There was a race condition between client initiated DTLS setup, and handling
of server side ice completion that caused the underlying SSL object to get
cleared during DTLS initialization. If this happened Asterisk would be left
in a partial DTLS setup state. RTP packets were sent and received, but were
not being encrypted and decrypted. This resulted in no audio, or static.
Specifically, this occurred when '__rtp_recvfrom' was processing the handshake
sequence from the client to the server, and then 'ast_rtp_on_ice_complete'
gets called from another thread and clears the SSL object when calling the
'dtls_perform_setup' function. The timing had to be just right in the sense
that from the external SSL library perspective SSL initialization completed
(rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks
to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks
it is not finished, thus not completing 'dtls_srtp_setup'.
This patch removes calls to 'dtls_perform_setup', which clears the SSL object,
in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear
the underlying SSL object. If an ice candidate changes a full protocol level
renegotiation occurs. Also, in the case of bundled ICE candidates are reused
when a stream is added. So no real reason to have to clear, and reset in this
instance.
Also, this patch adds a bit of extra logging to aid in diagnosis of any future
problems.
ASTERISK-28742 #close
Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f
The change to res_config_odbc that allowed empty strings to be
returned to realtime consumers¹ causes a warning to be emitted when
loading MoH classes. So we need to treat an empty 'format' as if it
was not specified to avoid the warning.
ASTERISK-28735 #close
Reported by: Ross Beer
[1] https://gerrit.asterisk.org/c/asterisk/+/13722
Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5
Given a scenario where session refreshes occur close to
each other while another is finishing it was possible for
the session refreshes to occur out of order. It was
also possible for session refreshes to be delayed for
quite some time if a session refresh did not result in
a topology change.
For the out of order session refreshes the first session
refresh would be queued due to a transaction in progress.
This transaction would then finish. When finished a
separate task to process the delayed requests queue
would be queued for handling. A second refresh would
be requested internally before this delayed request
queued task was processed. As no transaction was in
progress this session refresh would be immediately
handled before the queued session refresh.
The code will now check if any delayed requests exist
before allowing a session refresh to immediately occur.
If any exist then the session refresh is queued.
For the delayed session refreshes if a session refresh
did not result in a topology change the attempt would
be immediately stopped and no other delayed requests would
be processed.
The code will now go through the entire delayed requests
queue until a delayed request results in a request
actually being sent.
ASTERISK-28730
Change-Id: Ied640280133871f77d3f332be62265e754605088
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
if asterisk offer T38 SDP with none error correction scheme and
the endpoint respond with redundancy EC scheme, asterisk switch
to that mode. Since we configure the endpoint as none EC mode
we should not switch to any other mode except none.
following logic implemented in code.
1. If asterisk offer none, and anything except none in answer
will be ignored.
2. If asterisk offer fec, answer with fec, redundancy and none will
be accepted.
3. If asterisk offer redundancy, answer with redundancy and none
will be accepted.
ASTERISK-28621
Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19
Fixes: error: ‘domain_name’ may be used uninitialized in this function
Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008
Change-Id: I44413b49ea1205aa25538142161deb73883c79e8
OpenSSL can not tolerate if the packet sent out does not
match the length that it provided to the sender. This change
lies and says that each time the full packet was sent. If
a problem does occur then a retransmission will occur as
appropriate.
ASTERISK-28576
Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
resource_events:stasis_app_message_handler() was locking the session,
then attempting to determine if the app had debug enabled which
locked the app_registry container. res_stasis:__stasis_app_register
was locking the app_registry container then calling app_update
which caused app_handler (which locks the session) to run.
The result was a deadlock.
* Updated resource_events:stasis_app_message_handler() to determine
if debug was set (which locks the app_registry) before obtaining the
session lock.
* Updated res_stasis:__stasis_app_register to release the app_registry
container lock before calling app_update (which locks the sesison).
ASTERISK-28423
Reported by Ross Beer
Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4
There exists a scenario where a thread can hold a lock on the
channels container while trying to lock a bridge. At the same
time another thread can hold the lock for said bridge while
attempting to retrieve a channel. This causes a deadlock.
This change fixes this scenario by retrieving a channel snapshot
instead of a channel, as information present in the snapshot
is all that is needed.
ASTERISK-28616
Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2
Found during some testing, there is a race condition between selecting an
appropriate bridge type for a call versus the applying of media on the callee's
session. In some instances a native bridge type would have been chosen, but
due to the callee's media not yet being established at bridge compatibility
check time the simple bridge type is picked instead.
