This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The member reg in the peercnt structure is an unsigned char and peercnt_modify()
is expecting an unsigned char argument which gets assigned to peercnt->reg.
This patch fixes that by casting the integer argument being passed to
peercnt_modify to unsigned char.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Clean up some twisted code in the iax2_bridge() loop.
* Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames
to prevent the native bridge loop from breaking.
* Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over
a native IAX2 bridge.
(issue ABE-2912)
Review: https://reviewboard.asterisk.org/r/2870/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client. The provided expiry time of the client is
updated after inserting the astdb entry. As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after minregexpire
seconds until they reregister.
* Move updating of the expiry time to before inserting into the astdb.
(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Reduce indentation in __attempt_transmit().
* Don't update the static last error time variable every time in
__schedule_action() and socket_read().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix stray reference to idle_list in cleanup_thread_list(). This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.
* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix bridgecallno deadlock avoidance. When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.
* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.
* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list. defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1). When touching the bridgecallno, we need to lock it.
2). Remove magic number '0' and replace with TRANSFER_NONE.
3). Exit early if no bridgecallno.
4). Reduce indentation.
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2613/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@391333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1). When touching the bridgecallno, we need to lock it.
2). stop_stuff() which calls iax2_destroy_helper()
Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called.
Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno);
3). When evaluating the state of 'callno->transferring' of the current leg,
we can't change it to READY unless the bridgecallno is locked.
Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED',
the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs.
(closes issue ASTERISK-21409)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2594/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@391062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On startup, it's possible for a frame to arrive before the processing threads were ready.
In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive
before we are at ast_cond_wait, the signal will be ignored.
The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued.
Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled
after each thread is started.
(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2427/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
already holds the iax2 private lock and improperly fails to obtain the channel
lock before calling ast_set_callerid. By not safely obtaining the channel lock,
a locking inversion can take place, causing a deadlock.
This patch solves this by calling the required deadlock avoidance functions
that obtain the channel lock before setting the caller ID.
Thanks to Pavel for fixing my syntax errors and testing this patch out.
(closes issue ASTERISK-21128)
Reported by: Pavel Troller
Tested by: Pavel Troller
patches:
ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is currently an edge case where call number 32768 might be allocated for
a call, even though the IAX2 protocol requires call numbers be only 15 bits.
This resulted in some unpredictable behavior when call number 32678 is chosen.
This patch was mostly written by Richard Mudgett via ReviewBoard. I'm just
committing it.
Review: https://reviewboard.asterisk.org/r/2293/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.
(closes issue ASTERISK-20338)
Reported-by: John Covert
Patch-by: John Covert
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372015 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
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
Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.
(issue ASTERISK-19537)
(closes issue ASTERISK-19801)
Reported by: Alec Davis
(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made schedule_delivery() set the received frame f->data.ptr to NULL if
the datalen is zero.
* Fix queue_signalling() memcpy() size error.
* Made queue_signalling() not use C++ keyword variable names.
(closes issue ASTERISK-19597)
Reported by: mgrobecker
Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.
(Closes issue ASTERISK-19650)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most of the changes here are trivial NULL checks. There are a couple
optimizations to remove the need to check for NULL and outboundproxy parsing
in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was
found and fixed with the parsing of outboundproxy when "outboundproxy=," was
set.
(Closes issue ASTERISK-19654)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, setting trunkfreq had no effect on initial load or on reload and
only ever used the default value. This causes trunkfreq to be used
appropriately on initial load and reload.
(closes issue ASTERISK-19521)
Patch-by: Jaco Kroon
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Initialize a struct sockaddr_in in try_transfer() so that the code isn't
(potentially) trying to read from it while uninitialized.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These few places in the code used sizeof() on h_addr in struct hostent.
This is sizeof(char *). The correct way to get the size of this address is to
use h_length. This error would result in reads/writes of 8 bytes instead of 4
on 64-bit machines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but
this value is in milliseconds while ast_timer_set_rate() expects the rate argument
to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead.
With a default of 20ms, this change makes IAX2 send trunk packets every 20ms
instead of every 50ms.
Tracked down by myself and Bob Wienholt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
