The change to add setting hangupsource to sig_pri_queue_hangup()
made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued
deadlocks when a hangup request was received from the core at the
same time a hanguprequest was received from the remote end via the
D channel.
Although the PRI's channel private structure was being unlocked
before setting the hangupsource, the PRI's own lock was still being
held during the process. If channel actions were also coming from
the core, a deadlock on the PRI could result. This deadlock could
then escalate to the entire DAHDI subsystem via DAHDI's global
interface list lock, especially if someone used the PRI CLI commands.
Fix:
* We now unlock the PRI as well as the PRI's channel private
structure before setting the hangupsource, then relock both
afterwards.
ASTERISK-28605
Reported by: Dirk Wendland
Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9
Dialplan has to be careful about passing an empty device list or empty
positions in the list. As a result, dialplan has to check for these
conditions before using ChanIsAvail. Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.
* Made tolerate empty positions in the device list.
* Simplified the code and eliminated some unnecessary indention.
ASTERISK-28638
Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3
When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
When a topic is added to a pool, its name has the pool's topic
name prepended. For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.
When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.
Fix:
* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.
ASTERISK-28633
Reported by: Joeran Vinzens
Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Dial. Simplify dialplan by making Dial
handle these conditions gracefully.
* Made tolerate empty positions in the dialed device list.
* Reduced some message log levels from notice to verbose.
ASTERISK-28638
Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Page. Simplify dialplan by making Page
handle these conditions gracefully.
* Made tolerate empty positions in the paged device list.
* Reduced some warnings associated with the 's' option to verbose
messages. The warning level for those messages really serves no purpose
as that is why the 's' option exists.
ASTERISK-28638
Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3
The Bridge application was inconsistent if the channel to bridge with is
not specified. If no parameters are given then a warning is issued and
the current channel is hung up. If options are given but no channel is
specified then a warning is issued and the current channel is not hung up.
* Made the Bridge application give a verbose message instead of a warning
if the channel to bridge with is not specified and made not hang up the
current channel. As a result dialplan no longer needs to check if a
channel name is passed before calling Bridge and simply needs to check the
BRIDGERESULT channel variable instead. This is something you likely want
your dialplan to do anyway.
* Fixed up L() option warning message. It is up to the caller to
determine if the channel is hung up because of the warning. Dial() hangs
up the current channel while Bridge() does not.
Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73
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
Fix use of frame-level wildcard usage in suppression file.
ASTERISK-27243 #close
Reported-by: Richard Kenner
Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51
The SIP transaction state was reset when emitting an UPDATE or a re-INVITE
related to a COLP, preventing RTP packets to be emitted.
ASTERISK-28647
Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b
The db_init() function ultimately calls db_sync() which signals the
condition before it is initialized.
Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
ASTERISK-28604
Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
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
Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.
ASTERISK-28636
Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074
Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies. It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.
Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
add missing argument "rtt" and "status" to the documentation
The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.
This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)
ASTERISK-28626
Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d
This reverts commit 174e6426aa.
Reason for revert: Regression in XML validation.
validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)
As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.
Change-Id: I2c9656fb40b2d2f56f54caa35c8be02cc92babd0
