnsuspend

BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one
channel off the bridge. When kicking a channel, specifying the bridge
also (optional) insures it is not removed from the wrong bridge.  The
BridgeTechnology events allow viewing and changing suspension status,
which affects only subsequent not active bridging.

(closes ASTERISK-22356)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2973/
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7 years agoari wiki docs: add notes about allowMultiple parameters.
David M. Lee [Fri, 1 Nov 2013 16:31:57 +0000 (16:31 +0000)]
ari wiki docs: add notes about allowMultiple parameters.

This patch adds a note to any parameter that has 'allowMultiple' set in
the Swagger documentation.

(closes issue ASTERISK-22704)
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7 years agores_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early...
Joshua Colp [Fri, 1 Nov 2013 14:38:21 +0000 (14:38 +0000)]
res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media.

The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).

Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.

(closes issue ASTERISK-22701)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2916/
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7 years agochan_sip: Fix RTCP port for SRFLX ICE candidates
Kinsey Moore [Fri, 1 Nov 2013 12:40:40 +0000 (12:40 +0000)]
chan_sip: Fix RTCP port for SRFLX ICE candidates

This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.

(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
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Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agores_ari_channels: Fix a deadlock when originating multiple channels close to eachother.
Joshua Colp [Fri, 1 Nov 2013 12:33:09 +0000 (12:33 +0000)]
res_ari_channels: Fix a deadlock when originating multiple channels close to eachother.

If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.

(closes issue ASTERISK-22768)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2979/
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7 years agores_stasis: Ensure the channel is always departed from the bridge when it leaves.
Joshua Colp [Fri, 1 Nov 2013 12:13:09 +0000 (12:13 +0000)]
res_stasis: Ensure the channel is always departed from the bridge when it leaves.

This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.

(closes issue ASTERISK-22703)
Reported by: John Bigelow

(closes issue ASTERISK-22634)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2965/
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7 years agoUpdate the conversion script from sip.conf to pjsip.conf
Mark Michelson [Thu, 31 Oct 2013 22:09:47 +0000 (22:09 +0000)]
Update the conversion script from sip.conf to pjsip.conf

(closes issue ASTERISK-22374)
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2846
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7 years agocore/loader: Don't call dlclose in a while loop
Matthew Jordan [Thu, 31 Oct 2013 16:06:14 +0000 (16:06 +0000)]
core/loader: Don't call dlclose in a while loop

For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.

The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
    precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
    module

This results in Asterisk sitting forever.

Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.

Review: https://reviewboard.asterisk.org/r/2970
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7 years agomedix_index: Display errors when library calls fail
Matthew Jordan [Thu, 31 Oct 2013 15:52:32 +0000 (15:52 +0000)]
medix_index: Display errors when library calls fail

Based on feedback from ipengineer in #asterisk, when the media indexer
cannot access a sound file on the system (or otherwise fails) Asterisk
displays a "Cannot frob file" error but fails to tell you why. This is
especially problematic as the media_indexer failing will rpevent Asterisk
from starting, as it is in the core.

We now display the errno error messages so folks can figure out what they've
done wrong.
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7 years agostasis: add functions embarrassingly missing from r400522
David M. Lee [Thu, 31 Oct 2013 14:45:03 +0000 (14:45 +0000)]
stasis: add functions embarrassingly missing from r400522

I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.
........

Merged revisions 402276 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402277 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agopjsip_messaging: Added debug for in dialog messaging
Kevin Harwell [Wed, 30 Oct 2013 17:54:26 +0000 (17:54 +0000)]
pjsip_messaging: Added debug for in dialog messaging

(issue ASTERISK-22777)
Reported by: Matt Jordan
........

Merged revisions 402265 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402266 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdates for 1.4.25 core sounds and 1.4.14 extra sounds, plus new en_GB language set
Rusty Newton [Tue, 29 Oct 2013 23:43:58 +0000 (23:43 +0000)]
Updates for 1.4.25 core sounds and 1.4.14 extra sounds, plus new en_GB language set

The new sound packages relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, ASTERISK-20782
Modified sounds/Makefile for the new sound versions and to account for the new en_GB language set.

