Richard Mudgett [Tue, 12 Jun 2018 19:09:54 +0000 (14:09 -0500)]
channel.c: Fix usage of CHECK_BLOCKING()
The CHECK_BLOCKING() macro is used to indicate if a channel's handling
thread is about to do a blocking operation (poll, read, or write) of
media. A few operations such as ast_queue_frame(), soft hangup, and
masquerades use the indication to wake up the blocked thread to reevaluate
what is going on.
Richard Mudgett [Mon, 18 Jun 2018 23:04:54 +0000 (18:04 -0500)]
autoservice: Don't start channel autoservice if the thread is a user interface.
Executing dialplan functions from either AMI or ARI by getting a variable
could place the channel into autoservice. However, these user interface
threads do not handle the channel's media so we wind up with two threads
attempting to handle the media.
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
Richard Mudgett [Mon, 18 Jun 2018 21:07:47 +0000 (16:07 -0500)]
Dialplan functions: Fix some channel autoservice misuse.
* Fix off nominal paths leaving the channel in autoservice.
* Remove unnecessary start/stop channel autoservice.
* Fix channel locking around a channel datastore search.
Acccording to the man page for sprintf, using the same buffer for
output as one used as an input yields undefined behavior.
This patch should work around this problem.
ASTERISK-27903 Reported-by: Alexander Traud
Change-Id: I2213dcb454aff26457e2e4cc9c6821276463ae3a
Sam Wierema [Tue, 12 Jun 2018 14:30:37 +0000 (16:30 +0200)]
app_mp3: remove 10 seconds of silence after mp3 playback
This patch changes the way asterisk polls output from mpg123, instead
of waiting for 10 seconds(when playing an http url) it now uses a
timeout of one second and iterates 10 times using this same timeout.
The main difference is that for every timeout asterisk receives it now
checks if mpg123 is still running before poll again.
George Joseph [Thu, 31 May 2018 21:22:13 +0000 (15:22 -0600)]
app_confbridge: Enable sending events to participants
ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc. In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.
For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.
To control this behavior, the following options have been added to
confbridge.conf:
bridge_profile/enable_events: This must be enabled on any bridge where
events are desired.
user_profile/send_events: This must be set for a user profile to send
events. Different user profiles connected to the same bridge can have
different settings. This allows admins to get events but not normal
users for instance.
user_profile/echo_events: In some cases, you might not want the user
triggering the event to get the event sent back to them. To prevent it,
set this to false.
A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used. This allows participant A's video
stream to appear as the same label to all other participants.
Alexander Traud [Wed, 13 Jun 2018 10:06:10 +0000 (12:06 +0200)]
res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP.
Previously, Asterisk used its script ./configure, to test whether OpenSSL was
built with no-srtp (or was simply too old). However, the header file
<openssl/opensslconf.h> is the preferred way to detect the local configuration
of OpenSSL.
As a positive side-effect the script ./configure does not interleave the
detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of
individual features of OpenSSL anymore.
ktyerman [Tue, 5 Jun 2018 01:31:39 +0000 (11:31 +1000)]
chan_iax2: better handling for timeout and EINTR
The iax2 module is not handling timeout and EINTR case properly. Mainly when
there is an interupt to the kernel thread. In case of ast_io_wait recieves a
signal, or timeout it can be an error or return 0 which eventually escapes the
thread loop, so that it cant recieve any data. This then causes the modules
receive queue to build up on the kernel and stop any communications via iax in
asterisk.
The proposed patch is for the iax module, so that timeout and EINTR does not
exit the thread.
Richard Mudgett [Mon, 30 Apr 2018 22:38:58 +0000 (17:38 -0500)]
AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden. However, if an endpoint is not identified then a 401
unauthorized response is sent. This vulnerability just discloses which
requests hit a defined endpoint. The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.
* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified. The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.
Alexander Traud [Fri, 8 Jun 2018 20:02:38 +0000 (22:02 +0200)]
res_rtp_asterisk: Allow OpenSSL configured with no-deprecated.
Furthermore, allow OpenSSL configured with no-dh. Additionally, this change
allows auto-negotiation of the elliptic curve/group for servers, not only with
OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519
(since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect.
Alexei Gradinari [Thu, 31 May 2018 15:25:40 +0000 (11:25 -0400)]
func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql
The functions acf_odbc_read/cli_odbc_read ignore a number of columns
returned by the SQLNumResultCols.
