This patch adds support for receiving events regarding Peer status changes
and Contact status changes. This is particularly useful in scenarios where
we are subscribed to all endpoints and channels, where we often want to know
more about the state of channel technology specific items than a single
endpoint's state.
ASTERISK-24870
Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is
shown with each file descriptor. This helps to debug leaked UDP/TCP ports on
long-lived servers, for example.
ASTERISK-25405 #close
Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b
When changing a hint extension without removing the hint first the
device state and presence state is not updated. This causes the state
of the hint to be that of the previous extension and not the current
one. This state is kept until a state change occurs as a result of
something (presence state change, device state change).
This change updates the hint with the current device and presence
state of the new extension when it is changed. Any state callbacks
which may have been added before the hint extension is changed are
also informed of the new device and presence state if either have
changed.
ASTERISK-25394 #close
Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f
When Asterisk translates between codecs, each with a different frame size (for
example between iLBC 30 and Speex-WB), too large frames were created by
ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
length, creating several frames when necessary. Affects all transcoding modules
which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.
ASTERISK-25353 #close
Change-Id: I2e229569d73191d66a4e43fef35432db24000212
It has been observed that on long-running busy systems, a scheduler
context can eventually hit INT_MAX for its assigned IDs and end up
overflowing into a very low negative number. When this occurs, this can
result in odd behaviors, because a negative return is interpreted by
callers as being a failure. However, the item actually was successfully
scheduled. The result may be that a freed item remains in the scheduler,
resulting in a crash at some point in the future.
The scheduler can overflow because every time that an item is added to
the scheduler, a counter is bumped and that counter's current value is
assigned as the new item's ID.
This patch introduces a new method for assigning scheduler IDs. Instead
of assigning from a counter, a queue of available IDs is maintained.
When assigning a new ID, an ID is pulled from the queue. When a
scheduler item is released, its ID is pushed back onto the queue. This
way, IDs may be reused when they become available, and the growth of ID
numbers is directly related to concurrent activity within a scheduler
context rather than the uptime of the system.
Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2
Asterisk can load and register an object type while still having an invalid
sorcery mapping. This can cause an issue when a creation call is invoked.
For example, mis-configuring PJSIP's endpoint identifier by IP address mapping
in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a
subsequent ARI invocation to create the object will cause a crash, as the
internal type may not be registered as sorcery expects.
Merely checking for a NULL pointer here solves the issue.
Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac
In ast_endpoint_snapshot_create(), a failure to init the
string fields results in two attempts to ao2_cleanup the
same pointer. Removed RAII_VAR to eliminate problem.
ASTERISK-25375 #close
Reported by: Scott Griepentrog
Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979
When issuing the "core show hints" CLI command a combination of both
the hint extension and context is created. This uses a fixed size
buffer expecting that the extension will not exceed maximum extension
length. When the extension is actually a pattern match this constraint
does not hold true, and the extension may exceed the maximum extension
length. In this case extra characters are written past the end of the
fixed size buffer.
This change makes it so the construction of the combined hint extension
and context can not exceed the size of the buffer.
ASTERISK-25367 #close
Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499
When unreferencing a taskprocessor its reference count is checked
to determine if it should be unlinked from the taskprocessors
container and its listener shut down. In between the time when the
reference count is checked and unlinking it is possible for
another thread to jump in, find it, and get a reference to it. If
the thread then uses the taskprocessor it may find that it is not
in the state it expects.
This change locks the taskprocessors container during almost the
entire unreference operation to ensure that any other thread which
may attempt to find the taskprocessor has to wait.
ASTERISK-25295
Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c
When deleting a scheduled item if the item in question is currently
executing the ast_sched_del function waits until it has completed.
This is accomplished using ast_cond_wait. Unfortunately the
ast_cond_wait function can suffer from spurious wakeups so the
predicate needs to be checked after it returns to make sure it has
really woken up as a result of being signaled.
This change adds a loop around the ast_cond_wait to make sure that
it only exits when the executing task has really completed.
