| Age | Commit message (Collapse) | Author |
|---|
|
updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs
Change-Id: I279335a2625261a8492206c37219698f42591c2e
(cherry picked from commit 6f448f32fe9b7379e2630fab7b06205f901f2ded)
|
|
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
|
|
The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.
This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.
With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
pbx_dundi, res_xmpp
Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".
ASTERISK-26164 #close
Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
|
|
|
|
|
|
A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk. Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c. This gets more code out of Asterisk's core that isn't
used when SRTP is not available. This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.
ASTERISK-26253 #close
Reported by: Ben Merrills
Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
|
|
If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'
On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.
To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1
Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
a deadlock is happened.
This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.
ASTERISK-26145 #close
Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
|
|
libunbound at version 1.4.20 (which CentOS still uses) declared all
of their string function parameters as as 'char *'. 1.4.21 changed
them all to 'const char *'. Thankfully 1.4.21 also introduced the
UNBOUND_VERSION_MAJOR define so configure now checks for that and
sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks
that and casts away the 'const' if it's not set.
Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
Fedora24 (1.5.4). There are a few failing tests to be addressed though.
ASTERISK-26283 #close
Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
|
|
The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.
This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.
ASTERISK-26230 #close
Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
|
|
This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init
ASTERISK-26265 #close
Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
|
|
We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity. Otherwise, we could never
execute dangerous functions.
ASTERISK-25996 #close
Reported by: Andrew Nagy
Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba
|
|
|
|
Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
this.
ASTERISK-26218 #close
ASTERISK-26221 #close
Reported by: Aaron Meriwether
Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa
|
|
Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
|
|
|
|
|
|
|
|
|
|
|
|
This changes context switches from a linked list to a vector, makes
'struct ast_sw' opaque to pbx.c.
Although ast_walk_context_switches is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_switches_count (AST_VECTOR_SIZE)
* ast_context_switches_get (AST_VECTOR_GET)
As with ast_walk_context_switches callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the switches, they have been converted to use the new functions.
Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998
|
|
ASTERISK-26190 #close
Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b
|
|
This adds support for tagging functions with the noreturn attribute.
If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE
and DO_CRASH are enabled then failed assertions never return. This can
resolve a large number of false positives with static analyzers.
ASTERISK-26220 #close
Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753
|
|
The fax detection timeout option did not work because basically the wrong
variable was checked in fax_detect_framehook(). As a result, the timer
would timeout immediately and disable fax detection.
* Fixed ignoring negative timeout values. We'd complain and then go right
on using the negative value.
* Fixed destroy_faxdetect() in the off-nominal case of an incomplete
object creation.
* Added more range checking to FAXOPT(gateway) timeout parameter.
ASTERISK-26214 #close
Reported by: Richard Mudgett
Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976
|
|
The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
|
|
This changes context ignore patterns from a linked list to a vector,
makes 'struct ast_ignorepat' opaque to pbx.c.
Although ast_walk_context_ignorepats is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_ignorepats_count (AST_VECTOR_SIZE)
* ast_context_ignorepats_get (AST_VECTOR_GET)
As with ast_walk_context_ignorepats callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the ignorepats, they have been converted to use the new functions.
Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a
|
|
This changes context includes from a linked list to a vector, makes
'struct ast_include' opaque to pbx.c.
Although ast_walk_context_includes is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_includes_count (AST_VECTOR_SIZE)
* ast_context_includes_get (AST_VECTOR_GET)
As with ast_walk_context_includes callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the includes, they have been converted to use the new functions.
const have been applied where possible to parameters for ast_include
functions.
Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60
|
|
This commit adds scaffolding in order to support the SILK audio format
on calls. Roughly, this is what is added:
* Cached silk formats. One for each possible sample rate.
* ast_codec structures for each possible sample rate.
* RTP payload mappings for "SILK".
In addition, this change overhauls the res_format_attr_silk file in the
following ways:
* The "samplerate" attribute is scrapped. That's native to the format.
