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
This is based on the review request posted by Walter Doekes
(referenced lower in the commit message)
The main fix here is to treat the IPorHost portion of the dial
string as a temporary outbound proxy. This ensures requests
get sent to the proper location.
Due to the age of the request, some parts were no longer relevant.
For instance, the request moved outbound proxy parsing code into
a single method. This is done in a previous commit, so it was not
necessary to do again.
Also, the review request fixed some errors with regards to request
routing for CANCEL and ACK requests. This has also been fixed in
more recent commits.
(closes issue ASTERISK-19677)
reported by Walter Doekes
Review https://reviewboard.asterisk.org/r/1859
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets. These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.
This fix prevents this "packet storm" and schedules the pokes for a random
time. That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.
The committed patch has some very small modifications to the patch schmidts
wrote for the review.
(closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon
patches:
issue19154.patch license #6034 uploaded by schmidts
Review: https://reviewboard.asterisk.org/r/1652
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.
This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.
For those who are having some deja vu, that's because this
patch was originally committed to trunk since there is a
new configuration option added. After seeing a bug report
about audio being slow to set up on SIP calls, it became
apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will
have no effect unless the option is explicitly enabled.
(closes issue AST-896)
reported by Thomas Arimont
(closes issue ASTERISK-19857)
reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The while loop responsible for reading AGI messages from a fastAGI service
can end up looping indefinitely when an AGI script fails to indicate the end
of a message with a \n character. This patch adds an indication that we are
expecting a \n character to end the message to make it more clear to users
that this is necessary if they are receiving this warning over and over.
(issue ASTERISK-20061)
Reported by: Eike Kuiper
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision 370205 added the use of a datastore attached to a dummy channel to
resolve a memory leak, but ast_dummy_channel_destructor() in this branch did
not free datastores, resulting in a continued (but slightly smaller) memory
leak. This patch backports the change to free said datastores from the Asterisk
trunk.
(related to issue AST-916)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
To fix a memory leak in CEL, a channel datastore was introduced whose
destruction function pointer was pointed to the ast_free macro. Without
MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free.
With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a
different place then utils.h, and became undefined. This patch resolves this
by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
ast_free, which is defined to be free.
(issue AST-916)
Reported by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will,
if a packet arrives out of order, drop the packet. This is to prevent
duplicate ton generation in the Asterisk core. Since the RTP layer does not
buffer data itself, this is the only option the RTP layer currently has for
handling packets that arrive out of order.
For the most part, this doesn't matter. For a particular digit, so long as a
BEGIN packet arrives before the first END packet, the digit will be produced.
If subsequent BEGIN packets arrive interleaved with the ENDs, they will be
dropped; likewise, if the BEGIN or END packets themselves are out of order,
those packets are dropped but sufficient information is conveyed to the
Asterisk core to produce the appropriate digit.
For certain sequences of DTMF packets - most notably when, for a particular
digit, an END packet arrives before any BEGIN packet for that digit - this
is a real problem. When an END arrives before any BEGINs, the END packet is
dropped - but at the same time, it causes subsequent BEGIN packets for that
digit to be ignored. When the next in order END packet arrives, it too is
dropped - Asterisk believes that there was no initial BEGIN.
The solution this patch provides is to trust the END packet to convey the
information needed for the Asterisk core to produce the DTMF digit. If we
receive an END packet, and it:
* Has a timestamp greater then the last timestamp received from an END
packet
* Does not have the same sequence number as the last received sequence
number (and is thus not an END packet retransmission)
Then we send the END frame up to the Asterisk core. It contains enough
DTMF information for Asterisk to produce the digit.
On the other hand, if we receive a BEGIN or continuation packet that occurs
with a timestamp equal to or less then the last END timestamp, then we've
received something out of order - but we already have received enough
information to produce the digit. These packets are dropped.
Much thanks goes to Olle Johansson (oej) for providing the idea for this
solution.
Review: https://reviewboard.asterisk.org/r/2033/
(closes issue ASTERISK-18404)
Reported by: Stephane Chazelas
Tested by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.
The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.
(closes issue AST-916)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2053/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If pedantic mode is enabled, outbound invites will have double-escaped
contacts. This avoids setting an already-escaped string into a field
where it is expected to be unescaped.
(closes issue ASTERISK-20023)
Reported by: Walter Doekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for DEC in func_math.c was incorrect. Looks like a copy and
paste error.
(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
func_math.patch uploaded by Billy Chia (license 6381)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Correct documentation on labeliftrue and labeliffalse parameters of
GotoIf() and update several other locations that use the same syntax.
(closes issue ASTERISK-20007)
Patch-by: Leif Madsen
Reported-by: WIMPy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also
inadvertently changed the return value, which would likely make the indication
not be sent in audio. This fixes that while still removing the warning message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip channels can receive flash control frames when connected to analog
phones and possibly for other reasons. There really isn't a reason to warn when
these frames are received, we can safely ignore them.
Patches:
dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function.
The code being removed is superfluous because the SSL_CTX
structures for servers will be properly freed when ast_ssl_teardown
is called.
(closes issue ASTERISK-20074)
Reported by Trevor Helmsley
Patches:
ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
Testers:
Trevor Helmsley
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The heard and deleted arrays in the voicemail state structure were not
handled properly following the memory leak fix in r354890 and a fix for
an invalid free in r356797. This could result in accessing and writing
into freed memory. The allocation for these arrays has been reworked
to avoid the possibility of invalid frees, access of freed memory, and
crashes that were occurring as a result of this.
Locking around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added to
prevent simultaneous modification and access when IMAP storage is in
use. If IMAP storage is not in use, this locking is not compiled in.
Review: https://reviewboard.asterisk.org/r/1994/
(closes issue ASTERISK-19923)
Reported by: Dan Delaney
Tested by: Dan Delaney, Julian Yap
Patches:
vm_alloc_fix.diff uploaded by kmoore (license 6273)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Commits r369557 and r369579 were done to improve handling of re-INVITEs
when the UA that was supposed to receive the re-INVITE fails to respond.
A limitation of those patches occurred when a UA sent a provisional
response to the re-INVITE. This triggered a sending of a BYE in
check_pending. This patch tweaks the handling of the re-INVITE such that
a BYE is not sent in response to those messages.
(issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies
patches:
(reinvite_tweak.diff license #5012 by Steve Davies)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is no need to call check_pendings() on a final response to an INVITE
when destroying the scheduler entry as it will be done later during normal
processing.
(issue ASTERISK-19992)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
