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)
........
Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 17 Aug 2012 20:21:30 +0000 (20:21 +0000)]
Fix memory leak in XML documentation
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
........
Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 17 Aug 2012 15:51:06 +0000 (15:51 +0000)]
Add instrumentation to subsystem reloads
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)
........
Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Thu, 16 Aug 2012 22:50:12 +0000 (22:50 +0000)]
Handle integer over/under-flow in ast_parse_args
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/
........
Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 16 Aug 2012 16:16:04 +0000 (16:16 +0000)]
chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
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
........
Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Wed, 15 Aug 2012 23:19:09 +0000 (23:19 +0000)]
Fix bug where final queue member would not be removed from memory.
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
........
Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Wed, 15 Aug 2012 20:15:08 +0000 (20:15 +0000)]
Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction
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.
Michael L. Young [Wed, 15 Aug 2012 01:35:57 +0000 (01:35 +0000)]
Fix Segfault When Registering SIP Over WebSockets
The helper function, get_address_family_filter, in chan_sip for dns resolution
by address family was not recognizing the websockets transport and resulting in
a null pointer being sent to functions in netsock2, in an attempt to determine
if we are bound to ANY address ([::]) or not.
This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and
SIP_TRANSPORT_WSS which results in a sock address being set properly for use in
determining the address family.
(closes issue ASTERISK-20221)
Reported by: Sven Beisiegel
Tested by: Sven Beisiegel, James Mortensen
Patches:
asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026)
Kinsey Moore [Mon, 13 Aug 2012 20:04:15 +0000 (20:04 +0000)]
Add test instrumentation
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)
........
Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 10 Aug 2012 21:23:52 +0000 (21:23 +0000)]
Fix a couple of documentation problems in app_queue.c
* 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
........
Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Alexandr Anikin [Fri, 10 Aug 2012 16:46:38 +0000 (16:46 +0000)]
remove ALREADYGONE flag on ooh323 call data by ooh323_indicate
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
Kinsey Moore [Thu, 9 Aug 2012 17:39:52 +0000 (17:39 +0000)]
Correct documentation for the MeetMe x flag
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.
........
Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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)
........
Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Wed, 8 Aug 2012 20:29:16 +0000 (20:29 +0000)]
Do not define a cause that doesn't actually exist
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.
........
Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 8 Aug 2012 20:04:44 +0000 (20:04 +0000)]
Fix the analog dial *0 flash-hook of bridged peer feature.
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.
........
Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix error in the "IPorHost" section of a SIP dialstring.
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
Matthew Jordan [Tue, 31 Jul 2012 21:19:41 +0000 (21:19 +0000)]
Schedule pokes of registered SIP peers within a given timespan after SIP reload
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.
Kinsey Moore [Tue, 31 Jul 2012 19:57:09 +0000 (19:57 +0000)]
Clean up and ensure proper usage of alloca()
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)
........
Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Tue, 31 Jul 2012 15:31:57 +0000 (15:31 +0000)]
Help mitigate potential reinvite glare scenarios.
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
........
Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Wed, 25 Jul 2012 21:12:50 +0000 (21:12 +0000)]
res_agi: Add message indicating need for \n character in verbose message
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.
Kevin P. Fleming [Mon, 23 Jul 2012 14:51:21 +0000 (14:51 +0000)]
Free any datastores attached to dummy channels.
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)
........
Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 19 Jul 2012 22:01:32 +0000 (22:01 +0000)]
Fix compilation error when MALLOC_DEBUG is enabled
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
........
Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 19 Jul 2012 21:37:09 +0000 (21:37 +0000)]
Handle extremely out of order RFC 2833 DTMF
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
........
Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Wed, 18 Jul 2012 19:14:09 +0000 (19:14 +0000)]
Resolve severe memory leak in CEL logging modules.
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/
........
Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Wed, 18 Jul 2012 17:13:07 +0000 (17:13 +0000)]
Ensure that all ast_datastore_info structures are 'const'.
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.
........
Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Thu, 12 Jul 2012 18:55:17 +0000 (18:55 +0000)]
Prevent double uri_escaping in chan_sip when pedantic is enabled
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
........
Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Michael L. Young [Thu, 12 Jul 2012 14:25:45 +0000 (14:25 +0000)]
Correct Documentation For DEC Function
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)
........
Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Allow the REALTIME() function to report errors back to the caller.
Also, do more error checking on the arguments specified to the REALTIME()
function and clarify the documentation. While I was editing the file, a
few coding guidelines fixups, as well.
Jonathan Rose [Mon, 9 Jul 2012 14:43:49 +0000 (14:43 +0000)]
chan_sip: Fix small behavioral change accidentally introduced in r369750
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.
........
Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Fri, 6 Jul 2012 21:02:37 +0000 (21:02 +0000)]
chan_sip: Add case for FLASH control frames so that we don't display a warning.
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)
........
Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 6 Jul 2012 18:47:05 +0000 (18:47 +0000)]
Remove a superfluous and dangerous freeing of an SSL_CTX.
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
........
Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 6 Jul 2012 15:23:28 +0000 (15:23 +0000)]
Fix bridging thread leak.
The bridge thread was exiting but was never being
reaped using pthread_join(). This has been fixed now
by calling pthread_join() in ast_bridge_destroy().
(closes issue ASTERISK-19834)
Reported by Marcus Hunger
Kinsey Moore [Thu, 5 Jul 2012 19:12:33 +0000 (19:12 +0000)]
AST-2012-011: Resolve heap corruption issue with voicemail
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)
........
Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 5 Jul 2012 17:02:53 +0000 (17:02 +0000)]
Do not send a BYE when a provisional response arrives during a re-INVITE
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)
........
Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Tue, 3 Jul 2012 17:02:18 +0000 (17:02 +0000)]
More improvements to re-INVITEs timing out after a provisional response
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)
........
Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Tue, 3 Jul 2012 14:34:22 +0000 (14:34 +0000)]
Better handle re-INVITEs with provisional but no final repsonses
A previous attempt at fixing this issue had negative side effects related
to attended transfers which this patch should resolve. Many thanks to
Steve Davies for all of the good suggestions and testing.
(closes issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies, Terry Wilson
Review: https://reviewboard.asterisk.org/r/2009/
........
Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Wed, 27 Jun 2012 21:10:01 +0000 (21:10 +0000)]
AST-2012-010: Clean up after a reinvite that never gets a final response
The basic problem is that if a re-INVITE is sent by Asterisk and it receives a
provisional response, but no final response, then the dialog is never torn
down. In addition to leaking memory, this also leaks file descriptors and will
eventually lead to Asterisk no longer being able to process calls.
This patch just keeps track of whether there is an outstanding re-INVITE, and if
there is goes ahead and cleans up everything as though there was no outstanding
reinvite.
Review: https://reviewboard.asterisk.org/r/2009/
(closes issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies, Terry Wilson
........
Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 26 Jun 2012 13:22:42 +0000 (13:22 +0000)]
Fix crash in unloading of res_adsi module
When res_adsi is unloaded, it removes the ADSI functions that it previously installed
by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not
checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing
it to check whether or not the version of the functions matches what the core was
expecting it.
This patch makes it so that the version is only checked if a potentially valid adsi_funcs
pointer was passed in. Passing in NULL removes the installed functions, bypassing the
version check.
........
Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Mon, 25 Jun 2012 19:36:02 +0000 (19:36 +0000)]
Fix incorrect duration reporting in CDRs created in batch mode
Certain places in core/cdr.c would, if the duration value were 0, calculate the
duration as being the delta between the current time and the time at which the
CDR record was started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR records are
gathered and written long after those calls have ended. In particular, this
affects calls that were never answered, as those are expected to have a duration
of 0. Often, this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY".
Note that this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value. The affected
core backends include cdr_apative_odbc and cdr_custom; other extended or
deprecated CDR backends may potentially still directly manipulate the duration
values.