When using chan_pjsip this initiates a topology change event. The topologies
are then compared for the two sessions. However, when the topology was created
for the caller its streams are initialized to "inactive". This topology is then
used as a base when creating the callee's topology, and streams. Soon after
the caller's topology's stream(s) get updated based on the sdp (get set to
sendrecv in the failing scenario).
Now when the topology change event is raised, and the two topologies are
compared, the comparison fails due to a stream state mismatch (sendrecv vs
inactive). And since they differ a reinvite is sent out (to the caller in
this case).
This patch makes it such that when the caller's topology is initially created
it gets created based on its configured endpoint's media topology. When the
endpoint's topology is created its stream's state(s) are initialized to
sendrecv instead of inactive. Subsequently, now when the callee's topology is
created its topology streams are now initialized to sendrecv. Thus when the
topology change event occurs due to the mentioned scenario the stream states
match for the given sessions, and the reinvite is not sent unless due to some
other valid mismatch.
Note, this patch only changes one pending media state's creation point. It's
possible other places *could* be changed, however for now it was deemed best
to only alter what's here.
Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7
If the "max_retries" option is set to 0 then upon failure no
further attemps are made, so explicitly document the behavior.
ASTERISK-28602
Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf
Calling ne_uri_parse allocates memory that needs to be freed with a
corresponding call to ne_uri_free.
ASTERISK-28572 #close
Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61
This change ensures that the module isn't unloaded when a
WebSocket is open. Previously it was possible to unload the
module manually or during shutdown which could cause a crash
when any active WebSockets were terminated.
ASTERISK-28585
Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c
When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia
REST endpoint. This object contained the channel object that was
created plus local_address and local_port attributes (which are
also in the Channel variables). At the time, we thought that
creating an ExternalMedia object would give us more flexibility
in the future but as we created the sample speech to text
application, we discovered that it doesn't work so well with ARI
client libraries that a) don't have the ExternalMedia object
defined and/or b) can't promote the embedded channel structure
to a first-class Channel object.
This change causes the channels/externalMedia REST endpoint to
return a Channel object (like channels/create and channels/originate)
instead of the ExternalMedia object.
Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9
PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef
(column default values), which has been deprecated since version 8.0.
Since then, the official/correct/supported way to retrieve the column
default value from the catalog is function pg_catalog.pg_get_expr().
This change breaks compatibility with pre-8.0 PostgreSQL servers,
but has reached end-of-support more than a decade ago.
cdr_pgsql and res_config_pgsql still have support for pre-7.3
servers, but cleaning that up is perhaps a topic for a major release,
not this bugfix.
ASTERISK-28571
Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe
When creating an unsolicited MWI aggregate subscription it was possible for
the subscription object to be double unref'ed. This patch removes the explicit
unref as it is not needed since the RAII_VAR will handle it at function end.
Less concerning there was also a bug that could potentially allow the aggregate
subscription object to be added to the unsolicited container twice. This patch
ensures it is added only once.
ASTERISK-28575
Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763
On shutdown it's possible for the unsolicited mwi container to be freed before
other dependent threads are done using it. This patch ensures this can no
longer happen by wrapping the container in an ao2_global object. The solicited
container was also changed too.
ASTERISK-28552
Change-Id: I8f812286dc19a34916acacd71ce2ec26e1042047
Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they
both implemented their own serializer pool functionality that was pretty much
identical in each of the source files. This patch removes the duplicated code,
and uses the new 'ast_serializer_pool' object instead.
Additionally res_pjsip_mwi enables a shutdown group on the pool since if the
timing was right the module could be unloaded while taskprocessor threads still
needed to execute, thus causing a crash.
Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis
ASTERISK-28542 #close
Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8
There are some warning messages which are not informative without endpoint:
"No registered subscribe handler for event presence.winfo"
"No registered publish handler for event presence"
This patch adds an endpoint name to these messages.
Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827
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
This change fixes two bugs which both resulted in the packet loss
count exceeding 65,000.
The first issue is that the sequence number check to determine if
cycling had occurred was using the wrong variable resulting in the
check never seeing that cycling has occurred, throwing off the
packet loss calculation. It now uses the correct variable.
The second issue is that the packet loss calculation assumed that
the received number of packets in an interval could never exceed
the expected number. In practice this isn't true due to delayed
or retransmitted packets. The expected will now be updated to
the received number if the received exceeds it.
ASTERISK-28379
Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6
When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.
Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415
When we use early bridge with create and dial from stasis using Local channel
and the dialplan does not any entry the it is returned from core_local.c with
No such extension .
In such case asterisk locks up till the channel is not hangup with the error
Exceptionally long voice queue length
* Found that in such case app_control_dial fails on ast_call method and
return -1
* Since it is called from stasis_app_send_command_async and return -1 does
not cause resources to be freed and since no PBX exist it is not able to
read from channel causing exceptionally long queue
* After putting this code found that the channel was releasing immediately
and resources were freed.
ASTERISK-28399
Reported by: Abhay Gupta
Tested by: Abhay Gupta
Change-Id: I0a55c923fc6995559f808d63b9488762b4489318
Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.
ASTERISK-28402
Reported-by: Ross Beer
Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7
The dial bridge is meant to hold channels which have been created
and dialed in stasis. It handles the frames coming from them and raises
the appropriate events.
It was possible for the code to mistakenly place calls which came
from the dialplan into the dial bridge if they were not in an
answered state. These channels are not outgoing channels and
should not be placed into the dial bridge.
The code now checks to ensure that only stasis created channels are
placed into the dial bridge by checking that a PBX does not exist
on the channel.
ASTERISK-27756
Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6
The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.
For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.
The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.
ASTERISK-28400
Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer: Configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close
Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
An earlier contributor apparently forgot to run 'make ari-stubs'
before committing after making ARI model changes.
Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6
Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.
ASTERISK-17799 #close
Reported by: Kirill Katsnelson
Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5
In AEL2, if a 'for' statement contains macro* calls, like:
for (&iterator(${TRY},A); "${A}" != ""; &iterate(A)) {
The AEL2 parser will translate these into calls to the deprecated Macro
dialplan application and use the antiquated pipe delimiter.
Instead, convert these into calls to the Gosub dialplan application and
use commas as argument separators.
ASTERISK-18593 #close
Reported by: Luke-Jr
* 'macro' in this context means AEL2 macros, not the 'Macro' application
Change-Id: I3d73716033b8e3e42e0209d355bf5f10c97045fc
When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.
In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().
ASTERISK-14939 #close
Reported by: klaus3000
Change-Id: I8d2cccb3479168fba1b0a6704c52198b396468f1
REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using
an unsigned integer to represent the bitrate. However, that type is not large
enough to hold all potential bitrate values. If the bitrate is large enough
bits were being shifted off the "front" of the mantissa, which caused the
wrong value to be sent to the browser.
This patch makes it so it now uses a float type to hold the bitrate. Using a
float allows for all bitrate values to be correctly represented.
ASTERISK-28255
Change-Id: Ice00fdd16693b16b41230664be5d9f0e465b239e
It looks like we're not properly calculating jitter values on received
video streams. This patch enables the code that does jitter calculations
for those streams.
Change-Id: Iaac985808829c8f034db8c57318789c4c8c11392
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.
ASTERISK-28343
Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c
If Realtime @ variable value is NULL or empty or contains only whitespaces
then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING
pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint.
And the variable is missing in the result of CLI pjsip show endpoint.
This patch keeps empty sorcery extended field.
ASTERISK-28341 #close
Change-Id: I221fccc04cbfa2be17ce971f64ae0e74e465eea0
Because StasisEnd's timestamp created it's own timestamp, it makes
wrong timestamp, compare to other channel event(ChannelDestroyed).
Fixed to getting a timestamp from the Channel's timestamp.
ASTERISK-28333
Change-Id: I5eb8380fc472f1100535a6bc4983c64767026116
As part of an earlier voicemail refactor, ast_delete_mwi_state_full
was modified to remove the pool topic for a mailbox when the state
was deleted. This was an attempt to prevent stale topics from
accumulating when app_voicemail was reloaded and a mailbox went
away. Unfortunately because of the fact that when app_voicemail
reloads, ALL mailboxes are deleted then only current ones recreated,
topics were being removed from the pool that still had subscribers
on them, then recreated as new topics of the same name. So now
modules like res_pjsip_mwi are listening on a topic that will
never receive any messages because app_voicemail is publishing on
a different topic that happens to have the same name. The solutiuon
to this is not easy and given that accumulating topics for
deleted mailboxes is less evil that not sending NOTIFYs...
* Removed the call to stasis_topic_pool_delete_topic in
ast_delete_mwi_state_full.
Also:
* Fixed a topic reference leak in res_pjsip_mwi
mwi_stasis_subscription_alloc.
