Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.
Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa
If a permanent contact URI associated with an AOR is invalid, we add a
Contact header to REGISTER responses with a NULL URI, causing a crash.
ASTERISK-28463 #close
Change-Id: Id2b643e58b975bc560aab1c111e6669d54db9102
The following message:
"Subscription request from endpoint <blah> rejected. Expiration of 0 is invalid"
Would sometimes spam the log with warnings if Asterisk restarted and a bunch
of clients sent unsubscribes. This patch changes it from a warning to a debug
message.
Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467
When a stale item was being updated the object was being retrieved, but its
reference was not being decremented after the update. This patch makes it so
the object is now appropriately de-referenced.
ASTERISK-28523
Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7
This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.
ASTERISK-28533
Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
According to RFC3550, ALL RTCP packets must be sent in a compond packet
of at least two individual packets, including SR/RR and SDES. REMB,
FIR, and NACK were not following this format, and as a result, would
fail the packet check in ast_rtcp_interpret. This was found from writing
unit tests for RTCP. The browser would accept the way we were
constructing these RTCP packets, but when sending directly from one
Asterisk instance to another, the above mentioned problem would occur.
Change-Id: Ieb140e9c22568a251a564cd953dd22cd33244605
When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.
Introduce ast_variable_list_replace() and use it where appropriate.
ASTERISK-23756 #close
Patches:
setvar-multiplie.patch submitted by Michael Goryainov
Change-Id: Ie1897a96c82b8945e752733612ee963686f32839
This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.
ASTERISK-17808
Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce
Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.
While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.
Also made a minor fix to the data buffer unit test.
Change-Id: I56107c7411003a247589bbb6086d25c54719901b
The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server. Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.
Change-Id: I9618899198880b4c650354581b50c0401b58bc46
After receiving a 200 OK with a declined stream in response to a T.38
initiated re-invite Asterisk would crash when attempting to dereference
a NULL session media object.
This patch checks to make sure the session media object is not NULL before
attempting to use it.
ASTERISK-28495
patches:
ast-2019-004.patch submitted by Alexei Gradinari (license 5691)
Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572
res_pjsip_mwi allows both solicited and unsolicited MWI subscription types.
While both can be set in the configuration for a given endpoint/aor, only
one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor
is configured to allow both types then the solicited subscription is rejected
when it comes in. However, there is a configuration option to override that
behavior:
mwi_subscribe_replaces_unsolicited
When set to "yes" then when a solicited subscription comes in instead of
rejecting it Asterisk is suppose to replace the unsolicited one if it exists.
Prior to this patch there was a bug in Asterisk that allowed the solicted one
to be added, but did not remove the unsolicited. As a matter of fact a new
unsolicited subscription got added everytime a SIP register was received.
Over time this eventually could "flood" a phone with SIP notifies.
This patch fixes that behavior to now make it work as expected. If configured
to do so a solicited subscription now properly replaces the unsolicited one.
As well when an unsubscribe is received the unsolicited subscription is
restored. Logic was also put in to handle reloads, and any configuration changes
that might result from that. For instance, if a solicited subscription had
previously replaced an unsolicited one, but after reload it was configured to
not allow that then the solicited one needs to be shutdown, and the unsolicited
one added.
ASTERISK-28488
Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1
Given the following request path and 2 handler paths...
Request: /channels/externalMedia
Handler: /channels/{channelId} "wildcard"
Handler: /channels/externalmedia "non-wildcard"
...if /channels/externalMedia was registered as a handler after
/channels/{channelId} as shown above, the request would automatically
match the wildcard handler and attempt to parse "externalMedia" into
the channelId variable which isn't what was intended. It'd work
if the non-wildard entry was defined in rest-api/api-docs/channels.json
before the wildcard entry but that makes the json files
order-dependent which isn't a good thing.
To combat this issue, the search loop saves any wildcard match but
continues looking for exact matches at the same level. If it finds
one, it's used. If it hasn't found an exact match at the end of
the current level, the wildcard is used. Regardless, after
searching the current level, the wildcard is cleared so it won't
accidentally match for a different object or a higher level.
BTW, it's currently not possible for more than 1 wildcard entry
to be defined for a level. For instance, there couldn't be:
Handler: /channels/{channelId}
Handler: /channels/{channelName}
We wouldn't know which one to match.
Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925
In chan_sip, there was variable SIPFROMDOMAIN that allows to set
From header URI domain per channel. This patch introduces res_pjsip
variable SIPFROMDOMAIN for backward compatibility with chan_sip.
