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Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7
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Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e
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* Break some long lines.
* Fix doxygen comment.
Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305
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Due to the use of stasis_unsubscribe_and_join in the peer destructor
it is possible for a deadlock to occur when an event callback is
occurring at the same time.
This happens because the peer may be destroyed while holding the
peers container lock. If this occurs the event callback will never
be able to acquire the container lock and the unsubscribe will
never complete.
This change makes it so the peers that have been removed from the
peers container are not destroyed with the container lock held.
ASTERISK-25163 #close
Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33
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This resolves two observed race conditions.
First, a bit of background on what the Stasis application does:
1a Creates a stasis_app_control structure. This structure is linked into
a global container and can be looked up using a channel's unique ID.
2a Puts the channel in an event loop. The event loop can exit either
because the stasis_app_control structure has been marked done, or
because of some other factor, such as a hangup. In the event loop, the
stasis_app_control determines if any specific ARI commands need to be
run on the channel and will run them from this thread.
3a Checks if the channel is bridged. If the channel is bridged, then
ast_bridge_depart() is called since channels that are added to Stasis
bridges are always imparted as departable.
4a Unlink the stasis_app_control from the container.
When an ARI command is received by Asterisk, the following occurs
1b A thread is spawned to handle the HTTP request
2b The stasis_app_control(s) that corresponds to the channel(s) in the
request is/are retrieved. If the stasis_app_control cannot be
retrieved, then it is assumed that the channel in question has exited
the Stasis app or perhaps was never in Stasis in the first place.
3b A command is queued onto the stasis_app_control, and the channel's
event loop thread is signaled to run the command.
4b While most ARI commands do nothing further, some, such as adding or
removing channels from a bridge, will block until the command they
issued has been completed by the channel's event loop.
The first race condition that is solved by this patch involves a crash
that can occur due to faulty detection of the channel's bridged status
in step 3a. What can happen is that in step 2a, the event loop may run
the ast_bridge_impart() function to asynchronously place the channel
into a bridge, then immediately exit the event loop because the channel
has hung up. In step 3a, we would detect that the channel was not
bridged and would not call ast_bridge_depart(). The reason that the
channel did not appear to be bridged was that the depart_thread that is
spawned by ast_bridge_impart() had not yet started. That is the thread
where the channel is marked as being bridged. Since we did not call
ast_bridge_depart(), the Stasis application would exit, and then the
channel would be destroyed Then the depart_thread would start up and
try to manipulate the destroyed channel, causing a crash.
The fix for this is to switch from using ast_channel_is_bridged() to
checking the NULLity of ast_channel_internal_bridge_channel() to
determine if ast_bridge_depart() needs to be called. The channel's
internal bridge_channel is set when ast_bridge_impart() is called and
is NULLed by the call to ast_bridge_depart(). If the channel's internal
bridge_channel is non-NULL, then the channel must have been imparted
into the bridge and needs to be departed, even if the actual bridging
operation has not yet started. By departing the channel when necessary,
the thread that is running the Stasis application will block until the
bridge gives the okay that the depart_thread has exited.
The second race condition that is solved by this patch involves a leak
of HTTP handler threads. The problem was that step 2b would successfully
retrieve a stasis_app_control structure. Then step 2a would exit the
channel from the event loop due to a hangup. Steps 3a and 4a would
execute, and then finally steps 3b and 4b would. The problem is that at
step 4b, when attempting to add a channel to a bridge, the thread would
block forever since the channel would never execute the queued command
since it was finished with the event loop. This meant that the HTTP
handling thread would be leaked, along with any references that thread
may have owned (in my case, I was seeing bridges leaked).
The fix for this is to hone in better on when the channel has exited the
event loop. The stasis_app_control structure has an is_done field that
is now set at each point where the channel may exit the event loop. If
step 2b retrieves a valid stasis_app_control structure but the control
is marked as done, then the attempted operation exits immediately since
there will be nothing to service the attempted command.