* Added some debugging to mwi_stasis_subscription_alloc,
stasis_topic_create, and topic_dtor.
* Fixed a topic reference leak in an error path in
internal_stasis_subscribe.
ASTERISK-28306
Reported-by: Jared Hull
Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27
Added ARI resource for channel statistics.
GET /ari/channels/{channelId}/rtp_statistics : It returns given
channel's rtp statistics detail.
ASTERISK-28320
Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376
Changed to requirement to having timestamp for all of ARI events.
The below ARI events were changed to having timestamp.
PlaybackStarted, PlaybackContinuing, PlaybackFinished,
RecordingStarted, RecordingFinished, RecordingFailed,
ApplicationReplaced, ApplicationMoveFailed
ASTERISK-28326
Change-Id: I382c2fef58f5fe107e1074869a6d05310accb41f
Topic names now follow: <subsystem>:<functionality>[/<object>]
This ensures that they are all unique, and also provides better
insight in to what each topic is for.
Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.
Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.
Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.
ASTERISK-28335
Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d
Currently, when the Asterisk calculates rtp statistics, it uses
sample_count as a unsigned integer parameter. This would be fine
for most of cases, but in case of large enough number of sample_count,
this might be causing the divide by zero error.
ASTERISK-28321
Change-Id: If7e0629abaceddd2166eb012456c53033ea26249
chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP. This new flag allows chan_pjsip to have the same
behavior as chan_sip.
ASTERISK-28322 #close
Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a
strtok() uses a static buffer, making it not thread safe.
Also add a #define to cause a compile failure if strtok is used.
Change-Id: Icce265153e1e65adafa8849334438ab6d190e541
Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:
client.channels.move(channelId, app, appArgs)
The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.
ASTERISK-28267 #close
Change-Id: I43d12b10045a98a8d42541889b85695be26f288a
This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.
ASTERISK-28279
Change-Id: I460238c488eca4d216b9176576211cb03286e040
PJSIP assumes that these header names are not allocated, and does not
clone the name strings when reusing headers.
Block unload of res_pjsip_diversion until shutdown to ensure static
memory stays valid.
ASTERISK-28312 #close
Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9
Rather than calling ast_odbc_find_table() in the prepare callback, call
it beforehand and pass it in to the callback to avoid the need for a
second connection.
ASTERISK-28166 #close
Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202
apply_negotiated_sdp_stream was returning a "1" when no joint
capabilities were found on an outgoing call instead of a "-1".
This indicated to res_pjsip_session that the handler DID handle
the sdp when in fact it didn't. Without the appropriate setup,
a subsequent media frame coming in would have an invalid stream_num
and cause a seg fault when the stream was attempted to be retrieved.
apply_negotiated_sdp_stream now returns the correct "-1" and any
media is now discarded before it reaches the core stream processing.
ASTERISK-28260
Reported by: Sotiris Ganouris
Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f
This reverts commit d524ad523d.
Reason for revert: This causes Contact and Via headers to have the wrong
transport address.
ASTERISK-28309 #close
Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8
If both send_registrations and send_auth are both set to yes,
outbound_auth/username must be set or we crash.
ASTERISK-27992 #close
Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d
When a contact was removed by the registrar it did not always check to see if
the circumstances involved a monitored reliable transport. For instance, if the
'remove_existing' option was set to 'true' then when existing contacts were
removed due to 'max_contacts' being reached, those existing contacts being
removed did not unregister the transport monitor.
Also, it was possible to add more than one monitor on a reliable transport for
a given aor and contact.
This patch makes it so all contact removals done by the registrar also remove
any associated transport monitors if necessary. It also makes it so duplicate
monitors cannot be added for a given transport.
ASTERISK-28213
Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a
This module allows presence subscriptions to voicemail boxes. This
allows common BLF keys to act as voicemail waiting indicators.
ASTERISK-28301
Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd
Delivery timeval in the smoother object will fall behind while a DTMF is
being generated. This can eventually lead to invalid rtp timestamps.
To prevent this from happening the smoother needs to be reset after every
DTMF to keep the timing up to date.
ASTERISK-28303 #close
Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f
When listing the applications the apps lock was incorrectly
locked twice instead of being locked and then unlocked.
ASTERISK-28302
Change-Id: If7d064592a9e88c0f1049214c50e02be6dabf79e
When processing SSRC attributes we were iterating through
all of them, even though we only need to know the remote
SSRC once. This was problematic because some browsers group
SSRCs together on a stream, and due to our negotiation only
end up using the first one. Since we set the second one as
the remote SSRC we would drop the received media from them
instead of allowing it through.
