* Now conf_alloc() has more off nominal error checking.
* Eliminated RAII_VAR() use in conf_alloc().
* Eliminated a dubius shortcut when destroying cfg->general in
conf_destructor() that would cause a crash if cfg->general failed to get
allocated.
* Add some ACO registration section comments.
Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a
Need to finish initializing the string fields in the ao2 object before
putting any default strings into them.
ASTERISK-25383 #close
Reported by: yaron nahum
Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84
When b99a705262 was merged, subscribing to a
NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to
all bridges. Unfortunately, the res_stasis control loop did not check that
a bridge changing on a channel's control object was actually also non-NULL.
As a result, app_subscribe_bridge will be called with a NULL bridge when a
channel leaves a bridge. This causes a new subscription to be made to the
bridge. If an application has also subscribed to the bridge, the application
will now have two subscriptions:
(1) The explicit one created by the app
(2) The implicit one accidentally created by the control structure
As a result, the 'BridgeDestroyed' event can be sent multiple times. This
patch corrects the control loop such that it only subscribes an application
to a new bridge if the bridge pointer is non-NULL.
ASTERISK-24870
Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f
This patch adds support for receiving events regarding Peer status changes
and Contact status changes. This is particularly useful in scenarios where
we are subscribed to all endpoints and channels, where we often want to know
more about the state of channel technology specific items than a single
endpoint's state.
ASTERISK-24870
Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
This patch adds support for subscribing to all device state changes. This is
done either by subscribing to an empty device, e.g., 'eventSource=deviceState:',
or by the WebSocket connection specifying that it wants all state in the
system.
ASTERISK-24870
Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32
This patch adds the ability to subscribe to all events. There are two possible
ways to accomplish this:
(1) On initial WebSocket connection. This patch adds a new query parameter,
'subscribeAll'. If present and True, Asterisk will subscribe the
applications to all ARI events.
(2) Via the applications resource. When subscribing in this manner, an ARI
client should merely specify a blank resource name, i.e., 'channels:'
instead of 'channels:12354'. This will subscribe the application to all
resources of the 'channels' type.
ASTERISK-24870 #close
Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6
There is a slim chance of a race condition occurring where two threads
can both attempt to manipulate the same area.
Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
lets the specific subscription handler know that the subscription has
been established.
At this point, Thread B may detect a state change on the subscribed
resource and queue up a notification task on Thread C, the subscription
serializer thread.
Now Thread A attempts to generate the initial NOTIFY request to send to
the subscriber at the same time that Thread C attempts to generate a
state change NOTIFY request to send to the subscriber.
The result is that Threads A and C can step on the same memory area,
resulting in a crash. The crash has been observed as happening when
attempting to allocate more space to hold the body for the NOTIFY.
The solution presented here is to queue the subscription establishment
and initial NOTIFY generation onto the subscription serializer thread
(Thread C in the above scenario). This way, there is no way that a state
change notification can occur before the initial NOTIFY is sent, and if
there is a quick succession of NOTIFYs, we can guarantee that the two
NOTIFY requests will be sent in succession.
Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815
We should not try to send a SIP response message because we may be
restoring a persistent subscription where we are not responding to a SIP
request.
Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec
ast_sip_pubsub_register_body_generator() did not account for the null
terminator set by sprintf() in the allocated output buffer.
Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2
The default_from_user retrieval function was pulling the
default_from_user from the global configuration struct in an unsafe way.
If using a database as a backend configuration store, the global
configuration struct is short-lived, so grabbing a pointer from it
results in referencing freed memory.
The fix here is to copy the default_from_user value out of the global
configuration struct.
Thanks go to John Hardin for discovering this problem and proposing the
patch on which this fix is based.
ASTERISK-25390 #close
Reported by Mark Michelson
Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c
We will only rewrite the Contact header if there is no Record-Route header in
the received request. If a malfunctioning proxy places a Record-Route header
into a REGISTER request, we will decide that we shouldn't update the IP/port
in the Contact header, and we will end up storing a contact with an AoR that
contains the NAT'd IP address.
While it is nice to have the proxy *not* send a Record-Route in a REGISTER
request, it's also a good idea to not process the header in a non-dialog
message. This patch updates the code to explicitly ignore the Record-Route
header in REGISTER requests.
ASTERISK-25387 #close
Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f
Make certain that the pjsip session has not failed to
allocate the format capabilities structure, which can
otherwise cause a crash when referenced.
ASTERISK-25323
Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750
In Asterisk 11, the announcer channel would receive channel variables
from the channel being parked by means of normal channel inheritance.
This functionality was lost during the big res_parking project in
Asterisk 12. This patch restores that functionality.
ASTERISK-25369 #close
Review: https://gerrit.asterisk.org/#/c/1180/
Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e
In working through a recent ICE negotiation bug, I found the debug
logging in res_rtp_asterisk to be lacking. This patch adds a number of
debug and warning statements that were helpful.
Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80
In the wild it is possible for Contact URIs to be quite long as
parameters can exist on them. This can present a problem when storing
them in the AstDB as the URI is used as part of the object name and
there is a fixed length limit for the AstDB. This will cause
the contact to not get stored.
This change uses the MD5 hash of the Contact URI as part of the
object name instead. This has a fixed length which is guaranteed
to not exceed the AstDB length limit.
