Richard Mudgett [Wed, 5 Aug 2015 19:21:50 +0000 (14:21 -0500)]
chan_dahdi.c: Lock private struct for ast_write().
There is a window of opportunity for DTMF to not go out if an audio frame
is in the process of being written to DAHDI while another thread starts
sending DTMF. The thread sending the audio frame could be past the
currently dialing check before being preempted by another thread starting
a DTMF generation request. When the thread sending the audio frame
resumes it will then cause DAHDI to stop the DTMF tone generation. The
result is no DTMF goes out.
* Made dahdi_write() lock the private struct before writing to the DAHDI
file descriptor.
Joshua Colp [Wed, 5 Aug 2015 10:23:21 +0000 (07:23 -0300)]
res_rtp_asterisk: Don't leak temporary key when enabling PFS.
A change recently went in which enabled perfect forward secrecy for
DTLS in res_rtp_asterisk. This was accomplished two different ways
depending on the availability of a feature in OpenSSL. The fallback
method created a temporary instance of a key but did not free it.
This change fixes that.
Mark Duncan [Tue, 28 Jul 2015 10:33:39 +0000 (19:33 +0900)]
res/res_rtp_asterisk: Add ECDH support
This will add ECDH support to Asterisk. It will
detect auto ECDH support in OpenSSL
(1.0.2b and above) during ./configure. If this is
available, it will use it,
otherwise it will fall back to prime256v1 (this
behavior is consistent with
other projects such as Apache and nginx).
This fixes WebRTC being broken in Firefox 38+ due
to Firefox now only supporting
ciphers with perfect forward secrecy.
Mark Michelson [Wed, 29 Jul 2015 19:35:58 +0000 (14:35 -0500)]
res_http_websocket: Properly encode 64 bit payload
A test agent was continuously failing all ARI tests when run against
Asterisk 13. As it turns out, the reason for this is that on those test
runs, for some reason we decided to use the super extended 64 bit
payload length for websocket text frames instead of the extended 16 bit
payload length. For 64-bit payloads, the expected byte order over the
network is
7, 6, 5, 4, 3, 2, 1, 0
However, we were sending the payload as
3, 2, 1, 0, 7, 6, 5, 4
This meant that we were saying to expect an absolutely MASSIVE payload
to arrive. Since we did not follow through on this expected payload
size, the client would sit patiently waiting for the rest of the payload
to arrive until the test would time out.
With this change, we use the htobe64() function instead of htonl() so
that a 64-bit byte-swap is performed instead of a 32 bit byte-swap.
Mark Michelson [Thu, 23 Jul 2015 18:11:42 +0000 (13:11 -0500)]
Local channels: Alternate solution to ringback problem.
Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a
specific scenario involving local channels and a native local RTP bridge
could result in ringback still being heard on a calling channel even
after the call is bridged.
That commit caused many tests in the testsuite to fail with alarming
consequences, such as not sending DialBegin and DialEnd events, and
giving incorrect hangup causes during calls.
This commit reverts the previous commit and implements and alternate
solution. This new solution involves only passing AST_CONTROL_RINGING
frames across local channels if the local channel is in AST_STATE_RING.
Otherwise, the frame does not traverse the local channels. By doing
this, we can ensure that a playtones generator does not get started on
the calling channel but rather is started on the local channel on which
the ringing frame was initially indicated.
Mark Michelson [Wed, 22 Jul 2015 14:46:29 +0000 (09:46 -0500)]
Local channels: Do not block control -1 payloads.
Control frames with a -1 payload are used as a special signal to stop
playtones generators on channels. This indication is sent both by
app_dial as well as by ast_answer() when a call is answered in case any
tones were being generated on a calling channel.
This control frame type was made to stop traversing local channel pairs
as an optimization, because it was thought that it was unnecessary to
send these indications, and allowing such unnecessary control frames to
traverse the local channels would cause the local channels to optimize
away less quickly.
As it turns out, through some special magic dialplan code, it is
possible to have a tones being played on a non-local channel, and it is
important for the local channel to convey that the tones should be
stopped. The result of having tones continue to be played on the
non-local channel is that the tones play even once the channel has been
bridged. By not blocking the -1 control frame type, we can ensure that
this situation does not happen.
audiohook: Read the correct number of samples based on audiohook format.
