The RTP/RTCP read error message can report "fail: success" when the
read failure is because of an ICE failure.
* Changed __rtp_recvfrom() to generate a PJ ICE message when ICE fails.
* Changed RTP/RTCP read error message to indicate an unspecified error
when errno is zero.
(closes issue ASTERISK-20288)
Reported by: Joern Krebs
Patches:
jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded by rmudgett (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous fix still would look in the static_RTP_PT table, which
is inappropriate since we specifically want to find a codec that has
been negotiated.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
Patches:
codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372311 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
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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/
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372199 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)
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The problem had to do with logic used when checking for what the oldest ringing channel
was. The problem was that if no channel was found, then no notification would be sent.
For custom device states, there is no associated channel, so no notification would get
sent. This fixes the issue by still sending the notification even if no associated
channel can be found for a ringing device state change.
(closes issue ASTERISK-20297)
Reported by Noah Engelberth
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A change for Asterisk 11 caused a check for failure to incorrectly check the return
value. This resulted in the possibility of transmitting media that a party had not
negotiated. If this media happened to be G.729, then this could potentially result
in one-way audio if no G.729 translators are installed.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372118 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.
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372050 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/11@372028 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
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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)
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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)
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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)
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pj_thread_register() takes a parameter of type pj_thread_desc.
It was assumed that pj_thread_register either used this item
temporarily or made a copy of it. Unfortunately, all it does is
keep a pointer to the structure in thread-local storage. This
means that if our pj_thread_desc goes out of scope, then pjlib
will be referencing bogus data quite often, most commonly on
operations involving a pj_mutex_t.
In our case, our pj_thread_desc was on the stack and went out
of scope very shortly after registering our thread with pjlib.
With this change, the pj_thread_desc is stored in thread-local
storage so the pointer that pjlib keeps in thread-local storage
will reference legitimate memory.
(closes issue ASTERISK-20237)
reported by Jeremy Pepper
Patches:
ASTERISK-20237.patch uploaded by Mark Michelson (license #5049)
Tested by Jeremy Pepper
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371571 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)
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The name of the "HangupCauseClear" application is "HangupCauseClear",
not "HangupcauseClear". The incorrect case of 'cause' caused the
XML documentation to not register properly.
As an aside, this commit message felt very awkward, but I'm not sure
how else to note that "X", which has to be "X", was referred to as "x".
(closes issue ASTERISK-20253)
Reported by: Andrew Latham
Patches:
hangupcause.diff uploaded by Andrew Latham (license #5985)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some core support modules and compiler options were no longer tagged with a
module support level. This patch adds 'core' back to those options.
Note that this patch modifies a few of the patches provided by Andrew Latham
slightly. res_curl and res_fax are both 'core' supported modules.
(closes issue ASTERISK-20215)
Reported by: Andrew Latham
Tested by: mjordan
Patches:
astcanary.diff (license #5985) uploaded by Andrew Latham
cflagsxml.diff (license #5985) uploaded by Andrew Latham
curl_fax.diff (license #5985) uploaded by Andrew Latham
soundsxml.diff (license #5985) uploaded by Andrew Latham
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371507 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
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While building up a new install to test chan_motif, I ran into a failure
due to icesupport being disabled. This was due to me not having an
rtp.conf. It was intended in the code for it to be enabled by default,
but it was only applied if rtp.conf existed.
This patch updates res_rtp_asterisk to be consistent in how it handles
defaults. A few options didn't have their default values set globally,
including icesupport. They are now set and icesupport is enabled by
default, even if you do not have an rtp.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@371425 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
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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
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