Joshua C. Colp [Wed, 11 Aug 2021 11:15:16 +0000 (08:15 -0300)]
policy: Deprecate modules and add versions to others.
app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.
Alexander Traud [Mon, 2 Aug 2021 17:33:07 +0000 (19:33 +0200)]
aelparse: Accept an included context with timings.
With Asterisk 1.6.0, in the main parser for the configuration file
extensions.conf, the separator was changed from vertical bar to comma.
However, the first separator was not changed in aelparse; it still had
to be a vertical bar, and no comma was allowed.
Additionally, this change allows the vertical bar for the first and
last parameter again, even in the main parser, because the vertical bar
was still accepted for the other parameters.
Naveen Albert [Mon, 28 Jun 2021 13:48:16 +0000 (13:48 +0000)]
cdr_adaptive_odbc: Prevent filter warnings
Previously, if CDR filters were used so that
not all CDR records used all sections defined
in cdr_adaptive_odbc.conf, then warnings will
always be emitted (if each CDR record is unique
to a particular section, n-1 warnings to be
specific).
This turns the offending warning log into
a verbose message like the other one, since
this behavior is intentional and not
indicative of anything wrong.
Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.
res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern
PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
It may be used to get all X- headers in case the actual set and names of headers unknown.
Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.
A flag has been introduced to allow meters to fallback to counters.
Sean Bright [Fri, 23 Jul 2021 16:00:00 +0000 (12:00 -0400)]
res_http_media_cache: Cleanup audio format lookup in HTTP requests
Asterisk first looks at the end of the URL to determine the file
extension of the returned audio, which in many cases will not work
because the URL may end with a query string or a URL fragment. If that
fails, Asterisk then looks at the Content-Type header and then finally
parses the URL to get the extension.
The order has been changed such that we look at the Content-Type
header first, followed by looking for the extension of the parsed
URL. We no longer look at the end of the URL, which was error prone.
under [Thu, 22 Jul 2021 16:39:11 +0000 (19:39 +0300)]
codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother
If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
This makes the audio stream not-playable at the receiver side.
Linphone isn't able to play such an audio - lots of disruptions are heard.
Also I had complains of bad audio from users which use other types of phones.
After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).
Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).
However, this flag is never set in Asterisk-12 and newer.
Previously it has been set (see Asterisk-11).
Naveen Albert [Wed, 16 Jun 2021 20:26:46 +0000 (16:26 -0400)]
app_dtmfstore: New application to store digits
Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.
Kevin Harwell [Mon, 10 May 2021 22:59:00 +0000 (17:59 -0500)]
AST-2021-008 - chan_iax2: remote crash on unsupported media format
If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.
This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.
AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.
If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.
Kevin Harwell [Mon, 14 Jun 2021 18:28:25 +0000 (13:28 -0500)]
AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.
Andre Barbosa [Tue, 29 Jun 2021 16:07:44 +0000 (17:07 +0100)]
res_stasis_playback: Check for chan hangup on play_on_channels
Verify `ast_check_hangup` before looping to the next sound file.
If the call is already hangup we just break the cycle.
It also ensures that the PlaybackFinished event is sent if the call was hangup.
This is also use-full when we are playing a big list of file for a channel that is hangup.
Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.
With the patch we just break the playback cycle when the chan is hangup.
Sean Bright [Thu, 15 Jul 2021 20:04:01 +0000 (16:04 -0400)]
res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.
From RFC 8225 Section 5.2.1:
The "dest" claim is a JSON object with the claim name of "dest"
and MUST have at least one identity claim object. The "dest"
claim value is an array containing one or more identity claim JSON
objects representing the destination identities of any type
(currently "tn" or "uri"). If the "dest" claim value array
contains both "tn" and "uri" claim names, the JSON object should
list the "tn" array first and the "uri" array second. Within the
"tn" and "uri" arrays, the identity strings should be put in
lexicographical order, including the scheme-specific portion of
the URI characters.
Additionally, make it clear that there was a failure to sign the JWT
payload and not necessarily a memory allocation failure.
Naveen Albert [Sat, 22 May 2021 14:31:22 +0000 (10:31 -0400)]
app_waitforcond: New application
While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.
Andre Barbosa [Fri, 4 Jun 2021 11:11:10 +0000 (12:11 +0100)]
res_stasis_playback: Send PlaybackFinish event only once for errors
When we try to play a list of sound files in the same Play command,
we get only one PlaybackFinish event, after all sounds are played.