Due to changes in audiohooks to support different sample rates the
underlying storage of samples is in the format of the audiohook
itself and not of the format being requested. This means that if a
channel is using G722 the samples stored will be at 16kHz. If
something subsequently reads from the audiohook at a format which
is not the same sample rate as the audiohook the number of samples
needs to be adjusted.
Given the following example:
1. Channel writing into audiohook at 16kHz (as it is using G722).
2. Chanspy reading from audiohook at 8kHz.
The original code would read 160 samples from the audiohook for
each 20ms of audio. This is incorrect. Since the audio in the
audiohook is at 16kHz the actual number needing to be read is 320.
Failure to read this much would cause the audiohook to reset
itself constantly as the buffer became full.
This change adjusts the requested number of samples by determining
the duration of audio requested and then calculating how many
samples that would be in the audiohook format.
Rusty Newton [Mon, 20 Jul 2015 22:16:36 +0000 (17:16 -0500)]
Documentation: chan_sip doesn't support WS or WSS in outbound register.
* In sip.conf.sample fix sentence where we said that WS or WSS are supported
transports for use in an outbound register definition. They are not
supported in that case.
Matt Jordan [Sat, 27 Jun 2015 22:53:37 +0000 (17:53 -0500)]
tests/test_devicestate: Add additional tests for the device state API
This patch adds more tests that exercise the device state API. This includes:
* Tests that cover adding a device state provider, as well as deleting a
device state provider. This also verifies that you cannot add an
already added device state provider, and cannot delete an already
deleted device state provider.
* A test that covers changing device state and receiving said updates
from a device state subscriber. This also covers hitting both the
device state cache as well as a custom device state provider.
* A test that covers converting device state to channel state and device
state values to a string representation and back.
* A test that covers obtaining device state from an active channel and a
channel driver that provides its own device state.
Matt Jordan [Sat, 27 Jun 2015 22:51:43 +0000 (17:51 -0500)]
main/devicestate: Prevent duplicate registration of device state providers
Currently, the device state provider API will allow you to register a
device state provider with the same case insensitive name more than
once. This could cause strange issues, as the duplicate device state
providers will not be queried when a device's state has to be polled.
This patch updates the API such that a device state provider with the
same name as one that has already registered will be rejected.
res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.
This change moves logic for setting up the DTLS SSL contexts to
when the SDP is done being processed instead of when ICE negotiation
completes. It also stops handshakes from being initiated when we
are acting as a server.
Manipulating the SSL context when ICE negotiation has completed
is problematic as the SSL context is not protected and if acting
as a client the remote side may have started DTLS negotiation
already.
The retransmission timeout timer code has also been split up
and simplified some. Both RTP and RTCP now have their own timers
and the points at which the timer is stopped and started is now
more specific. When a packet is sent the timer is started. When
a response is received but before it is processed the timer is
stopped. This provides a guarantee that the timeout is not
occurring while the response is processed.
Walter Doekes [Thu, 2 Jul 2015 11:25:36 +0000 (13:25 +0200)]
chan_sip: Fix early call pickup caused deadlock.
If non-magic pickup (no "pickup-" in callid) is used, chan_sip locks the
channel it wants to pick up, and a bit further down, it locks the
channel list when allocating a new channel.
That causes a deadlock when another part of the code traverses over the
channel list, locking all the channels one by one.
This changeset fixes it by releasing the locks before calling sip_new
and reacquiring them afterwards. Unfortunately this involves doing the
checks we already did again (because the channel may have changed).
While trying to avoid duplicate code, I did some refactoring for
readability:
- if refer_locked == 1, we guarantee there is a locked channel
- magic_callid holds a cached version of !ast_strlen_zero(pickup.exten)
This is for branch 11 only. It appears that the changed code in 13 does
not lock the components like it does in 11 and below. Reproducing the
deadlock on 13 has thusfar failed.
Walter Doekes [Thu, 2 Jul 2015 11:10:59 +0000 (13:10 +0200)]
rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format.