ASTERISK-25091 #close
Reported by Ilya Trikoz
Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
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This event was added some time ago in order to clarify when a channel
took the place of another channel in a parking lot. However, there was
no XML documentation added for the event. This patch adds the XML
documentation.
ASTERISK-24900 #close
Reported by Rusty Newton
Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac
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into 13
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ASTERISK-25162 #close
Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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The CDR_PROP function (as well as the NoCDR application) set the
'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This
flag is supposed to be applied to all CDRs that are currently in the
chain, as well as all CDRs that may be created in the future. Currently,
however, the flag is only applied to the existing CDRs in the chain; new
CDRs do not receive the 'disable all' flag. In particular, this affects
parallel dials, which generate new CDRs for each pair of channels in
the dial attempt.
This patch carries over the 'disable all' flag when it is specified on a
CDR and a new CDR is generated for the chain.
ASTERISK-24344 #close
Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e
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When a parallel dial occurs, a new CDR will be created for each dial
attempt that is made. In most circumstances, the act of creating each
CDR in the chain will include a step that updates the Party A snapshot,
which causes the context/extension of the Party A to be copied onto the
CDR object.
However, when the Party A is in a subroutine, we explicitly do *not*
copy the context/extension onto the CDR. This prevents the Macro or
GoSub routine name from blowing away the context/extension that the
channel was originally executing in. For the original CDR, this is not a
problem: the original CDR already recorded the last known 'good' state
of the channel just prior to it going into the subroutine. However, for
newly generated CDRs in a chain, there is no context/extension set on
them. Since we are in a subroutine, we will never set the Party A's
context/extension on the CDR, and we end up with a CDR with no
destination recorded on it.
This patch updates the creation of a chained CDR such that it copies
over the original CDR's context/extension. This is the last known "good"
state of the CDR, and is a reasonable starting point for the newly
generated CDR. In the case where we are not in a subroutine, subsequent
code will update the location of the CDR from the Party A information;
in the case where we are in a subroutine, the context/extension on the
original CDR is the correct information.
ASTERISK-24443 #close
Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a
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information." into 13
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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.
ASTERISK-25154 #close
Reported by: Damian Ivereigh
Tested by: Damian Ivereigh
Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e
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Prior to this patch, chan_pjsip was failing to pass the endpoint's
context and the desired extension to the ast_channel_alloc_* routine.
This caused a new channel snapshot to be issued without a context and
extension, which can cause some reporting issues for users of AMI, CEL,
and other APIs. The channel driver would later set the context and
extension on the channel such that the channel would start in the
correct location in the dialplan, but the information reported in the
initial event would be incorrect.
This patch modifies the channel driver such that it now passes the
context and extension directly into the allocation routine. This
provides the information in the new channel snapshot published over
Stasis.
ASTERISK-25156 #close
Reported by: cloos
Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e
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When performing a blonde transfer the code uses the old masquerade
mechanism to move a channel around. As a result of this certain information,
such as connected line, is moved between the channels involved. Upon
completion of the move a frame is queued which is supposed to update the
connected line information on the channel. This does not occur as the
code considers it a redundant update since the masquerade operation
updated the channel (but did not inform it of the new connected line
information). The code also does not queue a connected line update
to be handled by the thread handling the channel. Without this any
other channel that may be loosely involved does not know it is
talking to a different caller.
This change does the following to resolve this:
1. The indicated connected line information is cleared upon
completion of the masquerade operation when doing a blonde transfer.
This prevents the connected line update from being considered
redundant.
2. A connected line update frame is now queued upon the completion
of the masquerade operation so any other channel loosely involved
knows that there is a different caller.
ASTERISK-25157 #close
Reported by: Joshua Colp
Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20
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The voicemail.conf mailbox key/value pair is defined as:
<mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
Where all fields in the value including the field values are optional.
Since the parsing code for the mailbox key/value pair is sloppy, this
patch tightens the parsing for the directory information.
* Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
respectively in search_directory_sub(). Those names make more sense.
* Made sure that search_directory_sub() is dealing with the voicemail.conf
mailbox options field if it even exists when looking for the 'hidefromdir'
and 'alias' options.
* Fix crash if a voicemail.conf mailbox is just
<mailbox>=<password>,<name> when the 'a' option is used. If there were no
fields after the name then the 'options' pointer was not checked for NULL.
* Fix users.conf alias processing if the 'a' option is used. The wrong
variable was used.
ASTERISK-25087 #close
Reported by: Chet Stevens
Change-Id: I86052ea77307beddddba5279824d39dc0d593374
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Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976
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Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60
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channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/
send_ping. This deadlock happens because the scheduled task send_lagrq(or
send_ping) starts execution after the call hangup procedure starts but before
it deletes the tasks in the scheduler.
The solution is to delete scheduled lagrq (and ping) task asynchronously
(i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will
be called in a new context (doesn't have callno locked).
This commit also cleans up the procedure of sending LAGRQ and PING.
main/sched.c: Do not assert when deleting non existant entry from scheduler.
This assert seems to be the reason for a lot of awkward code to avoid it.
ASTERISK-24983 #close
Reported by: Y Ateya
Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c
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This patch fixes use-after-free bugs caught by AddressSanitizer.
1. PJSIP transport manager may decide to destroy transport on its own.
For example, when the contact registered via websocket has not renewed
its registration in time. The transport was destoyed, but the websocket
listener thread was still active until the socket closes, and then tried
to call transport_shutdown on transport that has been freed.
Also, the transport destructor accessed wstransport->rdata.tp_info.pool
right after freeing memory that contained wstransport itself.
This patch converts transport to an ao2 object, allowing it to be
refcounted, so that it is available until both websocket listener and
pjsip transport manager are finished with it.
2. The websocket listener deletes the last reference on websocket session
when the tcp connection is closed, and it gets destroyed, but
the transport manager may still use it, for example when disconnect
happens in the middle of a SIP transaction.
A new reference to websocket session has been added that is released
with the transport to prevent this.
ASTERISK-25096 #close
Reported by: Josh Kitchens
ASTERISK-24963 #close
Reported by: Badalian Vyacheslav
Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b
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GCC 4.7 Manual:
http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html
weakref ("target")
A weak reference is an alias that does not by itself require a definition
to be given for the target symbol.
ASTERISK-22559 #close
Reported by: Ibercom
Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf
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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.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
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Show uptime information ends with an unnecessary space.
Now NEEDCOMMA is better defined.
Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1
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It is possible to receive incoming requests or responses after the channel
on an ast_sip_session has been destroyed and NULLed out. Handlers of these
sorts of requests or responses need to be prepared for the possibility
that the channel is NULL or else they could cause a crash.
While several places have been amended to deal with NULL channels, there
were still a couple of places that needed updating.
res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
return early if there is no channel on the session.
res_pjsip_session.c: When handling a 302 response, we need to stop the
redirecting attempt if there is no channel on the session.
ASTERISK-25148 #close
reported by Mark Michelson
Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9
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So this issue is a bit complicated. Since it is possible to pass values to AMI
that contain a '\r\n' (or other similar sequences) these values need to be
escaped. One way to solve this is to escape the values and then pass the escaped
values to the AMI variable parameter string building function. However, this
puts the onus on the pre-build function to escape all string values. This
potentially requires a fair amount of changes along with a lot of string
allocations/freeing for all values.
Surely there is a way to push this complexity down a level into the string
building function itself? This of course is possible, but ends up requiring a
way to distinguish between strings that need to be escaped and those that don't.
The best way to handle this is by introducing a new format specifier in the
format string. For instance a %s (no escape) and %S (escape). However, that is
a bit weird and unexpected.
So faced with those possibilities this patch implements a limited version of the
first option. Instead of attempting to escape all string values this patch only
escapes those values that make sense. This approach limits the number of changes
and doesn't suffer from the odd format specifier problem.
