Daniel Tryba [Mon, 2 Oct 2017 12:48:41 +0000 (14:48 +0200)]
res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
res_calendar_icalendar: Filter out occurrences superceded by another VEVENT
When we are loading the calendars, we call libical's
icalcomponent_foreach_recurrence method for each VEVENT component that
we have in our calendar.
That method has no knowledge concerning the existence of the other
VEVENT components and will feed our callback with all ocurrences
matching the requested time span.
The occurrences generated by icalcomponent_foreach_recurrence while
expanding a recurring VEVENT's RRULE and RDATE properties can be
superceded by an other VEVENT sharing the same UID.
I use an external iterator (in libical terminology) to avoid messing
with the internal ones from the calling function, and search for
VEVENTS which could supersede the current occurrence.
The event which can invalidate this occurence needs to have:
- the same UID as our recurrent component (comp)
- a RECURRENCE-ID property, which represents the start time of this
occurrence
Richard Mudgett [Thu, 28 Sep 2017 22:37:15 +0000 (17:37 -0500)]
app_queue.c: Fix announcements when announce-to-first-user not enabled.
The previous patch for ASTERISK-27216 made it so you wouldn't get any
position or periodic announcements unless you had announce-to-first-user
enabled. The announce-to-first-user feature was added by ASTERISK_21782
as a result of the patch which introduced the redundant announcements that
ASTERISK-27216 removes.
* By noting that the makeannouncement variable is used to suppresses the
first user announcement, we set its initial value to the
announce-to-first-user enable setting.
George Joseph [Wed, 27 Sep 2017 18:45:21 +0000 (12:45 -0600)]
logger: Bring back ability to turn debug on by source file
Somewhere along the way we lost the ability to debug individual
source files. For modules, this wasn't a big deal but all the
source files in ./main are in the one "core" module so debugging
individual core capabilities was almost impossible.
* Added a test to DEBUG_ATLEAST that also checks __FILE__ instead
of just module name. Any source file will work even if it's in
a module subdirectory.
The pjsip_publishc_init() call was referenced with a misplaced
parentheses. As a result, outbound publication messages went out with an
expiration of 1 second.
George Joseph [Tue, 26 Sep 2017 16:01:48 +0000 (10:01 -0600)]
pjsip_message_filter: Fix regression causing bad contact address
The "res_pjsip: Filter out non SIP(S) requests" commit moved the
filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER
in order to filter out incoming bad uri schemes as early as possible.
Since the change affected outgoing messages as well and the TRANSPORT
layer is the last to be run on outgoing messages, we were overwriting
the setting of external_signaling_address (which is set earlier by
res_pjsip_nat) with an internal address.
* pjsip_message_filter now registers itself as a pjproject module
twice. Once in the TSX layer for the outgoing messages (as it was
originally), then a second time in the TRANSPORT layer for the
incoming messages to catch the invalid uri schemes.
The bridge_p2p_rtp_write() has potential reentrancy problems.
* Accessing the bridged RTP members must be done with the instance1 lock
held. The DTMF and asymmetric codec checks must be split to be done with
the correct RTP instance struct locked. i.e., They must be done when
working on the appropriate side of the point to point bridge.
* Forcing the RTP mark bit was referencing the wrong side of the point to
point bridge. The set mark bit is used everywhere else to set the mark
bit when sending not receiving.
The patches for ASTERISK_26745 and ASTERISK_27158 did not take into
account that not everything carried by RTP uses a codec. The telephony
DTMF events are not exchanged with a codec. As a result when
RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is
enabled, the DTMF digits would always get passed to the core even though
the local native RTP bridge is active, and the DTMF digits would go out
using the wrong SSRC id.
* Add protection for non-format payload types like DTMF when updating the
lastrxformat and lasttxformat. Also protect against non-format payload
types when checking for asymmetric codecs.
app_queue: Only do announcement logic between ringing cycles
This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:
The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.