But in the case where the Play fails (because channel is destroyed
for example), Asterisk will send one PlaybackFinish event for each
sound file still to be played. If the list is big, Asterisk is
sending many events.
This patch adds a failed state so we can understand that the play
failed. On that case we don't send the event, if we still have a
list of sounds to be played.
When we reach the last sound, we send the PlaybackFinish with
the failed state.
If the system time has stepped backwards because of a time
adjustment between the time a frame is timestamped and the
time we check the timestamps in abstract_jb:hook_event_cb(),
we get a negative interval, but we don't check for that there.
abstract_jb:hook_event_cb() then calls
fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
and the first thing that does is assert(interval >= 0).
There are several issues with this...
* abstract_jb:hook_event_cb() saves the interval in a variable
named "now" which is confusing in itself.
* "now" is defined as an unsigned int which converts the negative
value returned from ast_tvdiff_ms() to a large positive value.
* fixed_jb_get()'s parameter is defined as a signed int so the
interval gets converted back to a negative value.
* fixed_jb_get()'s assert is NOT an ast_assert but a direct define
that points to the system assert() so it triggers even in
production mode.
So...
* hook_event_cb()'s "now" was renamed to "relative_frame_start" and
changed to an int64_t.
* hook_event_cb() now checks for a negative value right after
retrieving both the current and framedata timestamps and just
returns the frame if the difference is negative.
* fixed_jb_get()'s local define of ASSERT() was changed to call
ast_assert() instead of the system assert().
Naveen Albert [Sat, 22 May 2021 00:08:58 +0000 (20:08 -0400)]
app_dial: Expanded A option to add caller announcement
Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.
Joshua C. Colp [Mon, 21 Jun 2021 11:31:41 +0000 (08:31 -0300)]
core: Don't play silence for Busy() and Congestion() applications.
When using the Busy() and Congestion() applications the
function ast_safe_sleep is used by wait_for_hangup to safely
wait on the channel. This function may send silence if Asterisk
is configured to do so using the transmit_silence option.
In a scenario where an answered channel dials a Local channel
either directly or through call forwarding and the Busy()
or Congestion() dialplan applications were executed with the
transmit_silence option enabled the busy or congestion
tone would not be heard.
This is because inband generation of tones (such as busy
and congestion) is stopped when other audio is sent to
the channel they are being played to. In the given
scenario the transmit_silence option would result in
silence being sent to the channel, thus stopping the
inband generation.
This change adds a variant of ast_safe_sleep which can be
used when silence should not be played to the channel. The
wait_for_hangup function has been updated to use this
resulting in the tones being generated as expected.
Bernd Zobl [Fri, 7 May 2021 06:18:40 +0000 (08:18 +0200)]
res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress
With the fix for ASTERISK_28754 channels are no longer put on hold if an
outbound INVITE is answered with a "Session Progress" containing
"inactive" audio.
The previous change moved the evaluation of the media attributes to
`negotiate_incoming_sdp_stream()` to have the `remotely_held` status
available when building the SDP in `create_outgoing_sdp_stream()`.
This however means that an answer to an outbound INVITE, which does not
traverse `negotiate_incoming_sdp_stream()`, cannot set the
`remotely_held` status anymore.
This change moves the check so that both, `negotiate_incoming_sdp_stream()` and
`apply_negotiated_sdp_stream()` can do the checks.
George Joseph [Wed, 16 Jun 2021 13:50:54 +0000 (07:50 -0600)]
res_pjsip_messaging: Overwrite user in existing contact URI
When the MessageSend destination is in the form
PJSIP/<number>@<endpoint> and the endpoint's contact
URI already has a user component, that user component
will now be replaced with <number> when creating the
request URI.
Bernd Zobl [Tue, 16 Mar 2021 16:45:23 +0000 (17:45 +0100)]
res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter
Set preferred transport when querying the local address to use in
filter_on_tx_messages(). This prevents the module to erroneously select
the wrong transport if more than one transports of the same type (TCP or
TLS) are configured.
Naveen Albert [Thu, 10 Jun 2021 14:34:53 +0000 (10:34 -0400)]
pbx_builtins: Corrects SayNumber warning
Previously, SayNumber always emitted a warning if the caller hung up
during execution. Usually this isn't correct, so check if the channel
hung up and, if so, don't emit a warning.
