Sean Bright [Fri, 6 Mar 2020 20:59:37 +0000 (15:59 -0500)]
enum.c: Add support for regular expression flag in NAPTR record
A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.
Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.
Sebastian Kemper [Sun, 12 Jan 2020 11:37:46 +0000 (12:37 +0100)]
check_expr2: fix cross-compile/hardening issues
When building check_expr2 with ASLR PIE hardening enabled the linker
fails. This is resolved by adding the regular compiler flags when
building the object files from ast_expr2f.c and ast_expr2.c.
Note: The STANDALONE define is removed because it is already defined in
_ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives
'--enable-dev-mode'.
Also, a Makefile variable "CROSS_COMPILING" is added so that the
build system doesn't try to run check_expr2 when cross-compiling,
because that will fail the build as will.
ASTERISK-28685 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915
Joshua C. Colp [Thu, 20 Feb 2020 17:33:42 +0000 (17:33 +0000)]
res_rtp_asterisk: Improve video performance in certain networks.
The receive buffer will now grow if we end up flushing the
receive queue after not receiving the expected packet in time.
This is done in hopes that if this is encountered again the
extra buffer size will allow more time to pass and any missing
packets to be received.
The send buffer will now grow if we are asked for packets and
can't find them. This is done in hopes that the packets are
from the past and have simply been expired. If so then in
the future with the extra buffer space the packets should be
available.
Sequence number cycling has been handled so that the
correct sequence number is calculated and used in
various places, including for sorting packets and
for determining if a packet is old or not.
NACK sending is now more aggressive. If a substantial number
of missing sequence numbers are added a NACK will be sent
immediately. Afterwards once the receive buffer reaches 25%
a single NACK is sent. If the buffer continues to grow and
reaches 50% or greater a NACK will be sent for each received
future packet to aggressively ask the remote endpoint to
retransmit.
Kevin Harwell [Fri, 28 Feb 2020 18:54:14 +0000 (12:54 -0600)]
message & stasis/messaging: make text message variables work in ARI
When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.
Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.
That being the case, and since this is technically an API breaking change (no
one should really be affected since things never really worked) the ARI version
was updated to reflect that.
Jaco Kroon [Thu, 20 Feb 2020 12:52:06 +0000 (14:52 +0200)]
addons/res_config_mysql: silense warnings about printf format errors.
Warnings without this:
res_config_mysql.c: In function 'update2_mysql':
res_config_mysql.c:741:15: warning: format '%llu' expects argument of type
'long long unsigned int', but argument 6 has type 'my_ulonglong'
{aka 'long unsigned int'} [-Wformat=]
ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n",
numrows, tablename);
Joshua C. Colp [Tue, 18 Feb 2020 13:10:16 +0000 (13:10 +0000)]
pjsip: Update ACLs on named ACL changes.
This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.
This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.
George Joseph [Thu, 13 Feb 2020 19:39:58 +0000 (12:39 -0700)]
res_pjsip_outbound_registration: Fix SRV failover on timeout
In order to retry outbound registrations for some situations, we
need access to the tdata from the original request. For instance,
for 401/407 responses we need it to properly construct the
subsequent request with the authentication. We also need it if
we're iterating over a DNS SRV response record set so we can skip
entries we've already tried.
We've been getting the tdata from the server response rdata and
transaction but that only works for the failures where there was
actually a response (4XX, 5XX, etc). For timeouts there's no
response and therefore no rdata or transaction from which to get
the tdata. When processing a single A/AAAA record for a server,
this wasn't an issue as we just retried that same server after the
retry timer expired. If we got an SRV record set for the server
though, without the state from the tdata, we just kept trying the
first entry in the set repeatedly instead of skipping to the next
one in the list.
* Added a "last_tdata" member to the client state structure to keep
track of the sent tdata.
* Updated registration_client_send() to save the tdata it used into
the client_state.
* Updated sip_outbound_registration_response_cb() to use the tdata
saved in client_state when we don't get a response from the
server. We still use the tdata from the transaction when we DO
get a response from the server so we can properly handle 4XX
responses where our new request depends on it.
General note on timeouts:
Although res_pjsip_outbound_registration skips to the next record
immediately when a timeout occurs during SRV set traversal, it's
pjproject that determines how long to wait before a timeout is
declared. As with other SIP message types, pjproject will continue
trying the same server at an interval specified by "timer_t1" until
"timer_b" expires. Both of those timers are set in the pjsip.conf
"system" section.
