1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
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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
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The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
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Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:
1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.
Also added an SDP when an update is sent out.
(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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* 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.
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Fix the misdn debug output to remote consoles. chan_misdn uses
ast_console_puts() which doesn't know about verbose levels. Better to use
ast_verbose() instead. Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e. any undefined level.
(closes issue AST-1218)
Reported by: Guenther Kelleter
Patches:
misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter
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With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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Refactored cases where a combination of ast_verbose/options_verbose were
present. Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used. Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
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If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.
This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.
Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.
(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
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A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.
This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.
Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.
(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
issueA21064_fix.patch uploaded by wdoekes (License 5674)
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Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.
* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().
* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.
* Fixed some formatting in ast_bt_get_symbols().
* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.
* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.
* Moved __dump_backtrace() because of compile issues with the utils
directory.
(closes issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2778/
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This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a response to an initial incoming INVITE results in a transport error
the INVITE transaction is removed from the INVITE session. Any attempts
to answer the INVITE session after this results in a crash as it requires
the INVITE transaction to exist. This change explicitly locks the dialog
and checks to ensure that the INVITE transaction exists before answering.
(closes issue AST-1203)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a From header on an outbound out-of-call SIP MESSAGE were
malformed, the result could crash Asterisk.
In addition, if a From header on an incoming out-of-call SIP
MESSAGE request were malformed, the message was happily accepted
rather than being rejected up front. The incoming message path
would not result in a crash, but the behavior was bad nonetheless.
(closes issue ASTERISK-22185)
reported by Zhang Lei
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The SIP_CODEC family of variables let you set the preferred codec to be
offered on an outbound INVITE request. However, for video calls, you need to
be able to set both the audio and video codecs to be offered. This patch lets
the SIP_CODEC variables accept a comma delineated list of codecs. The first
codec in the list is set as the preferred codec; additional codecs are still
offered however.
This lets a dialplan writer set both audio and video codecs, e.g.,
Set(SIP_CODEC=ulaw,h264)
Note that this feature was written by both Dennis Guse and Frank Haase
Review: https://reviewboard.asterisk.org/r/2728
(closes issue ASTERISK-21976)
Reported by: Denis Guse
Tested by: mjordan, sysreq
patches:
patch-channels-chan__sip.c-393919 uploaded by dennis.guse (license 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping
the old recv address since recv was already set. This has caused a problem when
a proxy is involved since responses to incoming requests from the proxy server,
after an outbound call is established, are never sent to the correct recv
address.
In 11, r382322 introduced this regression.
The fix is to revert that change and always store the recv address on incoming
requests.
Thank you Walter Doekes for helping to point out this error and Mark Michelson
for your input/review of the fix.
(closes issue ASTERISK-22071)
Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer
Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026)
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This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes it so that we can detect failures to originate as with
earlier versions of Asterisk, which restores the Asterisk 11 behavior
for the originate manager action. This was causing the ACL tests for
SIP and IAX2 to fail since those tests expected originate failures
when ACLs would cause rejections. Also, this patch fixes crashes in
chan_sip when ACLs rejected peers during registration verification.
(closes issue ASTERISK-22212)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2753/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code adds chan_dahdi the command 'dahdi create channels <range>'
(where <range> is a single <n>-<m> or 'new') and updates 'dahdi destroy
channel' with a similar 'dahdi destroy channels'. It allows DAHDI
channels and spans to be added after the initial channel load
(without destroying all other channels as in 'dahdi restart').
It also includes some fixes to the D-Channel / span destruction code
(r394552).
This change is intended to provide a hook for a script running from
udev once a span has been assigned ("registered") / unassigned
("unregistered") for its channels. The udev hook configures the span's
channels with dahdi_cfg -S, and can then ask Asterisk to create ethe
channels. See the scripts added to DAHDI-tools in 2.7.0.
Review: https://reviewboard.asterisk.org/r/1598/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.
This patch sets the dialog's transport based on the transport that was defined
in the register line. If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026)
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This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