ASTERISK-25355 #close
Change-Id: I51198270eb0b637c956c61aa409f46283432be61
When executing an action in a bridge it is possible for the
channel to be hung up without the bridge becoming aware of it.
This is most easily reproducible by hanging up when the bridge
is streaming DTMF due to a feature timeout. This change makes
it so after action execution the channel is checked to determine
if it has been hung up and if it has it is kicked from the bridge.
ASTERISK-25341 #close
Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062
* Make ast_rtp_codecs_payload_code() get the current mapping or create a
rx payload type mapping.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56
There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk. It uses only one mapping structure to
associate payload types to codecs. The single mapping is overkill if all
of the payload type values are well known values. Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive. Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.
1) An independent payload type mapping is needed for sending and
receiving.
2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.
3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.
* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.
* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.
* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created. All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.
* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP. We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
When allocating a sorcery object, fail if the
id value was not allocated.
ASTERISK-25323
Reported by: Scott Griepentrog
Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e
The built frame format in audiohook_read_frame_both() is now set to a
signed linear format before the rx and tx frames are duplicated instead of
only for the mixed audio frame duplication.
ASTERISK-25322 #close
Reported by Sean Pimental
Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538
Some codecs that may be a third party library to Asterisk need to have
knowledge of the format attributes that were negotiated. Unfortunately,
when the great format migration of Asterisk 13 occurred, that ability
was lost.
This patch adds an API call, ast_format_attribute_get, to the core
format API, along with updates to the unit test to check the new API
call. A new callback is also now available for format attribute modules,
such that they can provide the format attribute values they manage.
Note that the API returns a void *. This is done as the format attribute
modules themselves may store format attributes in any particular manner
they like. Care should be taken by consumers of the API to check the
return value before casting and dereferencing. Consumers will obviously
need to have a priori knowledge of the type of the format attribute as
well.
Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3
clock_gettime() is, unfortunately, not portable. But I did like that
over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we
usually do when we want a timespec and all we have is ast_tvnow().
This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a
timespec. If clock_gettime() is available, it will use that. Otherwise
ast_tsnow() falls back to using ast_tvnow().
Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e
An http request can be sent to get the existing Asterisk logs.
The command "curl -v -u user:pass -X GET 'http://localhost:8088
/ari/asterisk/logging'" can be run in the terminal to access the
newly implemented functionality.
* Retrieve all existing log channels
ASTERISK-25252
Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808
An http request can be sent to create a log channel
in Asterisk.
The command "curl -v -u user:pass -X POST
'http://localhost:088/ari/asterisk/logging/mylog?
configuration=notice,warning'" can be run in the terminal
to access the newly implemented functionality for ARI.
* Ability to create log channels using ARI
ASTERISK-25252
Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782
An http request can be sent to delete a log channel
in Asterisk.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/logging/mylog'" can be run in the terminal
to access the newly implemented functionally for ARI.
* Able to delete log channels using ARI
ASTERISK-25252
Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
An http request can be sent to rotate a specified log channel.
If the channel does not exist, an error response will be
returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/logging/logChannelName/rotate'" can be run in the
terminal to access this new functionality.
* Added the ability to rotate log files through ARI
ASTERISK-25252
Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
A testsuite test recently failed due to a crash that occurred in the DNS
core. The problem was that the test could not resolve an address, did
not set a result on the DNS query, and then indicated the query was
completed. The DNS core does not handle the case of a query with no
result gracefully, and so there is a crash.
This changeset makes the DNS system resolver set a result with a
zero-length answer in the case that a DNS resolution failure occurs
early. The DNS core now also will accept such a response without
treating it as invalid input. A unit test was updated to no longer treat
setting a zero-length response as off-nominal.
Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6
ast_rtp_codecs_get_payload() gets called once or twice for every received
RTP frame so it would be nice to not allocate an ao2 object to then have
it destroyed shortly thereafter. The ao2 object gets allocated only if
the payload type is not set by the channel driver as a negotiated value.
The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323.