In the future this may be extended to allow SSRC groups
and to use information from the attributes.
Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270
To prevent one subsystem's taskprocessors from causing others
to stall, new capabilities have been added to taskprocessors.
* Any taskprocessor name that has a '/' will have the part
before the '/' saved as its "subsystem".
Examples:
"sorcery/acl-0000006a" and "sorcery/aor-00000019"
will be grouped to subsystem "sorcery".
"pjsip/distributor-00000025" and "pjsip/distributor-00000026"
will bn grouped to subsystem "pjsip".
Taskprocessors with no '/' have an empty subsystem.
* When a taskprocessor enters high-water alert status and it
has a non-empty subsystem, the subsystem alert count will
be incremented.
* When a taskprocessor leaves high-water alert status and it
has a non-empty subsystem, the subsystem alert count will be
decremented.
* A new api ast_taskprocessor_get_subsystem_alert() has been
added that returns the number of taskprocessors in alert for
the subsystem.
* A new CLI command "core show taskprocessor alerted subsystems"
has been added.
* A new unit test was addded.
REMINDER: The taskprocessor code itself doesn't take any action
based on high-water alerts or overloading. It's up to taskprocessor
users to check and take action themselves. Currently only the pjsip
distributor does this.
* A new pjsip/global option "taskprocessor_overload_trigger"
has been added that allows the user to select the trigger
mechanism the distributor uses to pause accepting new requests.
"none": Don't pause on any overload condition.
"global": Pause on ANY taskprocessor overload (the default and
current behavior)
"pjsip_only": Pause only on pjsip taskprocessor overloads.
* The core pjsip pool was renamed from "SIP" to "pjsip" so it can
be properly grouped into the "pjsip" subsystem.
* stasis taskprocessor names were changed to "stasis" as the
subsystem.
* Sorcery core taskprocessor names were changed to "sorcery" to
match the object taskprocessors.
Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56
Event type filtering is now enabled, and configurable per application. An app is
now able to specify which events are sent to the application by configuring an
allowed and/or disallowed list(s). This can be done by issuing the following:
PUT /applications/{applicationName}/eventFilter
And then enumerating the allowed/disallowed event types as a body parameter.
ASTERISK-28106
Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b
p2p_write updates txformat but doesn't require a smoother. If a smoother
was created by another bridge type the smoother could fall out of date causing
one way audio issues. To prevent this the smoother is now destroyed on the
start of native bridge.
ASTERISK-28284 #close
Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6
Currently, the Asterisk's pjsip_session module does not keeping the
rtcp's stats info after it was removed. But by adding the results
vector and keeping it until session is destroying, it can give more
useful information for other modules.
ASTERISK-28253
Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5
A previous patch attempt to mitigate blocked threads on transport shutdown for
a given contact. It was thought that a second lock could be avoided by checking
the 'removing' flag on the transport monitor twice (once before and once after
the normal named aor locking). However as with usual threading issues if the
timing was right the original problem still occured.
This patch adds locking around the first 'removing' flag check and set, thus
nullifying the secondary check, so it was removed.
ASTERISK-28213
Change-Id: Iaa8e36e5311789549b76d8de42dfcea96013b2ed
When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.
This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.
This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.
ASTERISK-28277
Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6
At AstriCon, there was a strong desire for the ability to completely
bypass dialplan when using ARI. This is possible through the automatic
creation of a context and a couple of extensions whenever an application
is started.
For example, if you have an application named 'ari-example', a context
named 'stasis-ari-example' will be automatically created whenever this
application is started as long as one does not already exist. Two
extensions (a match-all extension for Stasis and a 'h' extension) are
created within this context. Any endpoint that registers to Asterisk
within this context will send all calls to the corresponding Stasis
application. When the application is destroyed, the context is removed.
ASTERISK-28104 #close
Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac
Added ARI resource.
GET /ari/asterisk/ping : It returns "pong" message with timestamp
and asterisk id. It would be useful for simple heath check.
Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29
The context specified by 'regcontext' was not being created, so when Asterisk
attempted to later dynamically add an extension it would fail. This patch now
creates the context if a 'regcontext' is specified.
ASTERISK-28238
Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265
Testing revealed that the cache added no benefit but that it could
consume excessive memory.
Two new index related functions were created:
ast_sounds_get_index_for_file() and ast_media_index_update_for_file()
which restrict index updating to specific sound files.