If the number of columns is zero it means no data.
In this case, a SQLFetch function has to be not called,
because it will cause an error.
George Joseph [Thu, 7 Jun 2018 13:46:03 +0000 (07:46 -0600)]
chan_pjsip: Register for "BEFORE_MEDIA" responses
chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses. If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".
* Removed chan_pjsip_incoming_response from the original session
supplement (which was handling only "AFTER MEDIA") and added it to a
new session supplement which accepts both "BEFORE_MEDIA" and
"AFTER_MEDIA".
* Also cleaned up some cleanup code in load module.
Alexei Gradinari [Tue, 22 May 2018 21:21:10 +0000 (17:21 -0400)]
pjsip_options: handle modification of qualify options in realtime
Currentrly pjsip_options code does not handle the situation when the
qualify options were changed in realtime database.
Only 'module reload res_pjsip' helps.
This patch add a check on contact add/update observers if the contact
qualify options are different than local aor qualify options.
If the qualify options were modified then synchronize
the pjsip_options AOR local state.
Alexei Gradinari [Wed, 23 May 2018 21:20:39 +0000 (17:20 -0400)]
pjsip_options: show/reload AOR qualify options using CLI
Currentrly pjsip_options code does not handle the situation when the
AOR qualify options were changed.
Also there is no way to find out what qualify options are using.
This patch add CLI commands to show and synchronize Aor qualify options:
pjsip show qualify endpoint <id>
Show the current qualify options for all Aors on the PJSIP endpoint.
pjsip show qualify aor <id>
Show the PJSIP Aor current qualify options.
pjsip reload qualify endpoint <id>
Synchronize the qualify options for all Aors on the PJSIP endpoint.
pjsip reload qualify aor <id>
Synchronize the PJSIP Aor qualify options.
Pirmin Walthert [Wed, 30 May 2018 06:12:30 +0000 (08:12 +0200)]
bridge_channel.c: Fix Deadlock when using Local channels and fax gateway
ast_indicate is invoked with the bridge locked. As ast_indicate locks the
other end of the bridge as well this can lead to a deadlock in some situations.
(Especially when a different thread does the same in the reverse order).
This patch calls ast_indicate after unlocking the bridge which fixes the
deadlock. Calling ast_indicate with these parameters without locking the
bridge should be safe as this is done at different places without a
bridge lock.
ASTERISK-27094 #close Reported-by: David Brillert
Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f
Joshua Colp [Tue, 5 Jun 2018 09:36:35 +0000 (09:36 +0000)]
rtp: Don't negotiate dynamic codecs using payload.
In Asterisk there are some dynamic codecs that have
a fixed payload number. This number was being improperly
used to negotiate the codec, instead of using the name
and sample rate. This could result in the wrong payload
number being negotiated for a codec.
This change makes it so that only static payloads
will be negotiated using their payload number.
George Joseph [Mon, 4 Jun 2018 14:50:51 +0000 (08:50 -0600)]
app_sendtext: Allow content types other than text/plain
There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before. Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.
William McCall [Tue, 29 May 2018 00:17:52 +0000 (00:17 +0000)]
app_confbridge: Add talking indicator for ConfBridgeList AMI response
When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.
Richard Mudgett [Tue, 29 May 2018 17:28:48 +0000 (12:28 -0500)]
app_meetme: Fix manager event documentation for several events.
The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and
MeetmeTalkRequest AMI events were documented with sending out a Usernum
header when the User header was actually output.
* Change the online documentation to match reality.
Alexander Traud [Mon, 28 May 2018 15:29:23 +0000 (17:29 +0200)]
tcptls.h: Repair ./configure --with-ssl=PATH.
asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those
inclusions got replaced by forward declarations. As side effect, the inclusions
got completed.
Alexander Traud [Fri, 25 May 2018 14:55:26 +0000 (16:55 +0200)]
tcptls: Allow OpenSSL configured with no-dh.
Additionally, this change allows auto-negotiation of the elliptic curve/group
for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer.
This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a
side-effect.
Torrey Searle [Wed, 9 May 2018 13:31:47 +0000 (15:31 +0200)]
res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change
Certain race conditions between changing bridge types and DTMF can
cause the current FLAG_NEED_MARKER_BIT to send the marker bit before
the actual first packet of native bridging.