* Made static_RTP_PT[] an array of ao2 objects that
ast_rtp_codecs_get_payload() can return instead of an array of structs
that must be copied into a created ao2 object.
ASTERISK-25296 #close
Reported by: Richard Mudgett
Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0
* Fix off nominial ref leak of new_type in
ast_rtp_codecs_payloads_set_m_type().
* No need to lock static_RTP_PT_lock in
ast_rtp_codecs_payloads_set_m_type() and
ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type
parameter sanity check.
* No need to create ast_rtp_payload_type ao2 objects with a lock since the
lock is not used.
Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4
This event is necessary for the bridge_wait_e_options test to be able to
confirm that ringing is being played on the local channel that runs the
BridgeWait() application with the e(r) option.
ASTERISK-25292 #close
Reported by Kevin Harwell
Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e
Currently, if a blank musiconhold.conf is used, musiconhold will fail
to start for a channel going into a holding bridge with an anticipation
of getting music on hold. That being the case, no frames will be written
to the channel and that can pose a problem for blind transfers in PJSIP
which may rely on frames being written to get past the REFER framehook.
This patch makes holding bridges start a silence generator if starting
music on hold fails and makes it so that if no music on hold functions
are installed that the ast_moh_start function will report a failure so
that consumers of that function will be able to respond appropriately.
ASTERISK-25271 #close
Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
(cherry picked from commit 8458b8d441)
This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Commit 54b25c80c8 solved an issue where a
specific scenario involving local channels and a native local RTP bridge
could result in ringback still being heard on a calling channel even
after the call is bridged.
That commit caused many tests in the testsuite to fail with alarming
consequences, such as not sending DialBegin and DialEnd events, and
giving incorrect hangup causes during calls.
This commit reverts the previous commit and implements and alternate
solution. This new solution involves only passing AST_CONTROL_RINGING
frames across local channels if the local channel is in AST_STATE_RING.
Otherwise, the frame does not traverse the local channels. By doing
this, we can ensure that a playtones generator does not get started on
the calling channel but rather is started on the local channel on which
the ringing frame was initially indicated.
ASTERISK-25250 #close
Reported by Etienne Lessard
Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39
Previous changes to sample rate support in audiohooks accidentally
removed code responsible for allowing the manipulate audiohooks
to work. Without this code the manipulated frame would be dropped
and not used. This change restores it.
ASTERISK-25253 #close
Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13
Control frames with a -1 payload are used as a special signal to stop
playtones generators on channels. This indication is sent both by
app_dial as well as by ast_answer() when a call is answered in case any
tones were being generated on a calling channel.
This control frame type was made to stop traversing local channel pairs
as an optimization, because it was thought that it was unnecessary to
send these indications, and allowing such unnecessary control frames to
traverse the local channels would cause the local channels to optimize
away less quickly.
As it turns out, through some special magic dialplan code, it is
possible to have a tones being played on a non-local channel, and it is
important for the local channel to convey that the tones should be
stopped. The result of having tones continue to be played on the
non-local channel is that the tones play even once the channel has been
bridged. By not blocking the -1 control frame type, we can ensure that
this situation does not happen.
ASTERISK-25250 #close
Reported by Etienne Lessard
Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815
Due to changes in audiohooks to support different sample rates the
underlying storage of samples is in the format of the audiohook
itself and not of the format being requested. This means that if a
channel is using G722 the samples stored will be at 16kHz. If
something subsequently reads from the audiohook at a format which
is not the same sample rate as the audiohook the number of samples
needs to be adjusted.
Given the following example:
1. Channel writing into audiohook at 16kHz (as it is using G722).
2. Chanspy reading from audiohook at 8kHz.
The original code would read 160 samples from the audiohook for
each 20ms of audio. This is incorrect. Since the audio in the
audiohook is at 16kHz the actual number needing to be read is 320.
Failure to read this much would cause the audiohook to reset
itself constantly as the buffer became full.
This change adjusts the requested number of samples by determining
the duration of audio requested and then calculating how many
samples that would be in the audiohook format.
ASTERISK-25247 #close
Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d
This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