When running valgrind on Asterisk, it complained about:
==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304)
==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...)
==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292)
==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437)
The code in question is a struct assignment, which may be performed by
memcpy as a compiler optimization. It is changed to only copy the struct
contents if source and destination are different.
Walter Doekes [Thu, 2 Jul 2015 10:16:26 +0000 (12:16 +0200)]
astfd: Fix buffer overflow in DEBUG_FD_LEAKS.
If DEBUG_FD_LEAKS was used and more file descriptors than the default of
1024 were available, some DEBUG_FD_LEAKS-patched functions would
overwrite memory past the fixed-size (1024) fdleaks buffer.
This change:
- adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe
- consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024
- stores pointers to constants instead of copying the contents
- reorders the fdleaks struct for possibly tighter packing
- adds a tiny bit of documentation
Walter Doekes [Thu, 2 Jul 2015 09:57:44 +0000 (11:57 +0200)]
res_timing: Don't close FD 0 when out of open files.
This fixes so a failure to get a timer file descriptor does not cascade
to closing FD 0.
On error, both res_timing_kqueue and res_timing_timerfd would call the
destructor before setting the file handle. The file handle had been
initialized to 0, causing FD 0 to be closed. This in turn, resulted in
floods of "CLI>" messages and an unusable terminal.
Matt Jordan [Sat, 27 Jun 2015 23:47:19 +0000 (18:47 -0500)]
Makefile: Remove coverage files on 'make clean'
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Matt Jordan [Sat, 27 Jun 2015 01:38:58 +0000 (20:38 -0500)]
main/pbx: Resolve case sensitivity regression in PBX hints
When 8297136f was merged for ASTERISK-25040, a regression was introduced
surrounding the case sensitivity of device names within hints.
Previously, device names - such as 'sip/foo' - were compared in a case
insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
As a result, some dialplan hints stopped working.
This patch re-introduces case insensitive matching for device names in
hints.
Damian Ivereigh [Thu, 11 Jun 2015 13:18:48 +0000 (23:18 +1000)]
chan_sip.c: Update dialog fromtag after request with auth
If a client sends and INVITE which is 401 rejected, then subsequently
sends a new INVITE with the auth info and uses a different fromtag
from the first INVITE, Asterisk will accept the new INVITE as part of
the original dialog - match_req_to_dialog() specifically ignores the
fromtag. However it does not update the stored dialog with the new
fromtag.
This results in Asterisk being unable to match future packets that are
part of this dialog (such as the ACK to the OK or the OK to the BYE),
and the call is dropped.
This problem was originally found when using an NEC-i SV8100-GE (NEC SIP
Card).
* After a successful match of a packet to the dialog, if the packet is
not a SIP_RESPONSE, authentication is present and the fromtags are
different, the stored fromtag is updated with the one from the recent
INVITE.
Mark Michelson [Thu, 11 Jun 2015 21:44:19 +0000 (16:44 -0500)]
chan_sip: Prevent deadlock when performing BYE with Also transfer.
When a BYE with an Also header is successfully processed, and the sender
of the BYE is bridged with another channel, chan_sip will unlock the
owner of the dialog on which the BYE was received, call ast_async_goto()
on the bridged channel, and then re-lock the owner. The reason for this
locking behavior is that ast_async_goto() can result in a masquerade,
which requires that the involved channels are unlocked.
The problem here is that this causes a locking inversion since the
dialog's lock is held when re-locking the owner channel after the async
goto. The lock order is supposed to be channel and then sip_pvt.
The fix proposed is simple. In addition to unlocking the owner channel
before the ast_async_goto() call, also unlock the sip_pvt. Then relock
both after ast_async_goto() returns, being sure to lock the channel and
then the sip_pvt.
Corey Farrell [Sun, 17 May 2015 04:04:16 +0000 (00:04 -0400)]
Fix unsafe uses of ast_context pointers.
Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0.
However, abs(INT_MIN) = INT_MIN and is still negative, as well as
abs(INT_MIN) % num_buckets, and as a result this led to a crash.
