The ICalendar module had a systemic memory leak on each fetch of data from
the ICalendar source. The previous fetched data was not being properly
disposed. This patch makes it so that before each fetch of data, we dispose
of the previously fetched data.
(closes issue ASTERISK-21012)
Reported by: Joel Vandal
Tested by: Joel Vandal
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Sorcery is a unifying data access layer which provides a pluggable mechanism to allow
object creation, retrieval, updating, and deletion using different backends (or wizards).
This is a fancy way of saying "one interface to rule them all" where them is configuration,
realtime, and anything else that comes along.
Review: https://reviewboard.asterisk.org/r/2259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes the problem, but the issue includes a test which is still
being considered for the automated test suite.
(issue ASTERISK-20919)
Reported by: NITESH BANSAL
Patches:
patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418)
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This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
Add module tags to documentation for res_jabber/res_xmpp
Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
the XML documentation for each needs to call out which module is providing
the documentation. The module attribute has been added to the various XML
fragments for this purpose.
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r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all elements
Mea culpa.
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In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to
better account for out of order RTP packets. This was accomplished by using the
RTP timestamp and sequence number to check for out of order packets. However,
when a SSRC change occurs, the timestamp and sequence number will no longer
have any relation to the previously received packets. The variables tracking
the timestamp and sequence number therefore have to be reset.
(closes issue ASTERISK-20906)
Reported by: Eelco Brolman
patches:
dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442)
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Previously if an XMPP client reconnected any filters added by an external module were lost.
This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.
(closes issue ASTERISK-20916)
Reported by: kuj
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This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.
(closes issue ASTERISK-20499)
Reported by: Tootai
Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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On a fresh checkout of Asterisk 11, running make before ./configure
could cause the pjproject subdirectory to get in an odd state that
would prevent compilation. This patch by Tilghman prevents that from
occurring.
(closes issue ASTERISK-20681)
Reported by: Dinesh Ramjuttun
Tested by: danilo borges, Steve Lang
patches:
20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003)
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Similar to r378287, res_xmpp was marshaling data read from an external source
onto the stack. For a sufficiently large message, this could cause a stack
overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
removing the stack allocation, as it was unnecessary.
(issue ASTERISK-20658)
Reported by: wdoekes
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Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently the res_calendar_exchange module uses its own method of generating
UUIDs using ast_random(). Now that we have a UUID API we should use that
instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This provides a common API for dealing with unique identifiers.
The API provides methods to create, parse, copy, and stringify UUIDs.
An accompanying unit test is provided that tests all operations.
(closes issue ASTERISK-20726)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2217
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent memory leak fix in main/cli.c causes an ast_cli_entry's command
field to be freed and NULLed if ast_cli_register() fails. res_clialiases
was ignoring the return value of ast_cli_register() and was then passing
the NULL command off to a a hash function. This resulted in a crash.
The fix is not to ignore the erroneous return value. If ast_cli_register()
fails, then we do not continue trying to process the current alias.
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With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.
ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.
(closes issue ASTERISK-20643)
Reported by: coopvr
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its
functionality. This code changes the function names and prefixes "__" to the front. The
res_http_websocket exports file did not take this into account, thereby not allowing those functions
to be global and ultimately found.
(closes issue ASTERISK-20631)
Reported by: danjenkins
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Unlike all other calendar modules, res_calendar_ews fails to extract the Body
information for a calendar item. This is due, in part, to a quirk in the
schema in the XML - not only does a CalendarItem contain a Body element, but
the CalendarItem exists as a descendant of a different Body element. The neon
parser was erroneously skipping all Body elements.
This patch fixes that by bypassing Body elements that are not a child of
CalendarItem, and parsing the Body element out if it is a child.
Note that the original patch by Terry Wilson only needed slight modifications
to make it properly pull the Body information out; as such, while I've linked
to the patch that I uploaded for Dmitry, I've attributed the patch to Terry.
(closes issue ASTERISK-19738)
Reported by: Dmitry Burilov
Tested by: Dmitry Burilov
patches:
calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283)
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Remove the "RTP Read too short" warning for RTP keepalives. Remove the
the warning about the application delimiter switch from pipe to comma.
