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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
Richard Mudgett [Tue, 27 Mar 2018 16:04:42 +0000 (11:04 -0500)]
res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations.
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
Richard Mudgett [Thu, 22 Mar 2018 00:43:21 +0000 (19:43 -0500)]
pjsip_scheduler.c: Fix some corner cases.
* Fix the periodic interval wander because it may take significant time
between the sched thread queueing the task in the serializer and the
serializer actually executing the task. The time it takes to actually
execute the task was already taken into account.
* Pass a schtd ref to the serializer when we queue a scheduled task on
the serializer. We don't want it going away on us while it is in the
serializer queue.
* Skip the scheduled task if the task was canceled between queueing the
task to the serializer and the serializer actually executing the task.
* Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed
task_id and added next_periodic.
* Hold a ref to the passed in serializer so the serializer cannot go away
on the scheduled task.
cdr_mysql: Compile error because MYSQL_PORT definition is missing
If it is not defined, it will add MYSQL_PORT definition. After some
research on MySQL/MariaDB development tree, I couldn't find any reference
to MYSQL_PORT definition in include files.
res_pjsip_session: Rewrite o= with external_media_address.
It now appends the external IP address on the
o= line of the SDP packet. The decision was made to write
the numeric IP address as opposed to the RFC that states
the FQDN should be used if and when available. We believe
the usage of literal IP address will help avoid
potential problems.
Nathan Bruning [Thu, 22 Feb 2018 18:18:48 +0000 (19:18 +0100)]
res_pjsip_notify.c: enable in-dialog NOTIFY
This patch adds support to send in-dialog SIP NOTIFY commands on
chan_pjsip channels, similar to the functionality recently added
for chan_sip (ASTERISK_27461).
This extends res_pjsip_notify to allow for in-dialog messages.
projects / thirdparty / asterisk.git / log