One way to trigger the bug is using host=::80 or 0.0.0.128 in peer
configuration section in chan_sip or chan_iax.
This patch takes the remainder before applying abs, so that bucket
number is always in range.
Corey Farrell [Thu, 21 May 2015 01:53:46 +0000 (21:53 -0400)]
Logger: Reset defaults before processing config.
Reset options to default values before reloading config. This ensures
that if a setting is removed or commented out of the configuration file
it is unset on reload.
Kevin Harwell [Thu, 14 May 2015 20:13:58 +0000 (15:13 -0500)]
audiohook.c: Difference in read/write rates caused continuous buffer resets
Currently, everytime a sample rate change occurs (on read or write) the
associated factory buffers are reset. If the requested sample rate on a
read differed from that of a write then the buffers are continually reset
on every read and write. This has the side effect of emptying the buffer,
thus there being no data to read and then write to a file in the case of
call recording.
This patch fixes it so that an audiohook_list's rate always maintains the
maximum sample rate among hooks and formats. Audiohook sample rates are
only overwritten by this value when slin native compatibility is turned on.
Also, the audiohook sample rate can only overwrite the list's sample rate
when its rate is greater than that of the list or if compatibility is
turned off. This keeps the rate from constantly switching/resetting.
Jonathan Rose [Wed, 13 May 2015 20:41:07 +0000 (15:41 -0500)]
Message.c: Clear message channel frames on cleanup
The message channel is a special channel that doesn't actually process frames.
However, certain actions can cause frames to be placed in the channel's read
queue including the Hangup application which is called on the channel after
each message is processed. Since the channel will continually be reused for
many messages, it's necessary to flush these frames at some point.
main/manager.c: Bugfix sort action_manager by alphabetically
Fix the alphabetic order added on ast_manager_register_struct. The order
for struct manager_action added is not working, this change fixes the
problem.
MySQL configuration engine contains a bug in require_mysql(). This
function is used for column type checking in tables. This bug only
affects DATETIME, DATE and FLOAT types.
It came from mixing the first condition (switch-case-like
if/then/else), to check the expected column type, with the second
condition, to check the actual column type against the expected column
type. Both conditions must be checked separately in order to avoid the
execution of the wrong block.
ASTERISK-18252 #comment This patch might fix the issue
Reported by: Gareth Blades
Richard Mudgett [Tue, 12 May 2015 22:34:45 +0000 (17:34 -0500)]
chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision.
If an ISDN call is hungup by both sides at the same time a crash could
happen.
* Added missing NULL checks for the owner channel after calling
pri_queue_pvt_cause_data() in two places. Code after those calls need to
check the owner channel pointer for NULL before use because
pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the
owner and the owner may get hung up.
ASTERISK-21893 #close
Reported by: Alexandr Gordeev
MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the
target platform. But menuselect is to be run on the build system, so
BUILD_CC needs to be used instead - like it was in the past, before the
recent changes (https://reviewboard.asterisk.org/r/4370/). This is the
patch for ASTERISK-25074.
ASTERISK-25074 #close
Reported by: Sebastian Kemper
Tested by: Sebastian Kemper
Corey Farrell [Tue, 12 May 2015 06:31:58 +0000 (02:31 -0400)]
Fix processing of asterisk.conf debug=yes.
The code which reads asterisk.conf supports processing the debug
option with ast_true, but ast_true returns -1. This causes debug
to still be off, convert to 1 so debug will be on as requested.
Ivan Poddubny [Mon, 11 May 2015 12:07:31 +0000 (12:07 +0000)]
pbx/pbx_spool: Fix issue when call files were executed too early
pbx_spool used to delete/move the call file upon successful outgoing
call completion, but did not delete it from in-memory list of files
(dirlist, used only when compiled with inotify/kqueue support).
That resulted in an extra attempt to process that filename after
retrytime seconds.
Then, if a new file with the same name appears that is scheduled
in future further than the completed one plus its retrytime,
then it gets executed earlier than expected.
This patch fixes remove_from_queue function to also remove the entry
from the dirlist.
Sean Bright [Fri, 8 May 2015 15:39:42 +0000 (15:39 +0000)]
res_rtp_asterisk: Issue ERROR if res_srtp is not found.