The original ast_sounds_get_index() and ast_media_index_update()
calls are still available but since they no longer cache the results
internally, developers should re-use an index they may already have
instead of calling ast_sounds_get_index() repeatedly. If information
for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().
The media_index directory scan code was elimininated in favor of
using the existing ast_file_read_dirs() function.
Since there's no more cache, ast_sounds_index_init now only
registers the sounds cli commands instead of generating the
initial index and subscribing to stasis format register/unregister
messages.
"sounds" is no longer a valid target for the "module reload"
command.
Both the sounds cli commands and the sounds ari resources were
refactored to only call ast_sounds_get_index() once per invocation
and to use ast_sounds_get_index_for_file() when a specific sound
file is requested.
Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d
Control frames (PING / PONG / CLOSE) can be received in the middle of a
fragmented message. In order to ensure they do not interfere with the
reassembly buffer, we exit early and do not return the payload to the
caller.
ASTERISK-28257 #close
Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc
When a reliable transport is shutdown it's possible for the pjsip registrar
resource shutdown handler to get called multiple times. If this happens and one
of the threads is taking "too long" (slow database call for instance) then the
others get blocked waiting to delete.
Since it only takes one to delete the contact then the other threads should be
able to continue on if one of the threads is currently "deleting". This patch
makes it so now when a thread enters the shutdown handler it checks to see if a
thread is currently already "deleting". If so, then the thread does not attempt
to get the lock, and instead continues on thus avoiding the blockage.
ASTERISK-28213 #close
Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a
To avoid the stream name collide if there're more than one video track
in one client. If client has multi video tracks, the name of ast_stream
which represents each video track may be the same. Use the MSID:LABEL
here because it's identifiable.
ASTERISK-28196 #close
Reported-by: xiemchen
Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b
This ensures that Asterisk responds properly to frames received from a
client with opcode 8 (CLOSE) by echoing back the status code in its own
CLOSE frame.
Handling of the CLOSE opcode is moved up with the rest of the opcodes so
that unmasking gets applied. The payload is no longer returned to the
caller, but neither ARI nor the chan_sip nor pjsip made use of the
payload, which is a good thing since it was masked.
ASTERISK-28231 #close
Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf
The transport management code that checks for idle connections keeps a
reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by
default). Because of this, if the transport is closed before this
timeout, the idle checking code will keep the transport from actually
being shutdown until the timeout expires.
Rather than passing the AO2 object to the scheduler task, we just pass
its key and look it up when it is time to potentially close the idle
connection. The other transport management code handles cleaning up
everything else for us.
Additionally, because we use the address of the transport when
generating its name, we concatenate an incrementing ID to the end of the
name to guarantee uniqueness.
Related to ASTERISK~28231
Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb
Previously both AMI and ARI used a default route on
their stasis message router to handle some of the
messages for publishing out their respective
connection. This caused messages to be given to
their subscription that could not be formatted
into AMI or JSON.
This change adds an API call to the stasis message
router which allows a default route to be set as well
as formatters that the default route is expecting.
This allows both AMI and ARI to specify that their
default route only wants messages of their given
formatter. By doing so stasis can more intelligently
filter at publishing time so that they do not receive
messages which will not be turned into AMI or JSON.
ASTERISK-28244
Change-Id: I65272819a53ce99f869181d1d370da559a7d1703
When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert
if passed a payload length of 0, so treat empty frames as if we didn't
receive them.
Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48
The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out
the ContactStatus AMI event when a contact is updated.
Thist change broke things which rely on old behavior.
This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to preserve old
ContactStatus behavior.
By default new behavior, i.e. the ContactStatus event will not be sent when a
device refreshes its registration.
Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46
For video streams it was possible for the abs-send-time information
to be placed into RTP streams even if not negotiated. Depending on
the endpoint in use this could cause video to not flow.
We now only enable abs-send-time for negotiation if WebRTC is enabled.
ASTERISK-28230
Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c
The remote side may start a new stream when renegotiating RTP.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet on RTP renegotiation.
If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.
This bug was caught using Cisco ip-phone SPA5XX and codec g722.
On SIP session update the SPA50X resets stream and a new timestamp is twice
smaller then the previous.
ASTERISK-28162 #close
Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254
All of the fields that were removed were no longer referenced except for
'lastrxts' and 'rxseqno' which were only ever written to.
Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c
The profile-iop octet (the 2nd) of profile-level-id can be zero
according to RFC 6184 Section 8.1. So we ignore its value when deciding
to include profile-level-id in the outgoing SDP.
ASTERISK-27959 #close
Reported by: David Kuehling
Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73
The ARI DELETE /channels command takes a "reason" parameter
Previously, there were only five reasons implemented
This patch adds more reasons to choose from for more
complex setups
ASTERISK-28198 #close
Change-Id: I85996f1076c9946d65c778413f040a845a90fecc
* The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
* The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
* A topic pool is now used for individual bridge topics.
* The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
* A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
* The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
* A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
* The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
* The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.
* cdr, cel, manager and ari have been updated to use the new
arrangement.
Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369
When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.
This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.
The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.
Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.
ASTERISK-28119
Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423
Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.
As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.
The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.
ASTERISK-28102
Change-Id: I9334febff60a82d7c39703e49059fa3a68825786
The marker bit set on the voice packet indicates the start
of a new stream and a new time stamp.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet.
If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.
This bug was caught using Cisco ip-phone SPA50X and codec g722.
On SIP session update the SPA50X resets stream indicating it with market bit
and a new timestamp is twice smaller then the previous.
ASTERISK-28162 #close
Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
The res_ari(POST /channels/create handler) deos not check the endpoint
parameter length. And it causes core
dump.
Fixed it to check the parameter length. Also fixed memory leak.
ASTERISK-28169
Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993
PJSIP assumes that these header names are not allocated, does not clone
the name strings when reusing headers.
Block unload of res_pjsip_caller_id until shutdown to ensure static
memory stays valid. It was previously unsafe to unload while any
sessions are active.
Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f
The presence of Record-Route in re-invites is optional, thus it is
important to make sure the dialog doesn't have a routset before
rewriting the contact header.
ASTERISK-28129 #close
Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc
When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.
Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836
The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.
This patch replaces the pipe with a comma.
ASTERISK-28150
Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719
The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests. Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.
Longer running tasks with the round-robin method can delay processing
tasks.
* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.
Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd
This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.
The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
Change-Id: I16af967815efd904597ec2f033337e4333d097cd
The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.
For example if there are 10k endpoints the module makes 10k requests
of these 10k records.
Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.
This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.
ASTERISK-28137 #close
Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
This change implements a few different generic things which were brought
on by Google Voice SIP.
1. The concept of flow transports have been introduced. These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target. These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity). When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.
2. Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.
3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module. If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.
4. Configurable outbound extension support has been added to the outbound
registration module. When set the extension will be placed in the
Supported header.
5. Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.
6. Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.
All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.
ASTERISK-27971 #close
Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.
Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377
This patch sets the callerid_tag to empty string by default.
If the callerid_tag is set to NULL then the tag does not
become part of a connected line update.
For example:
Alice's tag is "Alice".
Bob's tag is empty.
Charlie's tag is "Charlie".
Alice calls Bob and then does attended transfer to Charlie.
When Alice hangs up the CONNECTEDLINE(tag) is "Alice"
on the interception routine on the Charlie's channel, but should be empty.
Ths patch also fix memory leaks if there are more then one options
"callerid", "callerid_tag", "voicemail_extension" and "contact_user"
in the pjsip.conf endpoint definition.
Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4
The return status when there was no change in statsd.conf was incorrect.
This resulted in the wrong status message on the CLI when reloading the
module.
* Fixed cleanup on initial load if initializing statsd failed.
Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481
I think this module is so screwed up that it doesn't work anymore. Even
with these attempts to fix things it still won't gracefully shut down.
The module refs will not go to zero to allow unloading the module.
* Fix module ref counting dealing with the SMDI interface object. There
were several off-nominal paths that unbalanced the module ref count. Also
the destructor freed the ao2 object itself which is bad. Made the
smdi_read thread not hold its own ref to the SMDI interface object so when
all refs go away the destructor will stop the listener thread.
* Fixed the smdi_load() return code of 1 concerning the number of
listeners. The test was inverted.
Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784
This module is an optional dependency of a couple of other modules. If it
declines to load, it then forces other modules that can optionally use
this module to also decline.
* Made use the default configuration if the config file does not exist and
simplified some of the logic.
Change-Id: Ib93191f1fe28c0dd9ebe3d84c7762b32f83c4eb9
This module is an optional dependency of many modules. If it declines to
load it then forces other modules that can optionally use this module to
also decline.
* Made use default configuration if there is a config error or the config
file does not exist.
Change-Id: If1068a582ec54ab7fb437265cb5370a97a825737
CLI command 'pjsip show contacts' inefficiently make a lot of DB requests.