Jaco Kroon [Sat, 22 May 2021 12:29:22 +0000 (14:29 +0200)]
func_lock: Add "dialplan locks show" cli command.
For example:
arthur*CLI> dialplan locks show
func_lock locks:
Name Requesters Owner
uls-autoref 0 (unlocked)
1 total locks listed.
Obviously other potentially useful stats could be added (eg, how many
times there was contention, how many times it failed etc ... but that
would require keeping the stats and I'm not convinced that's worth the
effort. This was useful to troubleshoot some other issues so submitting
it.
The scenario where a channel still has an associated datastore we
cannot unload since there is a function pointer to the destroy and fixup
functions in play. Thus increase the module ref count whenever we
allocate a datastore, and decrease it during destroy.
In order to tighten the race that still exists in spite of this (below)
add some extra failure cases to prevent allocations in these cases.
Race:
If module ref is zero, an LOCK or TRYLOCK is invoked (near)
simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
in such a case the datastore is created *prior* to unloading being set
to true (first step in module unload) then it's possible that the module
will unload with the destructor being called (and segfault) post the
module being unloaded. The module will however wait for such locks to
release prior to unloading.
If post that we can recheck the module ref before returning the we can
(in theory, I think) eliminate the last of the race. This race is
mostly theoretical in nature.
Jaco Kroon [Sat, 22 May 2021 12:42:04 +0000 (14:42 +0200)]
func_lock: Fix memory corruption during unload.
AST_TRAVERSE accessess current as current = current->(field).next ...
and since we free current (and ast_free poisons the memory) we either
end up on a ast_mutex_lock to a non-existing lock that can never be
obtained, or a segfault.
Incidentally add logging in the "we have to wait for a lock to release"
case, and remove an ineffective statement that sets memory that was just
cleared by ast_calloc to zero.
Jaco Kroon [Sat, 22 May 2021 12:48:18 +0000 (14:48 +0200)]
func_lock: Fix requesters counter in error paths.
In two places we bail out with failure after we've already incremented
the requesters counter, if this occured then it would effectively result
in unload to wait indefinitely, thus preventing clean shutdown.
Naveen Albert [Tue, 25 May 2021 15:36:04 +0000 (11:36 -0400)]
app_originate: Allow setting Caller ID and variables
Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.
Naveen Albert [Mon, 24 May 2021 00:20:28 +0000 (20:20 -0400)]
app_confbridge: New ConfKick() application
Adds a new ConfKick() application, which may
be used to kick a specific channel, all channels,
or all non-admin channels from a specified
conference bridge, similar to existing CLI and
AMI commands.
Naveen Albert [Wed, 2 Jun 2021 13:11:24 +0000 (09:11 -0400)]
sip_to_pjsip: Fix missing cases
Adds the "auto" case which is valid with
both chan_sip dtmfmode and chan_pjsip's
dtmf_mode, adds subscribecontext to
subscribe_context conversion, and accounts
for cipher = ALL being invalid.
George Joseph [Thu, 22 Apr 2021 18:07:22 +0000 (12:07 -0600)]
res_pjsip_messaging: Refactor outgoing URI processing
* Implemented the new "to" parameter of the MessageSend()
dialplan application. This allows a user to specify
a complete SIP "To" header separate from the Request URI.
* Completely refactored the get_outbound_endpoint() function
to actually handle all the destination combinations that
we advertized as supporting.
* We now also accept a destination in the same format
as Dial()... PJSIP/number@endpoint
Naveen Albert [Sun, 16 May 2021 15:21:39 +0000 (11:21 -0400)]
func_math: Three new dialplan functions
Introduces three new dialplan functions, MIN and MAX,
which can be used to calculate the minimum or
maximum of up to two numbers, and ABS, an absolute
value function.
Ben Ford [Wed, 19 May 2021 18:45:16 +0000 (13:45 -0500)]
STIR/SHAKEN: Add Date header, dest->tn, and URL checking.
STIR/SHAKEN requires a Date header alongside the Identity header, so
that has been added. Still on the outgoing side, we were missing the
dest->tn section of the JSON payload, so that has been added as well.
Moving to the incoming side, URL checking has been added to the public
cert URL to ensure that it starts with http.
Joshua C. Colp [Mon, 24 May 2021 18:38:26 +0000 (15:38 -0300)]
res_pjsip: On partial transport reload also move factories.