While trying to get WebRTC working with chan_pjsip, I was running
into the following error:
Attempted to set an invalid DTLS-SRTP configuration on RTP
instance...
Josh helpfully pointed out that res_srtp.so might not be loaded, and
sure enough, it wasn't. This patch adds a ERROR indiciating as much
to hopefully help others having a similar problem.
Alexander Traud [Tue, 5 May 2015 16:35:16 +0000 (18:35 +0200)]
tcptls: Avoiding ERR_remove_state in OpenSSL.
ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by
ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were
called by SSL_load_error_strings already and got removed. These changes allow
OpenSSL forks like BoringSSL to be used with Asterisk.
ASTERISK-25043 #close
Reported by: Alexander Traud
patches:
asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520)
George Joseph [Thu, 7 May 2015 19:54:35 +0000 (13:54 -0600)]
doc: Make progdocs play nice with git
Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in
Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly. Updated clean
targets as well.
Martin Tomec [Tue, 21 Apr 2015 16:52:22 +0000 (18:52 +0200)]
res_odbc: Use negative connection cache for all connections
Apply the negative connection cache setting to all connections,
even those that are not pooled. This ensures that the connection
will not be re-established before the negative connection cache
time is met.
Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which
checks compiler requirements for RAII:
gcc: -fnested-functions support
clang: -fblocks (and if required -lBlocksRuntime)
The original check was implemented in configure.ac and now has it's
own file. This function also sets C_COMPILER_FAMILY to either gcc or
clang for use by makefile
Created autoconf/ast_check_strsep_array_bounds.m4 (contains
AST_CHECK_STRSEP_ARRAY_BOUNDS):
which checks if clang is able to handle the optimized strsep & strcmp
functions (linux). If not, the standard libc implementation should be
used instead. Clang + the optimized macro's work with:
strsep(char *, char []), but not with strsepo(char *, char *).
Instead of replacing all the occurences throughout the source code,
not using the optimized macro version seemed easier
See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h':
llvm-comment: Normally, this array-bounds warning are suppressed for
macros, so that unused paths like the one that accesses __s1[3] are
not warned about. But if you preprocess manually, and feed the
result to another instance of clang, it will warn about all the
possible forks of this particular if statement. Instead of switching
of this optimization, another solution would be to run the preproces-
sing step with -frewrite-includes, which should preserve enough
information so that clang should still be able to suppress the diag-
nostic at the compile step later on.
See also "https://llvm.org/bugs/show_bug.cgi?id=20144"
See also "https://llvm.org/bugs/show_bug.cgi?id=11536"
Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning
suppressions:
-Wno-unused-value
-Wno-parentheses-equality
In an earlier review (reviewboard: 4550 and 4554), they were deemed a
nuisace and less than benefitial.
configure.ac:
Added AST_CHECK_RAII() see earlier
Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier
Removed moved content
Build System: Prevent unneeded changes to asterisk/buildopts.h.
* Add AST_DEVMODE to BUILDOPTS
* Remove CFLAGS that do not effect ABI from BUILDOPTS.
* Use BUILDOPTS to generate AST_BUILDOPT_SUM.
* Remove loop that defined AST_MODULE_*
These changes ensure that only ABI effecting options are considered for
AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused
by enabling or disabling one module that another is dependent on.
Matt Jordan [Wed, 29 Apr 2015 19:49:23 +0000 (14:49 -0500)]
main/pbx: Improve performance of dialplan reloads with a large number of hints
The PBX core maintains two hash tables for hints: a container of the
actual hints (hints), along with a container of devices that are watching that
hint (hintdevices). When a dialplan reload occurs, each hint in the hints
container is destroyed; this requires a lookup in the container of devices to
find the device => hint mapping object. In the current code, this performs an
ao2_callback, iterating over each of the device to hint objects in the
hintdevices container. For a large number of hints, this is extremely
expensive: dialplan reloads with 20000 hints could take several minutes
in just this phase.
This patch improves the performance of this step in the dialplan reloads
by caching which devices are watching a hint on the hint object itself.