(issue ASTERISK-22659)
(closes issue ASTERISK-22659)
(closes issue ASTERISK-22411)
(closes issue ASTERISK-22544)
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Merged revisions 402224 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 402225 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402226 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoRemove some spammy debug messages; improve clarity of others
Matthew Jordan [Tue, 29 Oct 2013 12:57:35 +0000 (12:57 +0000)]
Remove some spammy debug messages; improve clarity of others

Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:

* channel.c: When copying variables from a parent channel to a child channel,
  specify the channels involved. Do not log anything for a variable that is not
  inherited; the fact that it doesn't have an _ or __ already signifies that it
  won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
  to use these debug messages, and for each format that is registered (on
  startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
  For short tests in the Asterisk Test Suite, this should make finding the
  actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
  Often, description elements - which are not required - are not provided.
  This debug message adds no additional value, as it is not indicative of an
  error or helpful in debugging which element did not contain a 'blah' element
  as a child. If an element is supposed to contain a child element, then that
  XML tree should have failed validation in the first place.

Review: https://reviewboard.asterisk.org/r/2966/
........

Merged revisions 402150 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 402151 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402154 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoARI: Remove channels/{channelId}/dial
Kinsey Moore [Tue, 29 Oct 2013 12:51:57 +0000 (12:51 +0000)]
ARI: Remove channels/{channelId}/dial

This removes the /ari/channels/{channelId}/dial URI since it is
redundant, overly complex, is likely to become more externally complex
over time, and is too high-level compared with other ARI operations.
See the following for further information:
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html

(closes issue ASTERISK-22784)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2968/
........

Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agobridge_native_rtp: Ensure bridge is torn down
Kinsey Moore [Tue, 29 Oct 2013 12:30:21 +0000 (12:30 +0000)]
bridge_native_rtp: Ensure bridge is torn down

When a bridge transitions away from one tech to another, the tech going
away is provided a dummy bridge with no channels in it to tear down.
Currently this means that the teardown code exits prematurely and does
not tear anything down. This change tears down RTP bridging for the
channel provided in the leave bridge tech callback.

This also reverts the majority of r400403 since it is now redundant.

(closes issue ASTERISK-22628)
(closes issue ASTERISK-22676)
Reported by: John Bigelow
Reported by: Kevin Harwell
Tested by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2905/
Patches:
    native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)
........

Merged revisions 402148 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402149 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agores_ari_playback: Add missing 404 error response for GET and DELETE.
Joshua Colp [Tue, 29 Oct 2013 11:15:59 +0000 (11:15 +0000)]
res_ari_playback: Add missing 404 error response for GET and DELETE.

(closes issue ASTERISK-22722)
Reported by: Richard Mudgett
........

Merged revisions 402139 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402140 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoIgnore full docs
David M. Lee [Mon, 28 Oct 2013 22:10:16 +0000 (22:10 +0000)]
Ignore full docs
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Merged revisions 402127 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402130 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPut back several merge revisions that were lost in r402054
David M. Lee [Mon, 28 Oct 2013 22:09:49 +0000 (22:09 +0000)]
Put back several merge revisions that were lost in r402054

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402129 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPut back several merge revisions that were lost in r401962
David M. Lee [Mon, 28 Oct 2013 22:05:37 +0000 (22:05 +0000)]
Put back several merge revisions that were lost in r401962

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402128 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix UPGRADE.txt Due To Merging From Branch 11
Michael L. Young [Mon, 28 Oct 2013 15:08:00 +0000 (15:08 +0000)]
Fix UPGRADE.txt Due To Merging From Branch 11

When merging in the patch for ASTERISK-22728, the UPGRADE.txt file was changed
incorrectly.  That change should have gone into ASTERISK-11.txt.

This commit is to fix that.

Also, another comment in the UPGRADE-11.txt was missing and this commit adds
that as well.
........

Merged revisions 402115 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402117 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"
Michael L. Young [Mon, 28 Oct 2013 14:59:16 +0000 (14:59 +0000)]
chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"

While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing.  The display of
"N" used to mean NAT (i.e. yes).  The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear.  That was a great suggestion.

Therefore, this patch does the following:

* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.

* A column for the 'Comedia' setting has been added.  It too will display the
  setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.

* UPGRADE.txt has been updated to document this change.

(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
    asterisk-forcerport-display-clarification_v3.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2941
........

Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402112 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402113 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFilter out internal channels from dial message handling
Matthew Jordan [Sun, 27 Oct 2013 23:22:51 +0000 (23:22 +0000)]
Filter out internal channels from dial message handling

Surrogate channels would pop up from time to time in dial message handling.
This would cause a WARNING message to appear, indicating that the Surrogate
channel had no CDR. This patch filters out those channels that have the
internal implementation flag set, such that the WARNING message isn't
displayed.
........