Joshua C. Colp [Sun, 5 Jan 2020 00:11:20 +0000 (00:11 +0000)]
bridging: Add better support for adding/removing streams.
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
Ben Ford [Thu, 23 Jan 2020 19:17:06 +0000 (13:17 -0600)]
RTP/ICE: Send on first valid pair.
When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.
A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.
Sean Bright [Mon, 17 Feb 2020 14:05:16 +0000 (09:05 -0500)]
app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.
* If the user wants to write to a wav49 file, make sure that it is
reflected properly in MIXMONITOR_FILENAME.
* Add a note to the documentation describing this behavior.
* Add a note in main/file.c indicating that app_mixmonitor needs to be
changed if the logic in build_filename was changed.
Torrey Searle [Wed, 12 Feb 2020 16:05:11 +0000 (17:05 +0100)]
res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough
When moh_passthrough is used, asterisk is only generating invites
of type sendonly and sendrecv instead of taking fully into account
the on hold state of the local and remote parties
Kevin Harwell [Thu, 13 Feb 2020 21:08:10 +0000 (15:08 -0600)]
res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup
There was a race condition between client initiated DTLS setup, and handling
of server side ice completion that caused the underlying SSL object to get
cleared during DTLS initialization. If this happened Asterisk would be left
in a partial DTLS setup state. RTP packets were sent and received, but were
not being encrypted and decrypted. This resulted in no audio, or static.
Specifically, this occurred when '__rtp_recvfrom' was processing the handshake
sequence from the client to the server, and then 'ast_rtp_on_ice_complete'
gets called from another thread and clears the SSL object when calling the
'dtls_perform_setup' function. The timing had to be just right in the sense
that from the external SSL library perspective SSL initialization completed
(rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks
to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks
it is not finished, thus not completing 'dtls_srtp_setup'.
This patch removes calls to 'dtls_perform_setup', which clears the SSL object,
in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear
the underlying SSL object. If an ice candidate changes a full protocol level
renegotiation occurs. Also, in the case of bundled ICE candidates are reused
when a stream is added. So no real reason to have to clear, and reset in this
instance.
Also, this patch adds a bit of extra logging to aid in diagnosis of any future
problems.
Sean Bright [Tue, 11 Feb 2020 13:46:02 +0000 (08:46 -0500)]
res_musiconhold: Avoid spurious warning when 'format' is the empty string
The change to res_config_odbc that allowed empty strings to be
returned to realtime consumers¹ causes a warning to be emitted when
loading MoH classes. So we need to treat an empty 'format' as if it
was not specified to avoid the warning.
Given a scenario where session refreshes occur close to
each other while another is finishing it was possible for
the session refreshes to occur out of order. It was
also possible for session refreshes to be delayed for
quite some time if a session refresh did not result in
a topology change.
For the out of order session refreshes the first session
refresh would be queued due to a transaction in progress.
This transaction would then finish. When finished a
separate task to process the delayed requests queue
would be queued for handling. A second refresh would
be requested internally before this delayed request
queued task was processed. As no transaction was in
progress this session refresh would be immediately
handled before the queued session refresh.
The code will now check if any delayed requests exist
before allowing a session refresh to immediately occur.
If any exist then the session refresh is queued.
For the delayed session refreshes if a session refresh
did not result in a topology change the attempt would
be immediately stopped and no other delayed requests would
be processed.
The code will now go through the entire delayed requests
queue until a delayed request results in a request
actually being sent.
Currently aptitude is installed using interactive mode. This patch
changes this to use the non-interactive mode as it can block
automatic dependencies installation, ex: CI, Docker build.
Walter Doekes [Tue, 28 Jan 2020 08:34:45 +0000 (09:34 +0100)]
chan_sip: Clarify in sample docs how directmediapermit/-acl should be used
It said "restrict [...] which peers should be able to pass [audio]
to each other".
However, these settings are not global (for which you would expect
signaling IPs to be checked). These settings are available per peer
only, and the IPs being checked, are the RTP IPs.
Joshua C. Colp [Tue, 4 Feb 2020 14:18:13 +0000 (10:18 -0400)]
res_rtp_asterisk: Don't produce transport-cc if no packets.
The code assumed that when the transport-cc feedback
function was called at least one packet will have been
received. In practice this isn't always true, so now
we just reschedule the sending and do nothing.