Since we don't want to create a circular reference, we just cache the
name of the device. This allows us to perform a smarter ao2_callback on
the hintdevices container during hint removal, hashing on the name of the
device and returning an iterator to the matching names. The overall
performance improvement is rather large, taking this step down to a number of
seconds as opposed to minutes.
In addition, this patch also registers the hint containers in the PBX
core with the astobj2 library. This allows for reasonable debugging to
hash collisions in those containers.
Matt Jordan [Thu, 30 Apr 2015 16:04:04 +0000 (11:04 -0500)]
include/asterisk/vector.h: Backport vector.h to Asterisk 11
Vectors are very useful constructs. As a container, they prevent having
to calloc/realloc arrays manually. They also have advantages over linked
lists, which require elements in the list to be a struct. This patch
backports vectors to Asterisk 11 for use in future patches.
Richard Mudgett [Wed, 29 Apr 2015 22:28:07 +0000 (17:28 -0500)]
chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option.
Some telco switches occasionally ignore ISDN RESTART requests. The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request. If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco. The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.
chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code. Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards. I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.
* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.
ASTERISK-25034 #close
Reported by: Richard Mudgett
cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8
This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
columns added in Asterisk 1.8. The columns are:
* peeraccount
* linkedid
* sequence
When enabled, the columns in the database entry will be populated with the data
from the CDR.
Matt Jordan [Wed, 29 Apr 2015 21:15:43 +0000 (16:15 -0500)]
main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8
The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS
structures in the RTP engine. However, when a 'core reload' is issued, a
double free of the memory pointed to by the char *'s in the DTLS
configuration struct can occur, as ast_rtp_dtls_cfg_free does not set
the pointers to NULL when they are freed.
This patch sets those pointers to NULL, preventing a second call to
ast_rtp_dtls_cfg_free from corrupting memory.
Kevin Harwell [Wed, 29 Apr 2015 18:19:58 +0000 (13:19 -0500)]
res_fax: allow 2400 transmission rate according to v.27ter standard
A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so
a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits
per second. This reverts all or some of those patches since according to the
v.27ter standard a rate of 2400 bits per second is also supported.
One of the original patches also added 9600 bits per second support for v.27.
This patch also removes that since v.27ter only supports 2400/4800 bits per
second.
Also, since Asterisk specifically supports v.27ter the enum was renamed to
better reflect this.
Mark Michelson [Wed, 29 Apr 2015 16:46:48 +0000 (11:46 -0500)]
rtp_engine: Prevent unnecessary memory increases during calls.
The doxygen for ast_rtp_codecs_payloads_copy() states:
"This copies the payloads from the codecs0 structure to the codecs1
structure, overwriting any current values."
However, in practice, the overwriting of current values was not
happening. Instead, a new RTP codec payload object would be appended to
the codecs1 structure instead of replacing the corresponding object.
This patch corrects this behavior by overwriting the object in the
codecs1 structure if it exists already. If it does not already exist,
then create a new copy and link it in.
Tests of "memory show summary rtp_engine.c" had previously shown
additional allocations being performed any time that Asterisk processed
an incoming SDP. Scenarios involving lots of reinvites resulted in lots
of allocations. With this patch, I can perform as many reinvites as
I want and see no memory increases from the RTP engine.
ASTERISK-24916 #close
Reported by Christophe Osuna
The realtime API passes down the va_list argument to each RT engine in
failover chain until one succeeds. MySQL engine used to access the
variable argument list with va_arg, which mutates the va_list, so the
next engine in failover chain gets invalid agrument list.
This patch uses va_copy to preserve the original va_list argument intact.
ASTERISK-19538 #close
Reported by: alexat
Tested by: Ivan Poddubny
Steve Davies [Tue, 28 Apr 2015 10:38:30 +0000 (11:38 +0100)]
res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS
ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created.
The resources are linked into a table, but the original alloc refs
are never released. ast_strdup leak in rtp_engine.c. If
ast_rtp_dtls_cfg_copy() is called twice on the same destination struct,
a pointer to an alloc'd string is overwritten before the string is free'd.
projects / thirdparty / asterisk.git / log