(You should've done this by now.) Make cdr_odbc report more when an
insert fails. Make chan_sip warn less when the peer wants SRTP (and we
don't) or sends a zero port to disable a media type.
Review: https://reviewboard.asterisk.org/r/2167
(closes issue ASTERISK-20538)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update and extend the configuration_file group and enable linking to the resource. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since there are a number of legacy devices out there that fail to handle ICE
candidates properly (which is a nice way of saying something much uglier),
disable it by default.
Support for ICE candidates can be enabled in rtp.conf using the icesupport
setting.
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pjproject, in order to solve build problems on Windows [1], undefines s_addr in
one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr
is not a structure member, but defined to map to the real strucuture member,
therefore when building on Solaris it's possible to get build errors like:
[CC] res_rtp_asterisk.c -> res_rtp_asterisk.o
In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
from res_rtp_asterisk.c:51:
/export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp':
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt':
res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules
res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules
res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice':
res_rtp_asterisk.c:1085: error: structure has no member named `s_addr'
make[2]: *** [res_rtp_asterisk.o] Error 1
make[1]: *** [res] Error 2
make[1]: Leaving directory `/export/home/admin/asterisk-11-svn'
gmake: *** [_cleantest_all] Error 2
Unfortunately, in order to make this work, I also had to make sure pjproject
only used the typdef pj_in_addr and not the struct pj_in_addr so that when
building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible
then that the library and users of those interfaces in Asterisk have a different
idea about the type of the argument, while on the surface it looks like they are
all 32 bit big endian values.
[1] http://trac.pjsip.org/repos/changeset/484
(issues ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang, mjordan
patches:
0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417)
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While XEP-0115 states that the node and ver attributes are both required, some
devices fail to provide either field. Prior to this patch, failure to provide
the node or ver attribute would cause a crash in res_xmpp. While failing to
provide the node or ver attribute is technically invalid, since this
information is not utilized by Asterisk except for reporting purposes, for
interoperability reasons, we continue to process the capability stanza anyways.
(closes issue ASTERISK-20495)
Reported by: Martin W
Tested by: Martin W
patches:
20495.patch uploaded by Martin W (license #6434)
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When using the channel technology agnostic application/AMI command MessageSend,
the "From" field is technically optional for the SIP channel driver. However,
if being sent by the XMPP resource module (either res_xmpp or res_jabber), the
"From" field is necessary, and must correspond to a defined account. This
patch updates the documentation for this application/AMI command to reflect
this.
(closes issue ASTERISK-20405)
Reported by: Leif Madsen
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The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.
This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).
* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
results in successful result
(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
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The res_jabber resource module uses the ASTOBJ library for managing its ref
counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object,
the pointer to the object has to be checked to see if the buddy existed.
Prior to this patch, the buddy object was not checked for NULL; with this patch
in both aji_client_info_handler and aji_dinfo_handler the pointer is checked
before used and, if no buddy object was found, the handlers return an error
code.
This patch does not take the approach that our JID can be used to log in from
another resource. If that approach is desired, an improvement could be made to
this patch to create the buddy on the fly. This patch seeks only to prevent
Asterisk from crashing.
FYI: In Asterisk 11+, you really should be using res_xmpp. It does not have
this problem, as it moved to the astobj2 library.
Note that multiple people have proposed patches for this issue; the patch being
committed here is based on those.
(closes issue ASTERISK-19532)
Reported by: Karsten Wemheuer
Tested by: Byron Clark
patches:
fix-jabber uploaded by Karsten Wemheuer (license #5930)
xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157)
(closes issue ASTERISK-19557)
Reported by: ulugutz
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Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 374336 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 374337 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added AMI event documentation for AsyncAGI and AGIExec events.
(closes issue ASTERISK-20318)
Reported by: Dan Cropp
Patches:
res_agi_patch.txt (license #6422) patch uploaded by Dan Cropp
modified for trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