(issue ASTERISK-19860)
Reported by: Thomas Arimont
(issue AST-883)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Mark Michelson [Mon, 25 Jun 2012 19:16:52 +0000 (19:16 +0000)]
Re-fix how local tag is generated when sending a 481 to an INVITE.
Match our local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field.
(closes issue ASTERISK-19892)
reported by Walter Doekes
Richard Mudgett [Mon, 25 Jun 2012 15:59:28 +0000 (15:59 +0000)]
Fix Bridge application occasionally returning to the wrong location.
* Fix do_bridge_masquerade() getting the resume location from the zombie
channel. The code must not touch a clone channel after it has masqueraded
it. The clone channel has become a zombie and is starting to hangup.
Mark Michelson [Mon, 25 Jun 2012 14:23:16 +0000 (14:23 +0000)]
Be more consistent with the return code for requests received from invalid domain.
When Asterisk receives an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving
a REGISTER in this situation.
(Closes issue ASTERISK-19601)
Reported by Matthew Jordan
Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049)
........
Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Fri, 22 Jun 2012 19:34:59 +0000 (19:34 +0000)]
Don't crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed.
(closes issue ASTERISK-20040)
Reported by: Terry Wilson
........
Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 22 Jun 2012 17:23:26 +0000 (17:23 +0000)]
Don't parse media stream state for SIP video streams
The sendonly/recvonly/sendrecv/inactive media stream attributes were
parsed for video, but nothing was ever done with them. With this code
removed, an UNSUPPORTED message is produced when these attributes are
used in conjunction with a video stream which is the better behavior
since they were never really supported in the first place.
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Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Michael L. Young [Wed, 20 Jun 2012 02:04:58 +0000 (02:04 +0000)]
Fix NULL pointer segfault in ast_sockaddr_parse()
While working with ast_parse_arg() to perform a validity check, a segfault
occurred. The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in
config.h, "result pointer to the result. NULL is valid here, and can be used to
perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer. This patch also
adds documentation to netsock2.h about why it is necessary to check for a NULL
pointer.
(Closes issue ASTERISK-20006)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026)
Mark Michelson [Tue, 19 Jun 2012 15:37:37 +0000 (15:37 +0000)]
Fix request routing issue when outboundproxy is used.
Asterisk was incorrectly setting the destination of CANCELs
and ACKs for error responses to the URI of the initial INVITE.
This resulted in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead, when these CANCEL
or ACKs are to be sent, we should simply keep the destination the
same as what it previously was. There is no need to alter it any.
(closes issue ASTERISK-20008)
Reported by Marcus Hunger
Patches:
ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
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Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
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r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 14 Jun 2012 17:31:33 +0000 (17:31 +0000)]
AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling
AST-2012-008 (r367844) fixed a denial of service attack exploitable in the
Skinny channel driver that occurred when certain messages are sent after a
previously registered station sends an Off Hook message. Unresolved in that
patch is an issue in the Asterisk 10 releases, wherein, if a Station Key
Pad Button Message is processed after an Off Hook message, the channel driver
will inappropriately dereference a NULL pointer.
This patch fixes those places where the message handling or the channel
callback functions would attempt to dereference the line's pointer to the
device.
(issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Christoph Hebeisen
Patches:
AST-2012-009-10.diff uploaded by mjordan (license 6283)
Mark Michelson [Thu, 14 Jun 2012 15:25:23 +0000 (15:25 +0000)]
Revert Makefile change to remove embedding res_adsi.so
The change has resulted in a linking error for certain versions
of GCC. This is much worse than the original issue, so for now,
temporarily revert the change. A more thorough change will be
sought out.
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Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Wed, 13 Jun 2012 21:13:30 +0000 (21:13 +0000)]
Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.
With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.
(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)
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Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 13 Jun 2012 20:27:28 +0000 (20:27 +0000)]
Mark res_smdi/res_adsi as 'core' supported modules
Recently, various issues surrounding weak symbols have caused problems with
modules that rely on that feature to be enabled in menuselect. This includes
app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in menuselect.
Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this
patch marks both as 'core' supported modules. This will allow both
app_voicemail and chan_dahdi to be enabled as well, regardless of whether or
not that system supports weak symbols.
(issue AST-900)
Reported by: Thomas Arimont
(issue AST-885)
Reported by: Denis Alberto Martinez
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Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 13 Jun 2012 14:30:34 +0000 (14:30 +0000)]
Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a directory if that
directory does not exist. If some directory variable was empty, it would attempt
to create the empty location. It also failed to create the ASTLIBDIR directory.
This patch fixes it such that the correct directories are made and only created if
a value specifying them actually exists.
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Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 12 Jun 2012 18:30:06 +0000 (18:30 +0000)]
Do not perform install on existing directories
If a directory already exists, performing a 'make install' will remove the
permissions associated with the current directory and replace them with the
permissions of the user executing the install.
This patch changes this behavior to only perform an install on the directory
if the directory does not exist. Thus, if a user later changes the permissions
on that directory, those permissions will be preserved in subsequent installs.
Review: https://reviewboard.asterisk.org/r/1986
Review: https://reviewboard.asterisk.org/r/1864
(closes issue ASTERISK-19492)
Reported by: Karl Fife
Tested by: Paul Belanger, Tilghman Lesher
patches:
ASTERISK-19492 by pabelanger
(uploaded by mjordan)
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Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Tue, 12 Jun 2012 15:37:38 +0000 (15:37 +0000)]
Set the Caller ID "tag" on peers even if remote party information is present.
On incoming calls, we were setting the cid_tag on the dialog only if there was
no remote party information (Remote-Party-ID or P-Asserted-Identity) present.
The Caller ID tag is an invented parameter, though, and should be set no matter
the circumstance.
(closes issue ASTERISK-19859)
Reported by Thomas Arimont
(closes issue AST-884)
Reported by Trey Blancher
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Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 6 Jun 2012 21:32:09 +0000 (21:32 +0000)]
Fix POTS flash hook to orignate a second call deadlock.
A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.
* Release the channel and private locks when creating a new channel as a
result of a flash hook.
Mark Michelson [Wed, 6 Jun 2012 19:18:20 +0000 (19:18 +0000)]
Fix a specific scenario where ACKs are not matched.
If a dialog-starting INVITE contains a to-tag, then Asterisk
will respond with a 481. In this case, the resulting incoming
ACK would not be matched, so Asterisk would continue retransmitting
the 481 until the transaction times out.
There were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481, since there
was a to-tag in the INVITE, Asterisk would place this original to-tag
in the 481 response. When the ACK came in, Asterisk would attempt to
match the to-tag in the ACK to the generated local tag. Unfortunately,
Asterisk never actually transmitted a response with the generated local
tag, so the to-tag in the ACK would not match.
The other problem was that when the 481 was sent, nothing was set
on the sip_pvt to indicate what CSeq is expected in the ACK.
To fix the first problem, we zero out the to-tag seen in the incoming
INVITE. This way, Asterisk, when time to send a response, will send
its generated local tag instead.
To fix the second problem, we set the sip_pvt's pendinginvite to the
CSeq of the INVITE when we send a 481.
(closes issue ASTERISK-19892)
Reported by Mark Michelson
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Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 6 Jun 2012 17:21:20 +0000 (17:21 +0000)]
Add feature modifier to versions produced from branches
Certain branches, such as Certified Asterisk, may have a modifier added to
them that specifies the features available in that branch. For branches, this
modifier is expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of /certified/branches/1.8.11
would have a feature modifier of 'certified'. This is slightly different then
how features are determined for tags, where the feature is part of the actual
tag name, e.g., "10.5.0-digiumphones".
In keeping with the nomenclature used for tags, the feature specifier for
branches is translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
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Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
projects / thirdparty / asterisk.git / log