Reported-by: Stefan Engström Tested-by: Stefan Engström
ASTERISK-27216 #close
res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1
In PostgreSQL 9.1 the backslash are string literals and not the escape
of characters.
In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the
support for old version of Postgresql than 9.1 was dropped. The sentence
before make was "ESCAPE '\'" but in version before than 9.1 need it to be
as follow "ESCAPE '\\'".
Jean Aunis [Thu, 7 Sep 2017 09:41:09 +0000 (11:41 +0200)]
bridge : Fix one-way direct-media when early bridging with native_rtp
When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.
Alexander Traud [Mon, 18 Sep 2017 15:00:31 +0000 (17:00 +0200)]
res_srtp: lower log level of auth failures
Previously, sRTP authentication failures were reported on log level WARNING.
When such failures happen, each RT(C)P packet is affected, spamming the log.
Now, those failures are reported at log level VERBOSE 2. Furthermore, the
amount is further reduced (previously all two seconds, now all three seconds).
Additionally, the new log entry informs whether media (RTP) or statistics (RTCP)
are affected.
Richard Mudgett [Fri, 25 Aug 2017 22:01:57 +0000 (17:01 -0500)]
AST-2017-008: Improve RTP and RTCP packet processing.
Validate RTCP packets before processing them.
* Validate that the received packet is of a minimum length and apply the
RFC3550 RTCP packet validation checks.
* Fixed potentially reading garbage beyond the received RTCP record data.
* Fixed rtp->themssrc only being set once when the remote could change
the SSRC. We would effectively stop handling the RTCP statistic records.
* Fixed rtp->themssrc to not treat a zero value as special by adding
rtp->themssrc_valid to indicate if rtp->themssrc is available.
ASTERISK-27274
Make strict RTP learning more flexible.
Direct media can cause strict RTP to attempt to learn a remote address
again before it has had a chance to learn the remote address the first
time. Because of the rapid relearn requests, strict RTP could latch onto
the first remote address and fail to latch onto the direct media remote
address. As a result, you have one way audio until the call is placed on
and off hold.
The new algorithm learns remote addresses for a set time (1.5 seconds)
before locking the remote address. In addition, we must see a configured
number of remote packets from the same address in a row before switching.
* Fixed strict RTP learning from always accepting the first new address
packet as the new stream.
* Fixed strict RTP to initialize the expected sequence number with the
last received sequence number instead of the last transmitted sequence
number.
* Fixed the predicted next sequence number calculation in
rtp_learning_rtp_seq_update() to handle overflow.
Sean Bright [Wed, 13 Sep 2017 19:14:25 +0000 (15:14 -0400)]
res_calendar: On reload, update all configuration
This changes the behavior of res_calendar to drop all existing calendars
and re-create them whenever a reload is done. The Calendar API provides
no way for configuration information to be pushed down to calendar
'techs' so updated settings would not take affect until a module
unload/load was done or Asterisk was restarted.
Asterisk 15+ already has a configuration option 'fetch_again_at_reload'
that performs a similar function.
Also fix a tiny memory leak in res_calendar_caldav while we're at it.
George Joseph [Wed, 13 Sep 2017 21:23:54 +0000 (15:23 -0600)]
res_pjsip: Filter out non SIP(S) requests
Incoming requests with non sip(s) URIs in the Request, To, From
or Contact URIs are now rejected with
PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in
pjsip_message_filter (formerly pjsip_message_ip_updater) and is
done at pjproject's "TRANSPORT" layer before a request can even
reach the distributor.
URIs read by res_pjsip_outbound_publish from pjsip.conf are now
also checked for both length and sip(s) scheme. Those URIs read
by outbound registration and aor were already being checked for
scheme but their error messages needed to be updated to include
scheme failure as well as length failure.
Sean Bright [Wed, 13 Sep 2017 14:38:11 +0000 (10:38 -0400)]
chan_rtp: Use μ-law by default instead of signed linear
Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.