* There are far more checks to ensure that attributes have been
allocated before attempting to reference them.
* We do not SDP fmtp lines for attributes set to 0.
These changes make way to be able to install a codec_silk module and
have it actually work. It also should allow for passthrough silk calls
in Asterisk.
Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
|
|
|
|
|
|
Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
for DTLS. The source code from main/tcptls.c should have been re-used to ease
security audits. Therefore, this change rolls back the change from July 2015 and
re-uses the code from July 2014. This has the additional benefits to work under
CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.
ASTERISK-25659 #close
Reported by: StefanEng86, urbaniak, pay123
Tested by: sarumjanuch, traud
patches:
res_rtp_asterisk.patch submitted by sarumjanuch
dtls_centos_step_1.patch submitted by traud
dtls_centos_step_2.patch submitted by traud
Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
|
|
When res_corosync detects that a node leaves or joins, it currently is
informed of this via Corosync callbacks. However, there are a few
limitations with the information presented:
(1) While we have information that Corosync is aware of - such as the
Corosync nodeid - that information is really only useful inside of
Corosync or res_corosync. There's no way to translate a Corosync
nodeid to some other internally useful unique identifier for the
Asterisk instance that just joined or left the cluster.
(2) While res_corosync is notified of the instance joining or leaving
the cluster, it has no mechanism to inform the Asterisk core or
other modules of this event. This limits the usefulness of res_corosync
as a heartbeat mechanism for other modules.
This patch addresses both issues.
First, it adds the notion of a cluster discovery message both within the
Stasis message bus, as well as the binary event messages that
res_corosync uses to transmit data back and forth within the cluster.
When Asterisk joins the cluster, it sends a discovery message to the other
nodes in the cluster, which correlates the Corosync nodeid along with
the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids
to Asterisk EIDs, such that it can map changes in cluster state with the
Asterisk instance that has that nodeid. Likewise, when an Asterisk
instance receives a discovery message from a node in the cluster, it now
sends its own discovery message back to the originating node with the
local Asterisk EID. This lets Asterisk instances within the cluster
build a complete picture of the other Asterisk instances within the
cluster.
Second, it publishes the discovery messages onto the Stasis message bus.
Said messages are published whenever a node joins or leaves the cluster.
Interested modules can subscribe for the ast_cluster_discovery_type()
message under the ast_system_topic() and be notified when changes in
cluster state occur.
Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465
|
|
Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version.
ASTERISK-26046 #close
Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7
|
|
|
|
|
|
|
|
The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
|
|
The func_odbc module was modified to ensure that the
previous behavior of using a single database connection
was maintained. This was done by getting a single database
connection and holding on to it. With the new multiple
connection support in res_odbc this will actually starve
every other thread from getting access to the database as
it also maintains the previous behavior of having only
a single database connection.
This change disables the func_odbc specific behavior if
the res_odbc module is running with only a single database
connection active. The connection is only kept for the
duration of the request.
ASTERISK-26177 #close
Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f
|
|
If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
|
|
Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This
avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is
using AS_HELP_STRING everywhere else already.
ASTERISK-26046
Change-Id: I8299faf504ceaeee3e39930c59293809e116c631
|
|
|
|
ast_codec"
|
|
|
|
Adding format_name even to the end of ast_codec caused issued with
binary codec modules because the pointer would be garbage in asterisk
when they registered. So, the ast_codec structure was reverted and an
internal_ast_codec structure was created just for use in codec.c. A new
internal-only API was also added (__ast_codec_register_with_format) so
that codec_builtin could register codecs with the format_name in a
separate parameter rather than in the ast_codec structure.
ASTERISK-26144 #close
Reported-by: Alexei Gradinari
Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba
|
|
Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C
but requires ANSI C anyway.
ASTERISK-26046
Change-Id: I914c014385e1862102d90fe7650621def78db02e
|
|
|
|
The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.