ASTERISK-24934 #close
Reported by: warren smith
Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0
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contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status
to force the creation of a contact_status object whenever a new
contact is added but it didn't unref the returned object.
Added an ao2_cleanup(status) to plug the leak.
ASTERISK-25141
Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40
Reported-by: Corey Farrell
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* Add some type casting so tv_usec can really be a long, instead of
some strange platform specific type.
* Add some .dylib style files to .gitignore.
* Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
versions of GCC, when compiling the Homebrew formula for Asterisk,
are not properly passing the -Xlinker options to the linker. Given
that -Wl, does exactly the [same thing][], and does it properly, this
patch changes the -Xlinker options to use -Wl, instead.
[reasons unknown]: http://bit.ly/1SUbEYx
[same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html
Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd
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The loop to find the first available contact of an endpoint grabbed
contact from the iterator, then checked for offline state. This
caused the first contact after the state was found to leak a reference.
ASTERISK-25141
Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08
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The length of frames retured by sample functions was twice as large as
real, what caused global buffer overflow caught by AddressSanitizer.
ASTERISK-24717 #close
Reported by: Badalian Vyacheslav
Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6
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When permanent_uri_handler was creating the contact status
object for each contact, it wasn't unreffing it at the
end of the loop.
ASTERISK-25141 #close
Reported-by: Corey Farrell
Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12
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This reverts commit 35c699086ae2fd81b2473307ccb2ae79ad32375a.
Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7
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When an endpoint was created, it's messages were being forwarded to
both the tech endpoint topic and the all endpoints topic. Since
the tech topic was also forwarded to all, this was resulting in
duplicate messages whenever an endpoint published. This patch
causes the endpoint to only forward to the tech topic and lets
the tech topic forward to all.
To accomplish this, the existing stasis_cp_single_create function
(which both creates and forwards) was cloned and split into 2
functions, one that creates the topic and one that sets up the
forwarding. This allows endpoint_internal_create to create
the topic from the endpoint_all cache without forwarding it there,
then allows it to do the forward to the tech's topic.
ASTERISK-25137 #close
Reported-by: Vitezslav Novy
ASTERISK-25116 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c
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When the remote peer requires authentication for in-dialog requests then
re-INVITEs to the peer cause the call to be disconnected and other
in-dialog requests to the peer like MESSAGE just don't go through.
* Made session_inv_on_tsx_state_changed() handle in-dialog authentication
for re-INVITEs and other methods. Initial INVITEs cannot be handled here
because the INVITE transaction must be restarted earlier.
* Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in
preparation for removing the file. The generic outbound authentication
code did not work as well as anticipated.
* Created outbound_invite_auth() to only handle initial outbound INVITEs.
Re-INVITEs cannot be handled here. The re-INVITE transaction is still in
progress and the PJSIP library cannot handle the overlapping INVITE
transactions. Other method types should not be handled here as this code
only works on outgoing calls and we need to handle incoming and outgoing
calls.
ASTERISK-25131 #close
Reported by: Richard Mudgett
Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0
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Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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The cache creation callback function expects to receive a sorcery_details
structure and not just a standalone object.
Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450
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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.
ASTERISK-25100 #close
Reported by: Mark Petersen
Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899
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Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves
truncated before passing to pjsip_tpmgr_receive_packet, but the length
was passed unaltered, thus causing memory corruption and segfault.
ASTERISK-25122 #close
Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab
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Many uses of stasis_unsubscribe in modules can be reached through unload.
These have been switched to stasis_unsubscribe_and_join.
Some subscription callbacks do nothing, for these I've created a noop
callback function in stasis.c. This is used by some modules that monitor
MWI topics in order to enable cache, since the callback does not become
invalid after dlclose it is safe to use stasis_unsubscribe on these, even
during module unload.
ASTERISK-25121 #close
Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
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for RLS" into 13
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