For connection oriented transports PJSIP uses factories to
produce transports. When doing a partial transport reload
we need to also move the factory of the transport over so
that anything referencing the transport (such as an endpoint)
has the factory available.
Naveen Albert [Thu, 20 May 2021 13:18:03 +0000 (09:18 -0400)]
func_volume: Add read capability to function.
Up until now, the VOLUME function has been write
only, so that TX/RX values can be set but not
read afterwards. Now, previously set TX/RX values
can be read later.
stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing
When unsubscribing from an endpoint technology a FRACK
would occur due to incorrect reference counting. This fixes
that issue, along with some other issues.
Fixed a typo in get_subscription when calling ao2_find as it
needed to pass the endpoint ID and not the entire object.
Fixed scenario where a subscription would get returned when
it shouldn't have been when searching based on endpoint
technology.
A doulbe unreference has also been resolved by only explicitly
releasing the reference held by tech_subscriptions.
ASTERISK-28237 #close
Reported by: Lucas Tardioli Silveira
Joseph Nadiv [Thu, 20 May 2021 07:15:30 +0000 (03:15 -0400)]
res_pjsip.c: Support endpoints with domain info in username
In multidomain environments, it is desirable to create
PJSIP endpoints with the domain info in the endpoint name
in pjsip_endpoint.conf. This resulted in an error with
registrations, NOTIFY, and OPTIONS packet generation.
This commit will detect if there is an @ in the endpoint
identifier and generate the URI accordingly so NOTIFY and
OPTIONS From headers will generate correctly.
Jeremy Lainé [Wed, 19 May 2021 18:13:36 +0000 (20:13 +0200)]
res_rtp_asterisk: make it possible to remove SOFTWARE attribute
By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:
George Joseph [Thu, 15 Apr 2021 15:43:48 +0000 (09:43 -0600)]
res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs
RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm. The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred. We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.
The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.
It doesn't stop there though... Each realm can require a
different username from the others. There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.
So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header. We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.
In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.
NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5. We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5. When we support the
others, we'll move the check into the loop that searches the
objects.
Changes:
* Added a new API ast_sip_retrieve_auths_vector() that takes in
a vector of auth ids (usually supplied on a call to
ast_sip_create_request_with_auth()) and populates another
vector with the actual objects.
* Refactored res_pjsip_outbound_authenticator_digest to handle
multiple Authenticate headers and set the stage for handling
additional digest algorithms.
* Added a pjproject patch that allows them to ignore digest
algorithms they don't support. This patch has already been
merged upstream.
* Updated documentation for auth objects in the XML and
in pjsip.conf.sample.
* Although res_pjsip_authenticator_digest isn't affected
by this change, some debugging and a testsuite AMI event
was added to facilitate testing.
Joseph Nadiv [Wed, 14 Apr 2021 14:44:00 +0000 (10:44 -0400)]
res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml
RFC 4235 Section 4.1.6 describes XML elements that should be
sent to subscribed endpoints to identify the local and remote
participants in the dialog.
This patch adds this functionality to PJSIP by iterating through the
ringing channels causing the NOTIFY, and inserts the channel info
into the dialog so that information is properly passed to the endpoint
in dialog-info+xml.
ASTERISK-24601
Patch submitted: Joshua Elson
Modified by: Joseph Nadiv and Sean Bright
Tested by: Joseph Nadiv
Naveen Albert [Thu, 13 May 2021 15:32:06 +0000 (11:32 -0400)]
AMI: Add AMI event to expose hook flash events
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
Naveen Albert [Thu, 13 May 2021 14:47:08 +0000 (10:47 -0400)]
app_voicemail: Configurable voicemail beep
Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.
To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.
Naveen Albert [Thu, 13 May 2021 15:13:55 +0000 (11:13 -0400)]
main/file.c: Don't throw error on flash event.
AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.
Ben Ford [Mon, 26 Apr 2021 22:00:11 +0000 (17:00 -0500)]
STIR/SHAKEN: Switch to base64 URL encoding.
STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.
The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.
Ben Ford [Tue, 11 May 2021 17:26:13 +0000 (12:26 -0500)]
STIR/SHAKEN: OPENSSL_free serial hex from openssl.
We're getting the serial number of the certificate from openssl and
freeing it with ast_free(), but it needs to be freed with OPENSSL_free()
instead. Now we duplicate the string and free the one from openssl with
OPENSSL_free(), which means we can still use ast_free() on the returned
string.
