The code assumed that when the transport-cc feedback
function was called at least one packet will have been
received. In practice this isn't always true, so now
we just reschedule the sending and do nothing.
Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a
When res_config_odbc (and perhaps other realtime backends) reads a SQL
NULL from the database, it coalesces the value to the empty string
which prevents it from being returned to the realtime core.
However, if it instead reads the empty string from the database, it
needs a way to encode that fact without having the value omitted
entirely. It does this by changing the value to a string with a single
space. The realtime code in main/config.c recognizes this special case
and _turns the string back into the empty string_ before passing it to
realtime API consumers.
For all of this to work, we need to ensure that we actually pass the
single-space-string back to the realtime core, which is currently
failing because we are trimming the value before checking its
content. So instead we now special case the single-space-string case
so that empty values are returned properly.
ASTERISK-28719 #close
Reported by: EDV O-TON
Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de
Incrementing stasis_app_playback.media_index directly in our playback
loop means that when we reach the end of our playlist the index into
the vector will be outside of the bounds of the vector.
Instead use a temporary variable and only assign when we're sure that
we are in bounds.
ASTERISK-28713 #close
Reported by: Sébastien Duthil
Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928
Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.
ASTERISK-28714
Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a
Calling 'app_send' eventually calls the app's message handler. It's possible
for a handler to obtain a lock on another object, and then need/want to lock
the app object. If the caller of 'app_send' locks the app object prior to
calling then there's a potential for a deadlock, if another thread calls
'app_send' without locking.
This patch makes it so 'app_send' is not called with the app object locked in
the section of code doing such.
ASTERISK-28423 #close
Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27
The cleanup code in stasis shuts down applications if they are in a deactivated
state, and no longer have explicit subscriptions. When registering an app the
cleanup code was running before calling 'update'. When it should be executed
after 'update' since a call to register may re-activate the app. We don't want
it to shutdown before the 'update' otherwise the app won't be re-activated,
or registered.
This patch makes it so the cleanup code is executed post 'update'.
ASTERISK-28679 #close
Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b
We need to wait for the message sending callback to finish to know if
we succeeded or failed.
ASTERISK-25421 #close
Reported by: Dmitriy Serov
Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059
The change in 9b99ef50b5 updated the
syntax of the 'realtime update2' CLI command but did not correctly
update the calls to ast_update2_realtime().
The issue this addresses was originally opened because we aren't
allowing a SQL NULL to be set as part of the update, but this is a
limitation of the Realtime API and is not a bug.
Additionally, this patch:
* Corrects the example in the command documentation to reflect
'update2' instead of 'update.'
* Fixes the leading spacing of the command documentation.
* Checks that the required 'NULL' literal argument is present where we
expect it to be.
ASTERISK-21794 #close
Reported by: Cédric Bassaget
Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd
This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.
A description of the protocol can be found on the above referenced
GitHub page. A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
ASTERISK-28484 #close
Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
Some body generators, such as dialog-info+xml, require storing state
information which is then conveyed in the NOTIFY request itself. Up
until now there was no way for such body generators to persist this
information.
Two new API calls have been added to allow body generators to set and
get persisted data. This data is persisted out alongside the normal
persistence information and allows the body generator to restore
state information or to simply use this for normal storage of state.
State is stored in the form of JSON and it is up to the body
generator to interpret this as needed.
The dialog-info+xml body generator has been updated to take advantage
of this to persist the version number.
ASTERISK-27759
Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de
Adds source port matching support when IP matching is used:
[example]
type = identify
match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444
If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.
ASTERISK-28639 #close
Reported by: Mitch Claborn
Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92
ast_sorcery_changeset_create() is not commutative and will fail to detect
differences between two variable lists depending on what changed, so switch to
ast_variable_lists_match().
ASTERISK-28492 #close
Reported by: Jean-Denis Girard
Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1
When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.
ASTERISK-28562 #close
Reported by: Robert Sutton
Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d
This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.
ASTERISK-28629
Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc
A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.
This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.
ASTERISK-28659
Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355
A previous patch:
Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39
made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.
This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).
This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.
The default is disabled keeping with the old behavior.
ASTERISK-28660
Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a
In Asterisk 16+, there are a few places in ast_rtp_read where we've
allocated a frame list but return a null frame instead of the list.
In these cases, any frames left in the list won't be freed. In the
vast majority of the cases, the list is empty when we return so
there's nothing to free but there have been leaks reported in the
wild that can be traced back to frames left in the list before
returning.
The escape paths now all have logic to free frames left in the
list.
ASTERISK-28609
Reported by: Ted G
Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a
RFC3261 Section 10 "Registrations", specifically paragraph
"10.2.4: Refreshing Bindings", states that a user agent compares
each contact address (in a 200 REGISTER response) to see if it
created the contact. If the Asterisk endpoint has the
rewrite_contact option set however, the contact host and port sent
back in the 200 response will be the rewritten one and not the
one sent by the user agent. This prevents the user agent from
matching its own contact. Some user agents get very upset when
this happens and will not consider the registration successful.
While this is rare, it is acceptable behavior especially if more
than 1 user agent is allowed to register to a single endpoint/aor.
This commit updates res_pjsip_nat (where rewrite_contact is
implemented) to store the original incoming Contact header in
a new "x-ast-orig-host" URI parameter before rewriting it, and to
restore the original host and port to the Contact headers in the
outgoing response.
This is only done if the request is a REGISTER and rewrite_contact
is enabled.
pjsip_message_filter was also updated to ensure that if a request
comes in with any existing x-ast-* URI parameters, we remove them
so they don't conflict. Asterisk will never send a request
with those headers in it but someone might just decide to add them
to a request they craft and send to Asterisk.
NOTE: If a device changes its contact address and registers again,
it's a NEW registration. If the device didn't unregister the
original registration then all existing behavior based
on aor/remove_existing and aor/max_contacts apply.
ASTERISK-28502
Reported-by: Ross Beer
Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e
The simple fix here is simply to NULL out username and password after we call
ast_free on them. Unfortunately, I noticed that we weren't checking for
allocation failures for username and password, and adding those checks made
things noisy and cumbersome.
So instead we partially rollback the recent LGTM patch, and move the alloca
calls into find_aor_name().
ASTERISK-28641 #close
Reported by: Ross Beer
Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2
We're appropriately sizing the id_domain_alias buffer, but then copying the data
into the id_domain one. We were then using the uninitialized id_domain_alias
buffer we just allocated.
This is ASTERISK~28641 adjacent, but significant enough to warrant its own
patch.
Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9
We need to copy the endpoint name before we call ao2_cleanup() on it,
otherwise we might try to access memory that has been reclaimed.
ASTERISK-28445 #close
Reported by: Bernhard Schmidt
Change-Id: I404b952608aa606e0babd3c4108346721fb726b3
