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
Jaco Kroon [Tue, 8 May 2018 09:59:02 +0000 (11:59 +0200)]
manager: fix digest auth for ami/http mechanism.
Due to a fixed size buffer the digest authentication could be
incorrectly calculated if a large URI was provided, causing
authentication failure. The buffer is now dynamically allocated to allow
any size URI within the normal limits of the HTTP request size.
Corey Farrell [Fri, 4 May 2018 18:47:25 +0000 (14:47 -0400)]
app_macro: Prevent infinite loop in find_matching_priority.
Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will
prevent crash if dialplan contains an include loop. Log an error when
this occurs, at most one message per call to Macro() so we avoid logger
spam.
stream: Make the topology a reference counted object.
The stream topology has no lock of its own resulting in
another lock protecting it in some way (for example the
channel lock). If multiple channels are being juggled at
the same time this can be problematic. This change makes
the topology a reference counted object instead which
guarantees it will remain valid even without the channel
lock being held.
Joshua Colp [Thu, 3 May 2018 11:34:32 +0000 (11:34 +0000)]
res_ari: Remove requirement that body exists when debug is on.
The "ari set debug" code for incoming requests incorrectly assumed
that all requests would contain a body. If one did not exist the
request would be incorrectly rejected. The response that was sent
was also incomplete as an incorrect function was used to construct
the response.
The code has now been changed to no longer require a request to have
a body and the response updated to use the correct function.
res_rtp_asterisk: Always update SRTP on local SSRC change.
When the local SSRC changes we need to update the SRTP information
so that the proper key is used. This is commonly done as a result
of bridging two channels together. Previously we only updated
the SRTP information if media had already flowed, but in practice
the channel driver may have already performed SRTP negotiation and
set up the previous SSRC. We now always do it on a local SSRC
change.
Joshua Colp [Mon, 11 Dec 2017 18:34:53 +0000 (18:34 +0000)]
pjsip: Rewrite OPTIONS support with new eyes.
The OPTIONS support in PJSIP has organically grown, like many things in
Asterisk. It has been tweaked, changed, and adapted based on situations
run into. Unfortunately this has taken its toll. Configuration file
based objects have poor performance and even dynamic ones aren't that
great.
This change scraps the existing code and starts fresh with new eyes. It
leverages all of the APIs made available such as sorcery observers and
serializers to provide a better implementation.
1. The state of contacts, AORs, and endpoints relevant to the qualify
process is maintained. This state can be updated by external forces (such
as a device registering/unregistering) and also the reload process. This
state also includes the association between endpoints and AORs.
2. AORs are scheduled and not contacts. This reduces the amount of work
spent juggling scheduled items.
3. Manipulation of which AORs are being qualified and the endpoint states
all occur within a serializer to reduce the conflict that can occur with
multiple threads attempting to modify things.
4. Operations regarding an AOR use a serializer specific to that AOR.
5. AORs and endpoint state act as state compositors. They take input
from lower level objects (contacts feed AORs, AORs feed endpoint state)
and determine if a sufficient enough change has occurred to be fed further
up the chain.
6. Realtime is supported by using observers to know when a contact has
been registered. If state does not exist for the associated AOR then it
is retrieved and becomes active as appropriate.
The end result of all of this is best shown with a configuration file of
3000 endpoints each with an AOR that has a static contact. In the old
code it would take over a minute to load and use all 8 of my cores. This
new code takes 2-3 seconds and barely touches the CPU even while dealing
with all of the OPTIONS requests.
Kevin Harwell [Wed, 18 Apr 2018 20:59:38 +0000 (15:59 -0500)]
translate: generic plc not filled in after translation
If during translation a codec could not handle a given frame the translation
core would return NULL, thus not passing along the "missing" frame. Due to this
there was no frame to apply generic plc to, thus rendering it useless.
This patch makes it so the translation core produces an interpolated slin frame
in the cases where an attempt was made to translate to slin, but failed. This
interpolated frame is then passed along and can be used by the generic plc
algorithms to fill in the frame.
bridge_softmix: Fix sporadic incorrect video stream mapping.
When an externally initiated renegotiation occurred it was
possible for video streams to be incorrectly remapped,
resulting in no video flowing to some receivers.
This change ensures that only the video source sets up
mappings and also that removed streams do not have mappings
set up.
Alexander Traud [Fri, 20 Apr 2018 10:40:21 +0000 (12:40 +0200)]
menuselect: Add DragonFly BSD.
In DragonFly BSD, added libraries from ports are placed into /usr/local.
Therefore, this directory must be added for the preprocessor, compiler, and
linker.
Beside that, the script ./configure was updated:
* OSARCH list was outdated and not used, removed.
* AC_CANONICAL_BUILD was not used.
* _REENTRANT, this feature test macro is obsolete.
This change fixes a bug where a REMB collector may be
freed twice, and also tweaks REMB combining such that if
there is no bitrate from anyone (or there are no sources)
we report 0 instead of using an old bitrate.
Build System: Add missing ASTMM_LIBC to flex output.
Redirect libc allocation functions to use Asterisk functions for
main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors
produced by astmm.h when these files are regenerated, though other
issues still remain.
bridge_softmix / app_confbridge: Add support for REMB combining.
This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.
Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.
Support for configuring which behavior to use has been
added to app_confbridge.
George Joseph [Tue, 10 Apr 2018 21:09:49 +0000 (15:09 -0600)]
app_sendtext: Enhance SendText to support Enhanced Messaging
SendText now accepts new channel variables that can be used
to override the To and From display names and set the Content-Type
of a message. Since you can now set Content-Type, other text/*
content types are now valid.
George Joseph [Wed, 27 Sep 2017 16:44:53 +0000 (10:44 -0600)]
bridge_softmix: Forward TEXT frames
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Ben Ford [Mon, 9 Apr 2018 22:09:03 +0000 (17:09 -0500)]
res_rtp_asterisk: Add support for receiving and handling NACK requests.
Adds the ability to receive and handle incoming NACK requests if
retransmissions are enabled. If retransmissions are enabled, a data
buffer is allocated that stores packets being sent. If a NACK request
is received, the packet requested for retransmission is sent if it is
still in the buffer. In the same request, if any of the following 16
packets are marked as not received, those will be sent as well if
available, as outlined in RFC4585.
Also changes RTCP RR and SR to use media source SSRC instead of packet
source SSRC when determining which instance to use for RTCP reports.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
George Joseph [Fri, 13 Apr 2018 20:17:36 +0000 (14:17 -0600)]
utils: Add ast_assert_return
Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the
following...
If the assert passes... NoOp
If the assert fails and AST_DEVMODE is defined, execute ast_assert()
then, if DO_CRASH isn't set, return from the calling function with
the supplied value.
If the assert fails and AST_DEVMODE is not defined, return from the
calling function with the supplied value.
The macro will execute a return without a value if one isn't suppled.
Ben Ford [Fri, 13 Apr 2018 19:32:48 +0000 (14:32 -0500)]
res_musiconhold: Don't restart MOH from beginning after announcement.
This reverts a problem introduced by the fix for ASTERISK_24329.
Now, when an announcement is played while waiting in a queue, music on
hold will not restart from the beginning of the sound file and will
instead pick up where it left off. However, the incorrect behavior in
ASTERISK_24329 is now present again; if an announcement X seconds
long is played when music on hold starts, music on hold will start X
seconds into the file.
projects / thirdparty / asterisk.git / log
