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
Sorcery creates taskprocessors for object types to process object observer
callbacks. An API call is needed to be able to set the congestion levels
of these taskprocessors for selected object types.
* Updated PJSIP's contact and contact_status sorcery object type observer
default congestion levels based upon stress testing. Increased the
congestion levels to reduce the potential for bursty register/unregister
and subscribe/unsubscribe activity from triggering the taskprocessor
overload alert.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6
When taskprocessors get backed up, there is a good chance that we are
being overloaded and need to defer adding new work to the system.
* Implemented a high/low water alert mechanism for modules to check if the
system is being overloaded and take appropriate action. When a
taskprocessor is created it has default congestion levels set. A
taskprocessor can later have those congestion levels altered for specific
needs if stress testing shows that the taskprocessor is a symptom of
overloading or needs to handle bursty activity without triggering an
overload alert.
* Add CLI "core show taskprocessor" low/high water columns.
* Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was
never a good thing to use when creating a taskprocessor because of the
nature of how its references needed to be cleaned up on a partial
creation.
* Made res_pjsip's distributor check if the taskprocessor overload alert
is active before placing a message representing brand new work onto a
distributor serializer.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I182f1be603529cd665958661c4c05ff9901825fa
We must continue using the serializer that the original INVITE came in on
for the dialog. There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems.
Outgoing call legs create the pjsip/outsess/<endpoint> serializers for
their dialogs.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc
Incoming messages that are not part of a dialog or a recognized response
to one of our requests need to be sent to a consistent serializer. Under
load we may be queueing retransmissions before we can process the original
message. We don't need to throw these messages onto random serializers
and cause reentrancy and message sequencing problems.
* Created a pool of pjsip/distributor serializers that get picked by
hashing the call-id and remote tag strings of the received messages.
* Made ast_sip_destroy_distributor() destroy items in the reverse order of
creation.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I2ce769389fc060d9f379977f559026fbcb632407
A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not
loaded and does not have a configuration file. Previously when this
occurred, checks were put in to see if the configuration was loaded
successfully. While this is a good idea - and has been added to the
offending function in res_hep - the reality is res_hep_pjsip and
res_hep_rtcp have no business running if res_hep isn't also running.
As such, this patch also adds a function to res_hep that returns whether
or not it successfully loaded. Oddly enough, ast_module_check returns
"everything is peachy" even if a module declined its load - so it cannot
be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this
function to see if they should continue to load; if it fails, they
decline their load as well.
ASTERISK-26096 #close
Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea
If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192. While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.
In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.
To facilitate determination of format names, the format name has been
added to "core show codecs".
ASTERISK-26070 #close
Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b
POSIX defines poll.h, sys/poll.h should not be used at is c-library
internal header which may or may not exist. Notable in musl it
generates warning of being incorrect. And add explict include of
sys/cdefs.h where needed.
Change-Id: I142930df53fe7585a06b854b6faddc5301e024be
This adds a new parameter to the end of a multicast RTP dialing string.
This parameter defines the following options:
* i: Set the interface from which multicast RTP is sent
* l: Set whether multicast packets are looped back to the sender
* t: Set the TTL for multicast packets
* c: Set the codec to use for RTP
ASTERISK-26068 #close
Reported by Mark Michelson
Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
ARI dial had been implemented using the Dial API. This made great sense
when dialing was 100% separate from bridging. However, if a channel were
to be added to a bridge during the dial attempt, there would be a
conflict between the dialing thread and the bridging thread. Each would
be attempting to read frames from the dialed channel and act on them.
The initial attempt to make the two play nice was to have the Dial API
suspend the channel in the bridge and stay in charge of the channel
until the dial was complete. The problem with this was that it was
riddled with potential race conditions. It also was not well-suited for
the case where the channel changed which bridge it was in during the
dial.
This new approach removes the use of the Dial API altogether. Instead,
the channel we are dialing is placed into an invisible ARI dialing
bridge. The bridge channel thread handles incoming frames from the
channel. If the channel is added to a real bridge, it is departed from
the invisible bridge and then added to the real bridge. Similarly, if
the channel is removed from the real bridge, it is automatically added
back to the invisible bridge if the dial attempt is still active.
This approach keeps the threading simple by always having the channel
being handled by bridge channel threads.
ASTERISK-25925
Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
Invisible bridges function the same as normal bridges, but they have the
following restrictions:
* They never show up in CLI, AMI, or ARI queries.
* They do not have Stasis messages published about them.