Merged revisions 402090 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPrevent CDR backends from unregistering while billing data is in flight
Matthew Jordan [Sun, 27 Oct 2013 20:04:17 +0000 (20:04 +0000)]
Prevent CDR backends from unregistering while billing data is in flight

This patch makes it so that CDR backends cannot be unregistered while active
CDR records exist. This helps to prevent billing data from being lost during
restarts and shutdowns.

Review: https://reviewboard.asterisk.org/r/2880/
........

Merged revisions 402081 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate Alembic database scripts for external scripting and PostgreSQL, Oracle
Matthew Jordan [Sun, 27 Oct 2013 02:39:34 +0000 (02:39 +0000)]
Update Alembic database scripts for external scripting and PostgreSQL, Oracle

This patch does the following:
1) The env scripts have been updated to be tolerant of a NULL configuration
   file. This occurs when configuration is provided by an external script,
   such that the actual config.ini file is not used.
2) Enum types have all been given names. This is needed for PostgreSQL script
   generation.
3) The identifier meetme_confno_starttime_endtime is greater than 30
   characters, and hence invalid for Oracle databases. This has been truncated
   down to meetme_confno_start_end.
........

Merged revisions 400383 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402073 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_pjsip: Fix a crash when direct media is enabled and an ACK is received after...
Joshua Colp [Sat, 26 Oct 2013 12:56:08 +0000 (12:56 +0000)]
chan_pjsip: Fix a crash when direct media is enabled and an ACK is received after the channel is hung up.

(closes issue ASTERISK-22731)
Reported by: Kinsey Moore
........

Merged revisions 402064 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agores_stasis.c: Made use the ao2_container callback templates.
Richard Mudgett [Sat, 26 Oct 2013 00:36:31 +0000 (00:36 +0000)]
res_stasis.c: Made use the ao2_container callback templates.

* Made res_stasis.c use the OBJ_SEARCH_XXX defines.
........

Merged revisions 402055 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agortp_engine: fix rtp payloads copy and improve argument names
Scott Griepentrog [Sat, 26 Oct 2013 00:27:02 +0000 (00:27 +0000)]
rtp_engine: fix rtp payloads copy and improve argument names

In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order.  This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.

(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
........

Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Test shows rtpmap:119 being copied per this change, but is not in sip invite
........

Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402043 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402054 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agotaskprocessor: Made use pthread_equal() to compare thread ids.
Richard Mudgett [Fri, 25 Oct 2013 23:58:32 +0000 (23:58 +0000)]
taskprocessor: Made use pthread_equal() to compare thread ids.

* Removed another silly use of RAII_VAR().  RAII_VAR() and SCOPED_LOCK()
are not silver bullets that allow you to turn off your brain.
........

Merged revisions 402044 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402045 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoYou'd think that new files would be free of whitespace issues. But you would be...
Richard Mudgett [Fri, 25 Oct 2013 22:03:04 +0000 (22:03 +0000)]
You'd think that new files would be free of whitespace issues.  But you would be wrong.
........

Merged revisions 402003 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoARI: channel/bridge recording errors when invalid format specified
Jonathan Rose [Fri, 25 Oct 2013 22:01:43 +0000 (22:01 +0000)]
ARI: channel/bridge recording errors when invalid format specified

Asterisk will now issue 422 if recording is requested against channels
or bridges with an unknown format

(closes issue ASTERISK-22626)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2939/
........

Merged revisions 402001 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoARI recordings: Issue HTTP failures for recording requests with file conflicts
Jonathan Rose [Fri, 25 Oct 2013 21:28:32 +0000 (21:28 +0000)]
ARI recordings: Issue HTTP failures for recording requests with file conflicts

If a file already exists in the recordings directory with the same name as what
we would record, issue a 422 instead of relying on the internal failure and
issuing success.

(closes issue ASTERISK-22623)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2922/
........

Merged revisions 401973 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agopbx.c: fix confused match caller id that deleted exten still in hash
Scott Griepentrog [Fri, 25 Oct 2013 20:51:13 +0000 (20:51 +0000)]
pbx.c: fix confused match caller id that deleted exten still in hash

This fixes a bug where a zero length callerid match adjacent to a no
match callerid extension entry would be deleted together, which then
resulted in hashtable references to free'd memory.  A third state of
the matchcid value has been added to indicate match to any extension
which allows enforcing comparison of matchcid on/off without errors.

(closes issue AST-1235)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2930/
........

Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401960 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401961 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPJSIP: Add log messages when requests are received for non-existent endpoints
Jonathan Rose [Fri, 25 Oct 2013 17:41:38 +0000 (17:41 +0000)]
PJSIP: Add log messages when requests are received for non-existent endpoints

(closes issue ASTERISK-22552)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2934/
........

Merged revisions 401938 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPut clicompat-r2.patch back in
Jonathan Rose [Fri, 25 Oct 2013 17:32:17 +0000 (17:32 +0000)]
Put clicompat-r2.patch back in

We've figured out how to resolve the problems this was causing in 12/trunk,
so this can go back in now.

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    clicompat-r2.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 401914 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401935 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401936 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401937 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agorevert clicompat-r2.patch from r401704
Jonathan Rose [Fri, 25 Oct 2013 16:59:33 +0000 (16:59 +0000)]
revert clicompat-r2.patch from r401704

Patch caused the following build errors against testsuite
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244

(issue ASTERISK-22467)
Reported by: Corey Farrell
........

Merged revisions 401895 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401896 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401897 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agochan_sip: Allow a sip peer to accept both AVP and AVPF calls
Kevin Harwell [Fri, 25 Oct 2013 16:09:05 +0000 (16:09 +0000)]
chan_sip: Allow a sip peer to accept both AVP and AVPF calls

Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.

(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
     optional_avpf_trunk.patch uploaded by tsearle (license 5334)
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7 years agoBlocked revisions 401391
David M. Lee [Fri, 25 Oct 2013 13:50:57 +0000 (13:50 +0000)]
Blocked revisions 401391

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Blocked revisions 401379

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chan_dahdi: Fix unable to get index warning when transferring an analog call.

Transferring an analog call using flashhooks generated an unable to get
index WARNING message when the transfer is completed.

* Removed unnecessary analog subchannel shell games when transferring a
call using flashhooks.

Thanks to Tzafrir Cohen for mentioning this in a comment on issue
ASTERISK-22720.
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7 years agotest_json: Fix deprecation warnings
David M. Lee [Fri, 25 Oct 2013 13:49:20 +0000 (13:49 +0000)]
test_json: Fix deprecation warnings

After a series of upgrades over recent weeks, I've discovered that
test_json.c won't compile in dev mode any more for me.

One of gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
tempnam. Which, in general, is a good thing. But for test code that just
needs a temporary file, it's just annoying.

This patch replaces usage of tempname with mkstemp, avoiding the
deprecation warning. It also removes the temporary files when the test
is complete, which apparently we weren't doing before (oops).

Review: https://reviewboard.asterisk.org/r/2957/
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7 years agoLogging: Logging types ignored after specifying a verbose level
Kevin Harwell [Thu, 24 Oct 2013 21:06:14 +0000 (21:06 +0000)]
Logging: Logging types ignored after specifying a verbose level

If one specified a verbose level within a logging facility in
logger.conf then any component after it was ignored.  Fixed so
all values are correctly read.

(closes issue ASTERISK-22456)
Reported by: Kevin Harwell
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7 years agoThe Swagger 1.2 specification for type extension ended up being
David M. Lee [Thu, 24 Oct 2013 20:48:17 +0000 (20:48 +0000)]
The Swagger 1.2 specification for type extension ended up being
slightly different than my proposal. Instead of putting an 'extends'
field on the subtype, the base type has a 'subTypes' field, which is a
list of the subTypes. Given that its a messaging model and not an
object model, kinda makes sense.

This patch changes the events.json api-doc, and the python translators
to take the new format into account.

Other changes that are in Swagger 1.2 were not adopted, since the spec
is still in flux, and could change before it's finalized.

A summary of changes to the Swagger-1.2 spec can be found at
https://github.com/wordnik/swagger-core/wiki/1.2-transition.

(closes issue ASTERISK-22440)
Review: https://reviewboard.asterisk.org/r/2909/
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7 years agoutils: Fix memory leaks and missed unregistration of CLI commands on shutdown
Jonathan Rose [Thu, 24 Oct 2013 20:34:53 +0000 (20:34 +0000)]
utils: Fix memory leaks and missed unregistration of CLI commands  on shutdown

Final set of patches in a series of memory leak/cleanup patches by Corey Farrell

(closes issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
    main-utils-11.patch uploaded by coreyfarrell (license 5909)
    main-utils-12up.patch uploaded by coreyfarrell (license 5909)
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7 years agotest_linkedlists: Fix memory leak
Jonathan Rose [Thu, 24 Oct 2013 19:57:04 +0000 (19:57 +0000)]
test_linkedlists: Fix memory leak