ASTERISK-28489
Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e
Somehow it's possible for the srtp session object to be NULL even though the
Asterisk srtp object itself is valid. When this happened it would cause a
crash down in the srtp code when attempting to protect or unprotect data.
After looking at the code there is at least one spot that makes this situation
possible. If Asterisk fails to unprotect the data, and after several retries
it still can't then the srtp->session gets freed, and set to NULL while still
leaving the Asterisk srtp object around. However, according to the original
issue reporter this does not appear to be their situation since they found
no errors logged stating the above happened (which Asterisk does for that
situation).
An issue was found however, where a possible race condition could occur between
the pjsip incoming negotiation, and the receiving of RTP packets. Both places
could attempt to create/setup srtp for the same rtp instance at the same time.
This potentially could be the cause of the problem as well.
Given the above this patch adds locking around srtp setup for a given rtp, or
rtcp instance. NULL checks for the session have also been added within the
protect and unprotect functions as a precaution. These checks should at least
stop Asterisk from crashing if it gets in this situation again.
This patch also fixes one other issue noticed during investigation. When doing
a replace the old object was freed before creating the replacement. If the new
replacement object failed to create then the rtp/rtcp instance would now point
to freed srtp data which could potentially cause a crash as well when the next
attempt to reference it was made. This is now fixed so the old srtp object is
kept upon replacement failure.
Lastly, more logging has been added to help diagnose future issues.
ASTERISK-28472
Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc
The code for gathering contacts could result in the same contact
being retrieved and added to the list multiple times. The container
which stores the contacts to display will now only allow a contact
to be added to it once instead of multiple times.
ASTERISK-28228
Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df
We now check that a body exists and it has a length > 0 before
attempting to process it.
ASTERISK-28447
Reported-by: Gil Richard
Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f
The MWI core recently got some new API calls that make tracking MWI state
lifetime more reliable. This patch updates those modules that subscribe to
specific MWI topics to use the new API. Specifically, these modules now
subscribe to both MWI topics and MWI state.
ASTERISK-28442
Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd
Currently, DELETE /ari/channels/<channelID> supports only few hangup reasons.
It's good enough for simple use, but when it needs to set the detail reason,
it comes challenges.
Added reason_code query parameter for that.
ASTERISK-28385
Change-Id: I1cf1d991ffd759d0591b347445a55f416ddc3ff2
According T.38 Gateway 'Use case 3'
https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway
T.38 Gateway should send T.38 negotiation request to called endpoint
if FAX preamble (using V.21 detector) generated by called endpoint.
But it does not, because fax_gateway_detect_v21 constructs T.38
negotiation request, but forwards it only to other channel,
not to the channel on which FAX preamble is detected.
Some SIP endpoints could be improperly configured to rely on the other side
to initiate T.38 re-INVITEs.
With this patch the T.38 Gateway tries to negotiate with both sides
by sending T.38 negotiation request to both endpoints supported T.38.
Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39
This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.
This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.
ASTERISK-28018
Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06
The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.
The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.
The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.
Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.
This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.
This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.
ASTERISK-27981 #close
Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803
This patch adds a channel name to output of CLI 'fax show session'
and also expands the channel name field up to 30 characters on
CLI 'fax show sessions'
Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953
When monitoring Asterisk instances, it's often useful to know when an
outbound registration fails, as this often maps to the notion of a trunk
and having a trunk fail is usually a "bad thing". As such, this patch
adds monitoring metrics that track the state of PJSIP outbound registrations.
It does this by looking for the Registry events coming across the Stasis
system topic, and publishing those as metrics to Prometheus. Note that
while this may support other outbound registration types (IAX2, SIP, etc.)
those haven't been tested. Your mileage may vary.
(And why are you still using IAX2 and SIP? It's 2019 folks. Get with the
program.)
This patch also adds Sorcery observers to handle modifications to the
underlying PJSIP outbound registration objects. This is useful when a
reload is triggered that modifies the properties of an outbound registration,
or when ARI push configuration is used and an object is updated or
deleted. Because we rely on properties of the registration object to
define the metric (label key/value pairs), we delete the relevant metric when
we notice that something has changed and wait for a new Stasis message to
arrive to re-create the metric.