George Joseph [Mon, 11 Sep 2017 10:46:35 +0000 (04:46 -0600)]
res_pjsip: Add handling for incoming unsolicited MWI NOTIFY
A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.
res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.
Walter Doekes [Sun, 10 Sep 2017 11:17:27 +0000 (13:17 +0200)]
res/res_pjsip: Fix localnet checks in pjsip, part 2.
In 45744fc53, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.
Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)
MS-SQL has no native Enum-type support and therefore
needs to work with constraints.
Since these constraints need unique names the suggested approach
referenced in the following alembic documentation has been applied:
http://bit.ly/2x9r8pb
chan_sip: when getting sip pvt return failure if not found
In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.
This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.
Sean Bright [Wed, 6 Sep 2017 15:50:53 +0000 (11:50 -0400)]
app_waitforsilence: Cleanup & don't treat missing frames as 'noise'
* WaitForSilence completes successfully if it receives no media in the
specified timeout, but when acting as WaitForNoise that logic needs
to be reversed.
* Use standard argument parsing macros and add some error checking for
invalid values.
* The documentation indicated that the first argument to both
WaitForSilence and WaitForNoise was required when it was not. Update
the documentation to reflect that.
* Wrap up some behavior in structs to avoid boolean checks all over the
place.
George Joseph [Fri, 1 Sep 2017 10:17:02 +0000 (04:17 -0600)]
stasis/control: Fix possible deadlock with swap channel
If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.
* control_swap_channel_in_bridge now only holds the control
lock while it's actually modifying the control structure and
releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.
chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE
If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.
Alexander Traud [Wed, 6 Sep 2017 08:02:19 +0000 (10:02 +0200)]
res_srtp: Add support for libsrtp2.1.
Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro
SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with
previous versions of libSRTP, this change allows libSRTP 2.1.x as well.
Ben Ford [Tue, 5 Sep 2017 14:35:12 +0000 (09:35 -0500)]
chan_pjsip: Suppress frame warnings.
When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.
Sean Bright [Tue, 5 Sep 2017 15:05:48 +0000 (11:05 -0400)]
formats: Restore previous fread() behavior
Some formats are able to handle short reads while others are not, so
restore the previous behavior for the format modules so that we don't
have spurious errors when playing back files.
Walter Doekes [Tue, 5 Sep 2017 14:16:01 +0000 (16:16 +0200)]
res/res_pjsip: Standardize/fix localnet checks across pjsip.
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.
For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.
Therefore, checks like this look wrong, but are right:
/* See if where we are sending this request is local or not, and if
not that we can get a Contact URI to modify */
if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
ast_debug(5, "Request is being sent to local address, "
"skipping NAT manipulation\n");
(In the list == localnet == DENY == skip NAT manipulation.)
And conversely, other checks that looked right, were wrong.
This change adds two macro's to reduce the confusion and uses those
instead:
George Joseph [Tue, 5 Sep 2017 10:23:04 +0000 (04:23 -0600)]
res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel
t38_reinvite_response_cb can get called by res_pjsip_session's
session_inv_on_tsx_state_changed in situations where session->channel
is NULL. If it is, the ast_log warning segfaults because it tries
to get the channel name from a NULL channel.
* Check session->channel and print "unknown channel" when it's NULL.
sanitize_tdata was assuming all URIs were SIP URIs so when a non
SIP uri was in the From, To or Contact headers, the unconditional
cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
a segfault when trying to access uri->other_param.
* Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
checks before attempting to cast or use the returned uri.
ASTERISK-27152 Reported-by: Ross Beer
Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
Joshua Colp [Mon, 22 May 2017 15:36:38 +0000 (15:36 +0000)]
res_rtp_asterisk: Only learn a new source in learn state.
This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.
The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.
Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.
projects / thirdparty / asterisk.git / log
