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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before playback was the only non plural resource. It has been renamed to
playbacks for consistency.
(closes issue ASTERISK-22737)
Reported by: Paul Belanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a note to any parameter that has 'allowMultiple' set in
the Swagger documentation.
(closes issue ASTERISK-22704)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Removed another silly use of RAII_VAR(). RAII_VAR() and SCOPED_LOCK()
are not silver bullets that allow you to turn off your brain.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
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Test shows rtpmap:119 being copied per this change, but is not in sip invite
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