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    test_linkedlists-1.8.patch uploaded by coreyfarrell (license 5909)
    test_linkedlists-11up.patch uploaded by coreyfarrell (license 5909)
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7 years agojitterbuf: Fix memory leak on jitter buffer reset
Jonathan Rose [Thu, 24 Oct 2013 19:42:21 +0000 (19:42 +0000)]
jitterbuf: Fix memory leak on jitter buffer reset

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    jitterbuf-jb_reset-leak-1.8.patch
    jitterbuf-jb_reset-leak-11up.patch
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7 years agoastobj2: Unregister debug CLI commands at exit
Jonathan Rose [Thu, 24 Oct 2013 19:31:23 +0000 (19:31 +0000)]
astobj2: Unregister debug CLI commands at exit

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell (license 5909)
    astobj2-clean-debug-cli-12up.patch uploaded by coreyfarrell (license 5909)
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7 years agoapp_voicemail: Memory Leaks against tests
Jonathan Rose [Thu, 24 Oct 2013 18:46:56 +0000 (18:46 +0000)]
app_voicemail: Memory Leaks against tests

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
    app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401745 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agomemory leaks: Memory leak cleanup patch by Corey Farrell (second set)
Jonathan Rose [Thu, 24 Oct 2013 17:00:27 +0000 (17:00 +0000)]
memory leaks: Memory leak cleanup patch by Corey Farrell (second set)

Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
    clicompat-r2.patch uploaded by coreyfarrell (license 5909)
    codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
    data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
    main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
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7 years agomemory leaks: Memory leak cleanup patch by Corey Farrell (first set)
Jonathan Rose [Wed, 23 Oct 2013 20:10:30 +0000 (20:10 +0000)]
memory leaks: Memory leak cleanup patch by Corey Farrell (first set)

(issue ASTERSIK-22467)
Reported by: Corey Farrell
Patches:
    chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909)
    cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909)
    func_math.patch uploaded by corefarrell (license 5909)
    main-test-cleanup.patch uploaded by coreyfarrell (license 5909)
    test_dlinklists.patch uploaded by coreyfarrell (license 5909)
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7 years agores_rtp_asterisk: Address jittery DTMF events in RTP streams
Jonathan Rose [Wed, 23 Oct 2013 17:56:44 +0000 (17:56 +0000)]
res_rtp_asterisk: Address jittery DTMF events in RTP streams

(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
    dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
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7 years agocdr_adaptive_odbc: Also apply a filter when the CDR value is empty.
Richard Mudgett [Wed, 23 Oct 2013 16:52:11 +0000 (16:52 +0000)]
cdr_adaptive_odbc: Also apply a filter when the CDR value is empty.

Extra CDR records are written if a filtered CDR value is empty because the
filter is not checked.

(closes issue ASTERISK-22272)
Reported by: Jordi Llull Chavarria
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7 years agoAdd a test suite event to indicate when the atxfer 3-way feature is detected
John Bigelow [Wed, 23 Oct 2013 16:48:39 +0000 (16:48 +0000)]
Add a test suite event to indicate when the atxfer 3-way feature is detected

This adds a test suite event that indicates to tests when the attended transfer
three-way call feature is detected.

Review: https://reviewboard.asterisk.org/r/2912/
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7 years agochan_mgcp: Properly handle malformed media lines
Kinsey Moore [Wed, 23 Oct 2013 15:23:58 +0000 (15:23 +0000)]
chan_mgcp: Properly handle malformed media lines

This corrects a situation in which a media line was not parsed properly
and resulted in a crash.

(closes issue ASTERISK-21190)
Reported by: adomjan
Patches:
    chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
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Merged revisions 401537 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401539 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agochan_sip: Fix an issue where an incompatible audio format may be added to SDP.
Joshua Colp [Wed, 23 Oct 2013 11:16:44 +0000 (11:16 +0000)]
chan_sip: Fix an issue where an incompatible audio format may be added to SDP.

If preferred codecs included any non-audio format the code would
mistakenly add the audio format, even if it was not a joint capability
with the remote side.

(closes issue ASTERISK-21131)
Reported by: nbougues
Patches:
patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470)
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7 years agochan_iax2: Fix Binding To Multiple Addresses Again
Michael L. Young [Wed, 23 Oct 2013 02:36:01 +0000 (02:36 +0000)]
chan_iax2: Fix Binding To Multiple Addresses Again

When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake.  This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.