George Joseph [Mon, 3 Feb 2020 16:24:58 +0000 (09:24 -0700)]
message.c: Add option to suppress the Message channel AMI and ARI events
In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use. To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes" In Asterisk versions <18, the default
is "no" preserving existing behavior. Beginning with
Asterisk 18, the option will default to "yes".
NOTE: This change does not affect UserEvents or the ARI
TextMessageReceived events.
* Added the "hide_messaging_ami_events" option to asterisk.conf.
* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
the "Message/ast_msg_queue" channel if the option is set in
asterisk.conf. This suppresses the reporting of the events.
Sean Bright [Wed, 29 Jan 2020 14:57:38 +0000 (09:57 -0500)]
res_config_odbc: Preserve empty strings returned by the database
When res_config_odbc (and perhaps other realtime backends) reads a SQL
NULL from the database, it coalesces the value to the empty string
which prevents it from being returned to the realtime core.
However, if it instead reads the empty string from the database, it
needs a way to encode that fact without having the value omitted
entirely. It does this by changing the value to a string with a single
space. The realtime code in main/config.c recognizes this special case
and _turns the string back into the empty string_ before passing it to
realtime API consumers.
For all of this to work, we need to ensure that we actually pass the
single-space-string back to the realtime core, which is currently
failing because we are trimming the value before checking its
content. So instead we now special case the single-space-string case
so that empty values are returned properly.
Sean Bright [Tue, 28 Jan 2020 19:23:19 +0000 (14:23 -0500)]
res_stasis_playback: Prevent media_index from going out of bounds
Incrementing stasis_app_playback.media_index directly in our playback
loop means that when we reach the end of our playlist the index into
the vector will be outside of the bounds of the vector.
Instead use a temporary variable and only assign when we're sure that
we are in bounds.
Joshua C. Colp [Tue, 28 Jan 2020 15:18:45 +0000 (15:18 +0000)]
res_pjsip_pubsub: Increment persistence data ref when recreating.
Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.
Kevin Harwell [Mon, 27 Jan 2020 18:01:15 +0000 (12:01 -0600)]
stasis/app: don't lock an app before a call to send
Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.
This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.
Kevin Harwell [Mon, 27 Jan 2020 17:44:45 +0000 (11:44 -0600)]
res_stasis: trigger cleanup after update
The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.
This patch makes it so the cleanup code is executed post 'update'.
Walter Doekes [Mon, 13 Jan 2020 10:13:41 +0000 (11:13 +0100)]
chan_sip: Always process updated SDP on media source change
Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).
If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.
This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.
Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.
(An alternative fix would be to set ignoresdpversion=yes on the peer.)
Sean Bright [Thu, 23 Jan 2020 15:06:35 +0000 (10:06 -0500)]
chan_pjsip: Ignore RTP that we haven't negotiated
If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.
We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.
George Joseph [Wed, 22 Jan 2020 18:56:38 +0000 (11:56 -0700)]
cdr.c: Set event time on party b when leaving a parking bridge
When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.
* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.
NOTE: This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.
Sean Bright [Wed, 22 Jan 2020 15:39:47 +0000 (10:39 -0500)]
http: Add ability to disable /httpstatus URI
Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.
We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.
Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.
Additionally:
* Change 'enablestatic' to 'enable_static' but keep the former for
backwards compatibility.
* Improve some internal variable names
Sean Bright [Thu, 16 Jan 2020 19:47:01 +0000 (14:47 -0500)]
app_voicemail, say: Fix various leading whitespace problems
In af90afd90c64c5183c2207d061f9aa15138081b2, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.
Sean Bright [Thu, 16 Jan 2020 15:09:47 +0000 (10:09 -0500)]
chan_sip.c: Stop handling continuation lines after reading headers
lws2sws() does not stop trying to handle header continuation lines
even after all headers have been found. This is problematic if the
first character of a SIP message body is a space or tab character, so
we update to recognize the end of the message header.
Sean Bright [Tue, 14 Jan 2020 22:20:21 +0000 (17:20 -0500)]
app_voicemail: Set globals to default values when voicemail.conf missing
If voicemail.conf exists but is empty, the config parsing process will
default a number of global variables to non-zero values. On the other
hand, if voicemail.conf is missing (arguably semantically equivalent
to an empty file), this process is skipped and the globals are
defaulted to 0.
Set the globals to the same values they would be set to if a
configuration were present. This allows voicemail configuration to be
done completely by Realtime without the need to create an empty
voicemail.conf file.
ASTERISK-27622 #close
Reported by: Jim Van Meggelen
projects / thirdparty / asterisk.git / log