Invisible bridges' main use is for when use of the bridging system is
desired, but the bridge should not be known to users of the Asterisk
system.
ASTERISK-25925
Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670
The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of
a signed comparison.
ASTERISK-25669 #close
Reported by: Jesper
patches:
strings.curl.trim.patch submitted by Jesper (License 5518)
Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a
This patch adds a new feature to ARI that allows a client to download
the media associated with a stored recording. The new route is
/recordings/stored/{name}/file, and transmits the underlying binary file
using Asterisk's HTTP server's underlying file transfer facilities.
Because this REST route returns non-JSON, a few small enhancements had
to be made to the Python Swagger generation code, as well as the
mustache templates that generate the ARI bindings.
ASTERISK-26042 #close
Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181
This change uses the newly added multi-user support for
outbound publish to publish to the specific user that an
extension state change is for.
This also extends the res_pjsip_outbound_publish support
to include the user specific From and To URI information in
the outbound publishing of extension state. Since the URI
is used when constructing the body it is important to ensure
that the correct local and remote URIs are used.
Finally the max string growths for the dialog-info+xml
body generator has been increased as through testing it has
proven to be too conservative.
ASTERISK-25965
Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1
Added a new multi_user option that when specified allows a particular
configuration to be used for multiple users. It does this by replacing
the user portion of the server uri with a dynamically created one.
Two new API calls have been added in order to make use of the new
functionality:
ast_sip_publish_user_send - Sends an outgoing publish message based on the
given user. If state for the user already exists it uses that, otherwise
it dynamically creates new outbound publishing state for the user at that
time.
ast_sip_publish_user_remove - Removes all outbound publish state objects
associated with the user. This essentially stops outbound publishing for
the user.
ASTERISK-25965 #close
Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc
Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.
Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.
In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.
It's important to note the following:
- If an offset is provided to the 'play' operations, it only applies to the
first media URI, as it would be weird to skip n seconds forward in every
media resource.
- Operations that control the position of the media only affect the current
media being played. For example, once a media resource in the list
completes, a 'reverse' operation on a subsequent media resource will not
start a previously completed media resource at the appropiate offset.
- This patch does not add any new operations to control the list. Hopefully,
user feedback and/or future patches would add that if people want it.
ASTERISK-26022 #close
Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f
When 2d7a4a3357 was merged, it missed the fact that Verbose log messages
are formatted and handled by 'verbosers'. Verbosers are registered
functions that handle verbose messages only; they exist as a separate
class of callbacks. This was done to handle the 'magic' that must be
inserted into Verbose messages sent to remote consoles, so that the
consoles can format the messages correctly, i.e., the leading
tabs/characters.
In reality, verbosers are a weird appendage: they're a separate class of
formatters/message handlers outside of what handles all other log
messages in Asterisk. After some code inspection, it became clear that
simply passing a Verbose message along with its 'sublevel' importance
through the normal logging mechanisms removes the need for verbosers
altogether.
This patch removes the verbosers, and makes the default log formatter
aware that, if the log channel is a console log, it should simply insert
the 'verbose magic' into the log messages itself. This allows the
console handlers to interpret and format the verbose message
themselves.
This simplifies the code quite a lot, and should improve the performance
of printing verbose messages by a reasonable factor:
(1) It removes a number of memory allocations that were done on each
verobse message
(2) It removes the need to strip the verbose magic out of the verbose
log messages before passing them to non-console log channels
(3) It now performs fewer iterations over lists when handling verbose
messages
Since verbose messages are now handled like other log messages (for the
most part), the JSON formatting of the messages works as well.
ASTERISK-25425
Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.
In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
result, there is always an 'odd message out', leading it to be
potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
This causes RTCP information to be uncorrelated to the SIP message
traffic seen by those capture nodes.
In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.
For res_hep_pjsip:
- uuid_type = call-id: the module uses the SIP Call-ID header value
- uuid_type = channel: the module uses the channel name if available,
falling back to SIP Call-ID if not
For res_hep_rtcp:
- uuid_type = call-id: the module uses the SIP Call-ID header if the
channel type is PJSIP and we have a channel,
falling back to the Stasis event provided
channel name if not
- uuid_type = channel: the module uses the channel name
ASTERISK-25352 #close
Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
With the old SIP module we can use IP access controls per peer.
PJSIP module missing this feature.