ASTERISK-25295 #close
Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02
When an AoR is deleted by an external mechanism, such as through ARI, we
currently do not remove dynamic contacts that were created for that AoR as a
result of a received REGISTER request. As a result, re-creating the AoR will
cause the dynamic contact to be interpreted as a persistent contact, leading
to some rather strange state being created for the contacts/endpoints.
This patch adds a sorcery observer for the 'aor' object. When a delete is
issued on the underlying sorcery object, the observer is called, and all
contacts created and persisted in sorcery for that AoR are also removed. Note
that we don't want to perform this action when an AO2 object that is an AoR is
destroyed, as the AoR can still exist in the backing storage (and we would
thus be removing valid contacts from an AoR that still "exists".)
ASTERISK-25381 #close
Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328
We were passing the wrong count into pj_ice_sess_create_check_list(),
causing the create to fail if we ever received more than PJ_ICE_MAX_CAND
candidates.
Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378
When Asterisk sends an outbound SIP request, if there is no direct
reason to place a specific value for the username in the From header,
Asterisk would generate a UUID. For example, this would happen when
sending outbound OPTIONS requests when qualifying or when sending
outbound INVITE requests when originating (if no explicit caller ID were
provided). The issue is that some SIP providers reject these sorts of
requests with a "Name too long" error response.
This patch aims to fix this by changing the default outbound username in
From headers to "asterisk". This value can be overridden by changing the
default_from_user option in the global options if desired.
ASTERISK-25377 #close
Reported by Mark Michelson
Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190
Pjsip is refusing to use unsecure transport with "sips" in url.
WSS should be considered as secure transport.
ASTERISK-24602 #comment Partially fixed by setting WSS as secure
Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353
When sending a stateful response, creation of the transaction can fail,
most commonly because we are trying to create a transaction from a
retransmitted request. When creation of the transaction fails, we end up
leaking a reference to a contact that was bumped when the response was
created.
This patch adds the missing deref and fixes the reference leak.
Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07
A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
a means of writing an appropriate packet to persistent storage. While
this partially solved the issue, it had its own problems.
pjsip_msg_print will always add a Content-Length header to the message
it prints. Frequent restarts of Asterisk can result in persistent
subscriptions being written with five or more Content-Length headers. In
addition, sometimes some apparent corruption of individual headers could
be seen.
This aims to fix the problem by not running a parsed message through an
interpreter but rather by taking the raw message and saving it. The
logic for what to save is going to be different depending on whether a
SUBSCRIBE was received from the wire or if it was pulled from
persistence. When receiving a packet from the wire, when using a
streaming transport, the rdata->pkt_info.packet may contain multiple SIP
messages or fragments. However, the rdata->msg_info.msg_buf will always
contain the current SIP message to be processed. When pulling from
persistence, though, the rdata->msg_info.msg_buf will be NULL since no
transport actually handled the packet. However, since we know that we
will always ever pull one SIP message from persistence, we are free to
save directly from rdata->pkt_info.packet instead.
ASTERISK-25365 #close
Reported by Mark Michelson
Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b
The keepalive support in res_pjsip_sdp_rtp currently assumes
that a stream will only be negotiated once. This is false.
If the stream is replaced and later added back it can be
negotiated again causing multiple keepalive scheduled items
to exist. This change explicitly deletes the existing
keepalive scheduled item before adding the new one.
The res_pjsip_sdp_rtp module also does not stop RTP
keepalives or timeout timer if the stream has been
replaced. This change adds a callback to the session media
interface to allow a media stream to be stopped without
the resources being destroyed. This allows the scheduled
items and RTP to be stopped when the stream no longer
exists.
ASTERISK-25356 #close
Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de
When a BYE request is received the PJSIP invite session implementation
creates and sends a 200 OK response before we are aware of it. This
causes the INVITE session state callback to be called into and ultimately
the session supplements run on the BYE request. Once this response has
been sent the normal transaction state callback is invoked which
invokes the session supplements on the BYE request again. This can
be problematic in particular with res_pjsip_rfc3326 as it may
attempt to update the hangup cause code on the channel while it is
in the process of being hung up.
This change makes it so the session supplements are only invoked
once by the INVITE session state callback.
ASTERISK-25318 #close
Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a
If the ast_strndup() call fails to allocate a copy of the
transport string for parsing, fail gracefully.
ASTERISK-25323
Reported by: Scott Griepentrog
Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28
Modules commonly used the pj_gethostip function for retrieving the
IP address of the host. This function does not cache the result and may
result in a DNS lookup occurring, or additional work. If the DNS
server is unreachable or network issues arise this can cause the
pj_gethostip function to block for a period of time.
This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
function which does the same thing but caches the host IP address at
module load time. This results in no additional work being done each
time the local host IP address is needed.
ASTERISK-25342 #close
Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e
When recreating a subscription it is possible for a freed sub_tree
to be referenced when the initial NOTIFY fails to be created.
Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788
When an endpoint is backed by a non-static conf file backend (such as
the AstDB or Realtime), the 'auth' object may be returned as being an
empty string. Currently, res_pjsip will interpret that as being a valid
auth object, and will attempt to authenticate inbound requests. This
isn't desired; is an auth value is empty (which the name of an auth
object cannot be), we should instead interpret that as being an invalid
auth object and skip it.
ASTERISK-25339 #close
Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7
This is a type mismatch fix of the debugging commit
c63316eec1 made to find out why
a testsuite test was failing only on one of the continuous
integration build agents.
Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75