This logic keeps track of the ssrc the bridge is currently sending
and will correctly ensure the marker bit is set if SSRC as changed
from the previous sent packet.
rtp: Add support for RTP extension negotiation and abs-send-time.
When RTP was originally created it had the ability to place a single
extension in an RTP packet. In practice people wanted to potentially
put multiple extensions in one and so RFC 5285 (obsoleted by RFC
8285) came into existence. This allows RTP extensions to be negotiated
with a unique identifier to be used in the RTP packet, allowing
multiple extensions to be present in the packet.
This change extends the RTP engine API to add support for this. A
user of it can enable extensions and the API provides the ability to
retrieve the information (to construct SDP for example) and to provide
negotiated information (from SDP). The end result is that the RTP
engine can then query to see if the extension has been negotiated and
what unique identifier is to be used. It is then up to the RTP engine
implementation to construct the packet appropriately.
The first extension to use this support is abs-send-time which is
defined in the REMB draft[1] and is a second timestamp placed in an
RTP packet which is for when the packet has left the sending system.
It is used to more accurately determine the available bandwidth.
Richard Mudgett [Tue, 22 May 2018 22:17:31 +0000 (17:17 -0500)]
channel.c: Fix off nominal channel allocation failure path.
__ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd
descriptors to -1 yet. The destructor would then attempt to close these
fd's that had never been opened.
Alexei Gradinari [Fri, 18 May 2018 21:45:22 +0000 (17:45 -0400)]
config.c: Fix successful DELETE treated as failure
The config engine destroy_func callback function returns the number of
rows deleted or -1 on error. But the function
ast_destroy_realtime_fields treated non-zero return values as error.
Nic Colledge [Sat, 12 May 2018 11:53:13 +0000 (12:53 +0100)]
app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail
Correct the log warning message shown when ODBC voicemail
retrieve_file is called and there is a null value in the category
column.
A more meaningfull message is now written at debug level.
Brian P. Martin [Wed, 18 Apr 2018 02:15:08 +0000 (19:15 -0700)]
chan_mobile: support handling of caller-id names ("cnam").
Add support to handle caller-ID names ("cnam") in addition to caller-ID
numbers. The prior code ignored the caller-ID name altogether, and
used the local name for the cell phone (e.g. "my-iphone") in its place.
Note: as of this writing, at least some Android phones don't pass cnam to
us. This can be seen by issuing "core set debug 2" in the CLI and watching
the "CLIP" record when a call comes in. If cnam isn't in the CLIP record,
there's nothing we can do to provide one. We'll provide a null cnam field,
so later Asterisk processes know to try other sources (e.g. cidname database,
OpenCNAM, etc.).
Reported by: Brian Martin
Tested by: Brian Martin
ASTERISK-27726
Alexander Traud [Thu, 17 May 2018 06:58:43 +0000 (08:58 +0200)]
res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
Asterisk uses Reference Counting to track whether a module can be unloaded.
Every consumer who requires a module, increases the reference count. When the
consumer goes, is unloaded itself, it has to decrease the reference count on
all its used/required modules. That way
core stop gracefully
works on the command-line interface (CLI): One module after the other is
unloaded. A recent change broke this for the module res_pjsip.
Alexander Traud [Thu, 17 May 2018 05:34:03 +0000 (07:34 +0200)]
res_pjsip: Register pjsip_transport_management not externally but internally.
The module (res_)pjsip_transport_management got moved into res_pjsip. It is no
longer an independent/external module with (un)load_module and therefore has to
register just internally with res_pjsip.
Alexander Traud [Fri, 11 May 2018 17:49:12 +0000 (19:49 +0200)]
rtp_engine: Remove the double assigned RTP payload ID of H.263+.
Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format
H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8
assigned another payload ID 98 for this format in Asterisk 1.6.
Alexander Traud [Fri, 11 May 2018 12:10:51 +0000 (14:10 +0200)]
rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again.
This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf
which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples
of media formats which were affected.
res_hep: Adds hostname resolution support for capture_address
Previously, only an IP address would be accepted for the capture_address config
setting in hep.conf. This change allows capture_address to be a resolvable
hostname or an IP address.
ASTERISK-27796 #close Reported-By: Sebastian Gutierrez
Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f
projects / thirdparty / asterisk.git / log