Ben Ford [Wed, 21 Apr 2021 16:12:55 +0000 (11:12 -0500)]
STIR/SHAKEN: Fix certificate type and storage.
During OpenSIPit, we found out that the public certificates must be of
type X.509. When reading in public keys, we use the corresponding X.509
functions now.
We also discovered that we needed a better naming scheme for the
certificates since certificates with the same name would cause issues
(overwriting certs, etc.). Now when we download a public certificate, we
get the serial number from it and use that as the name of the cached
certificate.
The configuration option public_key_url in stir_shaken.conf has also
been renamed to public_cert_url, which better describes what the option
is for.
George Joseph [Thu, 22 Apr 2021 18:07:22 +0000 (12:07 -0600)]
Updates for the MessageSend Dialplan App
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
Sean Bright [Tue, 27 Apr 2021 17:31:30 +0000 (13:31 -0400)]
res_rtp_asterisk: More robust timestamp checking
We assume that a timestamp value of 0 represents an 'uninitialized'
timestamp, but 0 is a valid value. Add a simple wrapper to be able to
differentiate between whether the value is set or not.
This also removes the fix for ASTERISK~28812 which should not be
needed if we are checking the last timestamp appropriately.
chan_local: Skip filtering audio formats on removed streams.
When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.
Ben Ford [Mon, 29 Mar 2021 17:28:24 +0000 (12:28 -0500)]
res_aeap: Add basic config skeleton and CLI commands.
Added support for a basic AEAP configuration read from aeap.conf.
Also added 2 CLI commands for showing individual configurations as
well as all of them: aeap show server <id> and aeap show servers.
Only one configuration option is required at the moment, and that one is
server_url. It must be a websocket URL. The other option, codecs, is
optional and will be used over the codecs specified on the endpoint if
provided.
George Joseph [Fri, 2 Apr 2021 12:21:33 +0000 (06:21 -0600)]
bridge_channel_write_frame: Check for NULL channel
There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL. The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff. So...
bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.
As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.
ASTERISK-29379 Reported-by: Vyrva Igor Reported-by: Ross Beer
Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce
George Joseph [Thu, 1 Apr 2021 13:39:03 +0000 (07:39 -0600)]
res_prometheus: Clone containers before iterating
The channels, bridges and endpoints scrape functions were
grabbing their respective global containers, getting the
count of entries, allocating metric arrays based on
that count, then iterating over the container. If the
global container had new objects added after the count
was taken and the metric arrays were allocated, we'd run
out of metric entries and attempt to write past the end
of the arrays.
Now each of the scape functions clone their respective
global containers and all operations are done on the
clone. Since the clone is stable between getting the
count and iterating over it, we can't run past the end
of the metrics array.
Joshua C. Colp [Wed, 10 Mar 2021 15:03:11 +0000 (11:03 -0400)]
loader: Output warnings for deprecated modules.
Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.
Kevin Harwell [Mon, 29 Mar 2021 22:40:49 +0000 (17:40 -0500)]
res_rtp_asterisk: Don't count 0 as a minimum lost packets
The calculated minimum lost packets represents the lowest number of
lost packets missed during an RTCP report interval. Zero of course
is the lowest, but the idea is that this value contain the lowest
number of lost packets once some have been missed.
This patch checks to make sure the number of lost packets over an
interval is not zero before checking and setting the minimum value.
Also, this patch updates the rtp lost packet test to check for
packet loss over several reports vs one.
Joshua C. Colp [Mon, 29 Mar 2021 22:52:08 +0000 (19:52 -0300)]
res_rtp_asterisk: Only raise flash control frame on end.
Flash in RTP is conveyed the same as DTMF, just with a
specific digit. In Asterisk however we do flash as a
single control frame.
This change makes it so that only on end do we provide
the flash control frame to the core. Previously we would
provide a flash control frame on both begin and end,
causing flash to work improperly.
Kevin Harwell [Fri, 5 Mar 2021 18:53:47 +0000 (12:53 -0600)]
res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
This patch makes it so when Asterisk is compiled in DEVMODE a CLI
command is available that allows someone to drop incoming RTP
packets. The command allows for dropping of packets once, or on a
timed interval (e.g. drop 10 packets every 5 seconds). A user can
also specify to drop packets by IP address.
projects / thirdparty / asterisk.git / log