This patch added next configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
This patch also better logging failed request:
add custom message instead of "No matching endpoint found"
add SIP method to logging
ASTERISK-25900
Change-Id: I456dea3909d929d413864fb347d28578415ebf02
This migrates res_pjsip_pubsub over to using the newly
introduce common datastores management API instead of using
its own implementations for both subscriptions and
publications.
As well the extension state data now provides a generic
datastores container instead of a subscription. This allows
the dialog-info+xml body generator to work for both
subscriptions and publications.
ASTERISK-25999 #close
Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1
This change introduces a common container based datastores
management API. This has been done in a few places across
the tree but this consolidates all of the logic into one
place in a generic fashion.
ASTERISK-25999
Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a
The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.
libresample was also disabled.
ASTERISK-25993 #close
Reported-by: Joshua Colp
Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.
With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing
This patch added contact.updated event.
ASTERISK-25904
Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.
ASTERISK-25931
Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
Create PUBLISH messages to update a third party when an extension state
changes because of either a device or presence state change.
A configuration example:
[exten-state-publisher]
type=outbound-publish
server_uri=sip:instance1@172.16.10.2
event=presence
; Optional regex for context filtering, if specified only extension state
; for contexts matching the regex will cause a PUBLISH to be sent.
@context=^users
; Optional regex for extension filtering, if specified only extension
; state for extensions matching the regex will cause a PUBLISH to be sent.
@exten=^[0-9]*
; Required body type for the PUBLISH message.
;
; Supported values are:
; application/pidf+xml
; application/xpidf+xml
; application/cpim-pidf+xml
; application/dialog-info+xml (Planned support but not yet)
@body=application/pidf+xml
The '@' extended variables are used because the implementation can't
extend the outbound publish type as it is provided by the outbound publish
module. That means you either have to use extended variables, or
implement some sort of custom extended variable thing in the outbound
publish module. Another option would be to refactor that stuff to have an
option which specifies the use of an alternate implementation's
configuration and then have that passed to the implementation. JColp
opted for the extended variables method originally.
ASTERISK-25972 #close
Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca
When starting the extension state publishers, check if the requested
message body generator is available. If not available give error message
and skip starting that publisher.
* res_pjsip_pubsub.c: Create new API if type/subtype generator
registered.
* res_pjsip_exten_state.c: Use new body generator API for validation.
ASTERISK-25922
Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c
A feature of chan_sip that service providers relied upon was the ability to
identify by the Authorization username. This is most often used when customers
have a PBX that needs to register rather than identify by IP address. From my
own experiance, this is pretty common with small businesses who otherwise
don't need a static IP.
In this scenario, a register from the customer's PBX may succeed because From
will usually contain the PBXs account id but an INVITE will contain the caller
id. With nothing recognizable in From, the service provider's Asterisk can
never match to an endpoint and the INVITE just stays unauthorized.
The fixes:
A new value "auth_username" has been added to endpoint/identify_by that
will use the username and digest fields in the Authorization header
instead of username and domain in the the From header to match an endpoint,
or the To header to match an aor. This code as added to
res_pjsip_endpoint_identifier_user rather than creating a new module.
Although identify_by was always a comma-separated list, there was only
1 choice so order wasn't preserved. So to keep the order, a vector was added
to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
globals/endpoint_identifier_order.
Along the way, the logic in res_pjsip_registrar was corrected to match
most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
The order is:
username@domain
username@domain_alias
username
Auth by username does present 1 problem however, the first INVITE won't have
an Authorization header so the distributor, not finding a match on anything,
sends a securty_alert. It still sends a 401 with a challenge so the next
INVITE will have the Authorization header and presumably succeed. As a result
though, that first security alert is actually a false alarm.
To address this, a new feature has been added to pjsip_distributor that keeps
track of unidentified requests and only sends the security alert if a
configurable number of unidentified requests come from the same IP in a
configurable amout of time. Those configuration options have been added to
the global config object. This feature is only used when auth_username
is enabled.
Finally, default_realm was added to the globals object to replace the hard
coded "asterisk" used when an endpoint is not yet identified.
The testsuite tests all pass but new tests are forthcoming for this new
feature.
ASTERISK-25835 #close
Reported-by: Ross Beer
Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
This patch added new global pjsip option 'disable_multi_domain'.
Disabling Multi Domain can improve Realtime performance by reducing
number of database requests.
ASTERISK-25930 #close
Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
* changes:
test_message.c: Wait longer in case dialplan also processes the test message.
Manager: Short circuit AMI message processing.
manager.c: Eliminate most RAII_VAR usage.