ASTERISK-28403
Change-Id: If01420e38530fc20b6dd4aa15cd281d94cd2b87e
This patch adds a few CLI commands to the res_prometheus module to aid
system administrators setting up and configuring the module. This includes:
* prometheus show status: Display basic statistics about the Prometheus
module, including its essential configuration, when it was last scraped,
and how long the scrape took. The last two bits of information are useful
when Prometheus isn't generating metrics appropriately, as it will at
least tell you if Asterisk has had its HTTP route hit by the remote
server.
* prometheus show metrics: Dump the current metrics to the CLI. Useful for
system administrators to see what metrics are currently available without
having to cURL or go to Prometheus itself.
ASTERISK-28403
Change-Id: Ic09813e5e14b901571c5c96ebeae2a02566c5172
This patch adds basic Asterisk bridge statistics to the res_prometheus
module. This includes:
* asterisk_bridges_count: The current number of bridges active on the
system.
* asterisk_bridges_channels_count: The number of channels active in a
bridge.
In all cases, enough information is provided with each bridge metric
to determine a unique instance of Asterisk that provided the data, along
with the technology, subclass, and creator of the bridge.
ASTERISK-28403
Change-Id: Ie27417dd72c5bc7624eb2a7a6a8829d7551788dc
This patch adds basic Asterisk endpoint statistics to the res_prometheus
module. This includes:
* asterisk_endpoints_state: The current state (unknown, online, offline)
for each defined endpoint.
* asterisk_endpoints_channels_count: The current number of channels
associated with a given endpoint.
* asterisk_endpoints_count: The current number of defined endpoints.
In all cases, enough information is provided with each endpoint metric
to determine a unique instance of Asterisk that provided the data, as well
as the underlying technology and resource definition.
ASTERISK-28403
Change-Id: I46443963330c206a7d12722d08dcaabef672310e
Using timestamp with signed int will cause timestamps exceeding max value
to be negative.
This causes the jitterbuffer to do passthrough of the packet.
ASTERISK-28421
Change-Id: I9dabd0718180f2978856c50f43aac4e52dc3cde9
This patch adds basic Asterisk channel statistics to the res_prometheus
module. This includes:
* asterisk_calls_sum: A running sum of the total number of
processed calls
* asterisk_calls_count: The current number of calls
* asterisk_channels_count: The current number of channels
* asterisk_channels_state: The state of any particular channel
* asterisk_channels_duration_seconds: How long a channel has existed,
in seconds
In all cases, enough information is provided with each channel metric
to determine a unique instance of Asterisk that provided the data, as
well as the name, type, unique ID, and - if present - linked ID of each
channel.
ASTERISK-28403
Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59
Prometheus is the defacto monitoring tool for containerized applications.
This patch adds native support to Asterisk for serving up Prometheus
compatible metrics, such that a Prometheus server can scrape an Asterisk
instance in the same fashion as it does other HTTP services.
The core module in this patch provides an API that future work can build
on top of. The API manages metrics in one of two ways:
(1) Registered metrics. In this particular case, the API assumes that
the metric (either allocated on the stack or on the heap) will have
its value updated by the module registering it at will, and not
just when Prometheus scrapes Asterisk. When a scrape does occur,
the metrics are locked so that the current value can be retrieved.
(2) Scrape callbacks. In this case, the API allows consumers to be
called via a callback function when a Prometheus initiated scrape
occurs. The consumers of the API are responsible for populating
the response to Prometheus themselves, typically using stack
allocated metrics that are then formatted properly into strings
via this module's convenience functions.
These two mechanisms balance the different ways in which information is
generated within Asterisk: some information is generated in a fashion
that makes it appropriate to update the relevant metrics immediately;
some information is better to defer until a Prometheus server asks for
it.
Note that some care has been taken in how metrics are defined to
minimize the impact on performance. Prometheus's metric definition
and its support for nesting metrics based on labels - which are
effectively key/value pairs - can make storage and managing of metrics
somewhat tricky. While a naive approach, where we allow for any number
of labels and perform a lot of heap allocations to manage the information,
would absolutely have worked, this patch instead opts to try to place
as much information in length limited arrays, stack allocations, and
vectors to minimize the performance impacts of scrapes. The author of
this patch has worked on enough systems that were driven to their knees
by poor monitoring implementations to be a bit cautious.
Additionally, this patch only adds support for gauges and counters.
Additional work to add summaries, histograms, and other Prometheus
metric types may add value in the future. This would be of particular
interest if someone wanted to track SIP response types.
Finally, this patch includes unit tests for the core APIs.
ASTERISK-28403
Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address
This causes both 192.168.1.1 and 1.2.3.4 to be advertized.
Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db