The status of endpoints with qualified aors will be updated by 'qualify'
functions.
ASTERISK-26061 #close
Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
|
|
A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.
In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.
This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.
ASTERISK-26128 #close
Change-Id: I51574736a881189de695a824883a18d66a52dcef
|
|
Occasionally under load we'll attempt to send a final NOTIFY on a
subscription that's already been terminated and a SEGV will occur
down in pjproject's evsub_destroy function. This is a result of a
race condition between all the paths that can generate a notify
and/or destroy the underlying pjproject evsub object:
* The client can send a SUBSCRIBE with Expires: 0.
* The client can send a SUBSCRIBE/refresh.
* The subscription timer can expire.
* An extension state can change.
* An MWI event can be generated.
* The pjproject transaction timer (timer_b) can expire.
Normally when our pubsub_on_evsub_state is called with a terminate,
we push a task to the serializer and return at which point the dialog
is unlocked. This is usually not a problem because the task runs
immediately and locks the dialog again. When the system is heavily
loaded though, there may be a delay between the unlock and relock
during which another event may occur such as the subscription timer
or timer_b expiring, an extension state change, etc. These may also
cause a terminate to be processed and if so, we could cause pjproject
to try to destroy the evsub structure twice. There's no way for us to
tell that the evsub was already destroyed and the evsub's group lock
can't tolerate this and SEGVs.
The remedy is twofold.
* A patch has been submitted to Teluu and added to the bundled
pjproject which adds add/decrement operations on evsub's group lock.
* In res_pjsip_pubsub:
* configure.ac and pjproject-bundled's configure.m4 were updated
to check for the new evsub group lock APIs.
* We now add a reference to the evsub group lock when we create
the subscription and remove the reference when we clean up the
subscription. This prevents evsub from being destroyed before
we're done with it.
* A state has been added to the subscription tree structure so
termination progress can be tracked through the asyncronous tasks.
* The pubsub_on_evsub_state callback has been split so it's not doing
double duty. It now only handles the final cleanup of the
subscription tree. pubsub_on_rx_refresh now handles both client
refreshes and client terminates. It was always being called for
both anyway.
* The serialized_on_server_timeout task was removed since
serialized_pubsub_on_rx_refresh was almost identical.
* Missing state checks and ao2_cleanups were added.
* Some debug levels were adjusted to make seeing only off-nominal
things at level 1 and nominal or progress things at level 2+.
ASTERISK-26099 #close
Reported-by: Ross Beer.
Change-Id: I779d11802cf672a51392e62a74a1216596075ba1
|
|
Announcer channels were not being destroyed because the
stasis_app_control structure that referenced them was not being
destroyed. The control structure was not being destroyed because it was
not being unlinked from its container. It was not being unlinked from
its container because the after bridge callback for the announcer
channel was not being run. The after bridge callback was not being run
because the after bridge datastore was not being removed from the
channel on destruction. The channel was not being destroyed because the
hangup that used to destroy the channel was now only reducing the
reference count to one. The reference count of the channel was only
being reduced to one because the stasis_app_control structure was
holding the final reference...
The control structure used to not keep a reference to the channel, so
that loop described above did not happen.
The solution is to manually remove the control structure from its
container when the playback on a bridge is complete.
ASTERISK-26083 #close
Reported by Joshua Colp
Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4
|
|
Stasis subscriptions and message routers create taskprocessors to process
the event messages. API calls are needed to be able to set the congestion
levels of these taskprocessors for selected subscriptions and message
routers.
* Updated CDR, CEL, and manager's stasis subscription congestion levels
based upon stress testing. Increased the congestion levels to reduce the
potential for bursty call setup/teardown activity from triggering the
taskprocessor overload alert. CDRs in particular need an extra high
congestion level because they can take awhile to process the stasis
messages.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: Id0a716394b4eee746dd158acc63d703902450244
|
