Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.
(closes issue AST-958)
Reported-by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors,
PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors. If the connection
is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in
the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send
messages to PBX3. If it does, PBX2 will assume that PBX3 already received the
message and fail to forward the message on to PBX3 itself. This patch fixes
this by only including peers in a DPDISCOVER message that are reachable by the
sending node. This includes all peers with an empty address
(00:00:00:00:00:00) and that are have been reached by a qualify message.
This patch also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers.
The patch uploaded by Peter was modified slightly for this commit.
(closes issue ASTERISK-19309)
Reported by: Peter Racz
patches:
dundi_routing.patch uploaded by Peter Racz (license 6290)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters. This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the configuration
file.
Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256
characters. Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.
(closes issue ASTERISK-16879)
Reported by: Clod Patry
Tested by: mjordan
patches:
followme_no_limit.diff uploaded by Clod Patry (license #5138)
Slightly modified for this commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation incorrectly listed 'rtp' as a reloadable subsystem
and left out many other reloadable subsystems. It is now also
documented that subsystems may only be reloaded, not loaded or
unloaded.
(closes issue AST-977)
Reported-by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop. This function should
now generate correct context counts.
(closes issue AST-970)
Reported-by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two memory leaks:
1. When find_user is called with NULL as its first parameter, the voicemail
user returned is allocated on the heap. The inboxcount2 function uses
find_user in such a fashion when counting new messages, and fails to free
the resulting voicemail user object.
2. When populate_defaults is called on a voicemail user, it wipes whatever
flags have been set on the object by copying over the global flags object.
If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
that flag is removed. This leaks the voicemail user when free_user is later
called.
(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
asterisk.patch2 uploaded by Filip Jenicek (license 6277)
Patch slightly modified for this commit.
Review: https://reviewboard.asterisk.org/r/2096
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary;
1. reseting of hits=0, when no signal, only need to set it once.
2. incrementing of hits, when the hit is the same as the current hit.
3. setting of lasthit, when it's the same as before.
Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3
& 3 spelling mistakes
(closes issue ASTERISK-19610)
alecdavis (license 585)
Reported by: Jean-Philippe Lord
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2085/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In Asterisk 1.4+, a fix was put in place to increment the sequence number for
retransmitted DTMF end packets. With the introduction of the RTP engine API in
1.8, the sequence number was no longer being incremented. This patch fixes this
regression as well as cleans up a few lines that were not doing anything.
(closes issue ASTERISK-20295)
Reported by: Nitesh Bansal
Tested by: Michael L. Young
Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418)
asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2083/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
PQClear is not called when the result object of a call to PQExec has a
status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
handled properly, so this memory leak only occurred when CEL records were
successfully written.
This patch properly clears the result in the nominal code path.
(closes issue ASTERISK-19991)
Reported by: Etienne Lessard
Tested by: Etienne Lessard
patches:
mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When app_queue is unloaded, the queues container has its refcount
decremented, potentially to 0. Then the taskprocessor responsible
for handling device state changes is unreferenced. If the
taskprocessor happens to be just about to run its task, then it
will create and destroy an iterator on the queues container.
This can cause the refcount on the queues container to increase to
1 and then back to 0. Going back to 0 a second time results in
double frees.
This failure was seen periodically in the testsuite when Asterisk
would shut down.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queue member status would not always get updated properly when the member
was called, thus resulting in the member getting multiple calls. With this
change, we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call before
placing an outbound call.
(closes issue ASTERISK-16115)
reported by nik600
Patches:
app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As noted by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki. Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder. The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.
(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dummy channels usually aren't read from, but functions like SHELL and CURL
use autoservice on the channel.
(closes issue ASTERISK-20283)
Reported by: Gareth Palmer
Patches:
svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation made it sound as if the DTMF acknowledgment was needed
at the time the agent logs in, rather than when the agent is called. This
is likely a relic from the days when there were multiple ways of logging
in agents.
(closes issue AST-962)
reported by Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "Waiting" field was misdocumented as reporting the number of
messages waiting. In reality, it simply indicated the presence or
absence of waiting messages.
(closes issue AST-975)
reported by John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
(closes issue ASTERISK-20240)
Reported by: Egor Gorlin
Patches:
lock.c.patch uploaded by Egor Gorlin (license 6416)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(closes issue ASTERISK-20090)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes three main issues
* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.
* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.
* Fix some memory leaks that were spotted while taking
care of the first two points.
(Closes issue ASTERISK-20135)
reported by Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2071
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.
(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373)
Reported-by: Benjamin (bulkorok)
Reported-by: Rob Gagnon (rgagnon)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr,
extconfig, etc.
(issue PQ-1126)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds AMI events for module reloads when Asterisk is built with
TEST_FRAMEWORK enabled and corrects generation of the module load AMI
event.
(issue PQ-1126)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The strtol family of functions will return *_MIN/*_MAX on overflow. To
detect when an overflow has happened, errno must be set to 0 before
calling the function, then checked afterward.
(closes issue ASTERISK-20120)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2073/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously the pvt SIP_OUTGOING flag was used instead, which will frequently
flip during reinvites.
(closes issue AST-897)
Reported by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under certain conditions, a SIP transaction involving directmedia wouldn't
trigger a re-invite because the SDP answer was included in an ACK instead
of in a message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK.
(closes issue AST-913)
Reported by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a static queue had realtime members, then there could be a potential
for those realtime members not to be properly deleted from memory.
If the queue's members were loaded from realtime and then all the
members were deleted from the backend, then the queue would still
think these members existed. The reason was that there was a short-
circuit in code such that if there were no members found in the
backend, then the queue would not be updated to reflect this.
Note that this only affected static queues with realtime members.
Realtime queues with realtime members were unaffected by this issue.
(closes issue ASTERISK-19793)
reported by Marcus Haas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The other instance of this bug was fixed by jcolp/file in r121496. If
we are destroying a dialog only set the MWI dialog pointer on the
related peer to NULL if it is the dialog currently being destroyed.
(closes issue ASTERISK-20119)
Patch-by: Misha Vodsedalek
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds test instrumentation for loading and unloading of modules
and for certain actions in MeetMe to be used in the testsuite or any
other consumer of AMI events. These will only be generated when
Asterisk is built with TEST_FRAMEWORK enabled.
(issue PQ-1131)
(issue PQ-1133)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The RemoveQueueMember app made mention of options that could
be passed in, but no options are supported. I have removed the
listing of options from the documentation.
* The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
value that could be set.
(closes issue AST-949)
reported by Steve Pitts
(closes issue AST-954)
reported by Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
(closes issue ASTERISK-19308)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-19308.patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for the x flag for MeetMe incorrectly described its
function as closing down the conference when the last marked user left.
It actually causes the users with that flag to leave the conference
when the last marked user exits. The functionality of this flag is not
changing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel hangs up while being spied upon and the option to exit the
ChanSpy application when the spied on channel hangs up is set,
ast_autochan_destroy is not being called and therefore a reference to the spied
upon channel is not removed.
The symptom being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel was still
being shown while "core show channels" showed that the channel was not up.
This patch calls ast_autochan_destroy when a spied upon channel hangs up and
the option to exit the ChanSpy application is set, removing the reference to
the channel allowing the count for the group that the spied channel was part of
to be decremented.
(closes issue ASTERISK-17515)
Reported by: Arkadiusz Malka
Tested by: Alexandr Gordeev, Michael L. Young
Patches:
asterisk-17515-destroy-autochan.diff
uploaded by Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause
information. As such, it should not be defined and translatable as a
cause.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port. It now also
flash-hooks the correct channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