For example if there are 10k aors then asterisk requests these 10k records
of aor and then does 10k requests of contact - one request per aor.
Even if use 'like <pattern>' the asterisk requests all aor's and contact's
records and then filters them by itself.
This patch gathers contact's container by
- retrieving all dynamic contacts by regex (filtered by reg_server)
- retrieving all aors with permanent contacts
- finally filters container by regex
ASTERISK-28077 #close
Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618
Fixed an issue that resulted in "Allocation failed" each time an ARI
request was made to start playing MOH on a bridge.
In bridge_moh_create() we were attaching the after bridge callbacks to
chan which is the ;1 channel of the unreal channel pair. We should have
attached them to the ;2 channel which is pushed into the bridge by
ast_unreal_channel_push_to_bridge(). The callbacks are called when the
specific channel leaves the bridging system. Since the ;1 channel is
never put into a bridge the callbacks never get called. The callbacks
then never remove the moh_wrapper from the app_bridges_moh container. As
a result we cannot find the channel associated with the wrapper to start
MOH because it has hungup. This is the reason causing the reported issue.
* Rather than using after bridge callbacks to cleanup, we now have
moh_channel_thread() doing the cleanup when the channel hangs up.
* Fixed moh_channel_thread() accumulating control frames on the stasis
bridge MOH channel until MOH is stopped. Control frames are no longer
accumulated while MOH is playing.
* Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may
or may not return a channel ref. As a result ast_ari_bridges_start_moh()
wouldn't know it may have a channel ref to release.
stasis_app_bridge_moh_channel() will now return a ref with the channel it
returns.
* Eliminated RAII_VAR in bridge_moh_create().
ASTERISK-26094 #close
Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c
When networks experience disruptions, there can be large gaps of time
between receiving packets. When strictrtp is enabled, this created
issues where a flood of packets could come in and be seen as an attack.
Another option - seqno - has been added to the strictrtp option that
ignores the time interval and goes strictly by sequence number for
validity.
Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71
On SQL error there is not diagnostic information about this error.
There is only
WARNING res_odbc.c: SQL Execute error -1!
The function ast_odbc_print_errors calls a SQLGetDiagField to get the number
of available diagnostic records, but the SQLGetDiagField returns 0.
However SQLGetDiagRec could return one diagnostic records in this case.
Looking at many example of getting diagnostics error information
I found out that the best way it's to use only SQLGetDiagRec
while it returns SQL_SUCCESS.
Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR
to res_config_odbc.
ASTERISK-28065 #close
Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1
In order to do this and provide good feedback, a new macro was
created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and
path setups for the library then compiles, links and runs a supplied
code fragment to do the final determination. In this case, the
final code fragment compares UNBOUND_VERSION_MAJOR
and UNBOUND_VERSION_MINOR to determine if they're greater than or
equal to 1.5.
Since we require version 1.5, some code in res_resolver_unbound
was also simplified.
ASTERISK-28045
Reported by: Samuel Galarneau
Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72
This change raises a testsuite event to provide what port
Asterisk has actually allocated for RTP. This ensures that
testsuite tests can remove any assumption of ports and instead
use the actual port in use.
ASTERISK-28070
Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044
* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
safe to just use ast_json_ref.
Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped. This was previously
avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null(). The
"?" is a new feature of ast_json_pack starting with Asterisk 16.
Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48
ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call
the destroy callback. Remove call to ast_free from ast_rtp_new, leave
it to rtp_engine.c to initiate the full cleanup. Add error detection
for the ssrc_mapping vector initialization. In rtp_allocate_transport
set rtp->s = -1 in the failure path where we close that FD to ensure we
don't try closing it twice.
ASTERISK-27854 #close
Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451
'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults
if they are not present in the updated configuration file.
Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
ASTERISK-28013 #close
Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a
Kevin Harwell
Joshua Colp
Sean Bright
Friendly Automation
George Joseph
Seán C McCord
Jean Aunis
Joshua Colp
Salah Ahmed
Christoph Moench-Tegeder
Torrey Searle
Alexei Gradinari
Ben Ford
sungtae kim
Stas Kobzar
sungtae kim
Matt Jordan
Morten Tryfoss
agupta
Dan Cropp
Matthew Fredrickson
Jeremy Lainé
Xiemin Chen
Sebastian Damm
Corey Farrell
Chris-Savinovich
Nick French
Richard Mudgett
Moritz Fain