(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
    asterisk-22741-fix-binding-multiple-addr.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2945/
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7 years agores_rtp_asterisk: Fix crash when RTCP is not available during SSRC change
Matthew Jordan [Tue, 22 Oct 2013 23:10:22 +0000 (23:10 +0000)]
res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change

In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.

(issue AST-1174)

(closes issue ASTERISK-22667)
Reported by: John Bigelow
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Merged revisions 401447 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agoapp_queue: Fix CLI "queue remove member" queue_log entry.
Richard Mudgett [Tue, 22 Oct 2013 19:04:53 +0000 (19:04 +0000)]
app_queue: Fix CLI "queue remove member" queue_log entry.

The queue_log entry resulting from CLI "queue remove member" when
log_membername_as_agent is enabled is wrong.  It always uses the interface
name instead of the member name in the queue_log entry.

* Get the queue member before removing it from the queue so the member
name is available for the queue_log entry.

(closes issue ASTERISK-21826)
Reported by: Oscar Esteve
Patches:
      fix_membername.diff (license #6505) patch uploaded by Oscar Esteve
         (modified to fix potential ref leak)
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7 years agoBridging: Fix orphaned bridge if neither of the joining channels can join.
Richard Mudgett [Tue, 22 Oct 2013 17:06:21 +0000 (17:06 +0000)]
Bridging: Fix orphaned bridge if neither of the joining channels can join.

The original issue noted that the bridge is orphaned when res_parking.so
is not loaded and a call uses the dial kK flags.

A similar issue happens when only one of the park flags is used.  In this
case you have the bridge with one or the other channel left in it.  The
channel and bridge will stay around until the channel hangs up.

* Fixed the initial bridge channel push failure to act as if the channel
were kicked out of the bridge.  The bridge then decides if it needs to be
dissolved.

(closes issue ASTERISK-22629)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2928/
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7 years agores_parking: Give parking timeout comebacktoorigin channel DTMF features.
Richard Mudgett [Tue, 22 Oct 2013 16:33:16 +0000 (16:33 +0000)]
res_parking: Give parking timeout comebacktoorigin channel DTMF features.

Parking timeouts did not set any DTMF features for the channel calling the
parker back.

* Added code to set the parkedcalltransfers, parkedcallreparking,
parkedcallhangup, and parkedcallrecording options appropriately for the
channels when a parking timeout occurs.  The recall channel DTMF options
are set using the BRIDGE_FEATURES channel variable to allow the other
timeout options to have the DTMF features available.

(closes issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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7 years agores_parking: Update XML documention for DTMF features after parking timeout.
Richard Mudgett [Tue, 22 Oct 2013 16:28:05 +0000 (16:28 +0000)]
res_parking: Update XML documention for DTMF features after parking timeout.

* Updated the XML documentation to indicate that the parkedcalltransfers,
parkedcallreparking, parkedcallhangup, and parkedcallrecording
configuration options also apply to parking timeouts.

(issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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7 years agoAdd an 'R' option to Dial which sends ringing until early media has been received.
Joshua Colp [Tue, 22 Oct 2013 15:17:56 +0000 (15:17 +0000)]
Add an 'R' option to Dial which sends ringing until early media has been received.

(closes issue ASTERISK-10487)
Reported by: Gaspar Zoltan
Patches:
10487.patch uploaded by n8ideas (license 6075)

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7 years agoRemove a noisy debug message from bridging code.
Mark Michelson [Mon, 21 Oct 2013 21:06:41 +0000 (21:06 +0000)]
Remove a noisy debug message from bridging code.

This particular debug message, during a stress test, was logged so
often that it appeared that there may be a memory leak in the logger
code. In actuality, there was no memory leak, but the logger thread
was having a hard time keeping up with the demands of the rest of the
system.

Since this debug message has no value at all, the best way to fix the
problem was to just remove the message.

(closes issue AST-1225)
reported by John Bigelow

Patches:
spammy_log.diff uploaded by Mark Michelson (License #5049)
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7 years agoSegfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev
Kevin Harwell [Mon, 21 Oct 2013 19:50:28 +0000 (19:50 +0000)]
Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev
isn't installed

Include the appropriate declarations when not using termcap, but term+curses
and [n]curses do not exist.

(closes issue ASTERISK-22351)
Reported by: A. Iglesias
Patches:
    issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674)
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7 years agoFixing r401281; the model name is Channel, with a capital C
David M. Lee [Mon, 21 Oct 2013 18:59:51 +0000 (18:59 +0000)]
Fixing r401281; the model name is Channel, with a capital C
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